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[alsa-devel] [PATCH v2] ASoC: samsung/smartq: use dynamic registration
by Arnd Bergmann 14 Jul '14

14 Jul '14

As a prerequisite for moving s3c64xx into multiplatform configurations, we need to change the smartq audio driver to stop using hardcoded gpio numbers from the header file, and instead pass the gpio data through platform_data. In order to do that, we also move the code to use module_platform_driver and register the platform device using platform_device_register_simple and register the gpios through the gpiod API. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/arch/arm/mach-s3c64xx/mach-smartq.c b/arch/arm/mach-s3c64xx/mach-smartq.c index b3d13537a7f0..deed4dd2d0e4 100644 --- a/arch/arm/mach-s3c64xx/mach-smartq.c +++ b/arch/arm/mach-s3c64xx/mach-smartq.c @@ -12,6 +12,7 @@ #include <linux/delay.h> #include <linux/fb.h> #include <linux/gpio.h> +#include <linux/gpio/driver.h> #include <linux/init.h> #include <linux/platform_device.h> #include <linux/pwm_backlight.h> @@ -379,6 +380,15 @@ void __init smartq_map_io(void) smartq_lcd_mode_set(); } +static struct gpiod_lookup_table smartq_audio_gpios = { + .dev_id = "smartq-audio", + .table = { + GPIO_LOOKUP("GPL", 12, "headphone detect", 0), + GPIO_LOOKUP("GPK", 12, "amplifiers shutdown", GPIO_ACTIVE_HIGH), + { }, + }, +}; + void __init smartq_machine_init(void) { s3c_i2c0_set_platdata(NULL); @@ -397,4 +407,7 @@ void __init smartq_machine_init(void) WARN_ON(smartq_wifi_init()); platform_add_devices(smartq_devices, ARRAY_SIZE(smartq_devices)); + + gpiod_add_lookup_table(&smartq_audio_gpios); + platform_device_register_simple("smartq-audio", -1, NULL, 0); } diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 9b0ffacab790..b13f60e402ae 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -13,15 +13,12 @@ * */ -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/jack.h> -#include <mach/gpio-samsung.h> -#include <asm/mach-types.h> - #include "i2s.h" #include "../codecs/wm8750.h" @@ -110,7 +107,7 @@ static struct snd_soc_jack_pin smartq_jack_pins[] = { static struct snd_soc_jack_gpio smartq_jack_gpios[] = { { - .gpio = S3C64XX_GPL(12), + .gpio = -1, .name = "headphone detect", .report = SND_JACK_HEADPHONE, .debounce_time = 200, @@ -127,7 +124,9 @@ static int smartq_speaker_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event)); + struct gpio_desc *gpio = snd_soc_card_get_drvdata(&snd_soc_smartq); + + gpiod_set_value(gpio, SND_SOC_DAPM_EVENT_OFF(event)); return 0; } @@ -218,62 +217,39 @@ static struct snd_soc_card snd_soc_smartq = { .num_controls = ARRAY_SIZE(wm8987_smartq_controls), }; -static struct platform_device *smartq_snd_device; - -static int __init smartq_init(void) +static int smartq_probe(struct platform_device *pdev) { + struct gpio_desc *gpio; int ret; - if (!machine_is_smartq7() && !machine_is_smartq5()) { - pr_info("Only SmartQ is supported by this ASoC driver\n"); - return -ENODEV; - } - - smartq_snd_device = platform_device_alloc("soc-audio", -1); - if (!smartq_snd_device) - return -ENOMEM; - - platform_set_drvdata(smartq_snd_device, &snd_soc_smartq); - - ret = platform_device_add(smartq_snd_device); - if (ret) { - platform_device_put(smartq_snd_device); - return ret; - } + platform_set_drvdata(pdev, &snd_soc_smartq); /* Initialise GPIOs used by amplifiers */ - ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown"); - if (ret) { - dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n"); - goto err_unregister_device; + gpio = devm_gpiod_get(&pdev->dev, "amplifiers shutdown"); + if (IS_ERR(gpio)) { + dev_err(&pdev->dev, "Failed to register GPK12\n"); + ret = PTR_ERR(gpio); + goto out; } + snd_soc_card_set_drvdata(&snd_soc_smartq, gpio); - /* Disable amplifiers */ - ret = gpio_direction_output(S3C64XX_GPK(12), 1); - if (ret) { - dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n"); - goto err_free_gpio_amp_shut; - } - - return 0; - -err_free_gpio_amp_shut: - gpio_free(S3C64XX_GPK(12)); -err_unregister_device: - platform_device_unregister(smartq_snd_device); + ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_smartq); + if (ret) + dev_err(&pdev->dev, "Failed to register card\n"); +out: return ret; } -static void __exit smartq_exit(void) -{ - gpio_free(S3C64XX_GPK(12)); - - platform_device_unregister(smartq_snd_device); -} +static struct platform_driver smartq_driver = { + .driver = { + .name = "smartq-audio", + .owner = THIS_MODULE, + }, + .probe = smartq_probe, +}; -module_init(smartq_init); -module_exit(smartq_exit); +module_platform_driver(smartq_driver); /* Module information */ MODULE_AUTHOR("Maurus Cuelenaere <mcuelenaere(a)gmail.com>");

1 0

[alsa-devel] [PATCH] pcm: rate: fix hw_ptr exceed the boundary
by Shengjiu Wang 14 Jul '14

14 Jul '14

For long time test case, the hw_ptr will exceed the boundary, then cause the avail size wrong. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- src/pcm/pcm_rate.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 7f667d4..2563d82 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -574,6 +574,8 @@ static inline void snd_pcm_rate_sync_hwptr(snd_pcm_t *pcm) rate->hw_ptr = (slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size + rate->ops.input_frames(rate->obj, slave_hw_ptr % rate->gen.slave->period_size); + + rate->hw_ptr %= pcm->boundary; } static int snd_pcm_rate_hwsync(snd_pcm_t *pcm) -- 1.7.9.5

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[alsa-devel] [PATCH V3 1/2] doc: dt bindings: Document Odroid X2/U3 audio subsystem bindings
by Sylwester Nawrocki 14 Jul '14

14 Jul '14

Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v2: - none. --- .../bindings/sound/samsung,odroidx2-max98090.txt | 35 ++++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt new file mode 100644 index 0000000..9148f72 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt @@ -0,0 +1,35 @@ +Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec + +Required properties: + - compatible : "samsung,odroidx2-audio" - for Odroid X2 board, + "samsung,odroidu3-audio" - for Odroid U3 board + - samsung,model : the user-visible name of this sound complex + - samsung,i2s-controller : the phandle of the I2S controller + - samsung,audio-codec : the phandle of the MAX98090 audio codec + - samsung,audio-routing : a list of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board + For Odroid X2: + * Headphone Jack + * Mic Jack + * DMIC + + For Odroid U3: + * Headphone Jack + * Speakers + +Example: + +sound { + compatible = "samsung,odroidu3-audio"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98090>; + samsung,model = "Odroid-X2"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; +}; -- 1.7.9.5

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[alsa-devel] No speaker output with kernel 3.15 and later
by Steven Newbury 14 Jul '14

14 Jul '14

I sent Takashi an email mentioning this regression a month ago, and it's taken me this long to send this email... :-$ Since updating to kernel versions >=3.15 I've had no audio from my laptop speakers. Earlier versions worked fine. It's a CLEVO W270EUQ laptop. The output mixer device is showing up, but no sound comes out at any volume level. As requested I've attached compressed alsa-info.sh output for both the working and non-working cases. I'm perfectly willing to test patches etc.

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[alsa-devel] [PATCH V2 0/2] Sound support for Exynos4412 Odroid X2, U3 board
by Sylwester Nawrocki 14 Jul '14

14 Jul '14

This series adds basic sound support for the Odroid X2/U3 boards. It relies on specific Exynos Audio Subsystem clock parent and frequencies being pre-configured. My full testing git branch has been pushed to: git://linuxtv.org/snawrocki/samsung.git v3.16-rc1-odroid-sound-clk It can be browsed at: http://git.linuxtv.org/cgit.cgi/snawrocki/samsung.git/log/?h=v3.16-rc1-odro… I didn't use the simple-card since there is going to be added one more CPU interface using the secondary I2S and IDMA. Sylwester Nawrocki (2): ASoC: samsung: Add machine driver for Odroid X2/U3 doc: dt bindings: Document Odroid X2/U3 audio subsystem bindings .../bindings/sound/samsung,odroidx2-max98090.txt | 35 ++++ sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/odroidx2_max98090.c | 191 ++++++++++++++++++++ 4 files changed, 236 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt create mode 100644 sound/soc/samsung/odroidx2_max98090.c -- 1.7.9.5 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH 1/1] ice1712: check if snd card loaded, if not do not proceed
by Konstantinos Tsimpoukas 14 Jul '14

14 Jul '14

Just adding a simple if statement, which verifies if actually card is loaded, if not does not continue with the execution of the snd_card_free. Signed-off-by: Konstantinos Tsimpoukas <kostaslinuxxx(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index d9b9e45..431096b 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2819,7 +2819,8 @@ static void snd_ice1712_remove(struct pci_dev *pci) if (ice->card_info && ice->card_info->chip_exit) ice->card_info->chip_exit(ice); - snd_card_free(card); + if (card) + snd_card_free(card); } #ifdef CONFIG_PM_SLEEP -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: Intel: Enable BDW/HSW SRAM power gating
by Jie Yang 14 Jul '14

14 Jul '14

Hi Mark, Here is the patch series for enable Intel Broadwell/Haswell ADSP SRAM power gating feature. Patches 0001~0003 implement FW module persistent area allocate during FW parse; Patch 0004 add dummy read to workaround SRAM write missing bytes issue; Patch 0005 will make SRAM start with all memory banks disabled, and enabled required banks during boot procedure. thanks, Keyon Jie Yang (5): ASoC: Intel: Add persistent area alloc ASoC: Intel: Merge wild cat point ADSP DRAM regions ASoC: Intel: Use a table for ADSP SRAM shift ASoC: Intel: Add dummy read for SRAM block enable ASoC: Intel: Start with all memory banks disabled sound/soc/intel/sst-dsp-priv.h | 2 ++ sound/soc/intel/sst-firmware.c | 52 ++++++++++++++++++++++++++++++ sound/soc/intel/sst-haswell-dsp.c | 68 ++++++++++++++++++++++++++++++--------- 3 files changed, 106 insertions(+), 16 deletions(-) -- 1.8.3.2

1 5

[alsa-devel] C-Media USB Audio Device (0d8c:000c) volume scale
by Alexander Schlarb 13 Jul '14

13 Jul '14

Hi everybody! I've just opened what I think is a bug at GitHub[1] and have been told to send it to this list instead. System configuration: - http://www.alsa-project.org/db/?f=fb914e7a6b8d91993b00d61c94c885e8d2ede527 - https://github.com/opensrc/alsa/issues/4 (some additional PulseAudio stuff) My problem: I have a DeLock USB audio device with an C-Media audio chip. When changing the volume in alsamixer there is no problem, but when changing the volume through any PulseAudio frontend (in my case either pavucontrol or veromix-plasmoid), all volumes below 30% in the GUI are shown as 0% in alsamixer and no sound is played. The PulseAudio wiki[2] blames this on ALSA for exporting incorrect dB information. However there is a StackExchange answer[3] which suggests adding the parameter ignore_dB=1 in /etc/pulse/default.pa which fixed the issue, so I'm not sure if this is indeed ALSAs fault? Please tell me if this can be fixed in ALSA or if this might be a PulseAudio bug... With best regards, Alexander Schlarb [1] https://github.com/opensrc/alsa/issues/4 [2] http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Puls… [3] http://askubuntu.com/questions/15069/how-do-i-change-the-way-ubuntu-adjusts…

2 1

[alsa-devel] [PATCH] ASoC: rockchips: fix build error
by Arnd Bergmann 12 Jul '14

12 Jul '14

The newly added rockchip_i2s driver is lacking an include of linux/module.h, causing a build error in some cases: ../sound/soc/rockchip/rockchip_i2s.c:526:20: error: expected declaration specifiers or '...' before string constant MODULE_DESCRIPTION("ROCKCHIP IIS ASoC Interface"); Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jianqun Xu <xjq(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af1d8eb94f04..c3e53feba5b9 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -13,6 +13,7 @@ #include <linux/delay.h> #include <linux/of_gpio.h> #include <linux/clk.h> +#include <linux/module.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <sound/pcm_params.h> -- 1.8.3.2

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[alsa-devel] question about sound/soc/ux500/ux500_msp_dai.c
by Himangi Saraogi 11 Jul '14

11 Jul '14

Hi, The probe function calls prcmu_qos_add_requirement but on failure it does not call prcmu_qos_remove_requirement even though that is done in the remove function. The in-kernel definition of both functions just returns 0. There is a prototype that is in a header file, but it is controlled by an #ifdef for an option CONFIG_DBX500_PRCMU_QOS_POWER that does not seem to be supported by the kernel. I am not sure what the prcmu_qos_remove_requirement does. Should it be added in the probe function? Thanks Himangi

1 0

[alsa-devel] [PATCH] ASoC: Intel: Add Broadwell Machine support
by Jie Yang 11 Jul '14

11 Jul '14

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Add support for Broadwell based machines with SST DSP audio. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/Kconfig | 12 +++ sound/soc/intel/Makefile | 2 + sound/soc/intel/broadwell.c | 251 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/broadwell.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c30fedb..0b305f9 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH help This adds audio driver for Intel Baytrail platform based boards with the MAX98090 audio codec. + +config SND_SOC_INTEL_BROADWELL_MACH + tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_HASWELL + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT286 + help + This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell + Ultrabook platforms. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4bfca79..7acbfc4 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c new file mode 100644 index 0000000..0e550f1 --- /dev/null +++ b/sound/soc/intel/broadwell.c @@ -0,0 +1,251 @@ +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt286.h" + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphones", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + /* always connected - check HP for jack detect */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "DMIC1"); + snd_soc_dapm_enable_pin(dapm, "DMIC2"); + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = broadwell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Capture PCM", + .stream_name = "Capture", + .cpu_dai_name = "Capture Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .name = "broadwell-rt286", + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + broadwell_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&broadwell_rt286); + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); -- 1.8.3.2

2 1

[alsa-devel] [PATCH 0/2] add rockchip i2s driver
by jianqun 11 Jul '14

11 Jul '14

From: Jianqun Xu <xjq(a)rock-chips.com> This patch is to add driver for I2S controller in RK3xxx SoCs. The only one change in this patch against which has been reviewed before is to fix randconfig build error with next-20140710 by add "#include <linux/module.h>". Jianqun Xu (2): ASoC: dt-bindings: add rockchip i2s bindings changes since v2: - change the description of dma-names, adviced by Mark - change the description of interrupts, adviced by Mark - list clock-names to be easy to read, adviced by Mark - modify compatible for rk3288 example, adviced by Mark and Heiko changes since v1: - modify the description of clock-names property, adviced by Mark Rutland. - modify the example to fit for rk3288. ASoC: add driver for Rockchip RK3xxx I2S controller changes since v3: - fix randconfig build error with next-20140710 by add "#include <linux/module.h>" tested with config given by robot test. changes since v2: - replace 4 with DMA_SLAVE_BUSWIDTH_4_BYTES - modify supported rate to SNDRV_PCM_RATE_8000_192000 - rename head file to "rockchi_i2s.h" changes since v1: - proper indentation for function arguments, adviced by Varka Bhadram - replace sizeof(struct rk_i2s_dev) with sizeof(*i2s), adviced by Mark Rutland - not to include head files which are unused actually in driver, adviced by Mark Brown - modify i2s_tx_status/i2s_rx_status to more meaningful with tx_start/rx_start, adviced by Mark Brown - use regmap_update_bits to reduce the amount of time spent locked, adviced by Mark Brown - add warning for possible error while waiting for the hardware to be ready, adviced by Mark Brown - error checking for clk_set_rate, adviced by Mark Brown - error checking for i2s_runtime_resume, adviced by Mark Brown - remove set_clkdiv operation, clock setting will do it, adviced by Mark Brown - no need SND_SOC for SND_SOC_ROCKCHIP, also allow the driver to be select when COMPILE_TEST is selected, adviced by Lars-Peter Clausen - not to include head files which are unused actually in driver, adviced by Lars-Peter Clausen - I2S core just use one snd_soc_dai_driver struct, adviced by Lars-Peter Clausen - delete rockchip_pcm.c and related codes, use generic dmaengine API, adviced by Lars-Peter and Mark Brown - a driver should never attempt to change its own device name, just use the name that the kernel set, adviced by Lars-Peter and Mark Brown .../devicetree/bindings/sound/rockchip-i2s.txt | 37 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/rockchip/Kconfig | 12 + sound/soc/rockchip/Makefile | 4 + sound/soc/rockchip/rockchip_i2s.c | 531 ++++++++++++++++++++ sound/soc/rockchip/rockchip_i2s.h | 223 ++++++++ 7 files changed, 809 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 sound/soc/rockchip/Kconfig create mode 100644 sound/soc/rockchip/Makefile create mode 100644 sound/soc/rockchip/rockchip_i2s.c create mode 100644 sound/soc/rockchip/rockchip_i2s.h -- 1.7.9.5

2 3

[alsa-devel] wm8974 and sam9g45 atmel
by Michele Da Rold 11 Jul '14

11 Jul '14

Hello everybody, I'm starting a new development with a Atmel SAM9G45 and a WM8974. I'm using kernel 2.6.33.2 I have connected the SSC1 pin TD,TF,TK and the RD pin, so I want to use WM8974 like a slave. I wrote a interface driver based on sam9263_wm8731.c The driver seem to load ok, I recognize the codec but when I start a mp3 playing (I use mpg123) the playback not start. Putting some dedug messagge the driver not call the function "set_bias_level", and call the mute function with mute=0. Testing the MCLK signal I not see any clock on pin because I never call clk_enable!! There is a way to identify the problem? Someone could help me? Thank you in advance Michele Da Rold I put the code of driver (don't care about printk I used it for debug purpose!!) /* * sam9g45_wm8974 -- SoC audio for AT91SAM9G20-based * ATMEL AT91SAM9G20ek board. * * Copyright (C) 2005 SAN People * Copyright (C) 2008 Atmel * * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> * * Based on sam9263_wm8731.c. * Frank Mandarino <fmandarino(a)endrelia.com> * Copyright 2006 Endrelia Technologies Inc. * Based on corgi.c by: * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/clk.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/i2c.h> #include <linux/atmel-ssc.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include "../codecs/wm8974.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" #define MCLK_RATE 12000000 /* * As shipped the board does not have inputs. However, it is relatively * straightforward to modify the board to hook them up so support is left * in the driver. */ #undef ENABLE_MIC_INPUT #define ENABLE_MIC_INPUT 1 static struct clk *mclk; static int at91sam9g45ek_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_hw_params() called\n"); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; return 0; } static struct snd_soc_ops at91sam9g45ek_ops = { .hw_params = at91sam9g45ek_hw_params, }; static int at91sam9g45ek_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { static int mclk_on; int ret = 0; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_set_bias_level() called %d\n",level); switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: if (!mclk_on) ret = clk_enable(mclk); if (ret == 0) mclk_on = 1; break; case SND_SOC_BIAS_OFF: case SND_SOC_BIAS_STANDBY: if (mclk_on) clk_disable(mclk); mclk_on = 0; break; } return ret; } #if 0 static const struct snd_soc_dapm_widget at91sam9g45ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* Speaker output mixer */ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, /* speaker connected to SPKOUTP */ {"SpkN Out", NULL, "Speaker Mixer"}, {"SpkP Out", NULL, "Speaker Mixer"}, {"SPKOUTN", NULL, "SpkN Out"}, {"SPKOUTP", NULL, "SpkP Out"}, /* Input PGA */ {"Input PGA", "Aux Switch", "Aux Input"}, {"Input PGA", "MicN Switch", "MICN"}, {"Input PGA", "MicP Switch", "MICP"}, }; /* * Logic for a wm974 as connected on a at91sam9g45ek board. */ static int at91sam9g45ek_wm8974_init(struct snd_soc_codec *codec) { struct snd_soc_dai *codec_dai = &codec->dai[0]; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45_wm8974_init() called\n"); /* ret = snd_soc_dai_set_sysclk(codec_dai, 0, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8974 SYSCLK: %d\n", ret); return ret; } */ /* Add specific widgets */ snd_soc_dapm_new_controls(codec, at91sam9g45ek_dapm_widgets, ARRAY_SIZE(at91sam9g45ek_dapm_widgets)); /* Set up specific audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_nc_pin(codec, "MONOOUT"); snd_soc_dapm_nc_pin(codec, "AUX"); #ifdef ENABLE_MIC_INPUT snd_soc_dapm_enable_pin(codec, "Int Mic"); #else snd_soc_dapm_nc_pin(codec, "Int Mic"); #endif /* always connected */ snd_soc_dapm_enable_pin(codec, "Ext Spk"); snd_soc_dapm_sync(codec); return 0; } #endif static struct snd_soc_dai_link at91sam9g45ek_dai = { .name = "WM8974", .stream_name = "WM8974 PCM", .cpu_dai = &atmel_ssc_dai[1], .codec_dai = &wm8974_dai, #if 0 .init = at91sam9g45ek_wm8974_init, #endif .ops = &at91sam9g45ek_ops, }; static struct snd_soc_card snd_soc_at91sam9g45ek = { .name = "AT91SAMG45", .platform = &atmel_soc_platform, .dai_link = &at91sam9g45ek_dai, .num_links = 1, .set_bias_level = at91sam9g45ek_set_bias_level, }; static struct snd_soc_device at91sam9g45ek_snd_devdata = { .card = &snd_soc_at91sam9g45ek, .codec_dev = &soc_codec_dev_wm8974, }; static struct platform_device *at91sam9g45ek_snd_device; static int __init at91sam9g45ek_init(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; struct clk *main; int ret; at91_set_A_periph(AT91_PIN_PD27, 0); /* * Codec MCLK is supplied by PCK1 - set it up. */ mclk = clk_get(NULL, "pck1"); if (IS_ERR(mclk)) { printk(KERN_ERR "ASoC: Failed to get MCLK\n"); ret = PTR_ERR(mclk); goto err; } main = clk_get(NULL, "main"); if (IS_ERR(main)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk; } ret = clk_set_parent(mclk, main); clk_put(main); if (ret != 0) { printk(KERN_ERR "ASoC: Failed to set MCLK parent\n"); goto err_mclk; } clk_set_rate(mclk, MCLK_RATE); /* * Request SSC device */ ssc = ssc_request(1); if (IS_ERR(ssc)) { printk(KERN_ERR "ASoC: Failed to request SSC 1\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; } ssc_p->ssc = ssc; at91sam9g45ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g45ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } platform_set_drvdata(at91sam9g45ek_snd_device, &at91sam9g45ek_snd_devdata); at91sam9g45ek_snd_devdata.dev = &at91sam9g45ek_snd_device->dev; ret = platform_device_add(at91sam9g45ek_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(at91sam9g45ek_snd_device); } return ret; err_ssc: ssc_free(ssc); ssc_p->ssc = NULL; err_mclk: clk_put(mclk); mclk = NULL; err: return ret; } static void __exit at91sam9g45ek_exit(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc; if (ssc_p != NULL) { ssc = ssc_p->ssc; if (ssc != NULL) ssc_free(ssc); ssc_p->ssc = NULL; } platform_device_unregister(at91sam9g45ek_snd_device); at91sam9g45ek_snd_device = NULL; clk_put(mclk); mclk = NULL; } module_init(at91sam9g45ek_init); module_exit(at91sam9g45ek_exit); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); MODULE_DESCRIPTION("ALSA SoC AT91SAM9G45_WM8974"); MODULE_LICENSE("GPL");

3 8

[alsa-devel] [PATCH] alsa-lib: Provide a CLOCK_MONOTONIC_RAW timestamp type
by Mark Brown 10 Jul '14

10 Jul '14

From: Mark Brown <broonie(a)linaro.org> For applications which need to synchronise with external timebases such as broadcast TV applications the kernel monotonic time is not optimal as it includes adjustments from NTP and so may still include discontinuities due to that. A raw monotonic time which does not include any adjustments is available in the kernel from getrawmonotonic() so provide userspace with a new timestamp type SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW which provides timestamps based on this as an option. Reported-by: Daniel Thompson <daniel.thompson(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/asound.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 1774a5c..9061cdd 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -457,7 +457,8 @@ struct snd_xfern { enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; /* channel positions */ -- 2.0.0

4 6

[alsa-devel] [PATCH alsa-lib 0/5] Add monotonic raw timestamp support
by Takashi Iwai 10 Jul '14

10 Jul '14

Hi, this is a series of patches to add the monotonic raw timestamp support to alsa-lib. A couple of new API functions and the new type are introduced while the old hw_params monotonic function still remains. Maybe we should mark it depecated? Takashi

2 7

[alsa-devel] [PATCH RESEND v7 0/3] Add ES8328 audio codec
by Sean Cross 10 Jul '14

10 Jul '14

This patchset adds support for the Everest Semi ES8328 audio codec, used in the Novena open source laptop. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks, in addition to the i.MX audmux routing. Changes since v6: - Minor clarification of AUDMUX in imx-audio-es8328.txt - Use u32 instead of int in imx-es8328.c for int-mux and ext-mux - Bounds checking on mux-int-port and mux-ext-port in imx-es8328.c - Remove NULL checks before calling of_node_put() Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 61 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 719 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 233 +++++++ 12 files changed, 1510 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

3 5

[alsa-devel] [PATCH] ASoC: rt286: make rt286_i2c_driver static
by bardliao＠realtek.com 10 Jul '14

10 Jul '14

From: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- This is reported by kbuild test robot. I don't know if my sob line is needed or not. --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8103315..218f86e 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static int rt286_i2c_remove(struct i2c_client *i2c) } -struct i2c_driver rt286_i2c_driver = { +static struct i2c_driver rt286_i2c_driver = { .driver = { .name = "rt286", .owner = THIS_MODULE, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: omap-dmic: use managed interfaces
by Himangi Saraogi 10 Jul '14

10 Jul '14

This patch moves data allocated using clk_put and snd_soc_register_component to the corresponding managed interfaces and does away with the functions to free memory in the probe and remove functions. The probe function is no longer required and is completely removed. Signed-off-by: Himangi Saraogi <himangi774(a)gmail.com> Acked-by: Julia Lawall <julia.lawall(a)lip6.fr> --- sound/soc/omap/omap-dmic.c | 35 ++++++++--------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 6925d71..b315b27 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -466,7 +466,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) mutex_init(&dmic->mutex); - dmic->fclk = clk_get(dmic->dev, "fck"); + dmic->fclk = devm_clk_get(dmic->dev, "fck"); if (IS_ERR(dmic->fclk)) { dev_err(dmic->dev, "cant get fck\n"); return -ENODEV; @@ -475,8 +475,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); if (!res) { dev_err(dmic->dev, "invalid dma memory resource\n"); - ret = -ENODEV; - goto err_put_clk; + return -ENODEV; } dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG; @@ -484,34 +483,17 @@ static int asoc_dmic_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); dmic->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(dmic->io_base)) { - ret = PTR_ERR(dmic->io_base); - goto err_put_clk; - } - + if (IS_ERR(dmic->io_base)) + return PTR_ERR(dmic->io_base); - ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, - &omap_dmic_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &omap_dmic_component, + &omap_dmic_dai, 1); if (ret) - goto err_put_clk; + return ret; ret = omap_pcm_platform_register(&pdev->dev); if (ret) - goto err_put_clk; - - return 0; - -err_put_clk: - clk_put(dmic->fclk); - return ret; -} - -static int asoc_dmic_remove(struct platform_device *pdev) -{ - struct omap_dmic *dmic = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - clk_put(dmic->fclk); + return ret; return 0; } @@ -529,7 +511,6 @@ static struct platform_driver asoc_dmic_driver = { .of_match_table = omap_dmic_of_match, }, .probe = asoc_dmic_probe, - .remove = asoc_dmic_remove, }; module_platform_driver(asoc_dmic_driver); -- 1.9.1

3 2

[alsa-devel] [PATCH 0/7] Arizona FLL Bug Fixes
by Charles Keepax 10 Jul '14

10 Jul '14

Hi, This patch series contains several small bug fixes and improves the handling of the situation in which an already running FLL is reconfigured. In the case of a running FLL only some tranisitions are supported whereas previously all transitions were allowed. Thanks, Charles Charles Keepax (7): ASoC: arizona: Do not test ratio zero as it is not a valid setting ASoC: arizona: Correct checking of FLL ratio limitations ASoC: arizona: Correct relationship between VCO corner and Fref ASoC: arizona: Coding standards, remove unneeded brackets ASoC: arizona: Correct return value of arizona_is_enabled_fll ASoC: arizona: FLL freerun only required whilst disabling ASoC: arizona: Update handling for input change on an active FLL sound/soc/codecs/arizona.c | 63 ++++++++++++++++++++++++++++++-------------- 1 files changed, 43 insertions(+), 20 deletions(-) -- 1.7.2.5

2 8

[alsa-devel] [PATCH - alsa-utils 0/5] The amidicat program as a patch
by Josh Lehan 10 Jul '14

10 Jul '14

This is my "amidicat" program, which makes it easy and straightforward to interact with the ALSA MIDI sequencer from the command line, by simply using standard input and output. It took just over a year, to find the time to work on it again, but here it is, formatted as a patch to alsa-utils, as requested. http://mailman.alsa-project.org/pipermail/alsa-devel/2013-May/062353.html It has been cleaned up to match "checkpatch" coding standards, and all compiler warnings have been fixed. The documentation has also been cleaned up, in particular, the Perl examples have been brought up to date: the raw bytes no longer cause errors on Unicode systems. I respectfully submit this program to alsa-utils, and hope that it makes it in there :) Josh Lehan Josh Lehan (5): Adding amidicat subdirectory to SUBDIRS Add amidicat Makefile to AC_OUTPUT list Makefile for amidicat, includes manual page Documentation manual page for amidicat The amidicat program itself, better late than never Makefile.am | 2 +- amidicat/Makefile.am | 5 + amidicat/amidicat.1 | 392 ++++++++++++ amidicat/amidicat.c | 1708 ++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 2 +- 5 files changed, 2107 insertions(+), 2 deletions(-) create mode 100644 amidicat/Makefile.am create mode 100644 amidicat/amidicat.1 create mode 100644 amidicat/amidicat.c -- 1.8.3.2

3 16

[alsa-devel] [PATCH] ASoC: wm-adsp: Add missing MODULE_LICENSE
by Vinod Koul 09 Jul '14

09 Jul '14

From: Praveen Diwakar <praveen.diwakar(a)intel.com> Since MODULE_LICENSE is missing the module load fails, so add this for module. Signed-off-by: Praveen Diwakar <praveen.diwakar(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0600271..2537725 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1758,3 +1758,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_init); + +MODULE_LICENSE("GPL v2"); -- 1.7.0.4

3 4

[alsa-devel] Split a 7.1 device into 4 stereo devices
by Xavier Claessens 09 Jul '14

09 Jul '14

Hello, I have a 7.1 sound device and speaker-test -c8 can output on each speaker. But what I would like to do is using it as 4 stereo devices so I could play different sounds on each of them. Ideally I would like to avoid dmix and friends to be able to have gapless and bit-perfect output. On IRC "debianuser" said it is possible with hdajackretask to make it appear as 4 different hw devices in "aplay -l" but we didn't succeed doing that. Anyone knows what configuration is needed for that? Here is my setting: http://www.alsa-project.org/db/?f=bf0744873d5e6d42d3adc0076c5f66189e9191b7 Regards, Xavier Claessens.

3 2

[alsa-devel] [PATCH V2] ASoC: omap-hdmi: Use devm_snd_soc_register_component
by Manish Badarkhe 09 Jul '14

09 Jul '14

Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe <badarkhe.manish(a)gmail.com> --- Changes since V1: Aligned function parameter properly. :100644 100644 eb9c392... 8dcd7f7... M sound/soc/omap/omap-hdmi.c sound/soc/omap/omap-hdmi.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index eb9c392..8dcd7f7 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -322,8 +322,9 @@ static int omap_hdmi_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, hdmi_data); - ret = snd_soc_register_component(&pdev->dev, &omap_hdmi_component, - &omap_hdmi_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &omap_hdmi_component, + &omap_hdmi_dai, 1); if (ret) return ret; @@ -335,8 +336,6 @@ static int omap_hdmi_remove(struct platform_device *pdev) { struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - if (hdmi_data == NULL) { dev_err(&pdev->dev, "cannot obtain HDMi data\n"); return -ENODEV; -- 1.7.10.4

2 2

Re: [alsa-devel] [PATCH] ALSA: Provide a CLOCK_MONOTONIC_RAW timestamp type
by Takashi Iwai 09 Jul '14

09 Jul '14

At Tue, 8 Jul 2014 16:51:49 +0200, Mark Brown wrote: > > From: Mark Brown <broonie(a)linaro.org> > > For applications which need to synchronise with external timebases such > as broadcast TV applications the kernel monotonic time is not optimal as > it includes adjustments from NTP and so may still include discontinuities > due to that. A raw monotonic time which does not include any adjustments > is available in the kernel from getrawmonotonic() so provide userspace with > a new timestamp type SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW which provides > timestamps based on this as an option. > > Reported-by: Daniel Thompson <daniel.thompson(a)linaro.org> > Signed-off-by: Mark Brown <broonie(a)linaro.org> > --- > > Mostly an RFC at this point as I don't have hardware to test this week. This looks good to me. Or, thinking of the original use case, should we just replace ktime_get_ts() with getrawmonotonic()? The MONOTONIC mode is really to be monotonic. While we're at it: there's been long-standing request from V4L guys for improving the timestamps to sync with video streams. I suggested to add a new mode that can be set by the driver. Worth to consider it, too? thanks, Takashi > > include/sound/pcm.h | 11 +++++++++-- > include/uapi/sound/asound.h | 3 ++- > sound/core/pcm_native.c | 12 +++++++++--- > 3 files changed, 20 insertions(+), 6 deletions(-) > > diff --git a/include/sound/pcm.h b/include/sound/pcm.h > index d854fb3..6f3e10c 100644 > --- a/include/sound/pcm.h > +++ b/include/sound/pcm.h > @@ -931,10 +931,17 @@ void snd_pcm_timer_done(struct snd_pcm_substream *substream); > static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime, > struct timespec *tv) > { > - if (runtime->tstamp_type == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC) > + switch (runtime->tstamp_type) { > + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC: > ktime_get_ts(tv); > - else > + break; > + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: > + getrawmonotonic(tv); > + break; > + default: > getnstimeofday(tv); > + break; > + } > } > > /* > diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h > index 2249483..cbf7dc85 100644 > --- a/include/uapi/sound/asound.h > +++ b/include/uapi/sound/asound.h > @@ -462,7 +462,8 @@ struct snd_xfern { > enum { > SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ > SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ > - SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, > + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ > + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, > }; > > /* channel positions */ > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index b653ab0..fd6d7a7 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -2540,9 +2540,15 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) > return -EFAULT; > if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST) > return -EINVAL; > - runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY; > - if (arg == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC) > - runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC; > + switch (runtime->tstamp_type) { > + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC: > + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: > + runtime->tstamp_type = runtime->tstamp_type; > + break; > + default: > + runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY; > + break; > + } > return 0; > } > > -- > 2.0.0 >

4 4

[alsa-devel] [PATCH] ASoC: s3c64xx/smartq: use dynamic registration
by Arnd Bergmann 09 Jul '14

09 Jul '14

As a prerequisite for moving s3c64xx into multiplatform configurations, we need to change the smartq audio driver to stop using hardcoded gpio numbers from the header file, and instead pass the gpio data through platform_data. In order to do that, we also move the code to use module_platform_driver and register the platform device using platform_device_register_data. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Maurus Cuelenaere <mcuelenaere(a)gmail.com> Cc: Kukjin Kim <kgene.kim(a)samsung.com> --- arch/arm/mach-s3c64xx/mach-smartq.c | 10 +++++ sound/soc/samsung/smartq_wm8987.c | 73 ++++++++++++++----------------------- 2 files changed, 37 insertions(+), 46 deletions(-) diff --git a/arch/arm/mach-s3c64xx/mach-smartq.c b/arch/arm/mach-s3c64xx/mach-smartq.c index b3d1353..4c111f6 100644 --- a/arch/arm/mach-s3c64xx/mach-smartq.c +++ b/arch/arm/mach-s3c64xx/mach-smartq.c @@ -20,6 +20,7 @@ #include <linux/spi/spi_gpio.h> #include <linux/usb/gpio_vbus.h> #include <linux/platform_data/s3c-hsotg.h> +#include <linux/platform_data/asoc-s3c-smartq.h> #include <asm/mach-types.h> #include <asm/mach/map.h> @@ -379,6 +380,11 @@ void __init smartq_map_io(void) smartq_lcd_mode_set(); } +static const __initconst struct smartq_audio_pdata smartq_audio_pdata = { + .gpio_jack = S3C64XX_GPL(12), + .gpio_amp = S3C64XX_GPK(12), +}; + void __init smartq_machine_init(void) { s3c_i2c0_set_platdata(NULL); @@ -397,4 +403,8 @@ void __init smartq_machine_init(void) WARN_ON(smartq_wifi_init()); platform_add_devices(smartq_devices, ARRAY_SIZE(smartq_devices)); + + platform_device_register_data(NULL, "smartq-audio", -1, + &smartq_audio_pdata, + sizeof (smartq_audio_pdata)); } diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 9b0ffac..8c4a0a2 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -19,8 +19,7 @@ #include <sound/soc.h> #include <sound/jack.h> -#include <mach/gpio-samsung.h> -#include <asm/mach-types.h> +#include <linux/platform_data/asoc-s3c-smartq.h> #include "i2s.h" #include "../codecs/wm8750.h" @@ -110,7 +109,7 @@ static struct snd_soc_jack_pin smartq_jack_pins[] = { static struct snd_soc_jack_gpio smartq_jack_gpios[] = { { - .gpio = S3C64XX_GPL(12), + .gpio = -1, .name = "headphone detect", .report = SND_JACK_HEADPHONE, .debounce_time = 200, @@ -127,7 +126,10 @@ static int smartq_speaker_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event)); + struct smartq_audio_pdata *pdata; + pdata = snd_soc_smartq.dev->platform_data; + + gpio_set_value(pdata->gpio_amp, SND_SOC_DAPM_EVENT_OFF(event)); return 0; } @@ -218,62 +220,41 @@ static struct snd_soc_card snd_soc_smartq = { .num_controls = ARRAY_SIZE(wm8987_smartq_controls), }; -static struct platform_device *smartq_snd_device; - -static int __init smartq_init(void) +static int smartq_probe(struct platform_device *pdev) { + struct smartq_audio_pdata *pdata = pdev->dev.platform_data; int ret; - if (!machine_is_smartq7() && !machine_is_smartq5()) { - pr_info("Only SmartQ is supported by this ASoC driver\n"); - return -ENODEV; - } - - smartq_snd_device = platform_device_alloc("soc-audio", -1); - if (!smartq_snd_device) - return -ENOMEM; - - platform_set_drvdata(smartq_snd_device, &snd_soc_smartq); - - ret = platform_device_add(smartq_snd_device); - if (ret) { - platform_device_put(smartq_snd_device); - return ret; - } + platform_set_drvdata(pdev, &snd_soc_smartq); /* Initialise GPIOs used by amplifiers */ - ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown"); + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_amp, + GPIOF_DIR_OUT | GPIOF_INIT_HIGH, + "amplifiers shutdown"); if (ret) { - dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n"); - goto err_unregister_device; + dev_err(&pdev->dev, "Failed to register GPK12\n"); + goto out; } - /* Disable amplifiers */ - ret = gpio_direction_output(S3C64XX_GPK(12), 1); - if (ret) { - dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n"); - goto err_free_gpio_amp_shut; - } + smartq_jack_gpios[0].gpio = pdata->gpio_jack; - return 0; - -err_free_gpio_amp_shut: - gpio_free(S3C64XX_GPK(12)); -err_unregister_device: - platform_device_unregister(smartq_snd_device); + ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_smartq); + if (ret) + dev_err(&pdev->dev, "Failed to register card\n"); +out: return ret; } -static void __exit smartq_exit(void) -{ - gpio_free(S3C64XX_GPK(12)); - - platform_device_unregister(smartq_snd_device); -} +static struct platform_driver smartq_driver = { + .driver = { + .name = "smartq-audio", + .owner = THIS_MODULE, + }, + .probe = smartq_probe, +}; -module_init(smartq_init); -module_exit(smartq_exit); +module_platform_driver(smartq_driver); /* Module information */ MODULE_AUTHOR("Maurus Cuelenaere <mcuelenaere(a)gmail.com>"); -- 1.8.3.2

2 1

[alsa-devel] [PATCH V2] ASoC: omap-dmic: Use devm_snd_soc_register_component
by Manish Badarkhe 09 Jul '14

09 Jul '14

Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe <badarkhe.manish(a)gmail.com> --- Changes since V1: Aligned function parameter properly. :100644 100644 6925d71... f125eb9... M sound/soc/omap/omap-dmic.c sound/soc/omap/omap-dmic.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 6925d71..f125eb9 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -490,8 +490,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) } - ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, - &omap_dmic_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &omap_dmic_component, + &omap_dmic_dai, 1); if (ret) goto err_put_clk; @@ -510,7 +511,6 @@ static int asoc_dmic_remove(struct platform_device *pdev) { struct omap_dmic *dmic = platform_get_drvdata(pdev); - snd_soc_unregister_component(&pdev->dev); clk_put(dmic->fclk); return 0; -- 1.7.10.4

3 2

[alsa-devel] [PATCH V2] ASoC: omap-mcbsp: Use devm_snd_soc_register_component
by Manish Badarkhe 09 Jul '14

09 Jul '14

Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe <badarkhe.manish(a)gmail.com> --- Changes since V1: Aligned function parameter properly. :100644 100644 efe2cd6... bd3ef2a... M sound/soc/omap/omap-mcbsp.c sound/soc/omap/omap-mcbsp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index efe2cd6..bd3ef2a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -805,8 +805,9 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, - &omap_mcbsp_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &omap_mcbsp_component, + &omap_mcbsp_dai, 1); if (ret) return ret; @@ -817,8 +818,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) { struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - snd_soc_unregister_component(&pdev->dev); - if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix driver unbound
by Fabio Estevam 09 Jul '14

09 Jul '14

Using the sgtl5000 codec driver as a module and trying to remove it causes the followig kernel oops: root@freescale /$ rmmod snd-soc-imx-sgtl5000 [ 117.122920] ------------[ cut here ]------------ [ 117.127609] WARNING: CPU: 0 PID: 631 at drivers/regulator/core.c:3604 regula) [ 117.137046] Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 evbug [ 117.144315] CPU: 0 PID: 631 Comm: rmmod Not tainted 3.16.0-rc3-next-201407043 [ 117.153366] Backtrace: [ 117.155865] [<80011e5c>] (dump_backtrace) from [<80011ff8>] (show_stack+0x18) [ 117.163484] r6:802fcc48 r5:00000000 r4:00000000 r3:00000000 [ 117.169228] [<80011fe0>] (show_stack) from [<80668cc0>] (dump_stack+0x88/0xa) [ 117.176508] [<80668c38>] (dump_stack) from [<80029a38>] (warn_slowpath_commo) [ 117.184696] r5:00000009 r4:00000000 [ 117.188322] [<800299c8>] (warn_slowpath_common) from [<80029a80>] (warn_slow) [ 117.197150] r8:dd60d600 r7:ddfa6d00 r6:dd5d9064 r5:dd5e0f90 r4:dd5d9400 [ 117.203983] [<80029a5c>] (warn_slowpath_null) from [<802fcc48>] (regulator_u) [ 117.212828] [<802fcb74>] (regulator_unregister) from [<7f0047c4>] (ldo_regul) [ 117.223475] r4:dd59e300 r3:dd5e0f90 [ 117.227100] [<7f00479c>] (ldo_regulator_remove [snd_soc_sgtl5000]) from [<7f) [ 117.238959] r4:dd5d8000 r3:ddd51420 [ 117.242623] [<7f0047dc>] (sgtl5000_remove [snd_soc_sgtl5000]) from [<804e5b1) [ 117.252489] r5:00000000 r4:dd5d8000 [ 117.256111] [<804e5af8>] (soc_remove_codec) from [<804e5ed4>] (soc_remove_da) [ 117.264933] r4:ddfb640c r3:00000000 [ 117.268555] [<804e5c10>] (soc_remove_dai_links) from [<804e5fbc>] (snd_soc_u) [ 117.277810] r10:80359e48 r9:dd684000 r8:dd5ca800 r7:dd685e60 r6:00000001 r5c [ 117.285761] r4:dd5d9064 [ 117.288329] [<804e5f28>] (snd_soc_unregister_card) from [<804f29f0>] (devm_c) [ 117.297324] r6:00000004 r5:ddd0fc10 r4:dd5a7980 r3:804f29dc [ 117.303095] [<804f29dc>] (devm_card_release) from [<8035a418>] (release_node) [ 117.311369] [<8035a2a8>] (release_nodes) from [<8035aab4>] (devres_release_a) [ 117.319583] r10:00000000 r9:dd684000 r8:8000ed64 r7:00000081 r6:ddd0fc44 r54 [ 117.327543] r4:ddd0fc10 [ 117.330108] [<8035aa7c>] (devres_release_all) from [<803571c8>] (__device_re) [ 117.339214] r4:ddd0fc10 r3:dd5d9010 [ 117.342989] [<80357148>] (__device_release_driver) from [<80357a48>] (driver) [ 117.351607] r5:7f00c9e4 r4:ddd0fc10 [ 117.355285] [<8035798c>] (driver_detach) from [<80357030>] (bus_remove_drive) [ 117.363454] r6:00000880 r5:00000000 r4:7f00c9e4 r3:dd6c3a80 [ 117.369195] [<80356fdc>] (bus_remove_driver) from [<803580b8>] (driver_unreg) [ 117.377683] r4:7f00c9e4 r3:dd60d480 [ 117.381305] [<80358088>] (driver_unregister) from [<80358bb4>] (platform_dri) [ 117.390642] r4:7f00ca28 r3:7f00c35c [ 117.394309] [<80358ba0>] (platform_driver_unregister) from [<7f00c370>] (imx) [ 117.406188] [<7f00c35c>] (imx_sgtl5000_driver_exit [snd_soc_imx_sgtl5000]) f) [ 117.417452] [<8008c2b8>] (SyS_delete_module) from [<8000eba0>] (ret_fast_sys) [ 117.425753] r6:5f636f73 r5:5f646e73 r4:00016f40 [ 117.430428] ---[ end trace 8fd8a5cb39e46d0e ]--- This problem is well explained by Russell King: "The sgtl5000 uses a two-stage initialisation process. The first stage is when the platform driver is probed, where some resources are found and initialised. The second stage is via the codec driver's probe function, where regulators are found and initialised using the managed resource support. The problem here is that this works fine until the codec driver is unbound. When this occurs, sgtl5000_remove() is called which is a no-op as far as the managed resource code is concerned. The regulators remain allocated, and their pointers in sgtl5000_priv remain valid. If the codec is now re-probed, it will again try and find the regulators, which will now be busy. This will fail. That's not the only problem - if using the LDO regulator, the regulator is unregistered from the regulator core code, but it still has a user. When the user is cleaned up (eg, by removing the module) it hits the free'd regulator, and this can oops the kernel. This bug was originally introduced by 63e54cd9caa3ce ("ASoC: sgtl5000: Use devm_regulator_bulk_get()")." This reverts commit 63e54cd9caa3ce. Tested on a imx53-qsb board. Reported-by: Russell King <rmk+kernel(a)arm.linux.org.uk> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 249fadb..d739f83 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1275,7 +1275,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return ret; } - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) goto err_ldo_remove; @@ -1283,13 +1283,16 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_ldo_remove; + goto err_regulator_free; /* wait for all power rails bring up */ udelay(10); return 0; +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); err_ldo_remove: if (!external_vddd) ldo_regulator_remove(codec); @@ -1359,6 +1362,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return ret; @@ -1372,6 +1377,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return 0; -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Use devm_ functions
by Himangi Saraogi 09 Jul '14

09 Jul '14

This patch introduces the use of managed interfaces like devm_kzalloc, devm_kstrdup and devm_regulator_register and does avay with the calls to the functions to free the allocated memory in ldo_regulator_register and ldo_regulator_remove. The ldo_regulator_remove function is completely removed as it is no longer required. ldo_regulator_register is called from a probe function and on failure its value is returned as the result. Signed-off-by: Himangi Saraogi <himangi774(a)gmail.com> --- To send to: Liam Girdwood <lgirdwood(a)gmail.com>,Mark Brown <broonie(a)kernel.org>,Jaroslav Kysela <perex(a)perex.cz>,Takashi Iwai <tiwai(a)suse.de>,alsa-devel@alsa-project.org,linux-kernel@vger.kernel.org sound/soc/codecs/sgtl5000.c | 46 +++++++-------------------------------------- 1 file changed, 7 insertions(+), 39 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 249fadb..0efd6d6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -841,14 +841,15 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); struct regulator_config config = { }; - ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); + ldo = devm_kzalloc(codec->dev, sizeof(struct ldo_regulator), + GFP_KERNEL); if (!ldo) return -ENOMEM; - ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); + ldo->desc.name = devm_kstrdup(codec->dev, dev_name(codec->dev), + GFP_KERNEL); if (!ldo->desc.name) { - kfree(ldo); dev_err(codec->dev, "failed to allocate decs name memory\n"); return -ENOMEM; } @@ -865,35 +866,17 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, config.driver_data = ldo; config.init_data = init_data; - ldo->dev = regulator_register(&ldo->desc, &config); + ldo->dev = devm_regulator_register(codec->dev, &ldo->desc, &config); if (IS_ERR(ldo->dev)) { int ret = PTR_ERR(ldo->dev); dev_err(codec->dev, "failed to register regulator\n"); - kfree(ldo->desc.name); - kfree(ldo); - return ret; } sgtl5000->ldo = ldo; return 0; } - -static int ldo_regulator_remove(struct snd_soc_codec *codec) -{ - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - struct ldo_regulator *ldo = sgtl5000->ldo; - - if (!ldo) - return 0; - - regulator_unregister(ldo->dev); - kfree(ldo->desc.name); - kfree(ldo); - - return 0; -} #else static int ldo_regulator_register(struct snd_soc_codec *codec, struct regulator_init_data *init_data, @@ -902,11 +885,6 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, dev_err(codec->dev, "this setup needs regulator support in the kernel\n"); return -EINVAL; } - -static int ldo_regulator_remove(struct snd_soc_codec *codec) -{ - return 0; -} #endif /* @@ -1278,23 +1256,17 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_ldo_remove; + return ret; ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_ldo_remove; + return ret; /* wait for all power rails bring up */ udelay(10); return 0; - -err_ldo_remove: - if (!external_vddd) - ldo_regulator_remove(codec); - return ret; - } static int sgtl5000_probe(struct snd_soc_codec *codec) @@ -1359,8 +1331,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - ldo_regulator_remove(codec); - return ret; } @@ -1372,8 +1342,6 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - ldo_regulator_remove(codec); - return 0; } -- 1.9.1

5 11

[alsa-devel] [PATCH 1/2] ASoC: samsung: Update sound-card name for Snow
by Tushar Behera 09 Jul '14

09 Jul '14

Snow sound-card driver supports multiple boards with different audio codecs. Updating the sound card name per board basis would provide some more information to the end-user. Signed-off-by: Tushar Behera <tushar.b(a)samsung.com> --- Documentation/devicetree/bindings/sound/snow.txt | 4 ++++ sound/soc/samsung/snow.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index e0b7a82..6df74f1 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -8,11 +8,15 @@ Required properties: - samsung,i2s-controller: The phandle of the Samsung I2S controller - samsung,audio-codec: The phandle of the audio codec +Optional: +- samsung,model: The name of the sound-card + Example: sound { compatible = "google,snow-audio-max98095"; + samsung,model = "Snow-I2S-MAX98095"; samsung,i2s-controller = <&i2s0>; samsung,audio-codec = <&max98095>; }; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 8bbd348..0acf5d0 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -92,6 +92,9 @@ static int snow_probe(struct platform_device *pdev) card->dev = &pdev->dev; + /* Update card-name if provided through DT, else use default name */ + snd_soc_of_parse_card_name(card, "samsung,model"); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); -- 1.7.9.5

2 3

[alsa-devel] [PATCH 0/2] add rockchip i2s driver
by jianqun 09 Jul '14

09 Jul '14

From: Jianqun Xu <xjq(a)rock-chips.com> This patch is to add driver for I2S controller in RK3xxx SoCs. Jianqun Xu (2): ASoC: dt-bindings: add rockchip i2s bindings ASoC: add driver for Rockchip RK3xxx I2S controller .../devicetree/bindings/sound/rockchip-i2s.txt | 45 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/rockchip/Kconfig | 16 + sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/i2s.h | 222 +++++++ sound/soc/rockchip/pcm.h | 14 + sound/soc/rockchip/rockchip_i2s.c | 622 ++++++++++++++++++++ sound/soc/rockchip/rockchip_pcm.c | 64 ++ 9 files changed, 991 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 sound/soc/rockchip/Kconfig create mode 100644 sound/soc/rockchip/Makefile create mode 100644 sound/soc/rockchip/i2s.h create mode 100644 sound/soc/rockchip/pcm.h create mode 100644 sound/soc/rockchip/rockchip_i2s.c create mode 100644 sound/soc/rockchip/rockchip_pcm.c -- 1.7.9.5

5 17

[alsa-devel] [PATCH] ASoC: cs42l56: Move ADC/PCM Swap to DAPM
by Brian Austin 08 Jul '14

08 Jul '14

The Swap controls for ADC/PCM paths should be in the DAPM domain. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l56.c | 64 +++++++++++++++++++++++++++++++------------- 1 file changed, 45 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index fdc4bd2..e5ef223 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -318,24 +318,32 @@ static const struct soc_enum adca_swap_enum = ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); +static const struct snd_kcontrol_new adca_swap_mux = + SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 4, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); +static const struct snd_kcontrol_new pcma_swap_mux = + SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 2, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); +static const struct snd_kcontrol_new adcb_swap_mux = + SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 6, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); +static const struct snd_kcontrol_new pcmb_swap_mux = + SOC_DAPM_ENUM("Route", pcmb_swap_enum); static const struct snd_kcontrol_new hpa_switch = SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 6, 1, 1); @@ -467,11 +475,6 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_SINGLE("ADCA Invert", CS42L56_MISC_ADC_CTL, 2, 1, 1), SOC_SINGLE("ADCB Invert", CS42L56_MISC_ADC_CTL, 3, 1, 1), - SOC_ENUM("PCMA Swap", pcma_swap_enum), - SOC_ENUM("PCMB Swap", pcmb_swap_enum), - SOC_ENUM("ADCA Swap", adca_swap_enum), - SOC_ENUM("ADCB Swap", adcb_swap_enum), - SOC_DOUBLE("HPF Switch", CS42L56_HPF_CTL, 5, 7, 1, 1), SOC_DOUBLE("HPF Freeze Switch", CS42L56_HPF_CTL, 4, 6, 1, 1), SOC_ENUM("HPFA Corner Freq", hpfa_freq_enum), @@ -570,6 +573,16 @@ static const struct snd_soc_dapm_widget cs42l56_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCA", NULL, CS42L56_PWRCTL_1, 1, 1), SND_SOC_DAPM_ADC("ADCB", NULL, CS42L56_PWRCTL_1, 2, 1), + SND_SOC_DAPM_MUX("ADCA Swap Mux", SND_SOC_NOPM, 0, 0, + &adca_swap_mux), + SND_SOC_DAPM_MUX("ADCB Swap Mux", SND_SOC_NOPM, 0, 0, + &adcb_swap_mux), + + SND_SOC_DAPM_MUX("PCMA Swap Mux", SND_SOC_NOPM, 0, 0, + &pcma_swap_mux), + SND_SOC_DAPM_MUX("PCMB Swap Mux", SND_SOC_NOPM, 0, 0, + &pcmb_swap_mux), + SND_SOC_DAPM_DAC("DACA", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACB", NULL, SND_SOC_NOPM, 0, 0), @@ -607,8 +620,19 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = { {"Digital Output Mux", NULL, "ADCA"}, {"Digital Output Mux", NULL, "ADCB"}, - {"ADCB", NULL, "ADCB Mux"}, - {"ADCA", NULL, "ADCA Mux"}, + {"ADCB", NULL, "ADCB Swap Mux"}, + {"ADCA", NULL, "ADCA Swap Mux"}, + + {"ADCA Swap Mux", NULL, "ADCA"}, + {"ADCB Swap Mux", NULL, "ADCB"}, + + {"DACA", "Left", "ADCA Swap Mux"}, + {"DACA", "LR 2", "ADCA Swap Mux"}, + {"DACA", "Right", "ADCA Swap Mux"}, + + {"DACB", "Left", "ADCB Swap Mux"}, + {"DACB", "LR 2", "ADCB Swap Mux"}, + {"DACB", "Right", "ADCB Swap Mux"}, {"ADCA Mux", NULL, "AIN3A"}, {"ADCA Mux", NULL, "AIN2A"}, @@ -633,30 +657,32 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = { {"PGAB Input Mux", NULL, "AIN2B"}, {"PGAB Input Mux", NULL, "AIN3B"}, - {"LOB", NULL, "Lineout Right"}, - {"LOA", NULL, "Lineout Left"}, - - {"Lineout Right", "Switch", "LINEOUTB Input Mux"}, - {"Lineout Left", "Switch", "LINEOUTA Input Mux"}, + {"LOB", "Switch", "LINEOUTB Input Mux"}, + {"LOA", "Switch", "LINEOUTA Input Mux"}, {"LINEOUTA Input Mux", "PGAA", "PGAA"}, {"LINEOUTB Input Mux", "PGAB", "PGAB"}, {"LINEOUTA Input Mux", "DACA", "DACA"}, {"LINEOUTB Input Mux", "DACB", "DACB"}, - {"HPA", NULL, "Headphone Left"}, - {"HPB", NULL, "Headphone Right"}, - - {"Headphone Right", "Switch", "HPB Input Mux"}, - {"Headphone Left", "Switch", "HPA Input Mux"}, + {"HPA", "Switch", "HPB Input Mux"}, + {"HPB", "Switch", "HPA Input Mux"}, {"HPA Input Mux", "PGAA", "PGAA"}, {"HPB Input Mux", "PGAB", "PGAB"}, {"HPA Input Mux", "DACA", "DACA"}, {"HPB Input Mux", "DACB", "DACB"}, - {"DACB", NULL, "HiFi Playback"}, - {"DACA", NULL, "HiFi Playback"}, + {"DACA", NULL, "PCMA Swap Mux"}, + {"DACB", NULL, "PCMB Swap Mux"}, + + {"PCMB Swap Mux", "Left", "HiFi Playback"}, + {"PCMB Swap Mux", "LR 2", "HiFi Playback"}, + {"PCMB Swap Mux", "Right", "HiFi Playback"}, + + {"PCMA Swap Mux", "Left", "HiFi Playback"}, + {"PCMA Swap Mux", "LR 2", "HiFi Playback"}, + {"PCMA Swap Mux", "Right", "HiFi Playback"}, }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: core: Fix trivial typo
by Sachin Kamat 08 Jul '14

08 Jul '14

Replaced providing with provided. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c32d8399e770..c5ce67483abc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4396,7 +4396,7 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, ret = of_property_read_string_index(np, propname, 0, &card->name); /* - * EINVAL means the property does not exist. This is fine providing + * EINVAL means the property does not exist. This is fine provided * card->name was previously set, which is checked later in * snd_soc_register_card. */ -- 1.7.9.5

2 2

[alsa-devel] [PATCH 00/13] Remove mach-kirkwood and mach-dove
by Andrew Lunn 08 Jul '14

08 Jul '14

This patchset removes arch/arm/mach-kirkwood and arch/arm/mach-dove. These SoCs are now supported in arch/arm/mach-mvebu using device tree. Change the dependencies for a number of drivers, either to use ARCH_MVEBU where the drivers are generic, or MACH_KIRKWOOD and MACH_DOVE where the drivers are specific to a SoC. Andrew Lunn (13): ARM: Kirkwood: Remove mach-kirkwood ARM: Dove: Remove mach-dove sound: ASoC: kirkwood: Remove unused drivers sound: ASoC: kirkwood: Remove ARCH_KIRKWOOD and ARCH_DOVE dependency cpuidle: kirkwood: Replace ARCH_KIRKWOOD dependency ata: Remove ARCH_KIRKWOOD and ARCH_DOVE dependency thermal: Replace ARCH_KIRKWOOD and ARCH_DOVE dependency leds: Replace ARCH_KIRKWOOD dependency PCI: Remove ARCH_KIRKWOOD and ARCH_DOVE dependency phy: Replace ARCH_KIRKWOOD and ARCH_DOVE dependency rtc: Remove ARCH_KIRKWOOD and ARCH_DOVE dependency watchdog: Remove ARCH_KIRKWOOD and ARCH_DOVE dependency Remove ARCH_DOVE dependency arch/arm/Kconfig | 32 - arch/arm/Kconfig.debug | 10 +- arch/arm/Makefile | 2 - arch/arm/boot/dts/Makefile | 5 +- arch/arm/configs/dove_defconfig | 146 ----- arch/arm/configs/kirkwood_defconfig | 181 ------ arch/arm/mach-dove/Kconfig | 25 - arch/arm/mach-dove/Makefile | 5 - arch/arm/mach-dove/Makefile.boot | 3 - arch/arm/mach-dove/cm-a510.c | 97 --- arch/arm/mach-dove/common.c | 411 ------------ arch/arm/mach-dove/common.h | 49 -- arch/arm/mach-dove/dove-db-setup.c | 103 --- arch/arm/mach-dove/include/mach/bridge-regs.h | 57 -- arch/arm/mach-dove/include/mach/dove.h | 190 ------ arch/arm/mach-dove/include/mach/entry-macro.S | 33 - arch/arm/mach-dove/include/mach/hardware.h | 19 - arch/arm/mach-dove/include/mach/irqs.h | 96 --- arch/arm/mach-dove/include/mach/pm.h | 72 --- arch/arm/mach-dove/include/mach/uncompress.h | 36 -- arch/arm/mach-dove/irq.c | 178 ------ arch/arm/mach-dove/mpp.c | 162 ----- arch/arm/mach-dove/mpp.h | 196 ------ arch/arm/mach-dove/pcie.c | 220 ------- arch/arm/mach-kirkwood/Kconfig | 111 ---- arch/arm/mach-kirkwood/Makefile | 14 - arch/arm/mach-kirkwood/Makefile.boot | 3 - arch/arm/mach-kirkwood/board-dt.c | 223 ------- arch/arm/mach-kirkwood/common.c | 746 ---------------------- arch/arm/mach-kirkwood/common.h | 74 --- arch/arm/mach-kirkwood/d2net_v2-setup.c | 231 ------- arch/arm/mach-kirkwood/include/mach/bridge-regs.h | 86 --- arch/arm/mach-kirkwood/include/mach/entry-macro.S | 34 - arch/arm/mach-kirkwood/include/mach/hardware.h | 14 - arch/arm/mach-kirkwood/include/mach/irqs.h | 65 -- arch/arm/mach-kirkwood/include/mach/kirkwood.h | 142 ---- arch/arm/mach-kirkwood/include/mach/uncompress.h | 46 -- arch/arm/mach-kirkwood/irq.c | 82 --- arch/arm/mach-kirkwood/lacie_v2-common.c | 114 ---- arch/arm/mach-kirkwood/lacie_v2-common.h | 16 - arch/arm/mach-kirkwood/mpp.c | 43 -- arch/arm/mach-kirkwood/mpp.h | 348 ---------- arch/arm/mach-kirkwood/netxbig_v2-setup.c | 422 ------------ arch/arm/mach-kirkwood/openrd-setup.c | 255 -------- arch/arm/mach-kirkwood/pcie.c | 296 --------- arch/arm/mach-kirkwood/pm.c | 76 --- arch/arm/mach-kirkwood/pm.h | 26 - arch/arm/mach-kirkwood/rd88f6192-nas-setup.c | 89 --- arch/arm/mach-kirkwood/rd88f6281-setup.c | 128 ---- arch/arm/mach-kirkwood/t5325-setup.c | 216 ------- arch/arm/mach-kirkwood/ts219-setup.c | 142 ---- arch/arm/mach-kirkwood/ts41x-setup.c | 186 ------ arch/arm/mach-kirkwood/tsx1x-common.c | 113 ---- arch/arm/mach-kirkwood/tsx1x-common.h | 7 - arch/arm/mm/Kconfig | 2 +- drivers/ata/Kconfig | 4 +- drivers/cpuidle/Kconfig.arm | 2 +- drivers/leds/Kconfig | 4 +- drivers/mmc/host/Kconfig | 2 +- drivers/pci/host/Kconfig | 2 +- drivers/phy/Kconfig | 2 +- drivers/rtc/Kconfig | 2 +- drivers/thermal/Kconfig | 4 +- drivers/watchdog/Kconfig | 2 +- sound/soc/kirkwood/Kconfig | 19 +- sound/soc/kirkwood/Makefile | 4 - sound/soc/kirkwood/kirkwood-openrd.c | 109 ---- sound/soc/kirkwood/kirkwood-t5325.c | 116 ---- 68 files changed, 20 insertions(+), 6930 deletions(-) delete mode 100644 arch/arm/configs/dove_defconfig delete mode 100644 arch/arm/configs/kirkwood_defconfig delete mode 100644 arch/arm/mach-dove/Kconfig delete mode 100644 arch/arm/mach-dove/Makefile delete mode 100644 arch/arm/mach-dove/Makefile.boot delete mode 100644 arch/arm/mach-dove/cm-a510.c delete mode 100644 arch/arm/mach-dove/common.c delete mode 100644 arch/arm/mach-dove/common.h delete mode 100644 arch/arm/mach-dove/dove-db-setup.c delete mode 100644 arch/arm/mach-dove/include/mach/bridge-regs.h delete mode 100644 arch/arm/mach-dove/include/mach/dove.h delete mode 100644 arch/arm/mach-dove/include/mach/entry-macro.S delete mode 100644 arch/arm/mach-dove/include/mach/hardware.h delete mode 100644 arch/arm/mach-dove/include/mach/irqs.h delete mode 100644 arch/arm/mach-dove/include/mach/pm.h delete mode 100644 arch/arm/mach-dove/include/mach/uncompress.h delete mode 100644 arch/arm/mach-dove/irq.c delete mode 100644 arch/arm/mach-dove/mpp.c delete mode 100644 arch/arm/mach-dove/mpp.h delete mode 100644 arch/arm/mach-dove/pcie.c delete mode 100644 arch/arm/mach-kirkwood/Kconfig delete mode 100644 arch/arm/mach-kirkwood/Makefile delete mode 100644 arch/arm/mach-kirkwood/Makefile.boot delete mode 100644 arch/arm/mach-kirkwood/board-dt.c delete mode 100644 arch/arm/mach-kirkwood/common.c delete mode 100644 arch/arm/mach-kirkwood/common.h delete mode 100644 arch/arm/mach-kirkwood/d2net_v2-setup.c delete mode 100644 arch/arm/mach-kirkwood/include/mach/bridge-regs.h delete mode 100644 arch/arm/mach-kirkwood/include/mach/entry-macro.S delete mode 100644 arch/arm/mach-kirkwood/include/mach/hardware.h delete mode 100644 arch/arm/mach-kirkwood/include/mach/irqs.h delete mode 100644 arch/arm/mach-kirkwood/include/mach/kirkwood.h delete mode 100644 arch/arm/mach-kirkwood/include/mach/uncompress.h delete mode 100644 arch/arm/mach-kirkwood/irq.c delete mode 100644 arch/arm/mach-kirkwood/lacie_v2-common.c delete mode 100644 arch/arm/mach-kirkwood/lacie_v2-common.h delete mode 100644 arch/arm/mach-kirkwood/mpp.c delete mode 100644 arch/arm/mach-kirkwood/mpp.h delete mode 100644 arch/arm/mach-kirkwood/netxbig_v2-setup.c delete mode 100644 arch/arm/mach-kirkwood/openrd-setup.c delete mode 100644 arch/arm/mach-kirkwood/pcie.c delete mode 100644 arch/arm/mach-kirkwood/pm.c delete mode 100644 arch/arm/mach-kirkwood/pm.h delete mode 100644 arch/arm/mach-kirkwood/rd88f6192-nas-setup.c delete mode 100644 arch/arm/mach-kirkwood/rd88f6281-setup.c delete mode 100644 arch/arm/mach-kirkwood/t5325-setup.c delete mode 100644 arch/arm/mach-kirkwood/ts219-setup.c delete mode 100644 arch/arm/mach-kirkwood/ts41x-setup.c delete mode 100644 arch/arm/mach-kirkwood/tsx1x-common.c delete mode 100644 arch/arm/mach-kirkwood/tsx1x-common.h delete mode 100644 sound/soc/kirkwood/kirkwood-openrd.c delete mode 100644 sound/soc/kirkwood/kirkwood-t5325.c Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Daniel Lezcano <daniel.lezcano(a)linaro.org> Cc: Rafael J. Wysocki <rjw(a)rjwysocki.net> Cc: linux-pm(a)vger.kernel.org Cc: Tejun Heo <tj(a)kernel.org> Cc: linux-ide(a)vger.kernel.org Cc: Zhang Rui <rui.zhang(a)intel.com> Cc: linux-pm(a)vger.kernel.org Cc: Bryan Wu <cooloney(a)gmail.com> Cc: Richard Purdie <rpurdie(a)rpsys.net> Cc: linux-leds(a)vger.kernel.org Cc: Bjorn Helgaas <bhelgaas(a)google.com> Cc: linux-pci(a)vger.kernel.org Cc: Kishon Vijay Abraham I <kishon(a)ti.com> Cc: Alessandro Zummo <a.zummo(a)towertech.it> Cc: rtc-linux(a)googlegroups.com Cc: Wim Van Sebroeck <wim(a)iguana.be> Cc: linux-watchdog(a)vger.kernel.org -- 2.0.0

7 27

[alsa-devel] [PATCH] ASoC: omap-dmic: Use devm_snd_soc_register_component
by Manish Badarkhe 08 Jul '14

08 Jul '14

Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe <badarkhe.manish(a)gmail.com> --- :100644 100644 6925d71... 011e30e... M sound/soc/omap/omap-dmic.c sound/soc/omap/omap-dmic.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 6925d71..011e30e 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -490,7 +490,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) } - ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, + ret = devm_snd_soc_register_component(&pdev->dev, &omap_dmic_component, &omap_dmic_dai, 1); if (ret) goto err_put_clk; @@ -510,7 +510,6 @@ static int asoc_dmic_remove(struct platform_device *pdev) { struct omap_dmic *dmic = platform_get_drvdata(pdev); - snd_soc_unregister_component(&pdev->dev); clk_put(dmic->fclk); return 0; -- 1.7.10.4

2 2

[alsa-devel] [PATCH] ASoC: omap-hdmi: Use devm_snd_soc_register_component
by Manish Badarkhe 08 Jul '14

08 Jul '14

Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe <badarkhe.manish(a)gmail.com> --- :100644 100644 eb9c392... 1dc95b5... M sound/soc/omap/omap-hdmi.c sound/soc/omap/omap-hdmi.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index eb9c392..1dc95b5 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -322,7 +322,7 @@ static int omap_hdmi_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, hdmi_data); - ret = snd_soc_register_component(&pdev->dev, &omap_hdmi_component, + ret = devm_snd_soc_register_component(&pdev->dev, &omap_hdmi_component, &omap_hdmi_dai, 1); if (ret) @@ -335,8 +335,6 @@ static int omap_hdmi_remove(struct platform_device *pdev) { struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - if (hdmi_data == NULL) { dev_err(&pdev->dev, "cannot obtain HDMi data\n"); return -ENODEV; -- 1.7.10.4

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda - fix a typo by changing mute_led_nid to cap_mute_led_nid
by Hui Wang 08 Jul '14

08 Jul '14

BugLink: https://bugs.launchpad.net/bugs/1329580 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 327413e..73b1759 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3529,7 +3529,7 @@ static void alc269_fixup_hp_cap_mic_mute_hook(struct hda_codec *codec, struct alc_spec *spec = codec->spec; unsigned int pinval, enable, disable; - pinval = snd_hda_codec_get_pin_target(codec, spec->mute_led_nid); + pinval = snd_hda_codec_get_pin_target(codec, spec->cap_mute_led_nid); pinval &= ~AC_PINCTL_VREFEN; enable = pinval | AC_PINCTL_VREF_80; disable = pinval | AC_PINCTL_VREF_HIZ; -- 1.8.1.2

2 2

[alsa-devel] [PATCH] USB-Audio: Add second S/PDIF device on Phiree U2
by Anssi Hannula 08 Jul '14

08 Jul '14

Phiree U2 has an unusual configuration. It only has S/PDIF output, but there are still two devices presented: - device 0: PCM audio, subject to volume control - device 1: non-PCM data (passthrough), not subject to volume control It looks like the AES bits are set according to the selected device, since outputting PCM data via device 1 will not work (silence). Currently only the device 0 is shown via the "iec958" alias, and the second device is not accessible via hinted aliases. Simply provide access to both of these devices via the "iec958" alias. Reported-by: touc @ XBMC forum Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- This isn't really the optimal solution, but is probably one of the simplest (except maybe just using device 1 as the only iec958 device). I guess the kernel driver (or even config rules with sufficient @func support) could select the appropriate device according to AES0, but not sure if that is worth the effort? Also, I'm not 100% sure if this is actually uncommon or if we just hide it in on other such devices. src/conf/cards/USB-Audio.conf | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index ce3ae01..77a48b9 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -52,6 +52,11 @@ USB-Audio.pcm.iec958_device { "USB Device 0x46d:0x992" 999 } +# Second iec958 device number, if any. +USB-Audio.pcm.iec958_2_device { + "PHIREE U2" 1 # 0 = PCM S/PDIF, 1 = non-PCM S/PDIF +} + # If a device requires non-standard definitions for front, surround40, # surround51, surround71 or iec958, they can be defined here. @@ -422,4 +427,39 @@ USB-Audio.pcm.iec958.0 { } } +USB-Audio.pcm.iec958.1 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.USB-Audio." + { @func card_name card $CARD } + ".pcm.iec958_2:CARD=" $CARD + ",AES0=" $AES0 ",AES1=" $AES1 ",AES2=" $AES2 ",AES3=" $AES3 + ] + } + default { + # FIXME: we cannot set the AES parameters + type hw + card $CARD + device { + @func refer + name { + @func concat + strings [ + "cards.USB-Audio.pcm.iec958_2_device." + { @func card_name card $CARD } + ] + } + default 999 + } + } +} + # vim: ft=alsaconf -- 1.8.4.5

2 1

[alsa-devel] [PATCH v2] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 08 Jul '14

08 Jul '14

This patch adds a CODEC function to the NXP TDA998x HDMI transmitter. The CODEC handles both I2S and S/PDIF input and does dynamic input switch in the TDA998x I2C driver on start/stop audio streaming. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2: check double stream start (Mark Brown) --- .../devicetree/bindings/drm/i2c/tda998x.txt | 13 ++ drivers/gpu/drm/i2c/Makefile | 2 +- drivers/gpu/drm/i2c/tda998x_codec.c | 247 +++++++++++++++++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 74 ++++-- drivers/gpu/drm/i2c/tda998x_drv.h | 32 +++ include/drm/i2c/tda998x.h | 1 + 6 files changed, 348 insertions(+), 21 deletions(-) create mode 100644 drivers/gpu/drm/i2c/tda998x_codec.c create mode 100644 drivers/gpu/drm/i2c/tda998x_drv.h diff --git a/Documentation/devicetree/bindings/drm/i2c/tda998x.txt b/Documentation/devicetree/bindings/drm/i2c/tda998x.txt index d7df01c..7ce4eac 100644 --- a/Documentation/devicetree/bindings/drm/i2c/tda998x.txt +++ b/Documentation/devicetree/bindings/drm/i2c/tda998x.txt @@ -15,6 +15,15 @@ Optional properties: - video-ports: 24 bits value which defines how the video controller output is wired to the TDA998x input - default: <0x230145> + - audio-ports: one or two values corresponding to entries in + the audio-port-names property. + + - audio-port-names: must contain "i2s", "spdif" entries + matching entries in the audio-ports property. + + - #sound-dai-cells: must be set to <1> for use with the simple-card. + The DAI 0 is the I2S input and the DAI 1 is the S/PDIF input. + Example: tda998x: hdmi-encoder { @@ -24,4 +33,8 @@ Example: interrupts = <27 2>; /* falling edge */ pinctrl-0 = <&pmx_camera>; pinctrl-names = "default"; + + audio-ports = <0x03>, <0x04>; + audio-port-names = "i2s", "spdif"; + #sound-dai-cells = <1>; }; diff --git a/drivers/gpu/drm/i2c/Makefile b/drivers/gpu/drm/i2c/Makefile index 43aa33b..f2d625c 100644 --- a/drivers/gpu/drm/i2c/Makefile +++ b/drivers/gpu/drm/i2c/Makefile @@ -6,5 +6,5 @@ obj-$(CONFIG_DRM_I2C_CH7006) += ch7006.o sil164-y := sil164_drv.o obj-$(CONFIG_DRM_I2C_SIL164) += sil164.o -tda998x-y := tda998x_drv.o +tda998x-y := tda998x_drv.o tda998x_codec.o obj-$(CONFIG_DRM_I2C_NXP_TDA998X) += tda998x.o diff --git a/drivers/gpu/drm/i2c/tda998x_codec.c b/drivers/gpu/drm/i2c/tda998x_codec.c new file mode 100644 index 0000000..e22f974 --- /dev/null +++ b/drivers/gpu/drm/i2c/tda998x_codec.c @@ -0,0 +1,247 @@ +/* + * ALSA SoC TDA998X CODEC + * + * Copyright (C) 2014 Jean-Francois Moine + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <linux/of.h> +#include <linux/i2c.h> +#include <drm/drm_encoder_slave.h> +#include <drm/i2c/tda998x.h> + +#include "tda998x_drv.h" + +#define TDA998X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static int tda_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + u8 *eld = priv->eld; + struct snd_pcm_runtime *runtime = substream->runtime; + u8 *sad; + int sad_count; + unsigned eld_ver, mnl, rate_mask; + unsigned max_channels, fmt; + u64 formats; + struct snd_pcm_hw_constraint_list *rate_constraints = + &priv->rate_constraints; + static const u32 hdmi_rates[] = { + 32000, 44100, 48000, 88200, 9600, 176400, 192000 + }; + + /* check if streaming is already active */ + if (priv->dai_id != AFMT_NO_AUDIO) + return -EBUSY; + priv->dai_id = dai->id; + + if (!eld) + return 0; + + /* adjust the hw params from the ELD (EDID) */ + eld_ver = eld[0] >> 3; + if (eld_ver != 2 && eld_ver != 31) + return 0; + + mnl = eld[4] & 0x1f; + if (mnl > 16) + return 0; + + sad_count = eld[5] >> 4; + sad = eld + 20 + mnl; + + /* Start from the basic audio settings */ + max_channels = 2; + rate_mask = 0; + fmt = 0; + while (sad_count--) { + switch (sad[0] & 0x78) { + case 0x08: /* PCM */ + max_channels = max(max_channels, (sad[0] & 7) + 1u); + rate_mask |= sad[1]; + fmt |= sad[2] & 0x07; + break; + } + sad += 3; + } + + /* set the constraints */ + rate_constraints->list = hdmi_rates; + rate_constraints->count = ARRAY_SIZE(hdmi_rates); + rate_constraints->mask = rate_mask; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + rate_constraints); + + formats = 0; + if (fmt & 1) + formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (fmt & 2) + formats |= SNDRV_PCM_FMTBIT_S20_3LE; + if (fmt & 4) + formats |= SNDRV_PCM_FMTBIT_S24_LE; + snd_pcm_hw_constraint_mask64(runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + formats); + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 1, max_channels); + return 0; +} + +static int tda_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + + /* Requires an attached display */ + if (!priv->encoder->crtc) + return -ENODEV; + + /* if same input and same parameters, do not do a full switch */ + if (dai->id == priv->params.audio_format && + params_format(params) == priv->audio_sample_format) { + tda998x_audio_start(priv, 0); + return 0; + } + priv->params.audio_format = dai->id; + priv->audio_sample_format = params_format(params); + priv->params.audio_cfg = + priv->audio_ports[dai->id == AFMT_I2S ? 0 : 1]; + tda998x_audio_start(priv, 1); + return 0; +} + +static void tda_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + + tda998x_audio_stop(priv); + priv->dai_id = AFMT_NO_AUDIO; +} + +static const struct snd_soc_dai_ops tda_ops = { + .startup = tda_startup, + .hw_params = tda_hw_params, + .shutdown = tda_shutdown, +}; + +static struct snd_soc_dai_driver tda998x_dai[] = { + { + .name = "i2s-hifi", + .id = AFMT_I2S, + .playback = { + .stream_name = "HDMI I2S Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = TDA998X_FORMATS, + }, + .ops = &tda_ops, + }, + { + .name = "spdif-hifi", + .id = AFMT_SPDIF, + .playback = { + .stream_name = "HDMI SPDIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 22050, + .rate_max = 192000, + .formats = TDA998X_FORMATS, + }, + .ops = &tda_ops, + }, +}; + +static const struct snd_soc_dapm_widget tda_widgets[] = { + SND_SOC_DAPM_OUTPUT("hdmi-out"), +}; +static const struct snd_soc_dapm_route tda_routes[] = { + { "hdmi-out", NULL, "HDMI I2S Playback" }, + { "hdmi-out", NULL, "HDMI SPDIF Playback" }, +}; + +static int tda_probe(struct snd_soc_codec *codec) +{ + struct i2c_client *i2c_client = to_i2c_client(codec->dev); + struct tda998x_priv *priv = i2c_get_clientdata(i2c_client); + struct device_node *np = codec->dev->of_node; + int i, j, ret; + const char *p; + + if (!priv) + return -ENODEV; + snd_soc_codec_set_drvdata(codec, priv); + + if (!np) + return 0; + + /* get the audio input ports*/ + for (i = 0; i < 2; i++) { + u32 port; + + ret = of_property_read_u32_index(np, "audio-ports", i, &port); + if (ret) { + if (i == 0) + dev_err(codec->dev, + "bad or missing audio-ports\n"); + break; + } + ret = of_property_read_string_index(np, "audio-port-names", + i, &p); + if (ret) { + dev_err(codec->dev, + "missing audio-port-names[%d]\n", i); + break; + } + if (strcmp(p, "i2s") == 0) { + j = 0; + } else if (strcmp(p, "spdif") == 0) { + j = 1; + } else { + dev_err(codec->dev, + "bad audio-port-names '%s'\n", p); + break; + } + priv->audio_ports[j] = port; + } + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_tda998x = { + .probe = tda_probe, + .dapm_widgets = tda_widgets, + .num_dapm_widgets = ARRAY_SIZE(tda_widgets), + .dapm_routes = tda_routes, + .num_dapm_routes = ARRAY_SIZE(tda_routes), +}; + +int tda998x_codec_register(struct device *dev) +{ + return snd_soc_register_codec(dev, + &soc_codec_tda998x, + tda998x_dai, ARRAY_SIZE(tda998x_dai)); +} + +void tda998x_codec_unregister(struct device *dev) +{ + snd_soc_unregister_codec(dev); +} diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c index 240c331..7e9f9dc 100644 --- a/drivers/gpu/drm/i2c/tda998x_drv.c +++ b/drivers/gpu/drm/i2c/tda998x_drv.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/irq.h> #include <sound/asoundef.h> +#include <sound/pcm_params.h> #include <drm/drmP.h> #include <drm/drm_crtc_helper.h> @@ -28,24 +29,9 @@ #include <drm/drm_edid.h> #include <drm/i2c/tda998x.h> -#define DBG(fmt, ...) DRM_DEBUG(fmt"\n", ##__VA_ARGS__) +#include "tda998x_drv.h" -struct tda998x_priv { - struct i2c_client *cec; - struct i2c_client *hdmi; - uint16_t rev; - uint8_t current_page; - int dpms; - bool is_hdmi_sink; - u8 vip_cntrl_0; - u8 vip_cntrl_1; - u8 vip_cntrl_2; - struct tda998x_encoder_params params; - - wait_queue_head_t wq_edid; - volatile int wq_edid_wait; - struct drm_encoder *encoder; -}; +#define DBG(fmt, ...) DRM_DEBUG(fmt"\n", ##__VA_ARGS__) #define to_tda998x_priv(x) ((struct tda998x_priv *)to_encoder_slave(x)->slave_priv) @@ -640,12 +626,11 @@ static void tda998x_configure_audio(struct tda998x_priv *priv, struct drm_display_mode *mode, struct tda998x_encoder_params *p) { - uint8_t buf[6], clksel_aip, clksel_fs, cts_n, adiv; + uint8_t buf[6], clksel_aip, clksel_fs, cts_n, adiv, aclk; uint32_t n; /* Enable audio ports */ reg_write(priv, REG_ENA_AP, p->audio_cfg); - reg_write(priv, REG_ENA_ACLK, p->audio_clk_cfg); /* Set audio input source */ switch (p->audio_format) { @@ -654,13 +639,28 @@ tda998x_configure_audio(struct tda998x_priv *priv, clksel_aip = AIP_CLKSEL_AIP_SPDIF; clksel_fs = AIP_CLKSEL_FS_FS64SPDIF; cts_n = CTS_N_M(3) | CTS_N_K(3); + aclk = 0; /* no clock */ break; case AFMT_I2S: reg_write(priv, REG_MUX_AP, MUX_AP_SELECT_I2S); clksel_aip = AIP_CLKSEL_AIP_I2S; clksel_fs = AIP_CLKSEL_FS_ACLK; - cts_n = CTS_N_M(3) | CTS_N_K(3); + /* with I2S input, the CTS_N predivider depends on + * the sample width */ + switch (priv->audio_sample_format) { + case SNDRV_PCM_FORMAT_S16_LE: + cts_n = CTS_N_M(3) | CTS_N_K(1); + break; + case SNDRV_PCM_FORMAT_S24_LE: + cts_n = CTS_N_M(3) | CTS_N_K(2); + break; + default: + case SNDRV_PCM_FORMAT_S32_LE: + cts_n = CTS_N_M(3) | CTS_N_K(3); + break; + } + aclk = 1; /* clock enable */ break; default: @@ -672,6 +672,7 @@ tda998x_configure_audio(struct tda998x_priv *priv, reg_clear(priv, REG_AIP_CNTRL_0, AIP_CNTRL_0_LAYOUT | AIP_CNTRL_0_ACR_MAN); /* auto CTS */ reg_write(priv, REG_CTS_N, cts_n); + reg_write(priv, REG_ENA_ACLK, aclk); /* * Audio input somehow depends on HDMI line rate which is @@ -728,6 +729,24 @@ tda998x_configure_audio(struct tda998x_priv *priv, tda998x_write_aif(priv, p); } +/* tda998x codec interface */ +void tda998x_audio_start(struct tda998x_priv *priv, + int full) +{ + struct tda998x_encoder_params *p = &priv->params; + + if (!full) { + reg_write(priv, REG_ENA_AP, p->audio_cfg); + return; + } + tda998x_configure_audio(priv, &priv->encoder->crtc->hwmode, p); +} + +void tda998x_audio_stop(struct tda998x_priv *priv) +{ + reg_write(priv, REG_ENA_AP, 0); +} + /* DRM encoder functions */ static void @@ -1149,6 +1168,11 @@ tda998x_encoder_get_modes(struct drm_encoder *encoder, drm_mode_connector_update_edid_property(connector, edid); n = drm_add_edid_modes(connector, edid); priv->is_hdmi_sink = drm_detect_hdmi_monitor(edid); + + /* keep the EDID as ELD for the audio subsystem */ + drm_edid_to_eld(connector, edid); + priv->eld = connector->eld; + kfree(edid); } @@ -1191,6 +1215,8 @@ tda998x_encoder_destroy(struct drm_encoder *encoder) if (priv->hdmi->irq) free_irq(priv->hdmi->irq, priv); + tda998x_codec_unregister(&priv->hdmi->dev); + if (priv->cec) i2c_unregister_device(priv->cec); kfree(priv); @@ -1239,10 +1265,15 @@ tda998x_encoder_init(struct i2c_client *client, if (!priv) return -ENOMEM; + i2c_set_clientdata(client, priv); + priv->vip_cntrl_0 = VIP_CNTRL_0_SWAP_A(2) | VIP_CNTRL_0_SWAP_B(3); priv->vip_cntrl_1 = VIP_CNTRL_1_SWAP_C(0) | VIP_CNTRL_1_SWAP_D(1); priv->vip_cntrl_2 = VIP_CNTRL_2_SWAP_E(4) | VIP_CNTRL_2_SWAP_F(5); + priv->params.audio_frame[1] = 1; /* channels - 1 */ + priv->params.audio_sample_rate = 48000; /* 48kHz */ + priv->current_page = 0xff; priv->hdmi = client; priv->cec = i2c_new_dummy(client->adapter, 0x34); @@ -1340,6 +1371,9 @@ tda998x_encoder_init(struct i2c_client *client, /* enable EDID read irq: */ reg_set(priv, REG_INT_FLAGS_2, INT_FLAGS_2_EDID_BLK_RD); + /* register the audio CODEC */ + tda998x_codec_register(&client->dev); + if (!np) return 0; /* non-DT */ diff --git a/drivers/gpu/drm/i2c/tda998x_drv.h b/drivers/gpu/drm/i2c/tda998x_drv.h new file mode 100644 index 0000000..e6c8dd5 --- /dev/null +++ b/drivers/gpu/drm/i2c/tda998x_drv.h @@ -0,0 +1,32 @@ +/* tda998x private data */ + +struct tda998x_priv { + struct i2c_client *cec; + struct i2c_client *hdmi; + uint16_t rev; + uint8_t current_page; + int dpms; + bool is_hdmi_sink; + u8 vip_cntrl_0; + u8 vip_cntrl_1; + u8 vip_cntrl_2; + struct tda998x_encoder_params params; + + wait_queue_head_t wq_edid; + volatile int wq_edid_wait; + struct drm_encoder *encoder; + + u8 audio_ports[2]; + int audio_sample_format; + int dai_id; /* DAI ID when streaming active */ + + u8 *eld; + + struct snd_pcm_hw_constraint_list rate_constraints; +}; + +int tda998x_codec_register(struct device *dev); +void tda998x_codec_unregister(struct device *dev); + +void tda998x_audio_start(struct tda998x_priv *priv, int full); +void tda998x_audio_stop(struct tda998x_priv *priv); diff --git a/include/drm/i2c/tda998x.h b/include/drm/i2c/tda998x.h index 3e419d9..31757df 100644 --- a/include/drm/i2c/tda998x.h +++ b/include/drm/i2c/tda998x.h @@ -20,6 +20,7 @@ struct tda998x_encoder_params { u8 audio_frame[6]; enum { + AFMT_NO_AUDIO = 0, AFMT_SPDIF, AFMT_I2S } audio_format; -- 2.0.1 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 2

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID 0x10de0070 to snd-hda
by Aaron Plattner 08 Jul '14

08 Jul '14

Vendor ID 0x10de0070 is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- Another day, another Azalia codec ID. I verified that the patch_nvhdmi code is still needed for this chip. sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3e4417b0ddbe..1df5cf629a24 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3337,6 +3337,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, +{ .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3394,6 +3395,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); +MODULE_ALIAS("snd-hda-codec-id:10de0070"); MODULE_ALIAS("snd-hda-codec-id:10de0071"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); -- 2.0.1

2 1

[alsa-devel] Integrate WM8974 codec with AM1808 processor
by Sangilikumar 08 Jul '14

08 Jul '14

Hello Engineers I'm working with TI's AM1808 ARM9 processor. My design in based on OMAP-L138 LCDK board. Here i replaced the TLV320AIC310 codec with WM8974 mono codec. Added the codec driver in board file and registered the sound card by static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "wm8974-hifi", .codec_name = "wm8974-codec.1-001a", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, }; But got some errors Sound card registered successfully but i got some errors wm8974-codec 1-001a: Failed to add route HPLOUT->Headphone Jack wm8974-codec 1-001a: dapm: unknown pin MONO_LOUT wm8974-codec 1-001a: dapm: unknown pin HPLCOM wm8974-codec 1-001a: dapm: unknown pin HPRCOM asoc: wm8974-hifi <-> davinci-mcasp.0 mapping ok ALSA device list: #0: DA850/OMAP-L138 EVM TCP cubic registered .... ALSA: Restoring mixer settings... Configuring network interfaces... Found hardware: "DA850_OMAP-L138" "" "" "" "" Hardware is initialized using a generic method /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #1 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #2 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #3 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #4 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #5 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #6 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #7 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #8 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #9 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #10 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #11 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #12 (No such file or directory) davinci_mdio davinci_mdio.0: resetting idled controller root@omapl138-lcdk:/aud# speaker-test speaker-test 1.0.24.2 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 32768 Period size range asoc: machine hw_params failed from 8 to 2048 Using max buffer size 32768 Periods = 4 Unable to set hw params for playback: Invalid argument Setting of hwparams failed: Invalid argument root@omapl138-lcdk:/aud# Please let me know what are the changes i need to do in Linux files and drivers.

1 0

[alsa-devel] [PATCH v2 0/7] ACPICA: Enable ACPICA prototypes for CONFIG_ACPI=n builds.
by Lv Zheng 08 Jul '14

08 Jul '14

This is the last series for ACPICA header inclusion cleanups. The whole story can be found below: 1. Unsafe inclusions: origin: https://lkml.org/lkml/2013/11/22/510 merged: https://lkml.org/lkml/2013/12/6/79 2. Header orders: origin: https://lkml.org/lkml/2014/4/8/57 merged: https://lkml.org/lkml/2014/5/12/111 3. Header stubs: origin: https://lkml.org/lkml/2014/12/18/149 merged: https://lkml.org/lkml/2014/4/29/716 This patchset contains patches that are parts of the last series sent before, so it is marked as v2 belonging to the same thread as this series: https://lkml.org/lkml/2014/12/18/149 Now we have cleaned up all mis-ordered ACPICA headers and implemented stubs for ACPICA prototypes (functions, global variables, macros), we finally can enable ACPICA headers for CONFIG_ACPI=n builds so that all ACPICA defined types are visible to all kernel source files. After exposing ACPICA types to all kernel sources, we can also sort the final wrong <acpi/acpi.h> inclusion out of <linux/sfi_acpi.h>. This patchset also includes a rule enforcement patch to avoid future inclusion mistakes. The patch set has passed a build/boot test on the following machines: Dell Inspiron Mini 1010 (i386) HP Compaq 8200 Elite SFF PC (x86-64) With the following kernel configuration items enabled: All drivers/acpi configurations All platform drivers All ACPI drivers 4 DRM drivers that implement ACPI opregion All buses with ACPI bindings All cpufreq configurations All xen configurations All ACPI table drivers They are tested twice with CONFIG_ACPI enabled/disabled. Also allyes/allno/allmod tests are done for ARCH=i386 and ARCH=x86_64. A boot test has been done to ARCH=x86_64 default configuration and it is confirmed that the ACPI devices are successfully created under /sys/bus/acpi/devices when CONFIG_ACPI is enabled. Lv Zheng (7): ACPICA: Linux: Add stub support for Linux specific variables and functions. ACPICA: Linux: Add stub implementation of ACPICA 64-bit mathematics. ACPICA: Linux: Add configuration item to indicate the architecture specific support. ACPICA: Linux: Allow ACPICA inclusion for CONFIG_ACPI=n builds. ACPI/SFI: Fix wrong <acpi/acpi.h> inclusion in SFI/ACPI wrapper - table definitions. ACPI: Cleanup useless ACPI inclusion. ACPI: Add support to force header inclusion rules for <acpi/acpi.h>. arch/ia64/Kconfig | 1 + arch/x86/Kconfig | 1 + drivers/acpi/Kconfig | 4 ++++ drivers/acpi/acpica/Makefile | 2 +- drivers/acpi/bus.c | 3 --- include/acpi/acpixf.h | 28 +++++++++++++++++----------- include/acpi/platform/aclinux.h | 12 ++++++++++++ include/acpi/platform/aclinuxex.h | 22 ++++++++++++++++++++++ include/linux/acpi.h | 6 +++--- include/linux/sfi_acpi.h | 3 --- sound/soc/intel/sst-haswell-dsp.c | 3 --- 11 files changed, 61 insertions(+), 24 deletions(-) -- 1.7.10

3 3

[alsa-devel] [PATCH v5] ASoC: tas2552: Support TI TAS2552 Amplifier
by Dan Murphy 07 Jul '14

07 Jul '14

Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v5 - Consolidated dai_fmt call back to write serial control register once - https://patchwork.kernel.org/patch/4474531/ .../devicetree/bindings/sound/tas2552.txt | 25 + include/sound/tas2552-plat.h | 25 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2552.c | 576 ++++++++++++++++++++ sound/soc/codecs/tas2552.h | 120 ++++ 6 files changed, 753 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 include/sound/tas2552-plat.h create mode 100644 sound/soc/codecs/tas2552.c create mode 100644 sound/soc/codecs/tas2552.h diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 0000000..00223dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,25 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - One of: + "ti,tas2552" - TAS2552 + +- reg - I2C slave address + +Optional properties: + +- enable-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS2552 diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 0000000..65e7627 --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0b9571c..cc09261 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -99,6 +99,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TAS2552 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE @@ -754,4 +755,8 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1c..33bc7228 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -162,6 +162,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -325,3 +326,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 0000000..fe305d6 --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,576 @@ +/* + * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/tas2552-plat.h> + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x22}, + {TAS2552_CFG_3, 0x80}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xc0}, + {TAS2552_PDM_CFG, 0x01}, + {TAS2552_PGA_GAIN, 0x00}, + {TAS2552_BOOST_PT_CTRL, 0x0f}, + {TAS2552_LIMIT_LVL_CTRL, 0x0c}, + {TAS2552_LIMIT_RATE_HYS, 0x08}, + {TAS2552_CFG_2, 0xef}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x04}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +#define TAS2552_NUM_SUPPLIES 3 +static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = { + "vbat", /* vbat voltage */ + "iovdd", /* I/O Voltage */ + "avdd", /* Analog DAC Voltage */ +}; + +struct tas2552_data { + struct mutex mutex; + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES]; + unsigned char regs[TAS2552_VBAT_DATA]; + struct gpio_desc *enable_gpio; + u8 power_state:1; +}; + +static int tas2552_power(struct tas2552_data *data, u8 power) +{ + int ret = 0; + + mutex_lock(&data->mutex); + + if (power) { + if (data->enable_gpio) + gpiod_set_value(data->enable_gpio, 1); + + data->power_state = 1; + } else { + if (data->enable_gpio) + gpiod_set_value(data->enable_gpio, 0); + + data->power_state = 0; + } + + mutex_unlock(&data->mutex); + return ret; +} + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg = 0x0; + + if (sw_shutdown) + cfg1_reg |= TAS2552_SWS_MASK; + else + cfg1_reg &= ~TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static int tas2552_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u8 wclk_reg; + struct snd_soc_codec *codec = dai->codec; + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (params_rate(params)) { + case 8000: + wclk_reg = TAS2552_8KHZ; + break; + case 11025: + wclk_reg = TAS2552_11_12KHZ; + break; + case 16000: + wclk_reg = TAS2552_16KHZ; + break; + case 32000: + wclk_reg = TAS2552_32KHZ; + break; + case 22050: + case 24000: + wclk_reg = TAS2552_22_24KHZ; + break; + case 44100: + case 48000: + wclk_reg = TAS2552_44_48KHZ; + break; + case 96000: + wclk_reg = TAS2552_88_96KHZ; + break; + default: + dev_vdbg(codec->dev, "Substream sample rate is not found\n"); + return -EINVAL; + } + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_3, TAS2552_WCLK_MASK, wclk_reg); + + pm_runtime_put(codec->dev); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + u8 serial_format; + u8 serial_control_mask = 0x00; + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + if (fmt & SND_SOC_DAIFMT_MASTER_MASK) + serial_control_mask = TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format &= TAS2552_DAIFMT_I2S_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format |= TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format |= TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format |= TAS2552_DAIFMT_LEFT_J; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + if (fmt & SND_SOC_DAIFMT_FORMAT_MASK) + serial_control_mask |= TAS2552_DATA_FORMAT_MASK; + + if (serial_control_mask) { + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, serial_control_mask, + serial_format); + + pm_runtime_put(codec->dev); + } + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + snd_soc_write(codec, TAS2552_PLL_CTRL_1, 0x10); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, 0x00); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, 0x00); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + pm_runtime_put(codec->dev); + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg = 0x0; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg |= TAS2552_MUTE_MASK; + else + cfg1_reg &= ~TAS2552_MUTE_MASK; + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + pm_runtime_put(codec->dev); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int tas2552_runtime_suspend(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_sw_shutdown(tas2552, 1); + tas2552_power(tas2552, 0); + + regcache_cache_only(tas2552->regmap, true); + regcache_mark_dirty(tas2552->regmap); + + return 0; +} + +static int tas2552_runtime_resume(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_power(tas2552, 1); + tas2552_sw_shutdown(tas2552, 0); + + regcache_cache_only(tas2552->regmap, false); + regcache_sync(tas2552->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops tas2552_pm = { + SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume, + NULL) +}; + +static int tas2552_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + pm_runtime_get_sync(codec->dev); + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + return 0; +} + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + pm_runtime_put(codec->dev); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .startup = tas2552_startup, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + tas2552->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + tas2552_power(tas2552, 1); + tas2552_sw_shutdown(tas2552, 0); + + pm_runtime_set_active(codec->dev); + pm_runtime_set_autosuspend_delay(codec->dev, 1000); + pm_runtime_use_autosuspend(codec->dev); + pm_runtime_enable(codec->dev); + pm_runtime_mark_last_busy(codec->dev); + pm_runtime_put_sync_autosuspend(codec->dev); + + snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK | + TAS2552_SWS_MASK | TAS2552_PLL_SRC_BCLK); + snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL | + TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ); + snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8); + snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL); + /* 0dB gain */ + snd_soc_write(codec, TAS2552_PGA_GAIN, 0x10); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 | + TAS2552_APT_THRESH_2_1_7); + /** + * Data sheet indicates to write 0x0c to 0x0d during init but no + * additional information is given to what it means. + */ + snd_soc_write(codec, TAS2552_LIMIT_LVL_CTRL, 0x0c); + /** + * Data sheet indicates to write 0x20 to 0x0e during init but no + * additional information is given to what it means. + */ + + snd_soc_write(codec, TAS2552_LIMIT_RATE_HYS, 0x20); + + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | + TAS2552_BOOST_EN | TAS2552_APT_EN | + TAS2552_PLL_ENABLE | TAS2552_LIM_EN); + + return 0; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_power(tas2552, 0); + + return 0; +}; + +#ifdef CONFIG_PM +static int tas2552_suspend(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + pm_runtime_put(codec->dev); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) + dev_err(codec->dev, "Failed to disable supplies: %d\n", + ret); + return 0; +} + +static int tas2552_resume(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + } + + pm_runtime_get_sync(codec->dev); + + return 0; +} +#else +#define tas2552_suspend NULL +#define tas2552_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .suspend = tas2552_suspend, + .resume = tas2552_resume, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + int ret; + int i; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + data->enable_gpio = devm_gpiod_get(dev, "enable"); + if (IS_ERR(data->enable_gpio)) { + ret = PTR_ERR(data->enable_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + data->enable_gpio = NULL; + } else { + gpiod_direction_output(data->enable_gpio, 0); + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas2552_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) + dev_err(dev, "Failed to request supplies: %d\n", ret); + + mutex_init(&data->mutex); + + dev_set_drvdata(&client->dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return 0; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552-codec", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + .pm = &tas2552_pm, + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 0000000..107b237 --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,120 @@ +/* + * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_LIMIT_LVL_CTRL 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) + +/* CFG2 Register Masks */ +#define TAS2552_CLASSD_EN (1 << 7) +#define TAS2552_BOOST_EN (1 << 6) +#define TAS2552_APT_EN (1 << 5) +#define TAS2552_PLL_ENABLE (1 << 3) +#define TAS2552_LIM_EN (1 << 2) +#define TAS2552_IVSENSE_EN (1 << 1) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK (1 << 7) +#define TAS2552_BIT_CLK_MASK (1 << 6) +#define TAS2552_DATA_FORMAT_MASK (0x11 << 2) + +#define TAS2552_DAIFMT_I2S_MASK 0xf3 +#define TAS2552_DAIFMT_DSP (1 << 3) +#define TAS2552_DAIFMT_RIGHT_J (1 << 4) +#define TAS2552_DAIFMT_LEFT_J (0x11 << 3) + +#define TAS2552_PLL_SRC_MCLK 0x00 +#define TAS2552_PLL_SRC_BCLK (1 << 3) +#define TAS2552_PLL_SRC_IVCLKIN (1 << 4) +#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3) + +#define TAS2552_DIN_SRC_SEL_MUTED 0x00 +#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4) +#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5) +#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4) + +#define TAS2552_PDM_IN_SEL (1 << 5) +#define TAS2552_I2S_OUT_SEL (1 << 6) +#define TAS2552_ANALOG_IN_SEL (1 << 7) + +/* CFG3 WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ (1 << 1) +#define TAS2552_16KHZ (1 << 2) +#define TAS2552_22_24KHZ (1 << 3) +#define TAS2552_32KHZ (1 << 4) +#define TAS2552_44_48KHZ (1 << 5) +#define TAS2552_88_96KHZ (1 << 6) +#define TAS2552_176_192KHZ (1 << 7) + +/* OUTPUT_DATA register */ +#define TAS2552_PDM_DATA_I 0x00 +#define TAS2552_PDM_DATA_V (1 << 6) +#define TAS2552_PDM_DATA_I_V (1 << 7) +#define TAS2552_PDM_DATA_V_I (0x11 << 6) + +/* PDM CFG Register */ +#define TAS2552_PDM_DATA_ES_RISE 0x4 + +#define TAS2552_PDM_PLL_CLK_SEL 0x00 +#define TAS2552_PDM_IV_CLK_SEL (1 << 1) +#define TAS2552_PDM_BCLK_SEL (1 << 2) +#define TAS2552_PDM_MCLK_SEL (1 << 3) + +/* Boost pass-through register */ +#define TAS2552_APT_DELAY_50 0x00 +#define TAS2552_APT_DELAY_75 (1 << 1) +#define TAS2552_APT_DELAY_125 (1 << 2) +#define TAS2552_APT_DELAY_200 (1 << 3) + +#define TAS2552_APT_THRESH_2_5 0x00 +#define TAS2552_APT_THRESH_1_7 (1 << 3) +#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4) +#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2) + +#endif -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: max98090: Fix build warning
by Thierry Reding 07 Jul '14

07 Jul '14

From: Thierry Reding <treding(a)nvidia.com> The max98090_{suspend,resume}() functions are used for system sleep and therefore need to be guarded by CONFIG_PM_SLEEP rather than CONFIG_PM. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ca74f502a4cb..e806065d68cc 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2415,7 +2415,7 @@ static int max98090_runtime_suspend(struct device *dev) } #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int max98090_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); -- 2.0.1

2 1

[alsa-devel] [PATCH] ASoC: RT286: Fix silent at the beginning of stream
by bardliao＠realtek.com 07 Jul '14

07 Jul '14

From: Bard Liao <bardliao(a)realtek.com> This patch fix the issue that the output is almost silent at the beginning of starting a playback. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 20 +++++++++++++++++++- sound/soc/codecs/rt286.h | 5 +++++ 2 files changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e6f33ab..8103315 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -54,6 +54,7 @@ static struct reg_default rt286_index_def[] = { { 0x09, 0xd810 }, { 0x0a, 0x0060 }, { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, { 0x19, 0x0a17 }, { 0x20, 0x0020 }, @@ -62,6 +63,9 @@ static struct reg_default rt286_index_def[] = { { 0x4f, 0x50e9 }, { 0x50, 0x2c00 }, { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, }; #define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) @@ -902,14 +906,23 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_PREPARE: - if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x200); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(10); break; case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); break; default: @@ -1152,6 +1165,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, /*Power down LDO2*/ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + /*Set depop parameter*/ + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); + if (rt286->i2c->irq) { regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index 21c570f..b539b73 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -127,6 +127,7 @@ #define RT286_I2S_CTRL1 0x09 #define RT286_I2S_CTRL2 0x0a #define RT286_CLK_DIV 0x0b +#define RT286_DC_GAIN 0x0d #define RT286_POWER_CTRL3 0x0f #define RT286_MIC1_DET_CTRL 0x19 #define RT286_MISC_CTRL1 0x20 @@ -135,6 +136,10 @@ #define RT286_CBJ_CTRL1 0x4f #define RT286_CBJ_CTRL2 0x50 #define RT286_PLL_CTRL 0x63 +#define RT286_DEPOP_CTRL1 0x66 +#define RT286_DEPOP_CTRL2 0x67 +#define RT286_DEPOP_CTRL3 0x68 +#define RT286_DEPOP_CTRL4 0x69 /* SPDIF (0x06) */ #define RT286_SPDIF_SEL_SFT 0 -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: add SOC_VALUE_ENUM_EXT
by Richard Fitzgerald 07 Jul '14

07 Jul '14

Adds an equivalent of SOC_ENUM_EXT for value enums Strictly speaking SOC_ENUM_EXT can also be used to define a value enum since the only difference is the get and set functions. But this doesn't look good in code because it is inconsistent with the normal control definitions. Adding a specific SOC_VALUE_ENUM_EXT is better for code clarity. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a5b4f6..1483920 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,6 +248,8 @@ .info = snd_soc_info_enum_double, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } +#define SOC_VALUE_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ + SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix build warning
by Thierry Reding 07 Jul '14

07 Jul '14

From: Thierry Reding <treding(a)nvidia.com> The hda_tegra_disable_clocks() function is only used by the suspend and resume code, so it needs to be included in the #ifdef CONFIG_PM_SLEEP block to prevent the following warning: CC sound/pci/hda/hda_tegra.o sound/pci/hda/hda_tegra.c:238:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function] static void hda_tegra_disable_clocks(struct hda_tegra *data) ^ Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index dc4bb21ed17b..227990bc02e3 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -235,6 +235,7 @@ disable_hda: return rc; } +#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -242,7 +243,6 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) clk_disable_unprepare(data->hda_clk); } -#ifdef CONFIG_PM_SLEEP /* * power management */ -- 2.0.1

2 1

[alsa-devel] [PATCH v2] ASoC: rt5640: add ASRC support
by bardliao＠realtek.com 07 Jul '14

07 Jul '14

From: Bard Liao <bardliao(a)realtek.com> This patch add ASRC support for rt5640 series codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 79 ++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5640.h | 3 ++ 2 files changed, 78 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6bc6efd..494b1b4 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1020,9 +1020,45 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_asrc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(w->codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (rt5640->asrc_en[RT5640_AIF1] || + rt5640->asrc_en[RT5640_AIF2]) { + snd_soc_update_bits(w->codec, + RT5640_DUMMY1, 0x70, 0x70); + snd_soc_update_bits(w->codec, + RT5640_JD_CTRL, 0x3, 0x3); + snd_soc_write(w->codec, RT5640_ASRC_1, 0x9b00); + snd_soc_write(w->codec, RT5640_ASRC_2, 0xf800); + } + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, RT5640_ASRC_1, 0x0); + snd_soc_write(w->codec, RT5640_ASRC_2, 0x0); + snd_soc_update_bits(w->codec, + RT5640_JD_CTRL, 0x3, 0); + snd_soc_update_bits(w->codec, + RT5640_DUMMY1, 0x70, 0); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY_S("ASRC Function", 1, SND_SOC_NOPM, + 0, 0, rt5640_asrc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), /* Input Side */ /* micbias */ SND_SOC_DAPM_SUPPLY("LDO2", RT5640_PWR_ANLG1, @@ -1276,6 +1312,9 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { + {"I2S1", NULL, "ASRC Function"}, + {"I2S2", NULL, "ASRC Function"}, + {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1636,10 +1675,17 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, rt5640->lrck[dai->id] = params_rate(params); pre_div = rl6231_get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); - if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", - rt5640->lrck[dai->id], dai->id); - return -EINVAL; + if ((rt5640->bclk_ratio[dai->id] != 64 && + rt5640->bclk_ratio[dai->id] != 32) || pre_div < 0) { + if (rt5640->sysclk < rt5640->lrck[dai->id] * 384) { + dev_err(codec->dev, "Unsupported clock setting"); + return -EINVAL; + } + dev_info(codec->dev, "Use ASRC\n"); + rt5640->asrc_en[dai->id] = true; + pre_div = 0; + } else { + rt5640->asrc_en[dai->id] = false; } frame_size = snd_soc_params_to_frame_size(params); if (frame_size < 0) { @@ -1698,6 +1744,17 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, return 0; } +static int rt5640_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + rt5640->asrc_en[dai->id] = false; + + return 0; +} + static int rt5640_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; @@ -1864,6 +1921,16 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, return 0; } +static int rt5640_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + rt5640->bclk_ratio[dai->id] = ratio; + + return 0; +} + static int rt5640_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1995,9 +2062,11 @@ static int rt5640_resume(struct snd_soc_codec *codec) static const struct snd_soc_dai_ops rt5640_aif_dai_ops = { .hw_params = rt5640_hw_params, + .hw_free = rt5640_hw_free, .set_fmt = rt5640_set_dai_fmt, .set_sysclk = rt5640_set_dai_sysclk, .set_pll = rt5640_set_dai_pll, + .set_bclk_ratio = rt5640_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5640_dai[] = { @@ -2214,6 +2283,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, } rt5640->hp_mute = 1; + rt5640->bclk_ratio[RT5640_AIF1] = 64; + rt5640->bclk_ratio[RT5640_AIF2] = 64; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, rt5640_dai, ARRAY_SIZE(rt5640_dai)); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 58ebe96..63b465d 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2095,6 +2095,9 @@ struct rt5640_priv { int pll_out; bool hp_mute; + + bool asrc_en[RT5640_AIFS]; + int bclk_ratio[RT5640_AIFS]; }; #endif -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH 1/3] ASoC: RT286: remove test code
by bardliao＠realtek.com 07 Jul '14

07 Jul '14

From: Bard Liao <bardliao(a)realtek.com> Remove test code. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index acfba9c..7c5f9d0 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -993,7 +993,6 @@ static int rt286_probe(struct snd_soc_codec *codec) codec->dapm.bias_level = SND_SOC_BIAS_OFF; rt286->codec = codec; - rt286->i2c->irq = 0; if (rt286->i2c->irq) { snd_soc_update_bits(codec, RT286_IRQ_CTRL, 0x2, 0x2); -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ALSA: hda - Add several entries for enabling HP mute led
by Hui Wang 07 Jul '14

07 Jul '14

BugLink: https://bugs.launchpad.net/bugs/1329580 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- So far, the mute led DMI string on these machines are HP_Mute_LED_0_0. sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 832c225..327413e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4854,6 +4854,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), /* ALC282 */ + SND_PCI_QUIRK(0x103c, 0x21f8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -4907,6 +4909,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8004, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), /* ALC290 */ SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x221c, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x221d, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2220, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2222, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Use devm_snd_soc_register_component
by Manish Badarkhe 07 Jul '14

07 Jul '14

Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe <badarkhe.manish(a)gmail.com> --- :100644 100644 efe2cd6... bfc8f8e... M sound/soc/omap/omap-mcbsp.c sound/soc/omap/omap-mcbsp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index efe2cd6..bfc8f8e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -805,7 +805,7 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, &omap_mcbsp_dai, 1); if (ret) return ret; @@ -817,8 +817,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) { struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - snd_soc_unregister_component(&pdev->dev); - if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); -- 1.7.10.4

2 1

[alsa-devel] [PATCH v5 0/5] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 07 Jul '14

07 Jul '14

Hi Lars, Mark and Liam, Here is the fifth version of the multicodec series. It mostly fixes the remaining comments from Lars on the series. It removes as well the support for soc-compress.c based on Vinod comment. Please note that I have a very simple setup based on BBB to check if the series is working, so it will be good to test it on various platforms to ensure this is not generating any regression. The series is based on asoc/for-next (v3.16-rc3). Test and comments are welcome. Thanks, Benoit v1: http://comments.gmane.org/gmane.linux.alsa.devel/120532 v2: http://comments.gmane.org/gmane.linux.alsa.devel/121024 - Split the first version in several patches. - Remove the TDM fixup v3: - Fixed all the places that were not iterating over all the codecs. - Change the structure and split it. v4: - Improve the TDM management - Split the orginal patch in smaller chunks - Ignore the CODEC to CODEC link case for the moment - Add iteration for multicodecs everywhere v5: - Remove multicodec support for soc-compress - Simplify the code a little thanks to Lars component series Benoit Cousson (5): ASoC: core: Add initial support for DAI multicodec ASoC: pcm: Add support for DAI multicodec ASoC: dapm: Add support for DAI multicodec ASoC: compress: Prevent multicodec for compressed stream ASoC: pcm: Add soc_dai_hw_params helper include/sound/soc-dai.h | 4 + include/sound/soc.h | 17 ++ sound/soc/soc-compress.c | 5 + sound/soc/soc-core.c | 293 +++++++++++++++++-------- sound/soc/soc-dapm.c | 95 ++++---- sound/soc/soc-pcm.c | 555 ++++++++++++++++++++++++++++++++--------------- 6 files changed, 665 insertions(+), 304 deletions(-) -- 1.9.1

3 11

[alsa-devel] [PATCH 00/19]: add mrfld DSP support
by Vinod Koul 07 Jul '14

07 Jul '14

Here is the split patch series for adding DSP support for Intel's merrfield platform. The last patch si mxer update patch which we have been discussing over the list. This is only for discussion and sake of complteness. For that patch we need the dapm_kcontrol_get/set series to do merged OR patch to be reworked after compenent series Vinod Koul (19): ASoC: Intel: add COMPILE_TEST to mfld machine ASoC: Intel: mfld_pcm: move stream handling to dai_ops ASoC: Intel: mfld-pcm rename period callback arg ASoc: Intel: mfld-pcm: report pcm delay ASoC: Intel: add the mrfld fw IPC definations ASoC: Intel: mfld-pcm: modularize stream allocation code ASoC: Intel: add mrfld pipelines ASoC: Intel: use common stream allocation method for compressed stream ASoC: Intel: mfld-pcm: add FE and BE ops ASoC: Intel: add mrfld DSP registers ASoC: intel: mfld-pcm: don't call trigger ops to DSP for internal streams ASoC: Intel: add generic parameter set interface ASoC: Intel: mrfld: add bytes control for modules ASoC: Intel: mrfld: add the gain controls ASoC: Intel: mfld-pcm: add control for powering up/down dsp ASoC: Intel: mrfld: add DSP core controls ASoC: Intel; mrfld: add the DSP DAPM widgets ASoC: Intel: mfld-pcm: add the fe & be dai ops ASoC: Intel: mrfld: add the DSP mixers arch/x86/include/asm/platform_sst_audio.h | 78 ++ sound/soc/intel/Kconfig | 2 +- sound/soc/intel/Makefile | 3 +- sound/soc/intel/sst-atom-controls.c | 1386 ++++++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.h | 906 +++++++++++++++++ sound/soc/intel/sst-mfld-dsp.h | 414 ++++++++- sound/soc/intel/sst-mfld-platform-compress.c | 11 +- sound/soc/intel/sst-mfld-platform-pcm.c | 492 +++++++--- sound/soc/intel/sst-mfld-platform.h | 47 +- 9 files changed, 3199 insertions(+), 140 deletions(-) create mode 100644 arch/x86/include/asm/platform_sst_audio.h create mode 100644 sound/soc/intel/sst-atom-controls.c create mode 100644 sound/soc/intel/sst-atom-controls.h

3 55

[alsa-devel] [PATCH] ALSA: hda - Fix and neaten print_nid_path/debug_badness
by Joe Perches 07 Jul '14

07 Jul '14

print_nid_path has a possible buffer overflow if struct nid_path.path values are > 256. Avoid this and neaten the output to remove the leading ':' Neaten debug_badness to always verify arguments. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/pci/hda/hda_generic.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 589e47c..b293583 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -350,16 +350,16 @@ static void print_nid_path(struct hda_codec *codec, const char *pfx, struct nid_path *path) { char buf[40]; + char *pos = buf; int i; + *pos = 0; + for (i = 0; i < path->depth; i++) + pos += scnprintf(pos, sizeof(buf) - (pos - buf), "%s%02x", + pos != buf ? ":" : "", + path->path[i]); - buf[0] = 0; - for (i = 0; i < path->depth; i++) { - char tmp[4]; - sprintf(tmp, ":%02x", path->path[i]); - strlcat(buf, tmp, sizeof(buf)); - } - codec_dbg(codec, "%s path: depth=%d %s\n", pfx, path->depth, buf); + codec_dbg(codec, "%s path: depth=%d '%s'\n", pfx, path->depth, buf); } /* called recursively */ @@ -1700,9 +1700,11 @@ static int fill_and_eval_dacs(struct hda_codec *codec, #define DEBUG_BADNESS #ifdef DEBUG_BADNESS -#define debug_badness(fmt, args...) codec_dbg(codec, fmt, ##args) +#define debug_badness(fmt, ...) \ + codec_dbg(codec, fmt, ##__VA_ARGS__) #else -#define debug_badness(...) +#define debug_badness(fmt, ...) \ + do { if (0) codec_dbg(codec, fmt, ##__VA_ARGS__); } while (0) #endif #ifdef DEBUG_BADNESS

2 1

[alsa-devel] USB-Audio.conf comment regarding asound.conf
by Anssi Hannula 07 Jul '14

07 Jul '14

Hi, USB-Audio.conf contains: > # DO NO EDIT; this is an internal ALSA file. > # If you want to add your own definitions, put them into /etc/asound.conf or > # ~/.asoundrc, with "cards." before the "USB-Audio", e.g.: > # > # cards.USB-Audio.pcm.use_dmix."NoiseBlaster 3000" no > # > # If your device requires such a definition to work correctly, please report it > # to <alsa-devel(a)alsa-project.org>. However, snd_config_hook_load_for_all_cards() will completely skip loading of /usr/share/alsa/cards/USB-Audio.conf if one has existing configuration elements in cards.USB-Audio. I was just hit with this while debugging an issue with a user. Should users just edit /usr/share/alsa/cards/USB-Audio.conf directly until such entries are added upstream, or should this be somehow fixed in alsa-lib to work as instructed? Related commits: http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=96da0c842d14b40c… http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=133819d420d962b6… -- Anssi Hannula

1 0

[alsa-devel] [PATCH] amixer: Fix convert_prange1 not rounding correctly
by Josef Gajdusek 06 Jul '14

06 Jul '14

The convert_prange1 macro was not the exact inverse of convert_prange. This was causing the volume percentage set by 'amixer set' to differ from the one displayed by 'amixer get'. Signed-off-by: Josef Gajdusek <atx(a)atx.name> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index cf82892..577a2ca 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -201,7 +201,7 @@ static int convert_prange(long val, long min, long max) /* Function to convert from percentage to volume. val = percentage */ #define convert_prange1(val, min, max) \ - ceil((val) * ((max) - (min)) * 0.01 + (min)) + rint((val) * ((max) - (min)) * 0.01 + (min)) struct volume_ops { int (*get_range)(snd_mixer_elem_t *elem, long *min, long *max); -- 1.8.5.5

1 0

[alsa-devel] [PATCH] ASoC: s6105-ipcam: Fix build for I2C dependencies
by Mark Brown 05 Jul '14

05 Jul '14

From: Mark Brown <broonie(a)linaro.org> The s6105-ipcam driver calls i2c_register_board_info() which means that not only does it need I2C to be enabled to be built but since that symbol is not exported it means that the driver also can't be linked as a module. Use a bool instead of a tristate and add a dependency on I2C to fix these issues. Commit b5dd60fd263 (ASoC: s6000: Allow to build when COMPILE_TEST is enabled) made this problem more visible since it enabled build of the driver on more common architectures but this has always been an issue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/s6000/Kconfig | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig index 4823e1e4daef..f244a2566f20 100644 --- a/sound/soc/s6000/Kconfig +++ b/sound/soc/s6000/Kconfig @@ -15,8 +15,9 @@ config SND_S6000_SOC_I2S tristate config SND_S6000_SOC_S6IPCAM - tristate "SoC Audio support for Stretch 6105 IP Camera" - depends on SND_S6000_SOC + bool "SoC Audio support for Stretch 6105 IP Camera" + depends on SND_S6000_SOC=y + depends on I2C=y depends on XTENSA_PLATFORM_S6105 || COMPILE_TEST select SND_S6000_SOC_I2S select SND_SOC_TLV320AIC3X -- 2.0.0

1 0

[alsa-devel] [PATCH 1/2] ASoC: max98090: Add check for CODEC type
by Tushar Behera 04 Jul '14

04 Jul '14

CODEC type (MAX98090/MAX98091) can be specified from device-tree file, it can also be obtained from the CODEC during runtime. Add an explicit check to figure out if both are matching, else print a message warning about the same. Signed-off-by: Tushar Behera <tushar.b(a)samsung.com> --- sound/soc/codecs/max98090.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ca74f502..1ed06d9 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2215,6 +2215,7 @@ static int max98090_probe(struct snd_soc_codec *codec) { struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); struct max98090_cdata *cdata; + enum max98090_type devtype; int ret = 0; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2250,16 +2251,21 @@ static int max98090_probe(struct snd_soc_codec *codec) } if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { - max98090->devtype = MAX98091; + devtype = MAX98091; dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); } else { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); } + if (max98090->devtype != devtype) { + dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n"); + max98090->devtype = devtype; + } + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: samsung: Correct I2S DAI suspend/resume ops
by Sylwester Nawrocki 04 Jul '14

04 Jul '14

We should save/restore relevant I2S registers regardless of the dai->active flag, otherwise some settings are being lost after system suspend/resume cycle. E.g. I2S slave mode set only during dai initialization is not preserved and the device ends up in master mode after system resume. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2ac76fa..5f9b255 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -920,11 +920,9 @@ static int i2s_suspend(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); - i2s->suspend_i2scon = readl(i2s->addr + I2SCON); - i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); - } + i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); + i2s->suspend_i2scon = readl(i2s->addr + I2SCON); + i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); return 0; } @@ -933,11 +931,9 @@ static int i2s_resume(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - writel(i2s->suspend_i2scon, i2s->addr + I2SCON); - writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); - writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); - } + writel(i2s->suspend_i2scon, i2s->addr + I2SCON); + writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); + writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: sirf: provide pm-runtime functions when needed
by Arnd Bergmann 04 Jul '14

04 Jul '14

The newly introduced sirf-usp driver defines sirf_usp_pcm_{suspend,resume} functions only when PM_RUNTIME is enabled, but also uses them when that is disabled and only PM_SLEEP is turned on, resulting in this error: ../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_suspend': ../sound/soc/sirf/sirf-usp.c:308:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_suspend' [-Werror=implicit-function-declaration] sirf_usp_pcm_runtime_suspend(dev); ^ ../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_resume': ../sound/soc/sirf/sirf-usp.c:319:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_resume' [-Werror=implicit-function-declaration] ret = sirf_usp_pcm_runtime_resume(dev); ^ cc1: some warnings being treated as errors To fix that, this patch changes the #ifdef to CONFIG_PM, which is enabled when at least one of PM_SLEEP or PM_RUNTIME are enabled. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index bdf6aae..9693bc2 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -282,7 +282,7 @@ static struct snd_soc_dai_driver sirf_usp_pcm_dai = { .ops = &sirf_usp_pcm_dai_ops, }; -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int sirf_usp_pcm_runtime_suspend(struct device *dev) { struct sirf_usp *usp = dev_get_drvdata(dev);

2 1

[alsa-devel] [PATCH 1/3] ASoC: s6000-i2s: Fix s6000_i2s_remove() return type
by Lars-Peter Clausen 04 Jul '14

04 Jul '14

The platform_driver remove callback return type is int not void. Fixes the following warning: sound/soc/s6000/s6000-i2s.c:604:19: warning: incorrect type in initializer (different base types) sound/soc/s6000/s6000-i2s.c:604:19: expected int ( *remove )( ... ) sound/soc/s6000/s6000-i2s.c:604:19: got void ( static [toplevel] *<noident>)( ... ) Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/s6000/s6000-i2s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 7eba797..1c8d011 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -570,7 +570,7 @@ err_release_none: return ret; } -static void s6000_i2s_remove(struct platform_device *pdev) +static int s6000_i2s_remove(struct platform_device *pdev) { struct s6000_i2s_dev *dev = dev_get_drvdata(&pdev->dev); struct resource *region; @@ -597,6 +597,8 @@ static void s6000_i2s_remove(struct platform_device *pdev) iounmap(mmio); region = platform_get_resource(pdev, IORESOURCE_IO, 0); release_mem_region(region->start, resource_size(region)); + + return 0; } static struct platform_driver s6000_i2s_driver = { -- 1.8.0

2 3

[alsa-devel] [PATCH v3 0/5] ALSA/ASoC/dmaengine: Fix 3 bytes physical sample support
by Peter Ujfalusi 04 Jul '14

04 Jul '14

Hi, Changes since v2 (in patch 3): - Replace BIT(bits / 8) with (1 << (bits / 8)) as pointed out by Takashi - Use switch (addr_widths) to filter non valid physical widths, like -EINVAL or widths not dividable by 8. The same check could be done with: if (bits > 0 && bits <=64 && !(bits % 8) && (addr_widths & (1 << (bits / 8)))) But I think the switch() implementation is more readable. Changes since v1: - the hw.formats mask preparation moved to dmaengine_pcm_set_runtime_hwparams() - we prepare the mask for all supported sample formats based on the physical width of the sample and on the slave_caps information. I have noticed while switching the daVinci audio to dmaengine_pcm that the _3LE and _3BE support is not working correctly via dmaengine_pcm: In case of _3LE/_3BE formats the samples are stored in 3 consecutive bytes without padding it to 4 bytes. This means that the DMA needs to be able to support 3 bytes word length in order to read/write the samples from memory correctly. Originally the code treated 24 bits physical length samples as they were 32 bits which leads to corruption when playing or recording audio. To fix the support for 3 bytes physical samples the dma driver also need to have support for such data arrangement. eDMA does have support for it in HW and the legacy davinci-pcm platform driver w/o dmaengine was able to support such formats. First step is to add DMA_SLAVE_BUSWIDTH_3_BYTES to dma_slave_buswidth for engines and users to select 3 bytes as bus width. Followed by: In dmaengine_pcm_set_runtime_hwparams() we should check the slave_caps of the dma if it supports the give sample physical width of the sample. Based on this information we initialize the hw.formats: masking out all non supported formats based on the physical width. In case the dma driver does not implement the slave_caps the default assumption is that it supports 1, 2 and 4 bytes widths. Regards, Peter --- Peter Ujfalusi (5): dma: Support for 3 bytes word size ALSA: pcm_dmaengine: Use the available wrapper to get physical width ASoC: generic-dmaengine-pcm: Prepare formats mask for valid physical sample sizes ALSA: pcm_dmaengine: Correct support for 24bits physical sample widths dma: edma: Declare DMA_SLAVE_BUSWIDTH_3_BYTES as supported buswidth drivers/dma/edma.c | 1 + include/linux/dmaengine.h | 1 + sound/core/pcm_dmaengine.c | 4 +++- sound/soc/soc-generic-dmaengine-pcm.c | 37 ++++++++++++++++++++++++++++++++++- 4 files changed, 41 insertions(+), 2 deletions(-) -- 2.0.0

5 17

[alsa-devel] [PATCH v2] regulator: arizona-ldo1: remove bypass functionality
by Nikesh Oswal 04 Jul '14

04 Jul '14

WM5110/8280 devices do not support bypass mode for LDO1 so remove the bypass callbacks registered with regulator core. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- drivers/regulator/arizona-ldo1.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/drivers/regulator/arizona-ldo1.c b/drivers/regulator/arizona-ldo1.c index 04f262a..4c9db58 100644 --- a/drivers/regulator/arizona-ldo1.c +++ b/drivers/regulator/arizona-ldo1.c @@ -143,8 +143,6 @@ static struct regulator_ops arizona_ldo1_ops = { .map_voltage = regulator_map_voltage_linear, .get_voltage_sel = regulator_get_voltage_sel_regmap, .set_voltage_sel = regulator_set_voltage_sel_regmap, - .get_bypass = regulator_get_bypass_regmap, - .set_bypass = regulator_set_bypass_regmap, }; static const struct regulator_desc arizona_ldo1 = { -- 1.7.9.5

2 1

[alsa-devel] [PATCH v7] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 04 Jul '14

04 Jul '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds Realtek ALC286 codec driver. ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. We map verb command to regmap structure. However, we read most registers from cache to prevent the asymmetry read/write issue in rt286. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)intel.com> --- The jack detection function is done by Lewandowski Gustaw. So I add Gustaw's Signed-off-by line in this patch The difference between this version and previous version is listed below. * Use regmap for IO control. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1208 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 193 ++++++++ 5 files changed, 1426 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..eb773d1 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7541fa7..1c5eee3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,6 +74,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C @@ -451,6 +452,9 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5651=m default m if SND_SOC_RT5677=m +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1c..3302f95 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -235,6 +236,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..acfba9c --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1208 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/rt286.h> +#include <sound/hda_verbs.h> + +#include "rt286.h" + +#define RT286_VENDOR_ID 0x10ec0286 + +struct rt286_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + struct reg_default *index_cache; +}; + +static struct reg_default rt286_index_def[] = { + { 0x01, 0xaaaa }, + { 0x02, 0x8aaa }, + { 0x03, 0x0002 }, + { 0x04, 0xaf01 }, + { 0x08, 0x000d }, + { 0x09, 0xd810 }, + { 0x0a, 0x0060 }, + { 0x0b, 0x0000 }, + { 0x0f, 0x0000 }, + { 0x19, 0x0a17 }, + { 0x20, 0x0020 }, + { 0x33, 0x0208 }, + { 0x49, 0x0004 }, + { 0x4f, 0x50e9 }, + { 0x50, 0x2c00 }, + { 0x63, 0x2902 }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) + +static const struct reg_default rt286_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x0000007f }, + { 0x0023a000, 0x0000007f }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x000000c3 }, + { 0x00936000, 0x000000c3 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000097 }, + { 0x00b37200, 0x00000097 }, + { 0x00b37300, 0x00000097 }, + { 0x00c37000, 0x00000000 }, + { 0x00c37100, 0x00000080 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01439000, 0x00000080 }, + { 0x0143a000, 0x00000080 }, + { 0x01470700, 0x00000000 }, + { 0x01470500, 0x00000400 }, + { 0x01470c00, 0x00000000 }, + { 0x01470100, 0x00000000 }, + { 0x01837000, 0x00000000 }, + { 0x01870500, 0x00000400 }, + { 0x02050000, 0x00000000 }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000000 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x02040000, 0x00004002 }, + { 0x01870700, 0x00000020 }, + { 0x00830000, 0x000000c3 }, + { 0x00930000, 0x000000c3 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt286_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_PROC_COEF: + return true; + default: + return false; + } + + +} + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_SET_AUDIO_POWER: + case RT286_SET_HPO_POWER: + case RT286_SET_SPK_POWER: + case RT286_SET_DMIC1_POWER: + case RT286_SPK_MUX: + case RT286_HPO_MUX: + case RT286_ADC0_MUX: + case RT286_ADC1_MUX: + case RT286_SET_MIC1: + case RT286_SET_PIN_HPO: + case RT286_SET_PIN_SPK: + case RT286_SET_PIN_DMIC1: + case RT286_SPK_EAPD: + case RT286_SET_AMP_GAIN_HPO: + case RT286_SET_DMIC2_DEFAULT: + case RT286_DACL_GAIN: + case RT286_DACR_GAIN: + case RT286_ADCL_GAIN: + case RT286_ADCR_GAIN: + case RT286_MIC_GAIN: + case RT286_SPOL_GAIN: + case RT286_SPOR_GAIN: + case RT286_HPOL_GAIN: + case RT286_HPOR_GAIN: + case RT286_F_DAC_SWITCH: + case RT286_F_RECMIX_SWITCH: + case RT286_REC_MIC_SWITCH: + case RT286_REC_I2S_SWITCH: + case RT286_REC_LINE_SWITCH: + case RT286_REC_BEEP_SWITCH: + case RT286_DAC_FORMAT: + case RT286_ADC_FORMAT: + case RT286_COEF_INDEX: + case RT286_PROC_COEF: + case RT286_SET_AMP_GAIN_ADC_IN1: + case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_POWER(RT286_DAC_OUT1): + case RT286_SET_POWER(RT286_DAC_OUT2): + case RT286_SET_POWER(RT286_ADC_IN1): + case RT286_SET_POWER(RT286_ADC_IN2): + case RT286_SET_POWER(RT286_DMIC2): + case RT286_SET_POWER(RT286_MIC1): + return true; + default: + return false; + } +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + struct rt286_priv *rt286 = i2c_get_clientdata(client); + u8 data[4]; + int ret, i; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + if (reg == rt286->index_cache[i].reg) { + rt286->index_cache[i].def = value; + break; + } + + } + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + + if (ret == 4) + return 0; + else + pr_err("ret=%d\n", ret); + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + } + + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static void rt286_index_sync(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + } +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL2, 0x4, 0x4); + snd_soc_write(codec, RT286_SET_MIC1, 0x24); + val = snd_soc_read(codec, RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = snd_soc_read(codec, + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + snd_soc_write(codec, + RT286_SET_MIC1, 0x20); + /* power off HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x1001); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } else { + *mic = false; + } + + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE); + *mic = buf & 0x80000000; + } + + return 0; +} + +static void rt286_jack_detect_work(struct work_struct *work) +{ + struct rt286_priv *rt286 = + container_of(work, struct rt286_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, + RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN, + RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK, + RT286_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN, + RT286_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN, + RT286_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt286_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum); + +static const char * const rt286_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, + RT286_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + else + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, + RT286_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + /* Clear IRQ */ + snd_soc_update_bits(rt286->codec, + RT286_IRQ_CTRL, 0x1, 0x1); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i, ret; + + ret = snd_soc_read(codec, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID)); + if (ret != RT286_VENDOR_ID) { + dev_err(codec->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + + snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + snd_soc_write(codec, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + snd_soc_write(codec, RT286_CBJ_CTRL2, 0x0000); + snd_soc_write(codec, RT286_MIC1_DET_CTRL, 0x0816); + snd_soc_write(codec, RT286_MISC_CTRL1, 0x0000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x8, 0x0); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + rt286->i2c->irq = 0; + if (rt286->i2c->irq) { + snd_soc_update_bits(codec, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(codec->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt286->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt286_suspend(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, true); + regcache_mark_dirty(rt286->regmap); + + return 0; +} + +static int rt286_resume(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, false); + rt286_index_sync(codec); + regcache_sync(rt286->regmap); + + return 0; +} +#else +#define rt286_suspend NULL +#define rt286_resume NULL +#endif + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt286-aif2", + .id = RT286_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .suspend = rt286_suspend, + .resume = rt286_resume, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x02370100, + .volatile_reg = rt286_volatile_register, + .readable_reg = rt286_readable_register, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt286_reg, + .num_reg_defaults = ARRAY_SIZE(rt286_reg), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static const struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt286->index_cache = rt286_index_def; + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt286); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..21c570f --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,193 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_SET_PIN_SFT 6 +#define RT286_SET_PIN_ENABLE 0x40 +#define RT286_SET_PIN_DISABLE 0 +#define RT286_SET_EAPD_HIGH 0x2 +#define RT286_SET_EAPD_LOW 0 + +#define RT286_MUTE_SFT 7 + +/* Verb commands */ +#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT) +#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT) +#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1) +#define RT286_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0) +#define RT286_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0) +#define RT286_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0) +#define RT286_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0) +#define RT286_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0) +#define RT286_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0) +#define RT286_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0) +#define RT286_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) +#define RT286_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000) +#define RT286_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000) +#define RT286_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000) +#define RT286_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000) +#define RT286_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000) +#define RT286_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000) +#define RT286_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000) +#define RT286_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000) +#define RT286_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100) +#define RT286_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000) +#define RT286_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100) +#define RT286_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200) +#define RT286_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300) +#define RT286_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0) +#define RT286_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0) +#define RT286_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0) +#define RT286_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ + -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v4 0/8] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 04 Jul '14

04 Jul '14

Hi Lars, Mark and Liam, Sorry for the delay. Here is the fourth version of the multicodec series. I had to split it in smaller chunks since the previous patch was exceeding the size limit of the ML :-( I think (and hope) I did address all the v3 comments from Lars. I did not address the codec to codec link in multicodec case as suggested by Lars. There is still an open point about enforcing the number of channels for the CPU dai in case the TDM slots are set for the CODEC dais. I guess this point could be addressed later. Please note that I have a very simple setup based on BBB to check if the series is working, so it will be good to test it on various platforms to ensure this is not generating any regression. The series is based on asoc/for-next (v3.16-rc1). Test and comments are welcome. Thanks, Benoit v1: http://comments.gmane.org/gmane.linux.alsa.devel/120532 v2: http://comments.gmane.org/gmane.linux.alsa.devel/121024 - Split the first version in several patches. - Remove the TDM fixup v3: - Fixed all the places that were not iterating over all the codecs. - Change the structure and split it. v4: - Improve the TDM management - Split the orginal patch in smaller chunks - Ignore the CODEC to CODEC link case for the moment - Add iteration for multicodecs everywhere Benoit Cousson (8): ASoC: core: Change soc_link_dai_widgets signature for multiple codecs ASoC: pcm: Refactor soc_pcm_apply_msb for multicodecs ASoC: core: Add initial support for DAI multicodec ASoC: pcm: Add support for DAI multicodec ASoC: dapm: Add support for DAI multicodec ASoC: compress: Add support for DAI multicodec ASoC: pcm: Add soc_dai_hw_params helper ASoC: core: Add a warning for link_dai_widget in the multicodec case include/sound/soc-dai.h | 4 + include/sound/soc.h | 17 ++ sound/soc/soc-compress.c | 94 +++++--- sound/soc/soc-core.c | 348 +++++++++++++++++++---------- sound/soc/soc-dapm.c | 93 ++++---- sound/soc/soc-pcm.c | 563 ++++++++++++++++++++++++++++++++--------------- 6 files changed, 751 insertions(+), 368 deletions(-) -- 1.9.1

5 42

[alsa-devel] ASoC: samsung: MACH_SMDK6450
by Paul Bolle 04 Jul '14

04 Jul '14

Kukjin, Your commit 0aeaa68cf509 ("ASoC: samsung: no more support for S5P6440 and S5P6450 SoCs") landed in next-20140702. It removed references to MACH_SMDK6440 and MACH_SMDK6450 from sound/soc/samsung/. It seems to have missed one reference to MACH_SMDK6450 in sound/soc/samsung/Kconfig (an optional dependency of SND_SOC_SMDK_WM8580_PCM). Is the oneliner to remove that optional dependency queued somewhere? Paul Bolle

5 7

[alsa-devel] [PATCH 1/2] doc: dt bindings: Document Odroid X2/U3 audio subsystem bindings
by Sylwester Nawrocki 04 Jul '14

04 Jul '14

Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../bindings/sound/samsung,odroidx2-max98090.txt | 32 ++++++++++++++++++++ 1 file changed, 32 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt new file mode 100644 index 0000000..b37e79a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt @@ -0,0 +1,32 @@ +Odroid X2/U3 audio complex + +Required properties: +- compatible : "samsung,odroidx2-audio" +- samsung,i2s-controller : phandle to the I2S controller +- samsung,audio-codec : phandle to the audio codec + +Example: + + i2s0: i2s@03830000 { + ... + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_DOUT_AUD_BUS>; + clock-names = "iis", "i2s_opclk0"; + status = "okay"; + }; + + i2c@13870000 { + ... + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx0>; + interrupts = <1 0>; + }; + }; + + sound { + compatible = "samsung,odroidx2-audio"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98090>; + }; -- 1.7.9.5

3 10

[alsa-devel] Fwd: Re: About your snd-firewire-improve
by siickwkc＠gmail.com 04 Jul '14

04 Jul '14

-------- Message original -------- Sujet: Re: About your snd-firewire-improve Date : Sun, 29 Jun 2014 00:41:04 +0900 De : Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Pour : siickwkc(a)gmail.com <siickwkc(a)gmail.com> Hi Siick, (Jun 28 2014 20:18), siickwkc(a)gmail.com wrote: > Firstly, sorry for my english, i'm a french so i will make some mistakes. Don't worry. I'm also not good at english because I'm Japanese. > I'm a ubuntu long time user and i have a M-audio firewire audiophile for > MAO. Right now, i can just use it on Windows. That's why your project > have my interest. M-Audio Firewire Audiophile is one of my test devices. It means I own it and use it to debug my drivers. As long as I know, my driver (snd-bebob) can drive it. > I've installed like you explain it on you readme file here > > https://github.com/takaswie/snd-firewire-improve You can also use DKMS package from David Henningsson's PPA. He's a developer for ALSA/Pulseaudio and familiar with my developing. This package is for trusty. https://launchpad.net/~diwic/+archive/snd-firewire-improve > And now when i switch on my firewire souncard, it's different, it seems > to "work" because i don't have flashing light as usual. > I switch on, lights flash, and then it turns blue without flashing, like > on windows. It's correct behaviour. In detail, please see these patches: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075885.html http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075888.html > But now i try to make it works with audacious and it doesn't. > I try to configure audacious with ALSA out but in parameters all > firewire device listed seems don't work.... About "snd_mixer_find_selem failed" message, this behaviour is correct because my drivers don't have ALSA Control Interface. This means users cannot control internal mixer via ALSA applications such like alsamixer. For this purpose, please use ffado-dbus-server/ffado-mixer. But Ubuntu package for FFADO is a bit old. I upload the package with latest source in my PPA for trusty. https://launchpad.net/~mocchi/+archive/instruments Or you should use upstream source. http://subversion.ffado.org/wiki/InstallingFfadoFromSource If you're successful to install, you can see GUI such like in this page: http://subversion.ffado.org/wiki/MaudioBebob About "snd_pcm_hw_params_set_channels failed", this behaviour is correct to your device because AudioPhile can just playback 6ch PCM, while you try to playback 2ch PCM. There is a mismatch of channels. Instead of using "hw:2,0" PCM device, use "plughw:2,0" PCM device. This PCM device can arrange channels and solve the mismatch. > I'm not familiar with all audio stuff on Linux, maybe i've done > something wrong? > I attached you screenshot maybe it can help.... > Thanks for your reading and your future answer. No, you did correct things. Thanks for your report. If possible, would you please post your questions to alsa-devel? I believe our conversation may help to the other users. http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] [PATCH] ASoC: rsnd: fixup index of SSI mod when SRC is used
by Jürg Billeter 04 Jul '14

04 Jul '14

The default index 1 was used as the loop was terminated before the following code could be reached: if (mod[i] == this) index = i; Signed-off-by: Jürg Billeter <j(a)bitron.ch> --- sound/soc/sh/rcar/core.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4e86265..d737fea 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -295,9 +295,11 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, mod[0] = NULL; /* for "mem" */ index = 1; for (i = 1; i < MOD_MAX; i++) { - if (!src) { - mod[i] = ssi; + if (!ssi) { break; + } else if (!src) { + mod[i] = ssi; + ssi = NULL; } else if (!dvc) { mod[i] = src; src = NULL; -- 2.0.1

2 4

[alsa-devel] [PATCH v3 2/2] ALSA: hda - restore BCLK M/N value as per CDCLK for HSW/BDW display HDA controller
by mengdong.lin＠intel.com 04 Jul '14

04 Jul '14

From: Mengdong Lin <mengdong.lin(a)intel.com> For HSW/BDW display HD-A controller, hda_set_bclk() is defined to set BCLK by programming the M/N values as per the core display clock (CDCLK) queried from i915 display driver. And the audio driver will also set BCLK in azx_first_init() since the display driver can turn off the shared power in boot phase if only eDP is connected and M/N values will be lost and must be reprogrammed. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index e9e8a4a..8b4940b 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -20,10 +20,20 @@ #include <linux/module.h> #include <sound/core.h> #include <drm/i915_powerwell.h> +#include "hda_priv.h" #include "hda_i915.h" +/* Intel HSW/BDW display HDA controller Extended Mode registers. + * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display + * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled. + */ +#define ICH6_REG_EM4 0x100c +#define ICH6_REG_EM5 0x1010 + static int (*get_power)(void); static int (*put_power)(void); +static int (*get_cdclk)(void); int hda_display_power(bool enable) { @@ -38,6 +48,43 @@ int hda_display_power(bool enable) return put_power(); } +void haswell_set_bclk(struct azx *chip) +{ + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!get_cdclk) + return; + + cdclk_freq = get_cdclk(); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } + + azx_writew(chip, EM4, bclk_m); + azx_writew(chip, EM5, bclk_n); +} + + int hda_i915_init(void) { int err = 0; @@ -55,6 +102,10 @@ int hda_i915_init(void) return -ENODEV; } + get_cdclk = symbol_request(i915_get_cdclk_freq); + if (!get_cdclk) /* may have abnormal BCLK and audio playback rate */ + pr_warn("hda-i915: get_cdclk symbol get fail\n"); + pr_debug("HDA driver get symbol successfully from i915 module\n"); return err; @@ -70,6 +121,10 @@ int hda_i915_exit(void) symbol_put(i915_release_power_well); put_power = NULL; } + if (get_cdclk) { + symbol_put(i915_get_cdclk_freq); + get_cdclk = NULL; + } return 0; } diff --git a/sound/pci/hda/hda_i915.h b/sound/pci/hda/hda_i915.h index bfd835f..e6072c6 100644 --- a/sound/pci/hda/hda_i915.h +++ b/sound/pci/hda/hda_i915.h @@ -18,10 +18,12 @@ #ifdef CONFIG_SND_HDA_I915 int hda_display_power(bool enable); +void haswell_set_bclk(struct azx *chip); int hda_i915_init(void); int hda_i915_exit(void); #else static inline int hda_display_power(bool enable) { return 0; } +static inline void haswell_set_bclk(struct azx *chip) { return; } static inline int hda_i915_init(void) { return -ENODEV; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 25753db..b6b4e71 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -62,9 +62,9 @@ #include <linux/vga_switcheroo.h> #include <linux/firmware.h> #include "hda_codec.h" -#include "hda_i915.h" #include "hda_controller.h" #include "hda_priv.h" +#include "hda_i915.h" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; @@ -288,21 +288,8 @@ static char *driver_short_names[] = { [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; - -/* Intel HSW/BDW display HDA controller Extended Mode registers. - * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display - * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled. - */ -#define ICH6_REG_EM4 0x100c -#define ICH6_REG_EM5 0x1010 - struct hda_intel { struct azx chip; - - /* HSW/BDW display HDA controller to restore BCLK from CDCLK */ - unsigned int bclk_m; - unsigned int bclk_n; }; @@ -598,22 +585,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -static void haswell_save_bclk(struct azx *chip) -{ - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - - hda->bclk_m = azx_readw(chip, EM4); - hda->bclk_n = azx_readw(chip, EM5); -} - -static void haswell_restore_bclk(struct azx *chip) -{ - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - - azx_writew(chip, EM4, hda->bclk_m); - azx_writew(chip, EM5, hda->bclk_n); -} - #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -641,12 +612,6 @@ static int azx_suspend(struct device *dev) chip->irq = -1; } - /* Save BCLK M/N values before they become invalid in D3. - * Will test if display power well can be released now. - */ - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - haswell_save_bclk(chip); - if (chip->msi) pci_disable_msi(chip->pci); pci_disable_device(pci); @@ -668,7 +633,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); - haswell_restore_bclk(chip); + haswell_set_bclk(chip); } pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -713,10 +678,9 @@ static int azx_runtime_suspend(struct device *dev) azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - haswell_save_bclk(chip); + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); - } + return 0; } @@ -736,7 +700,7 @@ static int azx_runtime_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); - haswell_restore_bclk(chip); + haswell_set_bclk(chip); } /* Read STATESTS before controller reset */ @@ -1426,6 +1390,10 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); + + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + haswell_set_bclk(chip); + azx_init_chip(chip, (probe_only[dev] & 2) == 0); /* codec detection */ -- 1.8.1.2

2 1

Re: [alsa-devel] ALC286 recording delay
by Takashi Iwai 04 Jul '14

04 Jul '14

At Thu, 3 Jul 2014 15:30:17 +0800, Kailang wrote: > > > > At Mon, 30 Jun 2014 16:25:46 +0800, > > Kailang wrote: > >> > >> Hi Takashi, > >> > >> It will be recording voice delay for resume back recording for Headset > >> Mic. > >> This patch doesn't disable pin for suspend state. The issue was fixed. > >> This is follow ALC286 programing guide. > > > > Please make alc_snd_hda_shutup_pins() static and name it a bit more > > specific. > > Could you have idea for the name? > alc_shutup_pins() The name can indicate the purpose, e.g. alc_shutup_pins_except_extmic(). > > Also, it'd be helpful if you explain a bit more *why* this fixes the > > recording voice delay. > > > Our AE try to tune the mic cap to clear issue. > But he say, keep MIC input enable will be better. But for which purpose should we keep MIC input? This should be more elaborated. Takashi

1 0

[alsa-devel] [PATCH 1/3] ASoC: twl4030: Remove unused variable
by Sachin Kamat 03 Jul '14

03 Jul '14

'status' is not used in the function. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/codecs/twl4030.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 955df35c5ab6..cf404505e7d1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -344,17 +344,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status = -1; if (enable) { twl4030->apll_enabled++; if (twl4030->apll_enabled == 1) - status = twl4030_audio_enable_resource( + twl4030_audio_enable_resource( TWL4030_AUDIO_RES_APLL); } else { twl4030->apll_enabled--; if (!twl4030->apll_enabled) - status = twl4030_audio_disable_resource( + twl4030_audio_disable_resource( TWL4030_AUDIO_RES_APLL); } } -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: adau1701: fix adau1701_reg_read()
by Daniel Mack 03 Jul '14

03 Jul '14

Fix a long standing bug in the read register routing of adau1701. The bytes arrive in the buffer in big-endian, so the result has to be shifted before and-ing the bytes in the loop. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/adau1701.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index defb277..15feb12 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -240,8 +240,10 @@ static int adau1701_reg_read(void *context, unsigned int reg, *value = 0; - for (i = 0; i < size; i++) - *value |= recv_buf[i] << (i * 8); + for (i = 0; i < size; i++) { + *value <<= 8; + *value |= recv_buf[i]; + } return 0; } -- 1.9.3

3 2

[alsa-devel] [PATCH] regulator: arizona-ldo1: remove bypass functionality
by Nikesh Oswal 03 Jul '14

03 Jul '14

From: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> WM5110/8280 devices do not support bypass mode for LDO1 so remove the bypass callbacks registered with regulator core. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- drivers/regulator/arizona-ldo1.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/drivers/regulator/arizona-ldo1.c b/drivers/regulator/arizona-ldo1.c index 04f262a..4c9db58 100644 --- a/drivers/regulator/arizona-ldo1.c +++ b/drivers/regulator/arizona-ldo1.c @@ -143,8 +143,6 @@ static struct regulator_ops arizona_ldo1_ops = { .map_voltage = regulator_map_voltage_linear, .get_voltage_sel = regulator_get_voltage_sel_regmap, .set_voltage_sel = regulator_set_voltage_sel_regmap, - .get_bypass = regulator_get_bypass_regmap, - .set_bypass = regulator_set_bypass_regmap, }; static const struct regulator_desc arizona_ldo1 = { -- 1.7.9.5

2 1

[alsa-devel] [PATCH v3] ASoC: tas5086: add regulator consumer support
by Daniel Mack 03 Jul '14

03 Jul '14

The TAS5086 has two power domains, DVDD and AVDD. Enable them both as long as the codec is in use. Also, switch on the power to identify the chip at device probe level, and switch it off again afterwards. The codec level will take care for power handling later. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- This has been sitting in my tree since I forgot to resend it in March. .../devicetree/bindings/sound/ti,tas5086.txt | 5 ++ sound/soc/codecs/tas5086.c | 67 +++++++++++++++++++--- 2 files changed, 63 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt index d2866a0..234dad2 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas5086.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -31,6 +31,9 @@ Optional properties: Most systems should not set any of these properties. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + Examples: i2c_bus { @@ -39,5 +42,7 @@ Examples: reg = <0x1b>; reset-gpio = <&gpio 23 0>; ti,charge-period = <156000>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; }; }; diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index d48491a..be7194b 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -36,6 +36,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> +#include <linux/regulator/consumer.h> #include <linux/spi/spi.h> #include <linux/of.h> #include <linux/of_device.h> @@ -240,6 +241,10 @@ static int tas5086_reg_read(void *context, unsigned int reg, return 0; } +static const char * const supply_names[] = { + "dvdd", "avdd" +}; + struct tas5086_private { struct regmap *regmap; unsigned int mclk, sclk; @@ -251,6 +256,7 @@ struct tas5086_private { int rate; /* GPIO driving Reset pin, if any */ int gpio_nreset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; @@ -773,6 +779,8 @@ static int tas5086_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; } @@ -781,6 +789,10 @@ static int tas5086_soc_resume(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) + return ret; + tas5086_reset(priv); regcache_mark_dirty(priv->regmap); @@ -812,6 +824,12 @@ static int tas5086_probe(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int i, ret; + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + priv->pwm_start_mid_z = 0; priv->charge_period = 1300000; /* hardware default is 1300 ms */ @@ -832,16 +850,22 @@ static int tas5086_probe(struct snd_soc_codec *codec) } } + tas5086_reset(priv); ret = tas5086_init(codec->dev, priv); if (ret < 0) - return ret; + goto exit_disable_regulators; /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) - return ret; + goto exit_disable_regulators; return 0; + +exit_disable_regulators: + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + + return ret; } static int tas5086_remove(struct snd_soc_codec *codec) @@ -852,6 +876,8 @@ static int tas5086_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(priv->gpio_nreset, 0); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; }; @@ -900,6 +926,16 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (!priv) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + priv->regmap = devm_regmap_init(dev, NULL, i2c, &tas5086_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); @@ -919,21 +955,34 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, gpio_nreset = -EINVAL; priv->gpio_nreset = gpio_nreset; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) { + dev_err(dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + tas5086_reset(priv); /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); - if (ret < 0) - return ret; - - if (i != 0x3) { + if (ret == 0 && i != 0x3) { dev_err(dev, "Failed to identify TAS5086 codec (got %02x)\n", i); - return -ENODEV; + ret = -ENODEV; } - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, - &tas5086_dai, 1); + /* + * The chip has been identified, so we can turn off the power + * again until the dai link is set up. + */ + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + + if (ret == 0) + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); + + return ret; } static int tas5086_i2c_remove(struct i2c_client *i2c) -- 1.9.3

3 2

[alsa-devel] [PATCH] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 03 Jul '14

03 Jul '14

This patch adds a CODEC function to the NXP TDA998x HDMI transmitter. The CODEC handles both I2S and S/PDIF inputs and does dynamic input switch in the TDA998x I2C driver on start/stop audio streaming. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- This patch applies against linux-next. --- .../devicetree/bindings/drm/i2c/tda998x.txt | 13 ++ drivers/gpu/drm/i2c/Makefile | 2 +- drivers/gpu/drm/i2c/tda998x_codec.c | 241 +++++++++++++++++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 74 +++++-- drivers/gpu/drm/i2c/tda998x_drv.h | 31 +++ include/drm/i2c/tda998x.h | 1 + 6 files changed, 341 insertions(+), 21 deletions(-) create mode 100644 drivers/gpu/drm/i2c/tda998x_codec.c create mode 100644 drivers/gpu/drm/i2c/tda998x_drv.h diff --git a/Documentation/devicetree/bindings/drm/i2c/tda998x.txt b/Documentation/devicetree/bindings/drm/i2c/tda998x.txt index d7df01c..7ce4eac 100644 --- a/Documentation/devicetree/bindings/drm/i2c/tda998x.txt +++ b/Documentation/devicetree/bindings/drm/i2c/tda998x.txt @@ -15,6 +15,15 @@ Optional properties: - video-ports: 24 bits value which defines how the video controller output is wired to the TDA998x input - default: <0x230145> + - audio-ports: one or two values corresponding to entries in + the audio-port-names property. + + - audio-port-names: must contain "i2s", "spdif" entries + matching entries in the audio-ports property. + + - #sound-dai-cells: must be set to <1> for use with the simple-card. + The DAI 0 is the I2S input and the DAI 1 is the S/PDIF input. + Example: tda998x: hdmi-encoder { @@ -24,4 +33,8 @@ Example: interrupts = <27 2>; /* falling edge */ pinctrl-0 = <&pmx_camera>; pinctrl-names = "default"; + + audio-ports = <0x03>, <0x04>; + audio-port-names = "i2s", "spdif"; + #sound-dai-cells = <1>; }; diff --git a/drivers/gpu/drm/i2c/Makefile b/drivers/gpu/drm/i2c/Makefile index 43aa33b..f2d625c 100644 --- a/drivers/gpu/drm/i2c/Makefile +++ b/drivers/gpu/drm/i2c/Makefile @@ -6,5 +6,5 @@ obj-$(CONFIG_DRM_I2C_CH7006) += ch7006.o sil164-y := sil164_drv.o obj-$(CONFIG_DRM_I2C_SIL164) += sil164.o -tda998x-y := tda998x_drv.o +tda998x-y := tda998x_drv.o tda998x_codec.o obj-$(CONFIG_DRM_I2C_NXP_TDA998X) += tda998x.o diff --git a/drivers/gpu/drm/i2c/tda998x_codec.c b/drivers/gpu/drm/i2c/tda998x_codec.c new file mode 100644 index 0000000..dff6afb --- /dev/null +++ b/drivers/gpu/drm/i2c/tda998x_codec.c @@ -0,0 +1,241 @@ +/* + * ALSA SoC TDA998X CODEC + * + * Copyright (C) 2014 Jean-Francois Moine + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <linux/of.h> +#include <linux/i2c.h> +#include <drm/drm_encoder_slave.h> +#include <drm/i2c/tda998x.h> + +#include "tda998x_drv.h" + +#define TDA998X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static int tda_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + u8 *eld = priv->eld; + struct snd_pcm_runtime *runtime = substream->runtime; + u8 *sad; + int sad_count; + unsigned eld_ver, mnl, rate_mask; + unsigned max_channels, fmt; + u64 formats; + struct snd_pcm_hw_constraint_list *rate_constraints = + &priv->rate_constraints; + static const u32 hdmi_rates[] = { + 32000, 44100, 48000, 88200, 9600, 176400, 192000 + }; + + if (!eld) + return 0; + + /* adjust the hw params from the ELD (EDID) */ + eld_ver = eld[0] >> 3; + if (eld_ver != 2 && eld_ver != 31) + return 0; + + mnl = eld[4] & 0x1f; + if (mnl > 16) + return 0; + + sad_count = eld[5] >> 4; + sad = eld + 20 + mnl; + + /* Start from the basic audio settings */ + max_channels = 2; + rate_mask = 0; + fmt = 0; + while (sad_count--) { + switch (sad[0] & 0x78) { + case 0x08: /* PCM */ + max_channels = max(max_channels, (sad[0] & 7) + 1u); + rate_mask |= sad[1]; + fmt |= sad[2] & 0x07; + break; + } + sad += 3; + } + + /* set the constraints */ + rate_constraints->list = hdmi_rates; + rate_constraints->count = ARRAY_SIZE(hdmi_rates); + rate_constraints->mask = rate_mask; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + rate_constraints); + + formats = 0; + if (fmt & 1) + formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (fmt & 2) + formats |= SNDRV_PCM_FMTBIT_S20_3LE; + if (fmt & 4) + formats |= SNDRV_PCM_FMTBIT_S24_LE; + snd_pcm_hw_constraint_mask64(runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + formats); + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 1, max_channels); + return 0; +} + +static int tda_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + + /* Requires an attached display */ + if (!priv->encoder->crtc) + return -ENODEV; + + /* if same input and same parameters, do not do a full switch */ + if (dai->id == priv->params.audio_format && + params_format(params) == priv->audio_sample_format) { + tda998x_audio_start(priv, 0); + return 0; + } + priv->params.audio_format = dai->id; + priv->audio_sample_format = params_format(params); + priv->params.audio_cfg = + priv->audio_ports[dai->id == AFMT_I2S ? 0 : 1]; + tda998x_audio_start(priv, 1); + return 0; +} + +static void tda_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tda998x_priv *priv = snd_soc_codec_get_drvdata(dai->codec); + + tda998x_audio_stop(priv); +} + +static const struct snd_soc_dai_ops tda_ops = { + .startup = tda_startup, + .hw_params = tda_hw_params, + .shutdown = tda_shutdown, +}; + +static struct snd_soc_dai_driver tda998x_dai[] = { + { + .name = "i2s-hifi", + .id = AFMT_I2S, + .playback = { + .stream_name = "HDMI I2S Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = TDA998X_FORMATS, + }, + .ops = &tda_ops, + }, + { + .name = "spdif-hifi", + .id = AFMT_SPDIF, + .playback = { + .stream_name = "HDMI SPDIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 22050, + .rate_max = 192000, + .formats = TDA998X_FORMATS, + }, + .ops = &tda_ops, + }, +}; + +static const struct snd_soc_dapm_widget tda_widgets[] = { + SND_SOC_DAPM_OUTPUT("hdmi-out"), +}; +static const struct snd_soc_dapm_route tda_routes[] = { + { "hdmi-out", NULL, "HDMI I2S Playback" }, + { "hdmi-out", NULL, "HDMI SPDIF Playback" }, +}; + +static int tda_probe(struct snd_soc_codec *codec) +{ + struct i2c_client *i2c_client = to_i2c_client(codec->dev); + struct tda998x_priv *priv = i2c_get_clientdata(i2c_client); + struct device_node *np = codec->dev->of_node; + int i, j, ret; + const char *p; + + if (!priv) + return -ENODEV; + snd_soc_codec_set_drvdata(codec, priv); + + if (!np) + return 0; + + /* get the audio input ports*/ + for (i = 0; i < 2; i++) { + u32 port; + + ret = of_property_read_u32_index(np, "audio-ports", i, &port); + if (ret) { + if (i == 0) + dev_err(codec->dev, + "bad or missing audio-ports\n"); + break; + } + ret = of_property_read_string_index(np, "audio-port-names", + i, &p); + if (ret) { + dev_err(codec->dev, + "missing audio-port-names[%d]\n", i); + break; + } + if (strcmp(p, "i2s") == 0) { + j = 0; + } else if (strcmp(p, "spdif") == 0) { + j = 1; + } else { + dev_err(codec->dev, + "bad audio-port-names '%s'\n", p); + break; + } + priv->audio_ports[j] = port; + } + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_tda998x = { + .probe = tda_probe, + .dapm_widgets = tda_widgets, + .num_dapm_widgets = ARRAY_SIZE(tda_widgets), + .dapm_routes = tda_routes, + .num_dapm_routes = ARRAY_SIZE(tda_routes), +}; + +int tda998x_codec_register(struct device *dev) +{ + return snd_soc_register_codec(dev, + &soc_codec_tda998x, + tda998x_dai, ARRAY_SIZE(tda998x_dai)); +} + +void tda998x_codec_unregister(struct device *dev) +{ + snd_soc_unregister_codec(dev); +} diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c index 240c331..7e9f9dc 100644 --- a/drivers/gpu/drm/i2c/tda998x_drv.c +++ b/drivers/gpu/drm/i2c/tda998x_drv.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/irq.h> #include <sound/asoundef.h> +#include <sound/pcm_params.h> #include <drm/drmP.h> #include <drm/drm_crtc_helper.h> @@ -28,24 +29,9 @@ #include <drm/drm_edid.h> #include <drm/i2c/tda998x.h> -#define DBG(fmt, ...) DRM_DEBUG(fmt"\n", ##__VA_ARGS__) +#include "tda998x_drv.h" -struct tda998x_priv { - struct i2c_client *cec; - struct i2c_client *hdmi; - uint16_t rev; - uint8_t current_page; - int dpms; - bool is_hdmi_sink; - u8 vip_cntrl_0; - u8 vip_cntrl_1; - u8 vip_cntrl_2; - struct tda998x_encoder_params params; - - wait_queue_head_t wq_edid; - volatile int wq_edid_wait; - struct drm_encoder *encoder; -}; +#define DBG(fmt, ...) DRM_DEBUG(fmt"\n", ##__VA_ARGS__) #define to_tda998x_priv(x) ((struct tda998x_priv *)to_encoder_slave(x)->slave_priv) @@ -640,12 +626,11 @@ static void tda998x_configure_audio(struct tda998x_priv *priv, struct drm_display_mode *mode, struct tda998x_encoder_params *p) { - uint8_t buf[6], clksel_aip, clksel_fs, cts_n, adiv; + uint8_t buf[6], clksel_aip, clksel_fs, cts_n, adiv, aclk; uint32_t n; /* Enable audio ports */ reg_write(priv, REG_ENA_AP, p->audio_cfg); - reg_write(priv, REG_ENA_ACLK, p->audio_clk_cfg); /* Set audio input source */ switch (p->audio_format) { @@ -654,13 +639,28 @@ tda998x_configure_audio(struct tda998x_priv *priv, clksel_aip = AIP_CLKSEL_AIP_SPDIF; clksel_fs = AIP_CLKSEL_FS_FS64SPDIF; cts_n = CTS_N_M(3) | CTS_N_K(3); + aclk = 0; /* no clock */ break; case AFMT_I2S: reg_write(priv, REG_MUX_AP, MUX_AP_SELECT_I2S); clksel_aip = AIP_CLKSEL_AIP_I2S; clksel_fs = AIP_CLKSEL_FS_ACLK; - cts_n = CTS_N_M(3) | CTS_N_K(3); + /* with I2S input, the CTS_N predivider depends on + * the sample width */ + switch (priv->audio_sample_format) { + case SNDRV_PCM_FORMAT_S16_LE: + cts_n = CTS_N_M(3) | CTS_N_K(1); + break; + case SNDRV_PCM_FORMAT_S24_LE: + cts_n = CTS_N_M(3) | CTS_N_K(2); + break; + default: + case SNDRV_PCM_FORMAT_S32_LE: + cts_n = CTS_N_M(3) | CTS_N_K(3); + break; + } + aclk = 1; /* clock enable */ break; default: @@ -672,6 +672,7 @@ tda998x_configure_audio(struct tda998x_priv *priv, reg_clear(priv, REG_AIP_CNTRL_0, AIP_CNTRL_0_LAYOUT | AIP_CNTRL_0_ACR_MAN); /* auto CTS */ reg_write(priv, REG_CTS_N, cts_n); + reg_write(priv, REG_ENA_ACLK, aclk); /* * Audio input somehow depends on HDMI line rate which is @@ -728,6 +729,24 @@ tda998x_configure_audio(struct tda998x_priv *priv, tda998x_write_aif(priv, p); } +/* tda998x codec interface */ +void tda998x_audio_start(struct tda998x_priv *priv, + int full) +{ + struct tda998x_encoder_params *p = &priv->params; + + if (!full) { + reg_write(priv, REG_ENA_AP, p->audio_cfg); + return; + } + tda998x_configure_audio(priv, &priv->encoder->crtc->hwmode, p); +} + +void tda998x_audio_stop(struct tda998x_priv *priv) +{ + reg_write(priv, REG_ENA_AP, 0); +} + /* DRM encoder functions */ static void @@ -1149,6 +1168,11 @@ tda998x_encoder_get_modes(struct drm_encoder *encoder, drm_mode_connector_update_edid_property(connector, edid); n = drm_add_edid_modes(connector, edid); priv->is_hdmi_sink = drm_detect_hdmi_monitor(edid); + + /* keep the EDID as ELD for the audio subsystem */ + drm_edid_to_eld(connector, edid); + priv->eld = connector->eld; + kfree(edid); } @@ -1191,6 +1215,8 @@ tda998x_encoder_destroy(struct drm_encoder *encoder) if (priv->hdmi->irq) free_irq(priv->hdmi->irq, priv); + tda998x_codec_unregister(&priv->hdmi->dev); + if (priv->cec) i2c_unregister_device(priv->cec); kfree(priv); @@ -1239,10 +1265,15 @@ tda998x_encoder_init(struct i2c_client *client, if (!priv) return -ENOMEM; + i2c_set_clientdata(client, priv); + priv->vip_cntrl_0 = VIP_CNTRL_0_SWAP_A(2) | VIP_CNTRL_0_SWAP_B(3); priv->vip_cntrl_1 = VIP_CNTRL_1_SWAP_C(0) | VIP_CNTRL_1_SWAP_D(1); priv->vip_cntrl_2 = VIP_CNTRL_2_SWAP_E(4) | VIP_CNTRL_2_SWAP_F(5); + priv->params.audio_frame[1] = 1; /* channels - 1 */ + priv->params.audio_sample_rate = 48000; /* 48kHz */ + priv->current_page = 0xff; priv->hdmi = client; priv->cec = i2c_new_dummy(client->adapter, 0x34); @@ -1340,6 +1371,9 @@ tda998x_encoder_init(struct i2c_client *client, /* enable EDID read irq: */ reg_set(priv, REG_INT_FLAGS_2, INT_FLAGS_2_EDID_BLK_RD); + /* register the audio CODEC */ + tda998x_codec_register(&client->dev); + if (!np) return 0; /* non-DT */ diff --git a/drivers/gpu/drm/i2c/tda998x_drv.h b/drivers/gpu/drm/i2c/tda998x_drv.h new file mode 100644 index 0000000..6db5160 --- /dev/null +++ b/drivers/gpu/drm/i2c/tda998x_drv.h @@ -0,0 +1,31 @@ +/* tda998x private data */ + +struct tda998x_priv { + struct i2c_client *cec; + struct i2c_client *hdmi; + uint16_t rev; + uint8_t current_page; + int dpms; + bool is_hdmi_sink; + u8 vip_cntrl_0; + u8 vip_cntrl_1; + u8 vip_cntrl_2; + struct tda998x_encoder_params params; + + wait_queue_head_t wq_edid; + volatile int wq_edid_wait; + struct drm_encoder *encoder; + + u8 audio_ports[2]; + int audio_sample_format; + + u8 *eld; + + struct snd_pcm_hw_constraint_list rate_constraints; +}; + +int tda998x_codec_register(struct device *dev); +void tda998x_codec_unregister(struct device *dev); + +void tda998x_audio_start(struct tda998x_priv *priv, int full); +void tda998x_audio_stop(struct tda998x_priv *priv); diff --git a/include/drm/i2c/tda998x.h b/include/drm/i2c/tda998x.h index 3e419d9..31757df 100644 --- a/include/drm/i2c/tda998x.h +++ b/include/drm/i2c/tda998x.h @@ -20,6 +20,7 @@ struct tda998x_encoder_params { u8 audio_frame[6]; enum { + AFMT_NO_AUDIO = 0, AFMT_SPDIF, AFMT_I2S } audio_format; -- 2.0.1 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

4 14

[alsa-devel] [PATCH v2 0/2] ASoC: add CSR SiRFSoC audio USP interface driver
by Rongjun Ying 03 Jul '14

03 Jul '14

This patchset adds the SiRF USP controller driver, this driver uses the USP as I2S or DSP_A mode interface. Rongjun Ying (2): ASoC: sirf: Add audio usp interface driver ASoC: sirf: Add device tree binding for the USP audio device .../devicetree/bindings/sound/sirf-usp.txt | 27 ++ sound/soc/sirf/Kconfig | 6 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-usp.c | 415 +++++++++++++++++++++ sound/soc/sirf/sirf-usp.h | 293 +++++++++++++++ 5 files changed, 743 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.9.3 Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom More information can be found at www.csr.com. Keep up to date with CSR on our technical blog, www.csr.com/blog, CSR people blog, www.csr.com/people, YouTube, www.youtube.com/user/CSRplc, Facebook, www.facebook.com/pages/CSR/191038434253534, or follow us on Twitter at www.twitter.com/CSR_plc. New for 2014, you can now access the wide range of products powered by aptX at www.aptx.com.

2 3

[alsa-devel] [PATCH v4] ASoC: tas2552: Support TI TAS2552 Amplifier
by Dan Murphy 03 Jul '14

03 Jul '14

Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v4 - Added pm_runtime support, removed magical numbers, added regulator support, changed from legacy gpio to new gpiod calls, fixed default register values, flipped on battery tracking support and added suspend/resume support - https://patchwork.kernel.org/patch/4465611/ .../devicetree/bindings/sound/tas2552.txt | 25 + include/sound/tas2552-plat.h | 25 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2552.c | 574 ++++++++++++++++++++ sound/soc/codecs/tas2552.h | 119 ++++ 6 files changed, 750 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 include/sound/tas2552-plat.h create mode 100644 sound/soc/codecs/tas2552.c create mode 100644 sound/soc/codecs/tas2552.h diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 0000000..00223dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,25 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - One of: + "ti,tas2552" - TAS2552 + +- reg - I2C slave address + +Optional properties: + +- enable-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS2552 diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 0000000..65e7627 --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0b9571c..cc09261 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -99,6 +99,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TAS2552 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE @@ -754,4 +755,8 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1c..33bc7228 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -162,6 +162,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -325,3 +326,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 0000000..6eab02f --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,574 @@ +/* + * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/tas2552-plat.h> + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x22}, + {TAS2552_CFG_3, 0x80}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xc0}, + {TAS2552_PDM_CFG, 0x01}, + {TAS2552_PGA_GAIN, 0x00}, + {TAS2552_BOOST_PT_CTRL, 0x0f}, + {TAS2552_LIMIT_LVL_CTRL, 0x0c}, + {TAS2552_LIMIT_RATE_HYS, 0x08}, + {TAS2552_CFG_2, 0xef}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x04}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +#define TAS2552_NUM_SUPPLIES 3 +static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = { + "vbat", /* vbat voltage */ + "iovdd", /* I/O Voltage */ + "avdd", /* Analog DAC Voltage */ +}; + +struct tas2552_data { + struct mutex mutex; + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES]; + unsigned char regs[TAS2552_VBAT_DATA]; + struct gpio_desc *enable_gpio; + u8 power_state:1; +}; + +static int tas2552_power(struct tas2552_data *data, u8 power) +{ + int ret = 0; + + mutex_lock(&data->mutex); + + if (power) { + if (data->enable_gpio) + gpiod_set_value(data->enable_gpio, 1); + + data->power_state = 1; + } else { + if (data->enable_gpio) + gpiod_set_value(data->enable_gpio, 0); + + data->power_state = 0; + } + + mutex_unlock(&data->mutex); + return ret; +} + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg = 0x0; + + if (sw_shutdown) + cfg1_reg |= TAS2552_SWS_MASK; + else + cfg1_reg &= ~TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static int tas2552_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u8 wclk_reg; + struct snd_soc_codec *codec = dai->codec; + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (params_rate(params)) { + case 8000: + wclk_reg = TAS2552_8KHZ; + break; + case 11025: + wclk_reg = TAS2552_11_12KHZ; + break; + case 16000: + wclk_reg = TAS2552_16KHZ; + break; + case 32000: + wclk_reg = TAS2552_32KHZ; + break; + case 22050: + case 24000: + wclk_reg = TAS2552_22_24KHZ; + break; + case 44100: + case 48000: + wclk_reg = TAS2552_44_48KHZ; + break; + case 96000: + wclk_reg = TAS2552_88_96KHZ; + break; + default: + dev_vdbg(codec->dev, "Substream sample rate is not found\n"); + return -EINVAL; + } + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_3, TAS2552_WCLK_MASK, wclk_reg); + + pm_runtime_put(codec->dev); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + u8 serial_format; + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + return -EINVAL; + } + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, + (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK), + serial_format); + + pm_runtime_put(codec->dev); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format = 0x0; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format = TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format = TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format = TAS2552_DAIFMT_LEFT_J; + break; + + default: + return -EINVAL; + } + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, TAS2552_DATA_FORMAT_MASK, + serial_format); + + pm_runtime_put(codec->dev); + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + snd_soc_write(codec, TAS2552_PLL_CTRL_1, 0x10); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, 0x00); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, 0x00); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + pm_runtime_put(codec->dev); + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg = 0x0; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg |= TAS2552_MUTE_MASK; + else + cfg1_reg &= ~TAS2552_MUTE_MASK; + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + pm_runtime_put(codec->dev); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int tas2552_runtime_suspend(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_sw_shutdown(tas2552, 1); + tas2552_power(tas2552, 0); + + regcache_cache_only(tas2552->regmap, true); + regcache_mark_dirty(tas2552->regmap); + + return 0; +} + +static int tas2552_runtime_resume(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_power(tas2552, 1); + tas2552_sw_shutdown(tas2552, 0); + + regcache_cache_only(tas2552->regmap, false); + regcache_sync(tas2552->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops tas2552_pm = { + SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume, + NULL) +}; + +static int tas2552_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + pm_runtime_get_sync(codec->dev); + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + return 0; +} + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + pm_runtime_put(codec->dev); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .startup = tas2552_startup, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + tas2552->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + tas2552_power(tas2552, 1); + tas2552_sw_shutdown(tas2552, 0); + + pm_runtime_set_active(codec->dev); + pm_runtime_set_autosuspend_delay(codec->dev, 1000); + pm_runtime_use_autosuspend(codec->dev); + pm_runtime_enable(codec->dev); + pm_runtime_mark_last_busy(codec->dev); + pm_runtime_put_sync_autosuspend(codec->dev); + + snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK | + TAS2552_SWS_MASK | TAS2552_PLL_SRC_BCLK); + snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL | + TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ); + snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8); + snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL); + /* 0dB gain */ + snd_soc_write(codec, TAS2552_PGA_GAIN, 0x10); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 | + TAS2552_APT_THRESH_2_1_7); + /** + * Data sheet indicates to write 0x0c to 0x0d during init but no + * additional information is given to what it means. + */ + snd_soc_write(codec, TAS2552_LIMIT_LVL_CTRL, 0x0c); + /** + * Data sheet indicates to write 0x20 to 0x0e during init but no + * additional information is given to what it means. + */ + + snd_soc_write(codec, TAS2552_LIMIT_RATE_HYS, 0x20); + + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | + TAS2552_BOOST_EN | TAS2552_APT_EN | + TAS2552_PLL_ENABLE | TAS2552_LIM_EN); + + return 0; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_power(tas2552, 0); + + return 0; +}; + +#ifdef CONFIG_PM +static int tas2552_suspend(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + pm_runtime_put(codec->dev); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) + dev_err(codec->dev, "Failed to disable supplies: %d\n", + ret); + return 0; +} + +static int tas2552_resume(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + } + + pm_runtime_get_sync(codec->dev); + + return 0; +} +#else +#define tas2552_suspend NULL +#define tas2552_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .suspend = tas2552_suspend, + .resume = tas2552_resume, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + int ret; + int i; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + data->enable_gpio = devm_gpiod_get(dev, "enable"); + if (IS_ERR(data->enable_gpio)) { + ret = PTR_ERR(data->enable_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + data->enable_gpio = NULL; + } else { + gpiod_direction_output(data->enable_gpio, 0); + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas2552_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) + dev_err(dev, "Failed to request supplies: %d\n", ret); + + mutex_init(&data->mutex); + + dev_set_drvdata(&client->dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return 0; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552-codec", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + .pm = &tas2552_pm, + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 0000000..52b2d64 --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,119 @@ +/* + * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_LIMIT_LVL_CTRL 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) + +/* CFG2 Register Masks */ +#define TAS2552_CLASSD_EN (1 << 7) +#define TAS2552_BOOST_EN (1 << 6) +#define TAS2552_APT_EN (1 << 5) +#define TAS2552_PLL_ENABLE (1 << 3) +#define TAS2552_LIM_EN (1 << 2) +#define TAS2552_IVSENSE_EN (1 << 1) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK (1 << 7) +#define TAS2552_BIT_CLK_MASK (1 << 6) +#define TAS2552_DATA_FORMAT_MASK (0x11 << 2) + +#define TAS2552_DAIFMT_DSP (1 << 3) +#define TAS2552_DAIFMT_RIGHT_J (1 << 4) +#define TAS2552_DAIFMT_LEFT_J (0x11 << 3) + +#define TAS2552_PLL_SRC_MCLK 0x00 +#define TAS2552_PLL_SRC_BCLK (1 << 3) +#define TAS2552_PLL_SRC_IVCLKIN (1 << 4) +#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3) + +#define TAS2552_DIN_SRC_SEL_MUTED 0x00 +#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4) +#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5) +#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4) + +#define TAS2552_PDM_IN_SEL (1 << 5) +#define TAS2552_I2S_OUT_SEL (1 << 6) +#define TAS2552_ANALOG_IN_SEL (1 << 7) + +/* CFG3 WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ (1 << 1) +#define TAS2552_16KHZ (1 << 2) +#define TAS2552_22_24KHZ (1 << 3) +#define TAS2552_32KHZ (1 << 4) +#define TAS2552_44_48KHZ (1 << 5) +#define TAS2552_88_96KHZ (1 << 6) +#define TAS2552_176_192KHZ (1 << 7) + +/* OUTPUT_DATA register */ +#define TAS2552_PDM_DATA_I 0x00 +#define TAS2552_PDM_DATA_V (1 << 6) +#define TAS2552_PDM_DATA_I_V (1 << 7) +#define TAS2552_PDM_DATA_V_I (0x11 << 6) + +/* PDM CFG Register */ +#define TAS2552_PDM_DATA_ES_RISE 0x4 + +#define TAS2552_PDM_PLL_CLK_SEL 0x00 +#define TAS2552_PDM_IV_CLK_SEL (1 << 1) +#define TAS2552_PDM_BCLK_SEL (1 << 2) +#define TAS2552_PDM_MCLK_SEL (1 << 3) + +/* Boost pass-through register */ +#define TAS2552_APT_DELAY_50 0x00 +#define TAS2552_APT_DELAY_75 (1 << 1) +#define TAS2552_APT_DELAY_125 (1 << 2) +#define TAS2552_APT_DELAY_200 (1 << 3) + +#define TAS2552_APT_THRESH_2_5 0x00 +#define TAS2552_APT_THRESH_1_7 (1 << 3) +#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4) +#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2) + +#endif -- 1.7.9.5

2 5

[alsa-devel] [PATCH v2 1/2] drm/i915: provide interface for audio driver to query cdclk
by mengdong.lin＠intel.com 03 Jul '14

03 Jul '14

From: Jani Nikula <jani.nikula(a)intel.com> Signed-off-by: Jani Nikula <jani.nikula(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/drivers/gpu/drm/i915/intel_pm.c b/drivers/gpu/drm/i915/intel_pm.c index a90fdbd..21170e5 100644 --- a/drivers/gpu/drm/i915/intel_pm.c +++ b/drivers/gpu/drm/i915/intel_pm.c @@ -6256,6 +6256,27 @@ int i915_release_power_well(void) } EXPORT_SYMBOL_GPL(i915_release_power_well); +/* + * Private interface for the audio driver to get CDCLK in kHz. + * + * Caller must request power well using i915_request_power_well() prior to + * making the call. + */ +int i915_get_cdclk_freq(void) +{ + struct drm_i915_private *dev_priv; + + if (!hsw_pwr) + return -ENODEV; + + dev_priv = container_of(hsw_pwr, struct drm_i915_private, + power_domains); + + return intel_ddi_get_cdclk_freq(dev_priv); +} +EXPORT_SYMBOL_GPL(i915_get_cdclk_freq); + + #define POWER_DOMAIN_MASK (BIT(POWER_DOMAIN_NUM) - 1) #define HSW_ALWAYS_ON_POWER_DOMAINS ( \ diff --git a/include/drm/i915_powerwell.h b/include/drm/i915_powerwell.h index 2baba99..baa6f11 100644 --- a/include/drm/i915_powerwell.h +++ b/include/drm/i915_powerwell.h @@ -32,5 +32,6 @@ /* For use by hda_i915 driver */ extern int i915_request_power_well(void); extern int i915_release_power_well(void); +extern int i915_get_cdclk_freq(void); #endif /* _I915_POWERWELL_H_ */ -- 1.8.1.2

5 7

[alsa-devel] abdul, please add me to your LinkedIn network
by Putinei Ionut 03 Jul '14

03 Jul '14

Hi abdul, I'd like to connect with you on LinkedIn. Putinei Ionut Linux Engineer at Wind River Accept: http://www.linkedin.com/blink?simpleRedirect=1LiR9Pc516jzRVpkJApn9xq7cCczcO… View Putinei Ionut's profile: http://www.linkedin.com/blink?simpleRedirect=0Vej8Qd38Tcz8Zh4BOpm9JpmQCpmRx… You are receiving Invitation emails. Unsubscribe here: http://www.linkedin.com/blink?simpleRedirect=54MkDhggDAUcjdcpQZSt3dCiDBPplp… This email was intended for abdul moiz (Software engineer at Vu TelePresence Inc.). Learn why we included this at the following link: http://www.linkedin.com/blink?simpleRedirect=e3wTd3RAimlIoSBQsC4Ce39vsClyrm… © 2014, LinkedIn Corporation. 2029 Stierlin Ct. Mountain View, CA 94043, USA

1 0

[alsa-devel] Non-I2C/SPI ASoC codec
by Matthew Bucknall 03 Jul '14

03 Jul '14

Hi, I'm trying to get audio support for a custom ARM board to work and have written codec and machine drivers - The platform driver already existed (atmel_ssc_dai). The codec I am supporting is extremely simple and has no SPI/I2C interface for configuration and just accepts fixed format/fixed rate I2S. Previously, when I've gotten ASoC support for another board working, the codec it used (TLV320AIC32) had an I2C interface. Declaring I2C_BOARD_INFO for this in my board file seemed sufficient to get the machine driver to probe and initialize. For this new board, I have no SPI/I2C device to declare in the board file and its associated machine driver is not probing. All I get from the kernel is 'No soundcards found.' My driver modules are getting compiled and are linked statically. What do I need to do to get the machine driver going? Matt.

2 1

[alsa-devel] [PATCH v2 2/2] ALSA: hda - restore BCLK M/N value as per CDCLK for HSW/BDW display HDA controller
by mengdong.lin＠intel.com 03 Jul '14

03 Jul '14

From: Mengdong Lin <mengdong.lin(a)intel.com> For HSW/BDW display HD-A controller, hda_set_bclk() is defined to set BCLK by programming the M/N values as per the core display clock (CDCLK) queried from i915 display driver. And the audio driver will also set BCLK in azx_first_init() since the display driver can turn off the shared power in boot phase if only eDP is connected and M/N values will be lost and must be reprogrammed. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index e9e8a4a..8b4940b 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -20,10 +20,20 @@ #include <linux/module.h> #include <sound/core.h> #include <drm/i915_powerwell.h> +#include "hda_priv.h" #include "hda_i915.h" +/* Intel HSW/BDW display HDA controller Extended Mode registers. + * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display + * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled. + */ +#define ICH6_REG_EM4 0x100c +#define ICH6_REG_EM5 0x1010 + static int (*get_power)(void); static int (*put_power)(void); +static int (*get_cdclk)(void); int hda_display_power(bool enable) { @@ -38,6 +48,43 @@ int hda_display_power(bool enable) return put_power(); } +void haswell_set_bclk(struct azx *chip) +{ + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!get_cdclk) + return; + + cdclk_freq = get_cdclk(); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } + + azx_writew(chip, EM4, bclk_m); + azx_writew(chip, EM5, bclk_n); +} + + int hda_i915_init(void) { int err = 0; @@ -55,6 +102,10 @@ int hda_i915_init(void) return -ENODEV; } + get_cdclk = symbol_request(i915_get_cdclk_freq); + if (!get_cdclk) /* may have abnormal BCLK and audio playback rate */ + pr_warn("hda-i915: get_cdclk symbol get fail\n"); + pr_debug("HDA driver get symbol successfully from i915 module\n"); return err; @@ -70,6 +121,10 @@ int hda_i915_exit(void) symbol_put(i915_release_power_well); put_power = NULL; } + if (get_cdclk) { + symbol_put(i915_get_cdclk_freq); + get_cdclk = NULL; + } return 0; } diff --git a/sound/pci/hda/hda_i915.h b/sound/pci/hda/hda_i915.h index bfd835f..9237340 100644 --- a/sound/pci/hda/hda_i915.h +++ b/sound/pci/hda/hda_i915.h @@ -18,6 +18,7 @@ #ifdef CONFIG_SND_HDA_I915 int hda_display_power(bool enable); +void haswell_set_bclk(struct azx *chip); int hda_i915_init(void); int hda_i915_exit(void); #else diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 25753db..b6b4e71 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -62,9 +62,9 @@ #include <linux/vga_switcheroo.h> #include <linux/firmware.h> #include "hda_codec.h" -#include "hda_i915.h" #include "hda_controller.h" #include "hda_priv.h" +#include "hda_i915.h" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; @@ -288,21 +288,8 @@ static char *driver_short_names[] = { [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; - -/* Intel HSW/BDW display HDA controller Extended Mode registers. - * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display - * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled. - */ -#define ICH6_REG_EM4 0x100c -#define ICH6_REG_EM5 0x1010 - struct hda_intel { struct azx chip; - - /* HSW/BDW display HDA controller to restore BCLK from CDCLK */ - unsigned int bclk_m; - unsigned int bclk_n; }; @@ -598,22 +585,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -static void haswell_save_bclk(struct azx *chip) -{ - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - - hda->bclk_m = azx_readw(chip, EM4); - hda->bclk_n = azx_readw(chip, EM5); -} - -static void haswell_restore_bclk(struct azx *chip) -{ - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - - azx_writew(chip, EM4, hda->bclk_m); - azx_writew(chip, EM5, hda->bclk_n); -} - #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -641,12 +612,6 @@ static int azx_suspend(struct device *dev) chip->irq = -1; } - /* Save BCLK M/N values before they become invalid in D3. - * Will test if display power well can be released now. - */ - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - haswell_save_bclk(chip); - if (chip->msi) pci_disable_msi(chip->pci); pci_disable_device(pci); @@ -668,7 +633,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); - haswell_restore_bclk(chip); + haswell_set_bclk(chip); } pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -713,10 +678,9 @@ static int azx_runtime_suspend(struct device *dev) azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - haswell_save_bclk(chip); + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); - } + return 0; } @@ -736,7 +700,7 @@ static int azx_runtime_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); - haswell_restore_bclk(chip); + haswell_set_bclk(chip); } /* Read STATESTS before controller reset */ @@ -1426,6 +1390,10 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); + + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + haswell_set_bclk(chip); + azx_init_chip(chip, (probe_only[dev] & 2) == 0); /* codec detection */ -- 1.8.1.2

3 2

[alsa-devel] [PATCH 0/3] ALSA: hda - restore BCLK from CDCLK for Haswell/Broadwell display audio
by mengdong.lin＠intel.com 03 Jul '14

03 Jul '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The 3 patches are for Haswell/Broadwell display HD-A controller. The 1st patch is for i915 display driver, providing a private interface for the audio driver to query CDCLK. The other 2 patches are for HD-A driver, will restore BCLK by setting M/N values as per CDCLK. Mengdong Lin (3): drm/i915: provide interface for audio driver to query cdclk ALSA: hda - define hda_get_display_clk to query CDCLK for Haswell/Broadwell ALSA: hda - restore BCLK M/N value as per CDCLK for HSW/BDW display HDA controller drivers/gpu/drm/i915/intel_pm.c | 21 +++++++++++++++ include/drm/i915_powerwell.h | 1 + sound/pci/hda/hda_i915.c | 22 +++++++++++++++ sound/pci/hda/hda_i915.h | 1 + sound/pci/hda/hda_intel.c | 60 +++++++++++++++++++++++++---------------- 5 files changed, 82 insertions(+), 23 deletions(-) -- 1.8.1.2

3 6

[alsa-devel] [PATCH 1/1] ice1712: Correcting/completing #defines for REGS
by Konstantinos Tsimpoukas 03 Jul '14

03 Jul '14

This small patch completes #defines for Control/Status Register, adds comments for the missing ones there and on the Interrupt Mask Register and additionally corrects "#define ICE1712_SERR_LEVEL 0x04 -> 0x08", according to documentation. Signed-off-by: Konstantinos Tsimpoukas <kostaslinuxxx(a)gmail.com> --- sound/pci/ice1712/ice1712.h | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index b209fc3..58f8f2a 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -41,14 +41,17 @@ #define ICEREG(ice, x) ((ice)->port + ICE1712_REG_##x) #define ICE1712_REG_CONTROL 0x00 /* byte */ -#define ICE1712_RESET 0x80 /* reset whole chip */ -#define ICE1712_SERR_LEVEL 0x04 /* SERR# level otherwise edge */ +#define ICE1712_RESET 0x80 /* soft reset whole chip */ +#define ICE1712_SERR_ASSERT_DS_DMA 0x40 /* disabled SERR# assertion for the DS DMA Ch-C irq otherwise enabled */ +#define ICE1712_DOS_VOL 0x10 /* DOS WT/FM volume control */ +#define ICE1712_SERR_LEVEL 0x08 /* SERR# level otherwise edge */ +#define ICE1712_SERR_ASSERT_SB 0x02 /* disabled SERR# assertion for SB irq otherwise enabled */ #define ICE1712_NATIVE 0x01 /* native mode otherwise SB */ #define ICE1712_REG_IRQMASK 0x01 /* byte */ -#define ICE1712_IRQ_MPU1 0x80 -#define ICE1712_IRQ_TIMER 0x40 -#define ICE1712_IRQ_MPU2 0x20 -#define ICE1712_IRQ_PROPCM 0x10 +#define ICE1712_IRQ_MPU1 0x80 /* MIDI irq mask */ +#define ICE1712_IRQ_TIMER 0x40 /* Timer mask */ +#define ICE1712_IRQ_MPU2 0x20 /* Secondary MIDI irq mask */ +#define ICE1712_IRQ_PROPCM 0x10 /* professional multi-track */ #define ICE1712_IRQ_FM 0x08 /* FM/MIDI - legacy */ #define ICE1712_IRQ_PBKDS 0x04 /* playback DS channels */ #define ICE1712_IRQ_CONCAP 0x02 /* consumer capture */ -- 1.9.1

1 0

[alsa-devel] [PATCH 0/4] ASoC: Aux dev initialization cleanup
by Lars-Peter Clausen 02 Jul '14

02 Jul '14

This patch cleans up the way a aux dev is initialized and bound to the rtd. This is useful for both componentization (moving aux devs to the component level) as well as for the multi-CODEC support patches. The series depends on topic/component. - Lars Lars-Peter Clausen (4): ASoC: Remove duplicated rtd->codec initialization ASoC: Replace soc_find_matching_codec() with soc_find_codec() ASoC: Bind aux devs early ASoC: Move non-shared code paths out of snd_soc_post_component_init() sound/soc/soc-core.c | 192 +++++++++++++++++---------------------------------- 1 file changed, 62 insertions(+), 130 deletions(-) -- 1.8.0

2 5

[alsa-devel] [PATCH v2] ASoC: add RT5670 CODEC driver
by bardliao＠realtek.com 02 Jul '14

02 Jul '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds a minimum support of Realtek ALC5670 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Currently, only playback with headphone or line out and capture with digital mic are tested. We will add other patches for other functions. This version do the following changes: * Rename some widgets/controls. * Merge all ASRC checking function to a single function. * Change _MICBIAS to _SUPPLY * Return error in unsupported slots or slot_width cases. --- include/sound/rt5670.h | 27 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5670-dsp.h | 54 + sound/soc/codecs/rt5670.c | 2692 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670.h | 2000 ++++++++++++++++++++++++++++++ 6 files changed, 4781 insertions(+) create mode 100644 include/sound/rt5670.h create mode 100644 sound/soc/codecs/rt5670-dsp.h create mode 100644 sound/soc/codecs/rt5670.c create mode 100644 sound/soc/codecs/rt5670.h diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h new file mode 100644 index 0000000..bd31119 --- /dev/null +++ b/include/sound/rt5670.h @@ -0,0 +1,27 @@ +/* + * linux/sound/rt5670.h -- Platform data for RT5670 + * + * Copyright 2014 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5670_H +#define __LINUX_SND_RT5670_H + +struct rt5670_platform_data { + int jd_mode; + bool in2_diff; + + bool dmic_en; + unsigned int dmic1_data_pin; + /* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/ + unsigned int dmic2_data_pin; + /* 0 = GPIO8; 1 = IN3N; */ + unsigned int dmic3_data_pin; + /* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2fc38d4..85e3d78 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C + select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE @@ -451,10 +452,12 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5640=y default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y + default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5640=m default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m + default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m config SND_SOC_RT5631 @@ -469,6 +472,9 @@ config SND_SOC_RT5645 config SND_SOC_RT5651 tristate +config SND_SOC_RT5670 + tristate + config SND_SOC_RT5677 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 97b80a1..d87b08e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -73,6 +73,7 @@ snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o +snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -241,6 +242,7 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o +obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o diff --git a/sound/soc/codecs/rt5670-dsp.h b/sound/soc/codecs/rt5670-dsp.h new file mode 100644 index 0000000..a34d0cd --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.h @@ -0,0 +1,54 @@ +/* + * rt5670-dsp.h -- RT5670 ALSA SoC DSP driver + * + * Copyright 2014 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5670_DSP_H__ +#define __RT5670_DSP_H__ + +#define RT5670_DSP_CTRL1 0xe0 +#define RT5670_DSP_CTRL2 0xe1 +#define RT5670_DSP_CTRL3 0xe2 +#define RT5670_DSP_CTRL4 0xe3 +#define RT5670_DSP_CTRL5 0xe4 + +/* DSP Control 1 (0xe0) */ +#define RT5670_DSP_CMD_MASK (0xff << 8) +#define RT5670_DSP_CMD_PE (0x0d << 8) /* Patch Entry */ +#define RT5670_DSP_CMD_MW (0x3b << 8) /* Memory Write */ +#define RT5670_DSP_CMD_MR (0x37 << 8) /* Memory Read */ +#define RT5670_DSP_CMD_RR (0x60 << 8) /* Register Read */ +#define RT5670_DSP_CMD_RW (0x68 << 8) /* Register Write */ +#define RT5670_DSP_REG_DATHI (0x26 << 8) /* High Data Addr */ +#define RT5670_DSP_REG_DATLO (0x25 << 8) /* Low Data Addr */ +#define RT5670_DSP_CLK_MASK (0x3 << 6) +#define RT5670_DSP_CLK_SFT 6 +#define RT5670_DSP_CLK_768K (0x0 << 6) +#define RT5670_DSP_CLK_384K (0x1 << 6) +#define RT5670_DSP_CLK_192K (0x2 << 6) +#define RT5670_DSP_CLK_96K (0x3 << 6) +#define RT5670_DSP_BUSY_MASK (0x1 << 5) +#define RT5670_DSP_RW_MASK (0x1 << 4) +#define RT5670_DSP_DL_MASK (0x3 << 2) +#define RT5670_DSP_DL_0 (0x0 << 2) +#define RT5670_DSP_DL_1 (0x1 << 2) +#define RT5670_DSP_DL_2 (0x2 << 2) +#define RT5670_DSP_DL_3 (0x3 << 2) +#define RT5670_DSP_I2C_AL_16 (0x1 << 1) +#define RT5670_DSP_CMD_EN (0x1) + +struct rt5670_dsp_param { + u16 cmd_fmt; + u16 addr; + u16 data; + u8 cmd; +}; + +#endif /* __RT5670_DSP_H__ */ + diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c new file mode 100644 index 0000000..879d42e --- /dev/null +++ b/sound/soc/codecs/rt5670.c @@ -0,0 +1,2692 @@ +/* + * rt5670.c -- RT5670 ALSA SoC audio codec driver + * + * Copyright 2014 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5670.h> + +#include "rl6231.h" +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define RT5670_DEVICE_ID 0x6271 + +#define RT5670_PR_RANGE_BASE (0xff + 1) +#define RT5670_PR_SPACING 0x100 + +#define RT5670_PR_BASE (RT5670_PR_RANGE_BASE + (0 * RT5670_PR_SPACING)) + +static const struct regmap_range_cfg rt5670_ranges[] = { + { .name = "PR", .range_min = RT5670_PR_BASE, + .range_max = RT5670_PR_BASE + 0xf8, + .selector_reg = RT5670_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5670_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + { RT5670_PR_BASE + 0x14, 0x9a8a }, + { RT5670_PR_BASE + 0x38, 0x3ba1 }, + { RT5670_PR_BASE + 0x3d, 0x3640 }, +}; +#define RT5670_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5670_reg[] = { + { 0x00, 0x0000 }, + { 0x02, 0x8888 }, + { 0x03, 0x8888 }, + { 0x0a, 0x0001 }, + { 0x0b, 0x0827 }, + { 0x0c, 0x0000 }, + { 0x0d, 0x0008 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0011 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x1f, 0x2f2f }, + { 0x20, 0x0000 }, + { 0x26, 0x7860 }, + { 0x27, 0x7860 }, + { 0x28, 0x7871 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5656 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, + { 0x2e, 0x2f2f }, + { 0x2f, 0x1002 }, + { 0x30, 0x0000 }, + { 0x31, 0x5f00 }, + { 0x32, 0x0000 }, + { 0x33, 0x0000 }, + { 0x34, 0x0000 }, + { 0x35, 0x0000 }, + { 0x36, 0x0000 }, + { 0x37, 0x0000 }, + { 0x38, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe00f }, + { 0x4c, 0x5380 }, + { 0x4f, 0x0073 }, + { 0x52, 0x00d3 }, + { 0x53, 0xf0f0 }, + { 0x61, 0x0000 }, + { 0x62, 0x0001 }, + { 0x63, 0x00c3 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6f, 0x8000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1110 }, + { 0x74, 0x0e00 }, + { 0x75, 0x1505 }, + { 0x76, 0x0015 }, + { 0x77, 0x0c00 }, + { 0x78, 0x4000 }, + { 0x79, 0x0123 }, + { 0x7f, 0x1100 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0000 }, + { 0x86, 0x0008 }, + { 0x87, 0x0000 }, + { 0x88, 0x0000 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0000 }, + { 0x8c, 0x0007 }, + { 0x8d, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c06 }, + { 0x93, 0x0000 }, + { 0x94, 0x0000 }, + { 0x95, 0x0000 }, + { 0x97, 0x0000 }, + { 0x98, 0x0000 }, + { 0x99, 0x0000 }, + { 0x9a, 0x2184 }, + { 0x9b, 0x010a }, + { 0x9c, 0x0aea }, + { 0x9d, 0x000c }, + { 0x9e, 0x0400 }, + { 0xae, 0x7000 }, + { 0xaf, 0x0000 }, + { 0xb0, 0x6000 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0000 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x1813 }, + { 0xd0, 0x0690 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0001 }, + { 0xdc, 0x0049 }, + { 0xdd, 0x0009 }, + { 0xe6, 0x8000 }, + { 0xe7, 0x0000 }, + { 0xec, 0xb300 }, + { 0xed, 0x0000 }, + { 0xee, 0xb300 }, + { 0xef, 0x0000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x8010 }, + { 0xfb, 0x0033 }, + { 0xfc, 0x0080 }, +}; + +static bool rt5670_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) { + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5670_RESET: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_PRIV_DATA: + case RT5670_ASRC_5: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_EQ_CTRL1: + case RT5670_ALC_CTRL_1: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_INT_IRQ_ST: + case RT5670_IL_CMD: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5670_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) { + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5670_RESET: + case RT5670_HP_VOL: + case RT5670_LOUT1: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_IN2: + case RT5670_INL1_INR1_VOL: + case RT5670_DAC1_DIG_VOL: + case RT5670_DAC2_DIG_VOL: + case RT5670_DAC_CTRL: + case RT5670_STO1_ADC_DIG_VOL: + case RT5670_MONO_ADC_DIG_VOL: + case RT5670_STO2_ADC_DIG_VOL: + case RT5670_ADC_BST_VOL1: + case RT5670_ADC_BST_VOL2: + case RT5670_STO2_ADC_MIXER: + case RT5670_STO1_ADC_MIXER: + case RT5670_MONO_ADC_MIXER: + case RT5670_AD_DA_MIXER: + case RT5670_STO_DAC_MIXER: + case RT5670_DD_MIXER: + case RT5670_DIG_MIXER: + case RT5670_DSP_PATH1: + case RT5670_DSP_PATH2: + case RT5670_DIG_INF1_DATA: + case RT5670_DIG_INF2_DATA: + case RT5670_PDM_OUT_CTRL: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL2: + case RT5670_PDM1_DATA_CTRL3: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL2: + case RT5670_PDM2_DATA_CTRL3: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_REC_L1_MIXER: + case RT5670_REC_L2_MIXER: + case RT5670_REC_R1_MIXER: + case RT5670_REC_R2_MIXER: + case RT5670_HPO_MIXER: + case RT5670_MONO_MIXER: + case RT5670_OUT_L1_MIXER: + case RT5670_OUT_R1_MIXER: + case RT5670_LOUT_MIXER: + case RT5670_PWR_DIG1: + case RT5670_PWR_DIG2: + case RT5670_PWR_ANLG1: + case RT5670_PWR_ANLG2: + case RT5670_PWR_MIXER: + case RT5670_PWR_VOL: + case RT5670_PRIV_INDEX: + case RT5670_PRIV_DATA: + case RT5670_I2S4_SDP: + case RT5670_I2S1_SDP: + case RT5670_I2S2_SDP: + case RT5670_I2S3_SDP: + case RT5670_ADDA_CLK1: + case RT5670_ADDA_CLK2: + case RT5670_DMIC_CTRL1: + case RT5670_DMIC_CTRL2: + case RT5670_TDM_CTRL_1: + case RT5670_TDM_CTRL_2: + case RT5670_TDM_CTRL_3: + case RT5670_DSP_CLK: + case RT5670_GLB_CLK: + case RT5670_PLL_CTRL1: + case RT5670_PLL_CTRL2: + case RT5670_ASRC_1: + case RT5670_ASRC_2: + case RT5670_ASRC_3: + case RT5670_ASRC_4: + case RT5670_ASRC_5: + case RT5670_ASRC_7: + case RT5670_ASRC_8: + case RT5670_ASRC_9: + case RT5670_ASRC_10: + case RT5670_ASRC_11: + case RT5670_ASRC_12: + case RT5670_ASRC_13: + case RT5670_ASRC_14: + case RT5670_DEPOP_M1: + case RT5670_DEPOP_M2: + case RT5670_DEPOP_M3: + case RT5670_CHARGE_PUMP: + case RT5670_MICBIAS: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL1: + case RT5670_VAD_CTRL2: + case RT5670_VAD_CTRL3: + case RT5670_VAD_CTRL4: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_ADC_EQ_CTRL2: + case RT5670_EQ_CTRL1: + case RT5670_EQ_CTRL2: + case RT5670_ALC_DRC_CTRL1: + case RT5670_ALC_DRC_CTRL2: + case RT5670_ALC_CTRL_1: + case RT5670_ALC_CTRL_2: + case RT5670_ALC_CTRL_3: + case RT5670_JD_CTRL: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_INT_IRQ_ST: + case RT5670_GPIO_CTRL1: + case RT5670_GPIO_CTRL2: + case RT5670_GPIO_CTRL3: + case RT5670_SCRABBLE_FUN: + case RT5670_SCRABBLE_CTRL: + case RT5670_BASE_BACK: + case RT5670_MP3_PLUS1: + case RT5670_MP3_PLUS2: + case RT5670_ADJ_HPF1: + case RT5670_ADJ_HPF2: + case RT5670_HP_CALIB_AMP_DET: + case RT5670_SV_ZCD1: + case RT5670_SV_ZCD2: + case RT5670_IL_CMD: + case RT5670_IL_CMD2: + case RT5670_IL_CMD3: + case RT5670_DRC_HL_CTRL1: + case RT5670_DRC_HL_CTRL2: + case RT5670_ADC_MONO_HP_CTRL1: + case RT5670_ADC_MONO_HP_CTRL2: + case RT5670_ADC_STO2_HP_CTRL1: + case RT5670_ADC_STO2_HP_CTRL2: + case RT5670_JD_CTRL3: + case RT5670_JD_CTRL4: + case RT5670_DIG_MISC: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_GEN_CTRL2: + case RT5670_GEN_CTRL3: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5670_data_select[] = { + "Normal", "Swap", "left copy to right", "right copy to left" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_DAC_SEL_SFT, rt5670_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); + +static const struct snd_kcontrol_new rt5670_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 39, 0, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, + RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL, + RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5670_DAC2_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost Volume", RT5670_CJ_CTRL1, + RT5670_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost Volume", RT5670_IN2, + RT5670_BST_SFT1, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5670_INL1_INR1_VOL, + RT5670_INL_VOL_SFT, RT5670_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5670_ADC_BST_VOL1, + RT5670_STO1_ADC_L_BST_SFT, RT5670_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_DOUBLE_TLV("STO2 ADC Boost Gain Volume", RT5670_ADC_BST_VOL1, + RT5670_STO2_ADC_L_BST_SFT, RT5670_STO2_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int idx = -EINVAL; + + idx = rl6231_calc_dmic_clk(rt5670->sysclk); + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5670_DMIC_CTRL1, + RT5670_DMIC_CLK_MASK, idx << RT5670_DMIC_CLK_SFT); + return idx; +} + +static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5670_GLB_CLK); + val &= RT5670_SCLK_SRC_MASK; + if (val == RT5670_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + + switch (source->shift) { + case 0: + reg = RT5670_ASRC_3; + shift = 0; + break; + case 1: + reg = RT5670_ASRC_3; + shift = 4; + break; + case 2: + reg = RT5670_ASRC_5; + shift = 12; + break; + case 3: + reg = RT5670_ASRC_2; + shift = 0; + break; + case 8: + reg = RT5670_ASRC_2; + shift = 4; + break; + case 9: + reg = RT5670_ASRC_2; + shift = 8; + break; + case 10: + reg = RT5670_ASRC_2; + shift = 12; + break; + default: + return 0; + } + + val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + switch (val) { + case 1: + case 2: + case 3: + case 4: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5670_DIG_MIXER, + RT5670_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5670_DIG_MIXER, + RT5670_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5670_rec_l_mix[] = { + SOC_DAPM_SINGLE("INL Switch", RT5670_REC_L2_MIXER, + RT5670_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST1_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_rec_r_mix[] = { + SOC_DAPM_SINGLE("INR Switch", RT5670_REC_R2_MIXER, + RT5670_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST1_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_OUT_L1_MIXER, + RT5670_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_OUT_R1_MIXER, + RT5670_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5670_HPO_MIXER, + RT5670_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvoll_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvolr_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX L Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX R Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpl_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL1 Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpr_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR1 Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new lout_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1, + RT5670_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1, + RT5670_R_MUTE_SFT, 1, 1); + +/* DAC1 L/R source */ /* MX-29 [9:8] [11:10] */ +static const char * const rt5670_dac1_src[] = { + "IF1 DAC", "IF2 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1l_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_L_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1l_mux = + SOC_DAPM_ENUM("DAC1 L source", rt5670_dac1l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1r_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_R_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1r_mux = + SOC_DAPM_ENUM("DAC1 R source", rt5670_dac1r_enum); + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +/* TODO Use SOC_VALUE_ENUM_SINGLE_DECL */ +static const char * const rt5670_dac12_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", + "Bass", "VAD_ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2l_enum, RT5670_DAC_CTRL, + RT5670_DAC2_L_SEL_SFT, rt5670_dac12_src); + +static const struct snd_kcontrol_new rt5670_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_dac2l_enum); + +static const char * const rt5670_dacr2_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", "TxDP ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2r_enum, RT5670_DAC_CTRL, + RT5670_DAC2_R_SEL_SFT, rt5670_dacr2_src); + +static const struct snd_kcontrol_new rt5670_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5670_dac2r_enum); + +/*RxDP source*/ /* MX-2D [15:13] */ +static const char * const rt5670_rxdp_src[] = { + "IF2 DAC", "IF1 DAC", "STO1 ADC Mixer", "STO2 ADC Mixer", + "Mono ADC Mixer L", "Mono ADC Mixer R", "DAC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SEL_SFT, rt5670_rxdp_src); + +static const struct snd_kcontrol_new rt5670_rxdp_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_rxdp_enum); + +/* MX-2D [1] [0] */ +static const char * const rt5670_dsp_bypass_src[] = { + "DSP", "Bypass" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_ul_enum, RT5670_DSP_PATH1, + RT5670_DSP_UL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_ul_mux = + SOC_DAPM_ENUM("DSP UL source", rt5670_dsp_ul_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_dl_enum, RT5670_DSP_PATH1, + RT5670_DSP_DL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_dl_mux = + SOC_DAPM_ENUM("DSP DL source", rt5670_dsp_dl_enum); + +/* Stereo2 ADC source */ +/* MX-26 [15] */ +static const char * const rt5670_stereo2_adc_lr_src[] = { + "L", "LR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER, + RT5670_STO2_ADC_SRC_SFT, rt5670_stereo2_adc_lr_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_lr_mux = + SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5670_stereo2_adc_lr_enum); + +/* Stereo1 ADC source */ +/* MX-27 MX-26 [12] */ +static const char * const rt5670_stereo_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = + SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = + SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); + +/* MX-27 MX-26 [11] */ +static const char * const rt5670_stereo_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = + SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = + SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = + SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = + SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); + +/* MX-27 MX26 [10] */ +static const char * const rt5670_stereo_adc_src[] = { + "ADC1L ADC2R", "ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_mux = + SOC_DAPM_ENUM("Stereo1 ADC source", rt5670_stereo1_adc_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_mux = + SOC_DAPM_ENUM("Stereo2 ADC source", rt5670_stereo2_adc_enum); + +/* MX-27 MX-26 [9:8] */ +static const char * const rt5670_stereo_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5670_stereo1_dmic_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto2_dmic_mux = + SOC_DAPM_ENUM("Stereo2 DMIC source", rt5670_stereo2_dmic_enum); + +/* MX-27 [0] */ +static const char * const rt5670_stereo_dmic3_src[] = { + "DMIC3", "PDM ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC3_SRC_SFT, rt5670_stereo_dmic3_src); + +static const struct snd_kcontrol_new rt5670_sto_dmic3_mux = + SOC_DAPM_ENUM("Stereo DMIC3 source", rt5670_stereo_dmic3_enum); + +/* Mono ADC source */ +/* MX-28 [12] */ +static const char * const rt5670_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L1_SRC_SFT, rt5670_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5670_mono_adc_l1_enum); +/* MX-28 [11] */ +static const char * const rt5670_mono_adc_l2_src[] = { + "Mono DAC MIXL", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L2_SRC_SFT, rt5670_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5670_mono_adc_l2_enum); + +/* MX-28 [9:8] */ +static const char * const rt5670_mono_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_L_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC left source", rt5670_mono_dmic_l_enum); +/* MX-28 [1:0] */ +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_R_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC Right source", rt5670_mono_dmic_r_enum); +/* MX-28 [4] */ +static const char * const rt5670_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADC2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R1_SRC_SFT, rt5670_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5670_mono_adc_r1_enum); +/* MX-28 [3] */ +static const char * const rt5670_mono_adc_r2_src[] = { + "Mono DAC MIXR", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R2_SRC_SFT, rt5670_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5670_mono_adc_r2_enum); + +/* MX-2D [3:2] */ +static const char * const rt5670_txdp_slot_src[] = { + "Slot 0-1", "Slot 2-3", "Slot 4-5", "Slot 6-7" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_txdp_slot_enum, RT5670_DSP_PATH1, + RT5670_TXDP_SLOT_SEL_SFT, rt5670_txdp_slot_src); + +static const struct snd_kcontrol_new rt5670_txdp_slot_mux = + SOC_DAPM_ENUM("TxDP Slot source", rt5670_txdp_slot_enum); + +/* MX-2F [15] */ +static const char * const rt5670_if1_adc2_in_src[] = { + "IF_ADC2", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF1_ADC2_IN_SFT, rt5670_if1_adc2_in_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN source", rt5670_if1_adc2_in_enum); + +/* MX-2F [14:12] */ +static const char * const rt5670_if2_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3", "TxDC_DAC", "TxDP_ADC", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_IN_SFT, rt5670_if2_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if2_adc_in_mux = + SOC_DAPM_ENUM("IF2 ADC IN source", rt5670_if2_adc_in_enum); + +/* MX-30 [5:4] */ +static const char * const rt5670_if4_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA, + RT5670_IF4_ADC_IN_SFT, rt5670_if4_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if4_adc_in_mux = + SOC_DAPM_ENUM("IF4 ADC IN source", rt5670_if4_adc_in_enum); + +/* MX-31 [15] [13] [11] [9] */ +static const char * const rt5670_pdm_src[] = { + "Mono DAC", "Stereo DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 L source", rt5670_pdm1_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 R source", rt5670_pdm1_r_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_l_mux = + SOC_DAPM_ENUM("PDM2 L source", rt5670_pdm2_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_r_mux = + SOC_DAPM_ENUM("PDM2 R source", rt5670_pdm2_r_enum); + +/* MX-FA [12] */ +static const char * const rt5670_if1_adc1_in1_src[] = { + "IF_ADC1", "IF1_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN1_SFT, rt5670_if1_adc1_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in1_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN1 source", rt5670_if1_adc1_in1_enum); + +/* MX-FA [11] */ +static const char * const rt5670_if1_adc1_in2_src[] = { + "IF1_ADC1_IN1", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN2_SFT, rt5670_if1_adc1_in2_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in2_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN2 source", rt5670_if1_adc1_in2_enum); + +/* MX-FA [10] */ +static const char * const rt5670_if1_adc2_in1_src[] = { + "IF1_ADC2_IN", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC2_IN1_SFT, rt5670_if1_adc2_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in1_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN1 source", rt5670_if1_adc2_in1_enum); + +/* MX-9D [9:8] */ +static const char * const rt5670_vad_adc_src[] = { + "Sto1 ADC L", "Mono ADC L", "Mono ADC R", "Sto2 ADC L" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_vad_adc_enum, RT5670_VAD_CTRL4, + RT5670_VAD_SEL_SFT, rt5670_vad_adc_src); + +static const struct snd_kcontrol_new rt5670_vad_adc_mux = + SOC_DAPM_ENUM("VAD ADC source", rt5670_vad_adc_enum); + +static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5670->regmap, RT5670_CHARGE_PUMP, + RT5670_PM_HP_MASK, RT5670_PM_HP_HV); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0400, 0x0400); + /* headphone amp power on */ + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HA | RT5670_PWR_FV1 | + RT5670_PWR_FV2, RT5670_PWR_HA | + RT5670_PWR_FV1 | RT5670_PWR_FV2); + /* depop parameters */ + regmap_write(rt5670->regmap, RT5670_DEPOP_M2, 0x3100); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8009); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_HP_DCC_INT1, 0x9f00); + mdelay(20); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x0004); + msleep(30); + break; + default: + return 0; + } + + return 0; +} + +static int rt5670_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* headphone unmute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x805d); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0300, 0x0300); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, 0); + msleep(80); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* headphone mute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x803d); + mdelay(10); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + mdelay(10); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, + RT5670_L_MUTE | RT5670_R_MUTE); + msleep(20); + regmap_update_bits(rt5670->regmap, + RT5670_GEN_CTRL2, 0x0300, 0x0); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0707); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xfc00); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, RT5670_PWR_BST1_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, RT5670_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5670_PWR_ANLG2, + RT5670_PWR_PLL_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S DSP", RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5670_PWR_VOL, + RT5670_PWR_MIC_DET_BIT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5670_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5670_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5670_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5670_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5670_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5670_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5670_ASRC_1, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5670_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5670_ASRC_1, + 0, 0, NULL, 0), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5670_PWR_ANLG2, + RT5670_PWR_MB1_BIT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_INPUT("DMIC L2"), + SND_SOC_DAPM_INPUT("DMIC R2"), + SND_SOC_DAPM_INPUT("DMIC L3"), + SND_SOC_DAPM_INPUT("DMIC R3"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_3_EN_SFT, 0, NULL, 0), + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5670_PWR_ANLG2, RT5670_PWR_BST1_BIT, + 0, NULL, 0, rt5670_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5670_PWR_ANLG2, RT5670_PWR_BST2_BIT, + 0, NULL, 0, rt5670_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_R_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5670_PWR_MIXER, RT5670_PWR_RM_L_BIT, 0, + rt5670_rec_l_mix, ARRAY_SIZE(rt5670_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5670_PWR_MIXER, RT5670_PWR_RM_R_BIT, 0, + rt5670_rec_r_mix, ARRAY_SIZE(rt5670_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 2", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PGA("ADC 1_2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC 1 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC 2 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", RT5670_PR_BASE + + RT5670_CHOP_DAC_ADC, 12, 0, NULL, 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_dmic_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_lr_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Stereo2 Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix, + ARRAY_SIZE(rt5670_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL, + RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix, + ARRAY_SIZE(rt5670_sto1_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_l_mix, + ARRAY_SIZE(rt5670_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_r_mix, + ARRAY_SIZE(rt5670_sto2_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("ADC Mono Left Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, 1, rt5670_mono_adc_l_mix, + ARRAY_SIZE(rt5670_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("ADC Mono Right Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", RT5670_MONO_ADC_DIG_VOL, + RT5670_R_MUTE_SFT, 1, rt5670_mono_adc_r_mix, + ARRAY_SIZE(rt5670_mono_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VAD_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DSP */ + SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDC_DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("TDM Data Mux", SND_SOC_NOPM, 0, 0, + &rt5670_txdp_slot_mux), + + SND_SOC_DAPM_MUX("DSP UL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_ul_mux), + SND_SOC_DAPM_MUX("DSP DL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_dl_mux), + + SND_SOC_DAPM_MUX("RxDP Mux", SND_SOC_NOPM, 0, 0, + &rt5670_rxdp_mux), + + /* IF2 Mux */ + SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if2_adc_in_mux), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5670_PWR_DIG1, + RT5670_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5670_PWR_DIG1, + RT5670_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 ADC1 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in1_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 IN2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in2_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in1_mux), + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_vad_adc_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5670_GPIO_CTRL1, RT5670_I2S2_PIN_SFT, 1), + + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dac_l_mix, ARRAY_SIZE(rt5670_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dac_r_mix, ARRAY_SIZE(rt5670_dac_r_mix)), + SND_SOC_DAPM_PGA("DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1l_mux), + SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mono Left Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mono Right Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_l_mix, + ARRAY_SIZE(rt5670_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_r_mix, + ARRAY_SIZE(rt5670_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_l_mix, + ARRAY_SIZE(rt5670_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_r_mix, + ARRAY_SIZE(rt5670_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dig_l_mix, + ARRAY_SIZE(rt5670_dig_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dig_r_mix, + ARRAY_SIZE(rt5670_dig_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0), + /* OUT Mixer */ + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5670_PWR_MIXER, RT5670_PWR_OM_L_BIT, + 0, rt5670_out_l_mix, ARRAY_SIZE(rt5670_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5670_PWR_MIXER, RT5670_PWR_OM_R_BIT, + 0, rt5670_out_r_mix, ARRAY_SIZE(rt5670_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5670_PWR_VOL, + RT5670_PWR_HV_L_BIT, 0, + rt5670_hpvoll_mix, ARRAY_SIZE(rt5670_hpvoll_mix)), + SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5670_PWR_VOL, + RT5670_PWR_HV_R_BIT, 0, + rt5670_hpvolr_mix, ARRAY_SIZE(rt5670_hpvolr_mix)), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, + rt5670_hpo_mix, ARRAY_SIZE(rt5670_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5670_PWR_ANLG1, RT5670_PWR_LM_BIT, + 0, rt5670_lout_mix, ARRAY_SIZE(rt5670_lout_mix)), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0, + &lout_l_enable_control), + SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0, + &lout_r_enable_control), + SND_SOC_DAPM_PGA("LOUT Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux), + SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("PDM1L"), + SND_SOC_DAPM_OUTPUT("PDM1R"), + SND_SOC_DAPM_OUTPUT("PDM2L"), + SND_SOC_DAPM_OUTPUT("PDM2R"), +}; + +static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { + { "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "ADC Mono Left Filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "ADC Mono Right Filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "DAC Mono Left Filter", NULL, "DAC MONO L ASRC", is_using_asrc }, + { "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc }, + { "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc }, + + { "I2S1", NULL, "I2S1 ASRC" }, + { "I2S2", NULL, "I2S2 ASRC" }, + + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + { "DMIC3", NULL, "DMIC L3" }, + { "DMIC3", NULL, "DMIC R3" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "Mic Det Power" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "INL VOL", NULL, "IN2P" }, + { "INR VOL", NULL, "IN2N" }, + + { "RECMIXL", "INL Switch", "INL VOL" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + + { "RECMIXR", "INR Switch", "INR VOL" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + + { "ADC 1", NULL, "RECMIXL" }, + { "ADC 1", NULL, "ADC 1 power" }, + { "ADC 1", NULL, "ADC clock" }, + { "ADC 2", NULL, "RECMIXR" }, + { "ADC 2", NULL, "ADC 2 power" }, + { "ADC 2", NULL, "ADC clock" }, + + { "DMIC L1", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 Power" }, + { "DMIC R1", NULL, "DMIC CLK" }, + { "DMIC R1", NULL, "DMIC1 Power" }, + { "DMIC L2", NULL, "DMIC CLK" }, + { "DMIC L2", NULL, "DMIC2 Power" }, + { "DMIC R2", NULL, "DMIC CLK" }, + { "DMIC R2", NULL, "DMIC2 Power" }, + { "DMIC L3", NULL, "DMIC CLK" }, + { "DMIC L3", NULL, "DMIC3 Power" }, + { "DMIC R3", NULL, "DMIC CLK" }, + { "DMIC R3", NULL, "DMIC3 Power" }, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo2 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", "DMIC3", "DMIC L3" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", "DMIC3", "DMIC R3" }, + + { "ADC 1_2", NULL, "ADC 1" }, + { "ADC 1_2", NULL, "ADC 2" }, + + { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo1 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo1 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo1 ADC R2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Mono ADC L2 Mux", "DMIC", "Mono DMIC L Mux" }, + { "Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "ADC1", "ADC 1" }, + + { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + { "Mono ADC R1 Mux", "ADC2", "ADC 2" }, + { "Mono ADC R2 Mux", "DMIC", "Mono DMIC R Mux" }, + { "Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + + { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter" }, + { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter" }, + { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux" }, + { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux" }, + { "Mono ADC MIXL", NULL, "ADC Mono Left Filter" }, + { "ADC Mono Left Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux" }, + { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux" }, + { "Mono ADC MIXR", NULL, "ADC Mono Right Filter" }, + { "ADC Mono Right Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC L2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo2 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo2 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo2 ADC R2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Sto2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC L1 Mux" }, + { "Sto2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC L2 Mux" }, + { "Sto2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC R1 Mux" }, + { "Sto2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC R2 Mux" }, + + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXL" }, + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXR" }, + + { "Stereo2 ADC LR Mux", "L", "Sto2 ADC MIXL" }, + { "Stereo2 ADC LR Mux", "LR", "Sto2 ADC LR MIX" }, + + { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, + { "Stereo2 ADC MIXL", NULL, "ADC Stereo2 Filter" }, + { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, + { "Stereo2 ADC MIXR", NULL, "ADC Stereo2 Filter" }, + { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "VAD ADC Mux", "Sto1 ADC L", "Stereo1 ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC L", "Mono ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC R", "Mono ADC MIXR" }, + { "VAD ADC Mux", "Sto2 ADC L", "Sto2 ADC MIXL" }, + + { "VAD_ADC", NULL, "VAD ADC Mux" }, + + { "IF_ADC1", NULL, "Stereo1 ADC MIXL" }, + { "IF_ADC1", NULL, "Stereo1 ADC MIXR" }, + { "IF_ADC2", NULL, "Mono ADC MIXL" }, + { "IF_ADC2", NULL, "Mono ADC MIXR" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXL" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXR" }, + + { "IF1 ADC1 IN1 Mux", "IF_ADC1", "IF_ADC1" }, + { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, + + { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, + { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, + { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, + + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" }, + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXL" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXR" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXL" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXR" }, + + { "RxDP Mux", "IF2 DAC", "IF2 DAC" }, + { "RxDP Mux", "IF1 DAC", "IF1 DAC2" }, + { "RxDP Mux", "STO1 ADC Mixer", "Stereo1 ADC MIX" }, + { "RxDP Mux", "STO2 ADC Mixer", "Stereo2 ADC MIX" }, + { "RxDP Mux", "Mono ADC Mixer L", "Mono ADC MIXL" }, + { "RxDP Mux", "Mono ADC Mixer R", "Mono ADC MIXR" }, + { "RxDP Mux", "DAC1", "DAC MIX" }, + + { "TDM Data Mux", "Slot 0-1", "Stereo1 ADC MIX" }, + { "TDM Data Mux", "Slot 2-3", "Mono ADC MIX" }, + { "TDM Data Mux", "Slot 4-5", "Stereo2 ADC MIX" }, + { "TDM Data Mux", "Slot 6-7", "IF2 DAC" }, + + { "DSP UL Mux", "Bypass", "TDM Data Mux" }, + { "DSP UL Mux", NULL, "I2S DSP" }, + { "DSP DL Mux", "Bypass", "RxDP Mux" }, + { "DSP DL Mux", NULL, "I2S DSP" }, + + { "TxDP_ADC_L", NULL, "DSP UL Mux" }, + { "TxDP_ADC_R", NULL, "DSP UL Mux" }, + { "TxDC_DAC", NULL, "DSP DL Mux" }, + + { "TxDP_ADC", NULL, "TxDP_ADC_L" }, + { "TxDP_ADC", NULL, "TxDP_ADC_R" }, + + { "IF1 ADC", NULL, "I2S1" }, + { "IF1 ADC", NULL, "IF1_ADC1" }, + { "IF1 ADC", NULL, "IF1_ADC2" }, + { "IF1 ADC", NULL, "IF_ADC3" }, + { "IF1 ADC", NULL, "TxDP_ADC" }, + + { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF2 ADC Mux", "IF_ADC3", "IF_ADC3" }, + { "IF2 ADC Mux", "TxDC_DAC", "TxDC_DAC" }, + { "IF2 ADC Mux", "TxDP_ADC", "TxDP_ADC" }, + { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF2 ADC L", NULL, "IF2 ADC Mux" }, + { "IF2 ADC R", NULL, "IF2 ADC Mux" }, + + { "IF2 ADC", NULL, "I2S2" }, + { "IF2 ADC", NULL, "IF2 ADC L" }, + { "IF2 ADC", NULL, "IF2 ADC R" }, + + { "AIF1TX", NULL, "IF1 ADC" }, + { "AIF2TX", NULL, "IF2 ADC" }, + + { "IF1 DAC1", NULL, "AIF1RX" }, + { "IF1 DAC2", NULL, "AIF1RX" }, + { "IF2 DAC", NULL, "AIF2RX" }, + + { "IF1 DAC1", NULL, "I2S1" }, + { "IF1 DAC2", NULL, "I2S1" }, + { "IF2 DAC", NULL, "I2S2" }, + + { "IF1 DAC2 L", NULL, "IF1 DAC2" }, + { "IF1 DAC2 R", NULL, "IF1 DAC2" }, + { "IF1 DAC1 L", NULL, "IF1 DAC1" }, + { "IF1 DAC1 R", NULL, "IF1 DAC1" }, + { "IF2 DAC L", NULL, "IF2 DAC" }, + { "IF2 DAC R", NULL, "IF2 DAC" }, + + { "DAC1 L Mux", "IF1 DAC", "IF1 DAC1 L" }, + { "DAC1 L Mux", "IF2 DAC", "IF2 DAC L" }, + + { "DAC1 R Mux", "IF1 DAC", "IF1 DAC1 R" }, + { "DAC1 R Mux", "IF2 DAC", "IF2 DAC R" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "DAC1 L Mux" }, + { "DAC1 MIXL", NULL, "DAC Stereo1 Filter" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "DAC1 R Mux" }, + { "DAC1 MIXR", NULL, "DAC Stereo1 Filter" }, + + { "DAC MIX", NULL, "DAC1 MIXL" }, + { "DAC MIX", NULL, "DAC1 MIXR" }, + + { "Audio DSP", NULL, "DAC1 MIXL" }, + { "Audio DSP", NULL, "DAC1 MIXR" }, + + { "DAC L2 Mux", "IF1 DAC", "IF1 DAC2 L" }, + { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" }, + { "DAC L2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC L2 Mux", "VAD_ADC", "VAD_ADC" }, + { "DAC L2 Volume", NULL, "DAC L2 Mux" }, + { "DAC L2 Volume", NULL, "DAC Mono Left Filter" }, + + { "DAC R2 Mux", "IF1 DAC", "IF1 DAC2 R" }, + { "DAC R2 Mux", "IF2 DAC", "IF2 DAC R" }, + { "DAC R2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC R2 Mux", "TxDP ADC", "TxDP_ADC" }, + { "DAC R2 Volume", NULL, "DAC R2 Mux" }, + { "DAC R2 Volume", NULL, "DAC Mono Right Filter" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Stereo DAC MIXL", NULL, "DAC Stereo1 Filter" }, + { "Stereo DAC MIXL", NULL, "DAC L1 Power" }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Stereo DAC MIXR", NULL, "DAC Stereo1 Filter" }, + { "Stereo DAC MIXR", NULL, "DAC R1 Power" }, + + { "Mono DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXL", NULL, "DAC Mono Left Filter" }, + { "Mono DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXR", NULL, "DAC Mono Right Filter" }, + + { "DAC MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + + { "DAC L1", NULL, "DAC L1 Power" }, + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R1", NULL, "DAC R1 Power" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R2", NULL, "Mono DAC MIXR" }, + { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, + + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "INL Switch", "INL VOL" }, + { "OUT MIXL", "DAC L2 Switch", "DAC L2" }, + { "OUT MIXL", "DAC L1 Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "INR Switch", "INR VOL" }, + { "OUT MIXR", "DAC R2 Switch", "DAC R2" }, + { "OUT MIXR", "DAC R1 Switch", "DAC R1" }, + + { "HPOVOL MIXL", "DAC1 Switch", "DAC L1" }, + { "HPOVOL MIXL", "INL Switch", "INL VOL" }, + { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" }, + { "HPOVOL MIXR", "INR Switch", "INR VOL" }, + + { "DAC 2", NULL, "DAC L2" }, + { "DAC 2", NULL, "DAC R2" }, + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + { "HPOVOL", NULL, "HPOVOL MIXL" }, + { "HPOVOL", NULL, "HPOVOL MIXR" }, + { "HPO MIX", "DAC1 Switch", "DAC 1" }, + { "HPO MIX", "HPVOL Switch", "HPOVOL" }, + + { "LOUT MIX", "DAC L1 Switch", "DAC L1" }, + { "LOUT MIX", "DAC R1 Switch", "DAC R1" }, + { "LOUT MIX", "OUTMIX L Switch", "OUT MIXL" }, + { "LOUT MIX", "OUTMIX R Switch", "OUT MIXR" }, + + { "PDM1 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM1 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM1 L Mux", NULL, "PDM1 Power" }, + { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM1 R Mux", NULL, "PDM1 Power" }, + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, + + { "HP Amp", NULL, "HPO MIX" }, + { "HP Amp", NULL, "Mic Det Power" }, + { "HPOL", NULL, "HP Amp" }, + { "HPOL", NULL, "HP L Amp" }, + { "HPOL", NULL, "Improve HP Amp Drv" }, + { "HPOR", NULL, "HP Amp" }, + { "HPOR", NULL, "HP R Amp" }, + { "HPOR", NULL, "Improve HP Amp Drv" }, + + { "LOUT Amp", NULL, "LOUT MIX" }, + { "LOUT L Playback", "Switch", "LOUT Amp" }, + { "LOUT R Playback", "Switch", "LOUT Amp" }, + { "LOUTL", NULL, "LOUT L Playback" }, + { "LOUTR", NULL, "LOUT R Playback" }, + { "LOUTL", NULL, "Improve HP Amp Drv" }, + { "LOUTR", NULL, "Improve HP Amp Drv" }, + + { "PDM1L", NULL, "PDM1 L Mux" }, + { "PDM1R", NULL, "PDM1 R Mux" }, + { "PDM2L", NULL, "PDM2 L Mux" }, + { "PDM2R", NULL, "PDM2 R Mux" }, +}; + +static int rt5670_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5670->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5670->sysclk, rt5670->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5670->lrck[dai->id], dai->id); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32; + rt5670->bclk[dai->id] = rt5670->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5670->bclk[dai->id], rt5670->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= RT5670_I2S_DL_20; + break; + case 24: + val_len |= RT5670_I2S_DL_24; + break; + case 8: + val_len |= RT5670_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + mask_clk = RT5670_I2S_BCLK_MS1_MASK | RT5670_I2S_PD1_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS1_SFT | + pre_div << RT5670_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5670_AIF2: + mask_clk = RT5670_I2S_BCLK_MS2_MASK | RT5670_I2S_PD2_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS2_SFT | + pre_div << RT5670_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5670->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5670_I2S_MS_S; + rt5670->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5670_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5670_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5670_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5670_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) + return 0; + + switch (clk_id) { + case RT5670_SCLK_S_MCLK: + reg_val |= RT5670_SCLK_SRC_MCLK; + break; + case RT5670_SCLK_S_PLL1: + reg_val |= RT5670_SCLK_SRC_PLL1; + break; + case RT5670_SCLK_S_RCCLK: + reg_val |= RT5670_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, reg_val); + rt5670->sysclk = freq; + rt5670->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5670->pll_src && freq_in == rt5670->pll_in && + freq_out == rt5670->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5670->pll_in = 0; + rt5670->pll_out = 0; + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5670_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_MCLK); + break; + case RT5670_PLL1_S_BCLK1: + case RT5670_PLL1_S_BCLK2: + case RT5670_PLL1_S_BCLK3: + case RT5670_PLL1_S_BCLK4: + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK1); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK2); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5670_PLL_CTRL1, + pll_code.n_code << RT5670_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5670_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5670_PLL_M_SFT | + pll_code.m_bp << RT5670_PLL_M_BP_SFT); + + rt5670->pll_in = freq_in; + rt5670->pll_out = freq_out; + rt5670->pll_src = source; + + return 0; +} + +static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 14); + + switch (slots) { + case 4: + val |= (1 << 12); + break; + case 6: + val |= (2 << 12); + break; + case 8: + val |= (3 << 12); + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= (1 << 10); + break; + case 24: + val |= (2 << 10); + break; + case 32: + val |= (3 << 10); + break; + case 16: + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5670_TDM_CTRL_1, 0x7c00, val); + + return 0; +} + +static int rt5670_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_FV1 | RT5670_PWR_FV2); + snd_soc_update_bits(codec, RT5670_CHARGE_PUMP, + RT5670_OSW_L_MASK | RT5670_OSW_R_MASK, + RT5670_OSW_L_DIS | RT5670_OSW_R_DIS); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x1); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x3); + } + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); + snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); + snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); + snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x1); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5670_probe(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + rt5670->codec = codec; + + return 0; +} + +static int rt5670_remove(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + return 0; +} + +#ifdef CONFIG_PM +static int rt5670_suspend(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, true); + regcache_mark_dirty(rt5670->regmap); + return 0; +} + +static int rt5670_resume(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, false); + regcache_sync(rt5670->regmap); + + return 0; +} +#else +#define rt5670_suspend NULL +#define rt5670_resume NULL +#endif + +#define RT5670_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5670_aif_dai_ops = { + .hw_params = rt5670_hw_params, + .set_fmt = rt5670_set_dai_fmt, + .set_sysclk = rt5670_set_dai_sysclk, + .set_tdm_slot = rt5670_set_tdm_slot, + .set_pll = rt5670_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5670_dai[] = { + { + .name = "rt5670-aif1", + .id = RT5670_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, + { + .name = "rt5670-aif2", + .id = RT5670_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { + .probe = rt5670_probe, + .remove = rt5670_remove, + .suspend = rt5670_suspend, + .resume = rt5670_resume, + .set_bias_level = rt5670_set_bias_level, + .idle_bias_off = true, + .controls = rt5670_snd_controls, + .num_controls = ARRAY_SIZE(rt5670_snd_controls), + .dapm_widgets = rt5670_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5670_dapm_widgets), + .dapm_routes = rt5670_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes), +}; + +static const struct regmap_config rt5670_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * + RT5670_PR_SPACING), + .volatile_reg = rt5670_volatile_register, + .readable_reg = rt5670_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5670_reg, + .num_reg_defaults = ARRAY_SIZE(rt5670_reg), + .ranges = rt5670_ranges, + .num_ranges = ARRAY_SIZE(rt5670_ranges), +}; + +static const struct i2c_device_id rt5670_i2c_id[] = { + { "rt5670", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); + +static int rt5670_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5670_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5670_priv *rt5670; + int ret; + unsigned int val; + + rt5670 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5670_priv), + GFP_KERNEL); + if (NULL == rt5670) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5670); + + if (pdata) + rt5670->pdata = *pdata; + + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); + if (IS_ERR(rt5670->regmap)) { + ret = PTR_ERR(rt5670->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5670->regmap, RT5670_VENDOR_ID2, &val); + if (val != RT5670_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5670/72\n", val); + return -ENODEV; + } + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HP_L | RT5670_PWR_HP_R | + RT5670_PWR_VREF2, RT5670_PWR_VREF2); + msleep(100); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + + ret = regmap_register_patch(rt5670->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5670->pdata.in2_diff) + regmap_update_bits(rt5670->regmap, RT5670_IN2, + RT5670_IN_DF2, RT5670_IN_DF2); + + if (i2c->irq) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); + + } + + if (rt5670->pdata.jd_mode) { + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_MB, RT5670_PWR_MB); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, + RT5670_PWR_JD1, RT5670_PWR_JD1); + regmap_update_bits(rt5670->regmap, RT5670_IRQ_CTRL1, + RT5670_JD1_1_EN_MASK, RT5670_JD1_1_EN); + regmap_update_bits(rt5670->regmap, RT5670_JD_CTRL3, + RT5670_JD_TRI_CBJ_SEL_MASK | + RT5670_JD_TRI_HPO_SEL_MASK, + RT5670_JD_CBJ_JD1_1 | RT5670_JD_HPO_JD1_1); + switch (rt5670->pdata.jd_mode) { + case 1: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_0); + break; + case 2: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_1); + break; + case 3: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_2); + break; + default: + break; + } + } + + if (rt5670->pdata.dmic_en) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP2_PIN_MASK, + RT5670_GP2_PIN_DMIC1_SCL); + + switch (rt5670->pdata.dmic1_data_pin) { + case RT5670_DMIC_DATA_IN2P: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_IN2P); + break; + + case RT5670_DMIC_DATA_GPIO6: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_GPIO6); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP6_PIN_MASK, + RT5670_GP6_PIN_DMIC1_SDA); + break; + + case RT5670_DMIC_DATA_GPIO7: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_GPIO7); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP7_PIN_MASK, + RT5670_GP7_PIN_DMIC1_SDA); + break; + + default: + break; + } + + switch (rt5670->pdata.dmic2_data_pin) { + case RT5670_DMIC_DATA_IN3N: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_2_DP_MASK, + RT5670_DMIC_2_DP_IN3N); + break; + + case RT5670_DMIC_DATA_GPIO8: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_2_DP_MASK, + RT5670_DMIC_2_DP_GPIO8); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP8_PIN_MASK, + RT5670_GP8_PIN_DMIC2_SDA); + break; + + default: + break; + } + + switch (rt5670->pdata.dmic3_data_pin) { + case RT5670_DMIC_DATA_GPIO5: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL2, + RT5670_DMIC_3_DP_MASK, + RT5670_DMIC_3_DP_GPIO5); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP5_PIN_MASK, + RT5670_GP5_PIN_DMIC3_SDA); + break; + + case RT5670_DMIC_DATA_GPIO9: + case RT5670_DMIC_DATA_GPIO10: + dev_err(&i2c->dev, + "Always use GPIO5 as DMIC3 data pin\n"); + break; + + default: + break; + } + + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, + rt5670_dai, ARRAY_SIZE(rt5670_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5670_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt5670_i2c_driver = { + .driver = { + .name = "rt5670", + .owner = THIS_MODULE, + }, + .probe = rt5670_i2c_probe, + .remove = rt5670_i2c_remove, + .id_table = rt5670_i2c_id, +}; + +module_i2c_driver(rt5670_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5670 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h new file mode 100644 index 0000000..a0b5c85 --- /dev/null +++ b/sound/soc/codecs/rt5670.h @@ -0,0 +1,2000 @@ +/* + * rt5670.h -- RT5670 ALSA SoC audio driver + * + * Copyright 2014 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5670_H__ +#define __RT5670_H__ + +#include <sound/rt5670.h> + +/* Info */ +#define RT5670_RESET 0x00 +#define RT5670_VENDOR_ID 0xfd +#define RT5670_VENDOR_ID1 0xfe +#define RT5670_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5670_HP_VOL 0x02 +#define RT5670_LOUT1 0x03 +/* I/O - Input */ +#define RT5670_CJ_CTRL1 0x0a +#define RT5670_CJ_CTRL2 0x0b +#define RT5670_CJ_CTRL3 0x0c +#define RT5670_IN2 0x0e +#define RT5670_INL1_INR1_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5670_DAC1_DIG_VOL 0x19 +#define RT5670_DAC2_DIG_VOL 0x1a +#define RT5670_DAC_CTRL 0x1b +#define RT5670_STO1_ADC_DIG_VOL 0x1c +#define RT5670_MONO_ADC_DIG_VOL 0x1d +#define RT5670_ADC_BST_VOL1 0x1e +#define RT5670_STO2_ADC_DIG_VOL 0x1f +/* Mixer - D-D */ +#define RT5670_ADC_BST_VOL2 0x20 +#define RT5670_STO2_ADC_MIXER 0x26 +#define RT5670_STO1_ADC_MIXER 0x27 +#define RT5670_MONO_ADC_MIXER 0x28 +#define RT5670_AD_DA_MIXER 0x29 +#define RT5670_STO_DAC_MIXER 0x2a +#define RT5670_DD_MIXER 0x2b +#define RT5670_DIG_MIXER 0x2c +#define RT5670_DSP_PATH1 0x2d +#define RT5670_DSP_PATH2 0x2e +#define RT5670_DIG_INF1_DATA 0x2f +#define RT5670_DIG_INF2_DATA 0x30 +/* Mixer - PDM */ +#define RT5670_PDM_OUT_CTRL 0x31 +#define RT5670_PDM_DATA_CTRL1 0x32 +#define RT5670_PDM1_DATA_CTRL2 0x33 +#define RT5670_PDM1_DATA_CTRL3 0x34 +#define RT5670_PDM1_DATA_CTRL4 0x35 +#define RT5670_PDM2_DATA_CTRL2 0x36 +#define RT5670_PDM2_DATA_CTRL3 0x37 +#define RT5670_PDM2_DATA_CTRL4 0x38 +/* Mixer - ADC */ +#define RT5670_REC_L1_MIXER 0x3b +#define RT5670_REC_L2_MIXER 0x3c +#define RT5670_REC_R1_MIXER 0x3d +#define RT5670_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5670_HPO_MIXER 0x45 +#define RT5670_MONO_MIXER 0x4c +#define RT5670_OUT_L1_MIXER 0x4f +#define RT5670_OUT_R1_MIXER 0x52 +#define RT5670_LOUT_MIXER 0x53 +/* Power */ +#define RT5670_PWR_DIG1 0x61 +#define RT5670_PWR_DIG2 0x62 +#define RT5670_PWR_ANLG1 0x63 +#define RT5670_PWR_ANLG2 0x64 +#define RT5670_PWR_MIXER 0x65 +#define RT5670_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5670_PRIV_INDEX 0x6a +#define RT5670_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5670_I2S4_SDP 0x6f +#define RT5670_I2S1_SDP 0x70 +#define RT5670_I2S2_SDP 0x71 +#define RT5670_I2S3_SDP 0x72 +#define RT5670_ADDA_CLK1 0x73 +#define RT5670_ADDA_CLK2 0x74 +#define RT5670_DMIC_CTRL1 0x75 +#define RT5670_DMIC_CTRL2 0x76 +/* Format - TDM Control */ +#define RT5670_TDM_CTRL_1 0x77 +#define RT5670_TDM_CTRL_2 0x78 +#define RT5670_TDM_CTRL_3 0x79 + +/* Function - Analog */ +#define RT5670_DSP_CLK 0x7f +#define RT5670_GLB_CLK 0x80 +#define RT5670_PLL_CTRL1 0x81 +#define RT5670_PLL_CTRL2 0x82 +#define RT5670_ASRC_1 0x83 +#define RT5670_ASRC_2 0x84 +#define RT5670_ASRC_3 0x85 +#define RT5670_ASRC_4 0x86 +#define RT5670_ASRC_5 0x87 +#define RT5670_ASRC_7 0x89 +#define RT5670_ASRC_8 0x8a +#define RT5670_ASRC_9 0x8b +#define RT5670_ASRC_10 0x8c +#define RT5670_ASRC_11 0x8d +#define RT5670_DEPOP_M1 0x8e +#define RT5670_DEPOP_M2 0x8f +#define RT5670_DEPOP_M3 0x90 +#define RT5670_CHARGE_PUMP 0x91 +#define RT5670_MICBIAS 0x93 +#define RT5670_A_JD_CTRL1 0x94 +#define RT5670_A_JD_CTRL2 0x95 +#define RT5670_ASRC_12 0x97 +#define RT5670_ASRC_13 0x98 +#define RT5670_ASRC_14 0x99 +#define RT5670_VAD_CTRL1 0x9a +#define RT5670_VAD_CTRL2 0x9b +#define RT5670_VAD_CTRL3 0x9c +#define RT5670_VAD_CTRL4 0x9d +#define RT5670_VAD_CTRL5 0x9e +/* Function - Digital */ +#define RT5670_ADC_EQ_CTRL1 0xae +#define RT5670_ADC_EQ_CTRL2 0xaf +#define RT5670_EQ_CTRL1 0xb0 +#define RT5670_EQ_CTRL2 0xb1 +#define RT5670_ALC_DRC_CTRL1 0xb2 +#define RT5670_ALC_DRC_CTRL2 0xb3 +#define RT5670_ALC_CTRL_1 0xb4 +#define RT5670_ALC_CTRL_2 0xb5 +#define RT5670_ALC_CTRL_3 0xb6 +#define RT5670_ALC_CTRL_4 0xb7 +#define RT5670_JD_CTRL 0xbb +#define RT5670_IRQ_CTRL1 0xbd +#define RT5670_IRQ_CTRL2 0xbe +#define RT5670_INT_IRQ_ST 0xbf +#define RT5670_GPIO_CTRL1 0xc0 +#define RT5670_GPIO_CTRL2 0xc1 +#define RT5670_GPIO_CTRL3 0xc2 +#define RT5670_SCRABBLE_FUN 0xcd +#define RT5670_SCRABBLE_CTRL 0xce +#define RT5670_BASE_BACK 0xcf +#define RT5670_MP3_PLUS1 0xd0 +#define RT5670_MP3_PLUS2 0xd1 +#define RT5670_ADJ_HPF1 0xd3 +#define RT5670_ADJ_HPF2 0xd4 +#define RT5670_HP_CALIB_AMP_DET 0xd6 +#define RT5670_SV_ZCD1 0xd9 +#define RT5670_SV_ZCD2 0xda +#define RT5670_IL_CMD 0xdb +#define RT5670_IL_CMD2 0xdc +#define RT5670_IL_CMD3 0xdd +#define RT5670_DRC_HL_CTRL1 0xe6 +#define RT5670_DRC_HL_CTRL2 0xe7 +#define RT5670_ADC_MONO_HP_CTRL1 0xec +#define RT5670_ADC_MONO_HP_CTRL2 0xed +#define RT5670_ADC_STO2_HP_CTRL1 0xee +#define RT5670_ADC_STO2_HP_CTRL2 0xef +#define RT5670_JD_CTRL3 0xf8 +#define RT5670_JD_CTRL4 0xf9 +/* General Control */ +#define RT5670_DIG_MISC 0xfa +#define RT5670_GEN_CTRL2 0xfb +#define RT5670_GEN_CTRL3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5670_DIG_VOL 0x00 +#define RT5670_PR_ALC_CTRL_1 0x01 +#define RT5670_PR_ALC_CTRL_2 0x02 +#define RT5670_PR_ALC_CTRL_3 0x03 +#define RT5670_PR_ALC_CTRL_4 0x04 +#define RT5670_PR_ALC_CTRL_5 0x05 +#define RT5670_PR_ALC_CTRL_6 0x06 +#define RT5670_BIAS_CUR1 0x12 +#define RT5670_BIAS_CUR3 0x14 +#define RT5670_CLSD_INT_REG1 0x1c +#define RT5670_MAMP_INT_REG2 0x37 +#define RT5670_CHOP_DAC_ADC 0x3d +#define RT5670_MIXER_INT_REG 0x3f +#define RT5670_3D_SPK 0x63 +#define RT5670_WND_1 0x6c +#define RT5670_WND_2 0x6d +#define RT5670_WND_3 0x6e +#define RT5670_WND_4 0x6f +#define RT5670_WND_5 0x70 +#define RT5670_WND_8 0x73 +#define RT5670_DIP_SPK_INF 0x75 +#define RT5670_HP_DCC_INT1 0x77 +#define RT5670_EQ_BW_LOP 0xa0 +#define RT5670_EQ_GN_LOP 0xa1 +#define RT5670_EQ_FC_BP1 0xa2 +#define RT5670_EQ_BW_BP1 0xa3 +#define RT5670_EQ_GN_BP1 0xa4 +#define RT5670_EQ_FC_BP2 0xa5 +#define RT5670_EQ_BW_BP2 0xa6 +#define RT5670_EQ_GN_BP2 0xa7 +#define RT5670_EQ_FC_BP3 0xa8 +#define RT5670_EQ_BW_BP3 0xa9 +#define RT5670_EQ_GN_BP3 0xaa +#define RT5670_EQ_FC_BP4 0xab +#define RT5670_EQ_BW_BP4 0xac +#define RT5670_EQ_GN_BP4 0xad +#define RT5670_EQ_FC_HIP1 0xae +#define RT5670_EQ_GN_HIP1 0xaf +#define RT5670_EQ_FC_HIP2 0xb0 +#define RT5670_EQ_BW_HIP2 0xb1 +#define RT5670_EQ_GN_HIP2 0xb2 +#define RT5670_EQ_PRE_VOL 0xb3 +#define RT5670_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5670_L_MUTE (0x1 << 15) +#define RT5670_L_MUTE_SFT 15 +#define RT5670_VOL_L_MUTE (0x1 << 14) +#define RT5670_VOL_L_SFT 14 +#define RT5670_R_MUTE (0x1 << 7) +#define RT5670_R_MUTE_SFT 7 +#define RT5670_VOL_R_MUTE (0x1 << 6) +#define RT5670_VOL_R_SFT 6 +#define RT5670_L_VOL_MASK (0x3f << 8) +#define RT5670_L_VOL_SFT 8 +#define RT5670_R_VOL_MASK (0x3f) +#define RT5670_R_VOL_SFT 0 + +/* Combo Jack Control 1 (0x0a) */ +#define RT5670_CBJ_BST1_MASK (0xf << 12) +#define RT5670_CBJ_BST1_SFT (12) +#define RT5670_CBJ_JD_HP_EN (0x1 << 9) +#define RT5670_CBJ_JD_MIC_EN (0x1 << 8) +#define RT5670_CBJ_BST1_EN (0x1 << 2) + +/* Combo Jack Control 1 (0x0b) */ +#define RT5670_CBJ_MN_JD (0x1 << 12) +#define RT5670_CAPLESS_EN (0x1 << 11) +#define RT5670_CBJ_DET_MODE (0x1 << 7) + +/* IN2 Control (0x0e) */ +#define RT5670_BST_MASK1 (0xf<<12) +#define RT5670_BST_SFT1 12 +#define RT5670_BST_MASK2 (0xf<<8) +#define RT5670_BST_SFT2 8 +#define RT5670_IN_DF1 (0x1 << 7) +#define RT5670_IN_SFT1 7 +#define RT5670_IN_DF2 (0x1 << 6) +#define RT5670_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5670_INL_SEL_MASK (0x1 << 15) +#define RT5670_INL_SEL_SFT 15 +#define RT5670_INL_SEL_IN4P (0x0 << 15) +#define RT5670_INL_SEL_MONOP (0x1 << 15) +#define RT5670_INL_VOL_MASK (0x1f << 8) +#define RT5670_INL_VOL_SFT 8 +#define RT5670_INR_SEL_MASK (0x1 << 7) +#define RT5670_INR_SEL_SFT 7 +#define RT5670_INR_SEL_IN4N (0x0 << 7) +#define RT5670_INR_SEL_MONON (0x1 << 7) +#define RT5670_INR_VOL_MASK (0x1f) +#define RT5670_INR_VOL_SFT 0 + +/* Sidetone Control (0x18) */ +#define RT5670_ST_SEL_MASK (0x7 << 9) +#define RT5670_ST_SEL_SFT 9 +#define RT5670_M_ST_DACR2 (0x1 << 8) +#define RT5670_M_ST_DACR2_SFT 8 +#define RT5670_M_ST_DACL2 (0x1 << 7) +#define RT5670_M_ST_DACL2_SFT 7 +#define RT5670_ST_EN (0x1 << 6) +#define RT5670_ST_EN_SFT 6 + +/* DAC1 Digital Volume (0x19) */ +#define RT5670_DAC_L1_VOL_MASK (0xff << 8) +#define RT5670_DAC_L1_VOL_SFT 8 +#define RT5670_DAC_R1_VOL_MASK (0xff) +#define RT5670_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5670_DAC_L2_VOL_MASK (0xff << 8) +#define RT5670_DAC_L2_VOL_SFT 8 +#define RT5670_DAC_R2_VOL_MASK (0xff) +#define RT5670_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5670_M_DAC_L2_VOL (0x1 << 13) +#define RT5670_M_DAC_L2_VOL_SFT 13 +#define RT5670_M_DAC_R2_VOL (0x1 << 12) +#define RT5670_M_DAC_R2_VOL_SFT 12 +#define RT5670_DAC2_L_SEL_MASK (0x7 << 4) +#define RT5670_DAC2_L_SEL_SFT 4 +#define RT5670_DAC2_R_SEL_MASK (0x7 << 0) +#define RT5670_DAC2_R_SEL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5670_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_ADC_L_VOL_SFT 8 +#define RT5670_ADC_R_VOL_MASK (0x7f) +#define RT5670_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5670_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_MONO_ADC_L_VOL_SFT 8 +#define RT5670_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5670_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5670_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5670_STO1_ADC_L_BST_SFT 14 +#define RT5670_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5670_STO1_ADC_R_BST_SFT 12 +#define RT5670_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5670_STO1_ADC_COMP_SFT 10 +#define RT5670_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5670_STO2_ADC_L_BST_SFT 8 +#define RT5670_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5670_STO2_ADC_R_BST_SFT 6 +#define RT5670_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5670_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5670_STO2_ADC_SRC_MASK (0x1 << 15) +#define RT5670_STO2_ADC_SRC_SFT 15 + +/* Stereo ADC Mixer Control (0x26 0x27) */ +#define RT5670_M_ADC_L1 (0x1 << 14) +#define RT5670_M_ADC_L1_SFT 14 +#define RT5670_M_ADC_L2 (0x1 << 13) +#define RT5670_M_ADC_L2_SFT 13 +#define RT5670_ADC_1_SRC_MASK (0x1 << 12) +#define RT5670_ADC_1_SRC_SFT 12 +#define RT5670_ADC_1_SRC_ADC (0x1 << 12) +#define RT5670_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5670_ADC_2_SRC_MASK (0x1 << 11) +#define RT5670_ADC_2_SRC_SFT 11 +#define RT5670_ADC_SRC_MASK (0x1 << 10) +#define RT5670_ADC_SRC_SFT 10 +#define RT5670_DMIC_SRC_MASK (0x3 << 8) +#define RT5670_DMIC_SRC_SFT 8 +#define RT5670_M_ADC_R1 (0x1 << 6) +#define RT5670_M_ADC_R1_SFT 6 +#define RT5670_M_ADC_R2 (0x1 << 5) +#define RT5670_M_ADC_R2_SFT 5 +#define RT5670_DMIC3_SRC_MASK (0x1 << 1) +#define RT5670_DMIC3_SRC_SFT 0 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5670_M_MONO_ADC_L1 (0x1 << 14) +#define RT5670_M_MONO_ADC_L1_SFT 14 +#define RT5670_M_MONO_ADC_L2 (0x1 << 13) +#define RT5670_M_MONO_ADC_L2_SFT 13 +#define RT5670_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5670_MONO_ADC_L1_SRC_SFT 12 +#define RT5670_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5670_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5670_MONO_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5670_MONO_ADC_L2_SRC_SFT 11 +#define RT5670_MONO_ADC_L_SRC_MASK (0x1 << 10) +#define RT5670_MONO_ADC_L_SRC_SFT 10 +#define RT5670_MONO_DMIC_L_SRC_MASK (0x3 << 8) +#define RT5670_MONO_DMIC_L_SRC_SFT 8 +#define RT5670_M_MONO_ADC_R1 (0x1 << 6) +#define RT5670_M_MONO_ADC_R1_SFT 6 +#define RT5670_M_MONO_ADC_R2 (0x1 << 5) +#define RT5670_M_MONO_ADC_R2_SFT 5 +#define RT5670_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_SFT 4 +#define RT5670_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5670_MONO_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5670_MONO_ADC_R2_SRC_SFT 3 +#define RT5670_MONO_DMIC_R_SRC_MASK (0x3) +#define RT5670_MONO_DMIC_R_SRC_SFT 0 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5670_M_ADCMIX_L (0x1 << 15) +#define RT5670_M_ADCMIX_L_SFT 15 +#define RT5670_M_DAC1_L (0x1 << 14) +#define RT5670_M_DAC1_L_SFT 14 +#define RT5670_DAC1_R_SEL_MASK (0x3 << 10) +#define RT5670_DAC1_R_SEL_SFT 10 +#define RT5670_DAC1_R_SEL_IF1 (0x0 << 10) +#define RT5670_DAC1_R_SEL_IF2 (0x1 << 10) +#define RT5670_DAC1_R_SEL_IF3 (0x2 << 10) +#define RT5670_DAC1_R_SEL_IF4 (0x3 << 10) +#define RT5670_DAC1_L_SEL_MASK (0x3 << 8) +#define RT5670_DAC1_L_SEL_SFT 8 +#define RT5670_DAC1_L_SEL_IF1 (0x0 << 8) +#define RT5670_DAC1_L_SEL_IF2 (0x1 << 8) +#define RT5670_DAC1_L_SEL_IF3 (0x2 << 8) +#define RT5670_DAC1_L_SEL_IF4 (0x3 << 8) +#define RT5670_M_ADCMIX_R (0x1 << 7) +#define RT5670_M_ADCMIX_R_SFT 7 +#define RT5670_M_DAC1_R (0x1 << 6) +#define RT5670_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5670_M_DAC_L1 (0x1 << 14) +#define RT5670_M_DAC_L1_SFT 14 +#define RT5670_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_STO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2 (0x1 << 12) +#define RT5670_M_DAC_L2_SFT 12 +#define RT5670_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_STO_L_VOL_SFT 11 +#define RT5670_M_DAC_R1_STO_L (0x1 << 9) +#define RT5670_M_DAC_R1_STO_L_SFT 9 +#define RT5670_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R1_STO_L_VOL_SFT 8 +#define RT5670_M_DAC_R1 (0x1 << 6) +#define RT5670_M_DAC_R1_SFT 6 +#define RT5670_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_STO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2 (0x1 << 4) +#define RT5670_M_DAC_R2_SFT 4 +#define RT5670_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_STO_R_VOL_SFT 3 +#define RT5670_M_DAC_L1_STO_R (0x1 << 1) +#define RT5670_M_DAC_L1_STO_R_SFT 1 +#define RT5670_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5670_DAC_L1_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5670_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5670_M_DAC_L1_MONO_L_SFT 14 +#define RT5670_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5670_M_DAC_L2_MONO_L_SFT 12 +#define RT5670_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5670_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5670_M_DAC_R2_MONO_L_SFT 10 +#define RT5670_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5670_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5670_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5670_M_DAC_R1_MONO_R_SFT 6 +#define RT5670_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5670_M_DAC_R2_MONO_R_SFT 4 +#define RT5670_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5670_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5670_M_DAC_L2_MONO_R_SFT 2 +#define RT5670_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5670_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5670_M_STO_L_DAC_L (0x1 << 15) +#define RT5670_M_STO_L_DAC_L_SFT 15 +#define RT5670_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5670_STO_L_DAC_L_VOL_SFT 14 +#define RT5670_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5670_M_DAC_L2_DAC_L_SFT 13 +#define RT5670_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5670_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5670_M_STO_R_DAC_R (0x1 << 11) +#define RT5670_M_STO_R_DAC_R_SFT 11 +#define RT5670_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5670_STO_R_DAC_R_VOL_SFT 10 +#define RT5670_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5670_M_DAC_R2_DAC_R_SFT 9 +#define RT5670_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R2_DAC_R_VOL_SFT 8 +#define RT5670_M_DAC_R2_DAC_L (0x1 << 7) +#define RT5670_M_DAC_R2_DAC_L_SFT 7 +#define RT5670_DAC_R2_DAC_L_VOL_MASK (0x1 << 6) +#define RT5670_DAC_R2_DAC_L_VOL_SFT 6 +#define RT5670_M_DAC_L2_DAC_R (0x1 << 5) +#define RT5670_M_DAC_L2_DAC_R_SFT 5 +#define RT5670_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) +#define RT5670_DAC_L2_DAC_R_VOL_SFT 4 + +/* DSP Path Control 1 (0x2d) */ +#define RT5670_RXDP_SEL_MASK (0x7 << 13) +#define RT5670_RXDP_SEL_SFT 13 +#define RT5670_RXDP_SRC_MASK (0x3 << 11) +#define RT5670_RXDP_SRC_SFT 11 +#define RT5670_RXDP_SRC_NOR (0x0 << 11) +#define RT5670_RXDP_SRC_DIV2 (0x1 << 11) +#define RT5670_RXDP_SRC_DIV3 (0x2 << 11) +#define RT5670_TXDP_SRC_MASK (0x3 << 4) +#define RT5670_TXDP_SRC_SFT 4 +#define RT5670_TXDP_SRC_NOR (0x0 << 4) +#define RT5670_TXDP_SRC_DIV2 (0x1 << 4) +#define RT5670_TXDP_SRC_DIV3 (0x2 << 4) +#define RT5670_TXDP_SLOT_SEL_MASK (0x3 << 2) +#define RT5670_TXDP_SLOT_SEL_SFT 2 +#define RT5670_DSP_UL_SEL (0x1 << 1) +#define RT5670_DSP_UL_SFT 1 +#define RT5670_DSP_DL_SEL 0x1 +#define RT5670_DSP_DL_SFT 0 + +/* DSP Path Control 2 (0x2e) */ +#define RT5670_TXDP_L_VOL_MASK (0x7f << 8) +#define RT5670_TXDP_L_VOL_SFT 8 +#define RT5670_TXDP_R_VOL_MASK (0x7f) +#define RT5670_TXDP_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5670_IF1_ADC2_IN_SEL (0x1 << 15) +#define RT5670_IF1_ADC2_IN_SFT 15 +#define RT5670_IF2_ADC_IN_MASK (0x7 << 12) +#define RT5670_IF2_ADC_IN_SFT 12 +#define RT5670_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5670_IF2_DAC_SEL_SFT 10 +#define RT5670_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5670_IF2_ADC_SEL_SFT 8 + +/* Digital Interface Data Control (0x30) */ +#define RT5670_IF4_ADC_IN_MASK (0x3 << 4) +#define RT5670_IF4_ADC_IN_SFT 4 + +/* PDM Output Control (0x31) */ +#define RT5670_PDM1_L_MASK (0x1 << 15) +#define RT5670_PDM1_L_SFT 15 +#define RT5670_M_PDM1_L (0x1 << 14) +#define RT5670_M_PDM1_L_SFT 14 +#define RT5670_PDM1_R_MASK (0x1 << 13) +#define RT5670_PDM1_R_SFT 13 +#define RT5670_M_PDM1_R (0x1 << 12) +#define RT5670_M_PDM1_R_SFT 12 +#define RT5670_PDM2_L_MASK (0x1 << 11) +#define RT5670_PDM2_L_SFT 11 +#define RT5670_M_PDM2_L (0x1 << 10) +#define RT5670_M_PDM2_L_SFT 10 +#define RT5670_PDM2_R_MASK (0x1 << 9) +#define RT5670_PDM2_R_SFT 9 +#define RT5670_M_PDM2_R (0x1 << 8) +#define RT5670_M_PDM2_R_SFT 8 +#define RT5670_PDM2_BUSY (0x1 << 7) +#define RT5670_PDM1_BUSY (0x1 << 6) +#define RT5670_PDM_PATTERN (0x1 << 5) +#define RT5670_PDM_GAIN (0x1 << 4) +#define RT5670_PDM_DIV_MASK (0x3) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5670_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5670_G_HP_L_RM_L_SFT 13 +#define RT5670_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5670_G_IN_L_RM_L_SFT 10 +#define RT5670_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_L_SFT 7 +#define RT5670_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_L_SFT 4 +#define RT5670_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5670_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_L_SFT 13 +#define RT5670_M_IN_L_RM_L (0x1 << 5) +#define RT5670_M_IN_L_RM_L_SFT 5 +#define RT5670_M_BST2_RM_L (0x1 << 3) +#define RT5670_M_BST2_RM_L_SFT 3 +#define RT5670_M_BST1_RM_L (0x1 << 1) +#define RT5670_M_BST1_RM_L_SFT 1 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5670_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5670_G_HP_R_RM_R_SFT 13 +#define RT5670_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5670_G_IN_R_RM_R_SFT 10 +#define RT5670_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_R_SFT 7 +#define RT5670_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_R_SFT 4 +#define RT5670_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5670_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_R_SFT 13 +#define RT5670_M_IN_R_RM_R (0x1 << 5) +#define RT5670_M_IN_R_RM_R_SFT 5 +#define RT5670_M_BST2_RM_R (0x1 << 3) +#define RT5670_M_BST2_RM_R_SFT 3 +#define RT5670_M_BST1_RM_R (0x1 << 1) +#define RT5670_M_BST1_RM_R_SFT 1 + +/* HPMIX Control (0x45) */ +#define RT5670_M_DAC2_HM (0x1 << 15) +#define RT5670_M_DAC2_HM_SFT 15 +#define RT5670_M_HPVOL_HM (0x1 << 14) +#define RT5670_M_HPVOL_HM_SFT 14 +#define RT5670_M_DAC1_HM (0x1 << 13) +#define RT5670_M_DAC1_HM_SFT 13 +#define RT5670_G_HPOMIX_MASK (0x1 << 12) +#define RT5670_G_HPOMIX_SFT 12 +#define RT5670_M_INR1_HMR (0x1 << 3) +#define RT5670_M_INR1_HMR_SFT 3 +#define RT5670_M_DACR1_HMR (0x1 << 2) +#define RT5670_M_DACR1_HMR_SFT 2 +#define RT5670_M_INL1_HML (0x1 << 1) +#define RT5670_M_INL1_HML_SFT 1 +#define RT5670_M_DACL1_HML (0x1) +#define RT5670_M_DACL1_HML_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5670_M_DAC_R2_MA (0x1 << 15) +#define RT5670_M_DAC_R2_MA_SFT 15 +#define RT5670_M_DAC_L2_MA (0x1 << 14) +#define RT5670_M_DAC_L2_MA_SFT 14 +#define RT5670_M_OV_R_MM (0x1 << 13) +#define RT5670_M_OV_R_MM_SFT 13 +#define RT5670_M_OV_L_MM (0x1 << 12) +#define RT5670_M_OV_L_MM_SFT 12 +#define RT5670_G_MONOMIX_MASK (0x1 << 10) +#define RT5670_G_MONOMIX_SFT 10 +#define RT5670_M_DAC_R2_MM (0x1 << 9) +#define RT5670_M_DAC_R2_MM_SFT 9 +#define RT5670_M_DAC_L2_MM (0x1 << 8) +#define RT5670_M_DAC_L2_MM_SFT 8 +#define RT5670_M_BST4_MM (0x1 << 7) +#define RT5670_M_BST4_MM_SFT 7 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5670_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5670_G_BST3_OM_L_SFT 13 +#define RT5670_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_L_SFT 10 +#define RT5670_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_L_SFT 7 +#define RT5670_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5670_G_IN_L_OM_L_SFT 4 +#define RT5670_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5670_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5670_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5670_G_DAC_R2_OM_L_SFT 13 +#define RT5670_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5670_G_DAC_L2_OM_L_SFT 10 +#define RT5670_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5670_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5670_M_BST1_OM_L (0x1 << 5) +#define RT5670_M_BST1_OM_L_SFT 5 +#define RT5670_M_IN_L_OM_L (0x1 << 4) +#define RT5670_M_IN_L_OM_L_SFT 4 +#define RT5670_M_DAC_L2_OM_L (0x1 << 1) +#define RT5670_M_DAC_L2_OM_L_SFT 1 +#define RT5670_M_DAC_L1_OM_L (0x1) +#define RT5670_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5670_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5670_G_BST4_OM_R_SFT 13 +#define RT5670_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_R_SFT 10 +#define RT5670_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_R_SFT 7 +#define RT5670_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5670_G_IN_R_OM_R_SFT 4 +#define RT5670_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5670_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5670_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5670_G_DAC_L2_OM_R_SFT 13 +#define RT5670_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5670_G_DAC_R2_OM_R_SFT 10 +#define RT5670_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5670_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5670_M_BST2_OM_R (0x1 << 6) +#define RT5670_M_BST2_OM_R_SFT 6 +#define RT5670_M_IN_R_OM_R (0x1 << 4) +#define RT5670_M_IN_R_OM_R_SFT 4 +#define RT5670_M_DAC_R2_OM_R (0x1 << 1) +#define RT5670_M_DAC_R2_OM_R_SFT 1 +#define RT5670_M_DAC_R1_OM_R (0x1) +#define RT5670_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5670_M_DAC_L1_LM (0x1 << 15) +#define RT5670_M_DAC_L1_LM_SFT 15 +#define RT5670_M_DAC_R1_LM (0x1 << 14) +#define RT5670_M_DAC_R1_LM_SFT 14 +#define RT5670_M_OV_L_LM (0x1 << 13) +#define RT5670_M_OV_L_LM_SFT 13 +#define RT5670_M_OV_R_LM (0x1 << 12) +#define RT5670_M_OV_R_LM_SFT 12 +#define RT5670_G_LOUTMIX_MASK (0x1 << 11) +#define RT5670_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5670_PWR_I2S1 (0x1 << 15) +#define RT5670_PWR_I2S1_BIT 15 +#define RT5670_PWR_I2S2 (0x1 << 14) +#define RT5670_PWR_I2S2_BIT 14 +#define RT5670_PWR_DAC_L1 (0x1 << 12) +#define RT5670_PWR_DAC_L1_BIT 12 +#define RT5670_PWR_DAC_R1 (0x1 << 11) +#define RT5670_PWR_DAC_R1_BIT 11 +#define RT5670_PWR_DAC_L2 (0x1 << 7) +#define RT5670_PWR_DAC_L2_BIT 7 +#define RT5670_PWR_DAC_R2 (0x1 << 6) +#define RT5670_PWR_DAC_R2_BIT 6 +#define RT5670_PWR_ADC_L (0x1 << 2) +#define RT5670_PWR_ADC_L_BIT 2 +#define RT5670_PWR_ADC_R (0x1 << 1) +#define RT5670_PWR_ADC_R_BIT 1 +#define RT5670_PWR_CLS_D (0x1) +#define RT5670_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5670_PWR_ADC_S1F (0x1 << 15) +#define RT5670_PWR_ADC_S1F_BIT 15 +#define RT5670_PWR_ADC_MF_L (0x1 << 14) +#define RT5670_PWR_ADC_MF_L_BIT 14 +#define RT5670_PWR_ADC_MF_R (0x1 << 13) +#define RT5670_PWR_ADC_MF_R_BIT 13 +#define RT5670_PWR_I2S_DSP (0x1 << 12) +#define RT5670_PWR_I2S_DSP_BIT 12 +#define RT5670_PWR_DAC_S1F (0x1 << 11) +#define RT5670_PWR_DAC_S1F_BIT 11 +#define RT5670_PWR_DAC_MF_L (0x1 << 10) +#define RT5670_PWR_DAC_MF_L_BIT 10 +#define RT5670_PWR_DAC_MF_R (0x1 << 9) +#define RT5670_PWR_DAC_MF_R_BIT 9 +#define RT5670_PWR_ADC_S2F (0x1 << 8) +#define RT5670_PWR_ADC_S2F_BIT 8 +#define RT5670_PWR_PDM1 (0x1 << 7) +#define RT5670_PWR_PDM1_BIT 7 +#define RT5670_PWR_PDM2 (0x1 << 6) +#define RT5670_PWR_PDM2_BIT 6 + +/* Power Management for Analog 1 (0x63) */ +#define RT5670_PWR_VREF1 (0x1 << 15) +#define RT5670_PWR_VREF1_BIT 15 +#define RT5670_PWR_FV1 (0x1 << 14) +#define RT5670_PWR_FV1_BIT 14 +#define RT5670_PWR_MB (0x1 << 13) +#define RT5670_PWR_MB_BIT 13 +#define RT5670_PWR_LM (0x1 << 12) +#define RT5670_PWR_LM_BIT 12 +#define RT5670_PWR_BG (0x1 << 11) +#define RT5670_PWR_BG_BIT 11 +#define RT5670_PWR_HP_L (0x1 << 7) +#define RT5670_PWR_HP_L_BIT 7 +#define RT5670_PWR_HP_R (0x1 << 6) +#define RT5670_PWR_HP_R_BIT 6 +#define RT5670_PWR_HA (0x1 << 5) +#define RT5670_PWR_HA_BIT 5 +#define RT5670_PWR_VREF2 (0x1 << 4) +#define RT5670_PWR_VREF2_BIT 4 +#define RT5670_PWR_FV2 (0x1 << 3) +#define RT5670_PWR_FV2_BIT 3 +#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5670_PWR_BST1 (0x1 << 15) +#define RT5670_PWR_BST1_BIT 15 +#define RT5670_PWR_BST2 (0x1 << 13) +#define RT5670_PWR_BST2_BIT 13 +#define RT5670_PWR_MB1 (0x1 << 11) +#define RT5670_PWR_MB1_BIT 11 +#define RT5670_PWR_MB2 (0x1 << 10) +#define RT5670_PWR_MB2_BIT 10 +#define RT5670_PWR_PLL (0x1 << 9) +#define RT5670_PWR_PLL_BIT 9 +#define RT5670_PWR_BST1_P (0x1 << 6) +#define RT5670_PWR_BST1_P_BIT 6 +#define RT5670_PWR_BST2_P (0x1 << 4) +#define RT5670_PWR_BST2_P_BIT 4 +#define RT5670_PWR_JD1 (0x1 << 2) +#define RT5670_PWR_JD1_BIT 2 +#define RT5670_PWR_JD (0x1 << 1) +#define RT5670_PWR_JD_BIT 1 + +/* Power Management for Mixer (0x65) */ +#define RT5670_PWR_OM_L (0x1 << 15) +#define RT5670_PWR_OM_L_BIT 15 +#define RT5670_PWR_OM_R (0x1 << 14) +#define RT5670_PWR_OM_R_BIT 14 +#define RT5670_PWR_RM_L (0x1 << 11) +#define RT5670_PWR_RM_L_BIT 11 +#define RT5670_PWR_RM_R (0x1 << 10) +#define RT5670_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5670_PWR_HV_L (0x1 << 11) +#define RT5670_PWR_HV_L_BIT 11 +#define RT5670_PWR_HV_R (0x1 << 10) +#define RT5670_PWR_HV_R_BIT 10 +#define RT5670_PWR_IN_L (0x1 << 9) +#define RT5670_PWR_IN_L_BIT 9 +#define RT5670_PWR_IN_R (0x1 << 8) +#define RT5670_PWR_IN_R_BIT 8 +#define RT5670_PWR_MIC_DET (0x1 << 5) +#define RT5670_PWR_MIC_DET_BIT 5 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5670_I2S_MS_MASK (0x1 << 15) +#define RT5670_I2S_MS_SFT 15 +#define RT5670_I2S_MS_M (0x0 << 15) +#define RT5670_I2S_MS_S (0x1 << 15) +#define RT5670_I2S_IF_MASK (0x7 << 12) +#define RT5670_I2S_IF_SFT 12 +#define RT5670_I2S_O_CP_MASK (0x3 << 10) +#define RT5670_I2S_O_CP_SFT 10 +#define RT5670_I2S_O_CP_OFF (0x0 << 10) +#define RT5670_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5670_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5670_I2S_I_CP_MASK (0x3 << 8) +#define RT5670_I2S_I_CP_SFT 8 +#define RT5670_I2S_I_CP_OFF (0x0 << 8) +#define RT5670_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5670_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5670_I2S_BP_MASK (0x1 << 7) +#define RT5670_I2S_BP_SFT 7 +#define RT5670_I2S_BP_NOR (0x0 << 7) +#define RT5670_I2S_BP_INV (0x1 << 7) +#define RT5670_I2S_DL_MASK (0x3 << 2) +#define RT5670_I2S_DL_SFT 2 +#define RT5670_I2S_DL_16 (0x0 << 2) +#define RT5670_I2S_DL_20 (0x1 << 2) +#define RT5670_I2S_DL_24 (0x2 << 2) +#define RT5670_I2S_DL_8 (0x3 << 2) +#define RT5670_I2S_DF_MASK (0x3) +#define RT5670_I2S_DF_SFT 0 +#define RT5670_I2S_DF_I2S (0x0) +#define RT5670_I2S_DF_LEFT (0x1) +#define RT5670_I2S_DF_PCM_A (0x2) +#define RT5670_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5670_I2S2_SDI_MASK (0x1 << 6) +#define RT5670_I2S2_SDI_SFT 6 +#define RT5670_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5670_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5670_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5670_I2S_BCLK_MS1_SFT 15 +#define RT5670_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5670_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5670_I2S_PD1_MASK (0x7 << 12) +#define RT5670_I2S_PD1_SFT 12 +#define RT5670_I2S_PD1_1 (0x0 << 12) +#define RT5670_I2S_PD1_2 (0x1 << 12) +#define RT5670_I2S_PD1_3 (0x2 << 12) +#define RT5670_I2S_PD1_4 (0x3 << 12) +#define RT5670_I2S_PD1_6 (0x4 << 12) +#define RT5670_I2S_PD1_8 (0x5 << 12) +#define RT5670_I2S_PD1_12 (0x6 << 12) +#define RT5670_I2S_PD1_16 (0x7 << 12) +#define RT5670_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5670_I2S_BCLK_MS2_SFT 11 +#define RT5670_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5670_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5670_I2S_PD2_MASK (0x7 << 8) +#define RT5670_I2S_PD2_SFT 8 +#define RT5670_I2S_PD2_1 (0x0 << 8) +#define RT5670_I2S_PD2_2 (0x1 << 8) +#define RT5670_I2S_PD2_3 (0x2 << 8) +#define RT5670_I2S_PD2_4 (0x3 << 8) +#define RT5670_I2S_PD2_6 (0x4 << 8) +#define RT5670_I2S_PD2_8 (0x5 << 8) +#define RT5670_I2S_PD2_12 (0x6 << 8) +#define RT5670_I2S_PD2_16 (0x7 << 8) +#define RT5670_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5670_I2S_BCLK_MS3_SFT 7 +#define RT5670_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5670_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5670_I2S_PD3_MASK (0x7 << 4) +#define RT5670_I2S_PD3_SFT 4 +#define RT5670_I2S_PD3_1 (0x0 << 4) +#define RT5670_I2S_PD3_2 (0x1 << 4) +#define RT5670_I2S_PD3_3 (0x2 << 4) +#define RT5670_I2S_PD3_4 (0x3 << 4) +#define RT5670_I2S_PD3_6 (0x4 << 4) +#define RT5670_I2S_PD3_8 (0x5 << 4) +#define RT5670_I2S_PD3_12 (0x6 << 4) +#define RT5670_I2S_PD3_16 (0x7 << 4) +#define RT5670_DAC_OSR_MASK (0x3 << 2) +#define RT5670_DAC_OSR_SFT 2 +#define RT5670_DAC_OSR_128 (0x0 << 2) +#define RT5670_DAC_OSR_64 (0x1 << 2) +#define RT5670_DAC_OSR_32 (0x2 << 2) +#define RT5670_DAC_OSR_16 (0x3 << 2) +#define RT5670_ADC_OSR_MASK (0x3) +#define RT5670_ADC_OSR_SFT 0 +#define RT5670_ADC_OSR_128 (0x0) +#define RT5670_ADC_OSR_64 (0x1) +#define RT5670_ADC_OSR_32 (0x2) +#define RT5670_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5670_DAC_L_OSR_MASK (0x3 << 14) +#define RT5670_DAC_L_OSR_SFT 14 +#define RT5670_DAC_L_OSR_128 (0x0 << 14) +#define RT5670_DAC_L_OSR_64 (0x1 << 14) +#define RT5670_DAC_L_OSR_32 (0x2 << 14) +#define RT5670_DAC_L_OSR_16 (0x3 << 14) +#define RT5670_ADC_R_OSR_MASK (0x3 << 12) +#define RT5670_ADC_R_OSR_SFT 12 +#define RT5670_ADC_R_OSR_128 (0x0 << 12) +#define RT5670_ADC_R_OSR_64 (0x1 << 12) +#define RT5670_ADC_R_OSR_32 (0x2 << 12) +#define RT5670_ADC_R_OSR_16 (0x3 << 12) +#define RT5670_DAHPF_EN (0x1 << 11) +#define RT5670_DAHPF_EN_SFT 11 +#define RT5670_ADHPF_EN (0x1 << 10) +#define RT5670_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5670_DMIC_1_EN_MASK (0x1 << 15) +#define RT5670_DMIC_1_EN_SFT 15 +#define RT5670_DMIC_1_DIS (0x0 << 15) +#define RT5670_DMIC_1_EN (0x1 << 15) +#define RT5670_DMIC_2_EN_MASK (0x1 << 14) +#define RT5670_DMIC_2_EN_SFT 14 +#define RT5670_DMIC_2_DIS (0x0 << 14) +#define RT5670_DMIC_2_EN (0x1 << 14) +#define RT5670_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5670_DMIC_1L_LH_SFT 13 +#define RT5670_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5670_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5670_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5670_DMIC_1R_LH_SFT 12 +#define RT5670_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5670_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5670_DMIC_2_DP_MASK (0x1 << 10) +#define RT5670_DMIC_2_DP_SFT 10 +#define RT5670_DMIC_2_DP_GPIO8 (0x0 << 10) +#define RT5670_DMIC_2_DP_IN3N (0x1 << 10) +#define RT5670_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5670_DMIC_2L_LH_SFT 9 +#define RT5670_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5670_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5670_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5670_DMIC_2R_LH_SFT 8 +#define RT5670_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5670_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5670_DMIC_CLK_MASK (0x7 << 5) +#define RT5670_DMIC_CLK_SFT 5 +#define RT5670_DMIC_3_EN_MASK (0x1 << 4) +#define RT5670_DMIC_3_EN_SFT 4 +#define RT5670_DMIC_3_DIS (0x0 << 4) +#define RT5670_DMIC_3_EN (0x1 << 4) +#define RT5670_DMIC_1_DP_MASK (0x3 << 0) +#define RT5670_DMIC_1_DP_SFT 0 +#define RT5670_DMIC_1_DP_GPIO6 (0x0 << 0) +#define RT5670_DMIC_1_DP_IN2P (0x1 << 0) +#define RT5670_DMIC_1_DP_GPIO7 (0x2 << 0) + +/* Digital Microphone Control2 (0x76) */ +#define RT5670_DMIC_3_DP_MASK (0x3 << 6) +#define RT5670_DMIC_3_DP_SFT 6 +#define RT5670_DMIC_3_DP_GPIO9 (0x0 << 6) +#define RT5670_DMIC_3_DP_GPIO10 (0x1 << 6) +#define RT5670_DMIC_3_DP_GPIO5 (0x2 << 6) + +/* Global Clock Control (0x80) */ +#define RT5670_SCLK_SRC_MASK (0x3 << 14) +#define RT5670_SCLK_SRC_SFT 14 +#define RT5670_SCLK_SRC_MCLK (0x0 << 14) +#define RT5670_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ +#define RT5670_PLL1_SRC_MASK (0x3 << 12) +#define RT5670_PLL1_SRC_SFT 12 +#define RT5670_PLL1_SRC_MCLK (0x0 << 12) +#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5670_PLL1_PD_MASK (0x1 << 3) +#define RT5670_PLL1_PD_SFT 3 +#define RT5670_PLL1_PD_1 (0x0 << 3) +#define RT5670_PLL1_PD_2 (0x1 << 3) + +#define RT5670_PLL_INP_MAX 40000000 +#define RT5670_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5670_PLL_N_MAX 0x1ff +#define RT5670_PLL_N_MASK (RT5670_PLL_N_MAX << 7) +#define RT5670_PLL_N_SFT 7 +#define RT5670_PLL_K_MAX 0x1f +#define RT5670_PLL_K_MASK (RT5670_PLL_K_MAX) +#define RT5670_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5670_PLL_M_MAX 0xf +#define RT5670_PLL_M_MASK (RT5670_PLL_M_MAX << 12) +#define RT5670_PLL_M_SFT 12 +#define RT5670_PLL_M_BP (0x1 << 11) +#define RT5670_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5670_STO_T_MASK (0x1 << 15) +#define RT5670_STO_T_SFT 15 +#define RT5670_STO_T_SCLK (0x0 << 15) +#define RT5670_STO_T_LRCK1 (0x1 << 15) +#define RT5670_M1_T_MASK (0x1 << 14) +#define RT5670_M1_T_SFT 14 +#define RT5670_M1_T_I2S2 (0x0 << 14) +#define RT5670_M1_T_I2S2_D3 (0x1 << 14) +#define RT5670_I2S2_F_MASK (0x1 << 12) +#define RT5670_I2S2_F_SFT 12 +#define RT5670_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5670_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5670_DMIC_1_M_MASK (0x1 << 9) +#define RT5670_DMIC_1_M_SFT 9 +#define RT5670_DMIC_1_M_NOR (0x0 << 9) +#define RT5670_DMIC_1_M_ASYN (0x1 << 9) +#define RT5670_DMIC_2_M_MASK (0x1 << 8) +#define RT5670_DMIC_2_M_SFT 8 +#define RT5670_DMIC_2_M_NOR (0x0 << 8) +#define RT5670_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5670_MDA_L_M_MASK (0x1 << 15) +#define RT5670_MDA_L_M_SFT 15 +#define RT5670_MDA_L_M_NOR (0x0 << 15) +#define RT5670_MDA_L_M_ASYN (0x1 << 15) +#define RT5670_MDA_R_M_MASK (0x1 << 14) +#define RT5670_MDA_R_M_SFT 14 +#define RT5670_MDA_R_M_NOR (0x0 << 14) +#define RT5670_MDA_R_M_ASYN (0x1 << 14) +#define RT5670_MAD_L_M_MASK (0x1 << 13) +#define RT5670_MAD_L_M_SFT 13 +#define RT5670_MAD_L_M_NOR (0x0 << 13) +#define RT5670_MAD_L_M_ASYN (0x1 << 13) +#define RT5670_MAD_R_M_MASK (0x1 << 12) +#define RT5670_MAD_R_M_SFT 12 +#define RT5670_MAD_R_M_NOR (0x0 << 12) +#define RT5670_MAD_R_M_ASYN (0x1 << 12) +#define RT5670_ADC_M_MASK (0x1 << 11) +#define RT5670_ADC_M_SFT 11 +#define RT5670_ADC_M_NOR (0x0 << 11) +#define RT5670_ADC_M_ASYN (0x1 << 11) +#define RT5670_STO_DAC_M_MASK (0x1 << 5) +#define RT5670_STO_DAC_M_SFT 5 +#define RT5670_STO_DAC_M_NOR (0x0 << 5) +#define RT5670_STO_DAC_M_ASYN (0x1 << 5) +#define RT5670_I2S1_R_D_MASK (0x1 << 4) +#define RT5670_I2S1_R_D_SFT 4 +#define RT5670_I2S1_R_D_DIS (0x0 << 4) +#define RT5670_I2S1_R_D_EN (0x1 << 4) +#define RT5670_I2S2_R_D_MASK (0x1 << 3) +#define RT5670_I2S2_R_D_SFT 3 +#define RT5670_I2S2_R_D_DIS (0x0 << 3) +#define RT5670_I2S2_R_D_EN (0x1 << 3) +#define RT5670_PRE_SCLK_MASK (0x3) +#define RT5670_PRE_SCLK_SFT 0 +#define RT5670_PRE_SCLK_512 (0x0) +#define RT5670_PRE_SCLK_1024 (0x1) +#define RT5670_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5670_I2S1_RATE_MASK (0xf << 12) +#define RT5670_I2S1_RATE_SFT 12 +#define RT5670_I2S2_RATE_MASK (0xf << 8) +#define RT5670_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5670_I2S1_PD_MASK (0x7 << 12) +#define RT5670_I2S1_PD_SFT 12 +#define RT5670_I2S2_PD_MASK (0x7 << 8) +#define RT5670_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5670_HP_OVCD_MASK (0x1 << 10) +#define RT5670_HP_OVCD_SFT 10 +#define RT5670_HP_OVCD_DIS (0x0 << 10) +#define RT5670_HP_OVCD_EN (0x1 << 10) +#define RT5670_HP_OC_TH_MASK (0x3 << 8) +#define RT5670_HP_OC_TH_SFT 8 +#define RT5670_HP_OC_TH_90 (0x0 << 8) +#define RT5670_HP_OC_TH_105 (0x1 << 8) +#define RT5670_HP_OC_TH_120 (0x2 << 8) +#define RT5670_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5670_CLSD_OC_MASK (0x1 << 9) +#define RT5670_CLSD_OC_SFT 9 +#define RT5670_CLSD_OC_PU (0x0 << 9) +#define RT5670_CLSD_OC_PD (0x1 << 9) +#define RT5670_AUTO_PD_MASK (0x1 << 8) +#define RT5670_AUTO_PD_SFT 8 +#define RT5670_AUTO_PD_DIS (0x0 << 8) +#define RT5670_AUTO_PD_EN (0x1 << 8) +#define RT5670_CLSD_OC_TH_MASK (0x3f) +#define RT5670_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5670_CLSD_RATIO_MASK (0xf << 12) +#define RT5670_CLSD_RATIO_SFT 12 +#define RT5670_CLSD_OM_MASK (0x1 << 11) +#define RT5670_CLSD_OM_SFT 11 +#define RT5670_CLSD_OM_MONO (0x0 << 11) +#define RT5670_CLSD_OM_STO (0x1 << 11) +#define RT5670_CLSD_SCH_MASK (0x1 << 10) +#define RT5670_CLSD_SCH_SFT 10 +#define RT5670_CLSD_SCH_L (0x0 << 10) +#define RT5670_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5670_SMT_TRIG_MASK (0x1 << 15) +#define RT5670_SMT_TRIG_SFT 15 +#define RT5670_SMT_TRIG_DIS (0x0 << 15) +#define RT5670_SMT_TRIG_EN (0x1 << 15) +#define RT5670_HP_L_SMT_MASK (0x1 << 9) +#define RT5670_HP_L_SMT_SFT 9 +#define RT5670_HP_L_SMT_DIS (0x0 << 9) +#define RT5670_HP_L_SMT_EN (0x1 << 9) +#define RT5670_HP_R_SMT_MASK (0x1 << 8) +#define RT5670_HP_R_SMT_SFT 8 +#define RT5670_HP_R_SMT_DIS (0x0 << 8) +#define RT5670_HP_R_SMT_EN (0x1 << 8) +#define RT5670_HP_CD_PD_MASK (0x1 << 7) +#define RT5670_HP_CD_PD_SFT 7 +#define RT5670_HP_CD_PD_DIS (0x0 << 7) +#define RT5670_HP_CD_PD_EN (0x1 << 7) +#define RT5670_RSTN_MASK (0x1 << 6) +#define RT5670_RSTN_SFT 6 +#define RT5670_RSTN_DIS (0x0 << 6) +#define RT5670_RSTN_EN (0x1 << 6) +#define RT5670_RSTP_MASK (0x1 << 5) +#define RT5670_RSTP_SFT 5 +#define RT5670_RSTP_DIS (0x0 << 5) +#define RT5670_RSTP_EN (0x1 << 5) +#define RT5670_HP_CO_MASK (0x1 << 4) +#define RT5670_HP_CO_SFT 4 +#define RT5670_HP_CO_DIS (0x0 << 4) +#define RT5670_HP_CO_EN (0x1 << 4) +#define RT5670_HP_CP_MASK (0x1 << 3) +#define RT5670_HP_CP_SFT 3 +#define RT5670_HP_CP_PD (0x0 << 3) +#define RT5670_HP_CP_PU (0x1 << 3) +#define RT5670_HP_SG_MASK (0x1 << 2) +#define RT5670_HP_SG_SFT 2 +#define RT5670_HP_SG_DIS (0x0 << 2) +#define RT5670_HP_SG_EN (0x1 << 2) +#define RT5670_HP_DP_MASK (0x1 << 1) +#define RT5670_HP_DP_SFT 1 +#define RT5670_HP_DP_PD (0x0 << 1) +#define RT5670_HP_DP_PU (0x1 << 1) +#define RT5670_HP_CB_MASK (0x1) +#define RT5670_HP_CB_SFT 0 +#define RT5670_HP_CB_PD (0x0) +#define RT5670_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5670_DEPOP_MASK (0x1 << 13) +#define RT5670_DEPOP_SFT 13 +#define RT5670_DEPOP_AUTO (0x0 << 13) +#define RT5670_DEPOP_MAN (0x1 << 13) +#define RT5670_RAMP_MASK (0x1 << 12) +#define RT5670_RAMP_SFT 12 +#define RT5670_RAMP_DIS (0x0 << 12) +#define RT5670_RAMP_EN (0x1 << 12) +#define RT5670_BPS_MASK (0x1 << 11) +#define RT5670_BPS_SFT 11 +#define RT5670_BPS_DIS (0x0 << 11) +#define RT5670_BPS_EN (0x1 << 11) +#define RT5670_FAST_UPDN_MASK (0x1 << 10) +#define RT5670_FAST_UPDN_SFT 10 +#define RT5670_FAST_UPDN_DIS (0x0 << 10) +#define RT5670_FAST_UPDN_EN (0x1 << 10) +#define RT5670_MRES_MASK (0x3 << 8) +#define RT5670_MRES_SFT 8 +#define RT5670_MRES_15MO (0x0 << 8) +#define RT5670_MRES_25MO (0x1 << 8) +#define RT5670_MRES_35MO (0x2 << 8) +#define RT5670_MRES_45MO (0x3 << 8) +#define RT5670_VLO_MASK (0x1 << 7) +#define RT5670_VLO_SFT 7 +#define RT5670_VLO_3V (0x0 << 7) +#define RT5670_VLO_32V (0x1 << 7) +#define RT5670_DIG_DP_MASK (0x1 << 6) +#define RT5670_DIG_DP_SFT 6 +#define RT5670_DIG_DP_DIS (0x0 << 6) +#define RT5670_DIG_DP_EN (0x1 << 6) +#define RT5670_DP_TH_MASK (0x3 << 4) +#define RT5670_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5670_CP_SYS_MASK (0x7 << 12) +#define RT5670_CP_SYS_SFT 12 +#define RT5670_CP_FQ1_MASK (0x7 << 8) +#define RT5670_CP_FQ1_SFT 8 +#define RT5670_CP_FQ2_MASK (0x7 << 4) +#define RT5670_CP_FQ2_SFT 4 +#define RT5670_CP_FQ3_MASK (0x7) +#define RT5670_CP_FQ3_SFT 0 +#define RT5670_CP_FQ_1_5_KHZ 0 +#define RT5670_CP_FQ_3_KHZ 1 +#define RT5670_CP_FQ_6_KHZ 2 +#define RT5670_CP_FQ_12_KHZ 3 +#define RT5670_CP_FQ_24_KHZ 4 +#define RT5670_CP_FQ_48_KHZ 5 +#define RT5670_CP_FQ_96_KHZ 6 +#define RT5670_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5670_OSW_L_MASK (0x1 << 11) +#define RT5670_OSW_L_SFT 11 +#define RT5670_OSW_L_DIS (0x0 << 11) +#define RT5670_OSW_L_EN (0x1 << 11) +#define RT5670_OSW_R_MASK (0x1 << 10) +#define RT5670_OSW_R_SFT 10 +#define RT5670_OSW_R_DIS (0x0 << 10) +#define RT5670_OSW_R_EN (0x1 << 10) +#define RT5670_PM_HP_MASK (0x3 << 8) +#define RT5670_PM_HP_SFT 8 +#define RT5670_PM_HP_LV (0x0 << 8) +#define RT5670_PM_HP_MV (0x1 << 8) +#define RT5670_PM_HP_HV (0x2 << 8) +#define RT5670_IB_HP_MASK (0x3 << 6) +#define RT5670_IB_HP_SFT 6 +#define RT5670_IB_HP_125IL (0x0 << 6) +#define RT5670_IB_HP_25IL (0x1 << 6) +#define RT5670_IB_HP_5IL (0x2 << 6) +#define RT5670_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5670_PVDD_DET_MASK (0x1 << 15) +#define RT5670_PVDD_DET_SFT 15 +#define RT5670_PVDD_DET_DIS (0x0 << 15) +#define RT5670_PVDD_DET_EN (0x1 << 15) +#define RT5670_SPK_AG_MASK (0x1 << 14) +#define RT5670_SPK_AG_SFT 14 +#define RT5670_SPK_AG_DIS (0x0 << 14) +#define RT5670_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5670_MIC1_BS_MASK (0x1 << 15) +#define RT5670_MIC1_BS_SFT 15 +#define RT5670_MIC1_BS_9AV (0x0 << 15) +#define RT5670_MIC1_BS_75AV (0x1 << 15) +#define RT5670_MIC2_BS_MASK (0x1 << 14) +#define RT5670_MIC2_BS_SFT 14 +#define RT5670_MIC2_BS_9AV (0x0 << 14) +#define RT5670_MIC2_BS_75AV (0x1 << 14) +#define RT5670_MIC1_CLK_MASK (0x1 << 13) +#define RT5670_MIC1_CLK_SFT 13 +#define RT5670_MIC1_CLK_DIS (0x0 << 13) +#define RT5670_MIC1_CLK_EN (0x1 << 13) +#define RT5670_MIC2_CLK_MASK (0x1 << 12) +#define RT5670_MIC2_CLK_SFT 12 +#define RT5670_MIC2_CLK_DIS (0x0 << 12) +#define RT5670_MIC2_CLK_EN (0x1 << 12) +#define RT5670_MIC1_OVCD_MASK (0x1 << 11) +#define RT5670_MIC1_OVCD_SFT 11 +#define RT5670_MIC1_OVCD_DIS (0x0 << 11) +#define RT5670_MIC1_OVCD_EN (0x1 << 11) +#define RT5670_MIC1_OVTH_MASK (0x3 << 9) +#define RT5670_MIC1_OVTH_SFT 9 +#define RT5670_MIC1_OVTH_600UA (0x0 << 9) +#define RT5670_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5670_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5670_MIC2_OVCD_MASK (0x1 << 8) +#define RT5670_MIC2_OVCD_SFT 8 +#define RT5670_MIC2_OVCD_DIS (0x0 << 8) +#define RT5670_MIC2_OVCD_EN (0x1 << 8) +#define RT5670_MIC2_OVTH_MASK (0x3 << 6) +#define RT5670_MIC2_OVTH_SFT 6 +#define RT5670_MIC2_OVTH_600UA (0x0 << 6) +#define RT5670_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5670_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5670_PWR_MB_MASK (0x1 << 5) +#define RT5670_PWR_MB_SFT 5 +#define RT5670_PWR_MB_PD (0x0 << 5) +#define RT5670_PWR_MB_PU (0x1 << 5) +#define RT5670_PWR_CLK25M_MASK (0x1 << 4) +#define RT5670_PWR_CLK25M_SFT 4 +#define RT5670_PWR_CLK25M_PD (0x0 << 4) +#define RT5670_PWR_CLK25M_PU (0x1 << 4) + +/* Analog JD Control 1 (0x94) */ +#define RT5670_JD1_MODE_MASK (0x3 << 0) +#define RT5670_JD1_MODE_0 (0x0 << 0) +#define RT5670_JD1_MODE_1 (0x1 << 0) +#define RT5670_JD1_MODE_2 (0x2 << 0) + +/* VAD Control 4 (0x9d) */ +#define RT5670_VAD_SEL_MASK (0x3 << 8) +#define RT5670_VAD_SEL_SFT 8 + +/* EQ Control 1 (0xb0) */ +#define RT5670_EQ_SRC_MASK (0x1 << 15) +#define RT5670_EQ_SRC_SFT 15 +#define RT5670_EQ_SRC_DAC (0x0 << 15) +#define RT5670_EQ_SRC_ADC (0x1 << 15) +#define RT5670_EQ_UPD (0x1 << 14) +#define RT5670_EQ_UPD_BIT 14 +#define RT5670_EQ_CD_MASK (0x1 << 13) +#define RT5670_EQ_CD_SFT 13 +#define RT5670_EQ_CD_DIS (0x0 << 13) +#define RT5670_EQ_CD_EN (0x1 << 13) +#define RT5670_EQ_DITH_MASK (0x3 << 8) +#define RT5670_EQ_DITH_SFT 8 +#define RT5670_EQ_DITH_NOR (0x0 << 8) +#define RT5670_EQ_DITH_LSB (0x1 << 8) +#define RT5670_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5670_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5670_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5670_EQ_HPF1_M_SFT 8 +#define RT5670_EQ_HPF1_M_HI (0x0 << 8) +#define RT5670_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5670_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5670_EQ_LPF1_M_SFT 7 +#define RT5670_EQ_LPF1_M_LO (0x0 << 7) +#define RT5670_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5670_EQ_HPF2_MASK (0x1 << 6) +#define RT5670_EQ_HPF2_SFT 6 +#define RT5670_EQ_HPF2_DIS (0x0 << 6) +#define RT5670_EQ_HPF2_EN (0x1 << 6) +#define RT5670_EQ_HPF1_MASK (0x1 << 5) +#define RT5670_EQ_HPF1_SFT 5 +#define RT5670_EQ_HPF1_DIS (0x0 << 5) +#define RT5670_EQ_HPF1_EN (0x1 << 5) +#define RT5670_EQ_BPF4_MASK (0x1 << 4) +#define RT5670_EQ_BPF4_SFT 4 +#define RT5670_EQ_BPF4_DIS (0x0 << 4) +#define RT5670_EQ_BPF4_EN (0x1 << 4) +#define RT5670_EQ_BPF3_MASK (0x1 << 3) +#define RT5670_EQ_BPF3_SFT 3 +#define RT5670_EQ_BPF3_DIS (0x0 << 3) +#define RT5670_EQ_BPF3_EN (0x1 << 3) +#define RT5670_EQ_BPF2_MASK (0x1 << 2) +#define RT5670_EQ_BPF2_SFT 2 +#define RT5670_EQ_BPF2_DIS (0x0 << 2) +#define RT5670_EQ_BPF2_EN (0x1 << 2) +#define RT5670_EQ_BPF1_MASK (0x1 << 1) +#define RT5670_EQ_BPF1_SFT 1 +#define RT5670_EQ_BPF1_DIS (0x0 << 1) +#define RT5670_EQ_BPF1_EN (0x1 << 1) +#define RT5670_EQ_LPF_MASK (0x1) +#define RT5670_EQ_LPF_SFT 0 +#define RT5670_EQ_LPF_DIS (0x0) +#define RT5670_EQ_LPF_EN (0x1) +#define RT5670_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5670_MT_MASK (0x1 << 15) +#define RT5670_MT_SFT 15 +#define RT5670_MT_DIS (0x0 << 15) +#define RT5670_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5670_DRC_AGC_P_MASK (0x1 << 15) +#define RT5670_DRC_AGC_P_SFT 15 +#define RT5670_DRC_AGC_P_DAC (0x0 << 15) +#define RT5670_DRC_AGC_P_ADC (0x1 << 15) +#define RT5670_DRC_AGC_MASK (0x1 << 14) +#define RT5670_DRC_AGC_SFT 14 +#define RT5670_DRC_AGC_DIS (0x0 << 14) +#define RT5670_DRC_AGC_EN (0x1 << 14) +#define RT5670_DRC_AGC_UPD (0x1 << 13) +#define RT5670_DRC_AGC_UPD_BIT 13 +#define RT5670_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5670_DRC_AGC_AR_SFT 8 +#define RT5670_DRC_AGC_R_MASK (0x7 << 5) +#define RT5670_DRC_AGC_R_SFT 5 +#define RT5670_DRC_AGC_R_48K (0x1 << 5) +#define RT5670_DRC_AGC_R_96K (0x2 << 5) +#define RT5670_DRC_AGC_R_192K (0x3 << 5) +#define RT5670_DRC_AGC_R_441K (0x5 << 5) +#define RT5670_DRC_AGC_R_882K (0x6 << 5) +#define RT5670_DRC_AGC_R_1764K (0x7 << 5) +#define RT5670_DRC_AGC_RC_MASK (0x1f) +#define RT5670_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5670_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5670_DRC_AGC_POB_SFT 8 +#define RT5670_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5670_DRC_AGC_CP_SFT 7 +#define RT5670_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5670_DRC_AGC_CP_EN (0x1 << 7) +#define RT5670_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5670_DRC_AGC_CPR_SFT 5 +#define RT5670_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5670_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5670_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5670_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5670_DRC_AGC_PRB_MASK (0x1f) +#define RT5670_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5670_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5670_DRC_AGC_NGB_SFT 12 +#define RT5670_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5670_DRC_AGC_TAR_SFT 7 +#define RT5670_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5670_DRC_AGC_NG_SFT 6 +#define RT5670_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5670_DRC_AGC_NG_EN (0x1 << 6) +#define RT5670_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5670_DRC_AGC_NGH_SFT 5 +#define RT5670_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5670_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5670_DRC_AGC_NGT_MASK (0x1f) +#define RT5670_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5670_JD_MASK (0x7 << 13) +#define RT5670_JD_SFT 13 +#define RT5670_JD_DIS (0x0 << 13) +#define RT5670_JD_GPIO1 (0x1 << 13) +#define RT5670_JD_JD1_IN4P (0x2 << 13) +#define RT5670_JD_JD2_IN4N (0x3 << 13) +#define RT5670_JD_GPIO2 (0x4 << 13) +#define RT5670_JD_GPIO3 (0x5 << 13) +#define RT5670_JD_GPIO4 (0x6 << 13) +#define RT5670_JD_HP_MASK (0x1 << 11) +#define RT5670_JD_HP_SFT 11 +#define RT5670_JD_HP_DIS (0x0 << 11) +#define RT5670_JD_HP_EN (0x1 << 11) +#define RT5670_JD_HP_TRG_MASK (0x1 << 10) +#define RT5670_JD_HP_TRG_SFT 10 +#define RT5670_JD_HP_TRG_LO (0x0 << 10) +#define RT5670_JD_HP_TRG_HI (0x1 << 10) +#define RT5670_JD_SPL_MASK (0x1 << 9) +#define RT5670_JD_SPL_SFT 9 +#define RT5670_JD_SPL_DIS (0x0 << 9) +#define RT5670_JD_SPL_EN (0x1 << 9) +#define RT5670_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5670_JD_SPL_TRG_SFT 8 +#define RT5670_JD_SPL_TRG_LO (0x0 << 8) +#define RT5670_JD_SPL_TRG_HI (0x1 << 8) +#define RT5670_JD_SPR_MASK (0x1 << 7) +#define RT5670_JD_SPR_SFT 7 +#define RT5670_JD_SPR_DIS (0x0 << 7) +#define RT5670_JD_SPR_EN (0x1 << 7) +#define RT5670_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5670_JD_SPR_TRG_SFT 6 +#define RT5670_JD_SPR_TRG_LO (0x0 << 6) +#define RT5670_JD_SPR_TRG_HI (0x1 << 6) +#define RT5670_JD_MO_MASK (0x1 << 5) +#define RT5670_JD_MO_SFT 5 +#define RT5670_JD_MO_DIS (0x0 << 5) +#define RT5670_JD_MO_EN (0x1 << 5) +#define RT5670_JD_MO_TRG_MASK (0x1 << 4) +#define RT5670_JD_MO_TRG_SFT 4 +#define RT5670_JD_MO_TRG_LO (0x0 << 4) +#define RT5670_JD_MO_TRG_HI (0x1 << 4) +#define RT5670_JD_LO_MASK (0x1 << 3) +#define RT5670_JD_LO_SFT 3 +#define RT5670_JD_LO_DIS (0x0 << 3) +#define RT5670_JD_LO_EN (0x1 << 3) +#define RT5670_JD_LO_TRG_MASK (0x1 << 2) +#define RT5670_JD_LO_TRG_SFT 2 +#define RT5670_JD_LO_TRG_LO (0x0 << 2) +#define RT5670_JD_LO_TRG_HI (0x1 << 2) +#define RT5670_JD1_IN4P_MASK (0x1 << 1) +#define RT5670_JD1_IN4P_SFT 1 +#define RT5670_JD1_IN4P_DIS (0x0 << 1) +#define RT5670_JD1_IN4P_EN (0x1 << 1) +#define RT5670_JD2_IN4N_MASK (0x1) +#define RT5670_JD2_IN4N_SFT 0 +#define RT5670_JD2_IN4N_DIS (0x0) +#define RT5670_JD2_IN4N_EN (0x1) + +/* IRQ Control 1 (0xbd) */ +#define RT5670_IRQ_JD_MASK (0x1 << 15) +#define RT5670_IRQ_JD_SFT 15 +#define RT5670_IRQ_JD_BP (0x0 << 15) +#define RT5670_IRQ_JD_NOR (0x1 << 15) +#define RT5670_IRQ_OT_MASK (0x1 << 14) +#define RT5670_IRQ_OT_SFT 14 +#define RT5670_IRQ_OT_BP (0x0 << 14) +#define RT5670_IRQ_OT_NOR (0x1 << 14) +#define RT5670_JD_STKY_MASK (0x1 << 13) +#define RT5670_JD_STKY_SFT 13 +#define RT5670_JD_STKY_DIS (0x0 << 13) +#define RT5670_JD_STKY_EN (0x1 << 13) +#define RT5670_OT_STKY_MASK (0x1 << 12) +#define RT5670_OT_STKY_SFT 12 +#define RT5670_OT_STKY_DIS (0x0 << 12) +#define RT5670_OT_STKY_EN (0x1 << 12) +#define RT5670_JD_P_MASK (0x1 << 11) +#define RT5670_JD_P_SFT 11 +#define RT5670_JD_P_NOR (0x0 << 11) +#define RT5670_JD_P_INV (0x1 << 11) +#define RT5670_OT_P_MASK (0x1 << 10) +#define RT5670_OT_P_SFT 10 +#define RT5670_OT_P_NOR (0x0 << 10) +#define RT5670_OT_P_INV (0x1 << 10) +#define RT5670_JD1_1_EN_MASK (0x1 << 9) +#define RT5670_JD1_1_EN_SFT 9 +#define RT5670_JD1_1_DIS (0x0 << 9) +#define RT5670_JD1_1_EN (0x1 << 9) + +/* IRQ Control 2 (0xbe) */ +#define RT5670_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5670_IRQ_MB1_OC_SFT 15 +#define RT5670_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5670_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5670_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5670_IRQ_MB2_OC_SFT 14 +#define RT5670_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5670_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5670_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5670_MB1_OC_STKY_SFT 11 +#define RT5670_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5670_MB1_OC_STKY_EN (0x1 << 11) +#define RT5670_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5670_MB2_OC_STKY_SFT 10 +#define RT5670_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5670_MB2_OC_STKY_EN (0x1 << 10) +#define RT5670_MB1_OC_P_MASK (0x1 << 7) +#define RT5670_MB1_OC_P_SFT 7 +#define RT5670_MB1_OC_P_NOR (0x0 << 7) +#define RT5670_MB1_OC_P_INV (0x1 << 7) +#define RT5670_MB2_OC_P_MASK (0x1 << 6) +#define RT5670_MB2_OC_P_SFT 6 +#define RT5670_MB2_OC_P_NOR (0x0 << 6) +#define RT5670_MB2_OC_P_INV (0x1 << 6) +#define RT5670_MB1_OC_CLR (0x1 << 3) +#define RT5670_MB1_OC_CLR_SFT 3 +#define RT5670_MB2_OC_CLR (0x1 << 2) +#define RT5670_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5670_GP1_PIN_MASK (0x1 << 15) +#define RT5670_GP1_PIN_SFT 15 +#define RT5670_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5670_GP1_PIN_IRQ (0x1 << 15) +#define RT5670_GP2_PIN_MASK (0x1 << 14) +#define RT5670_GP2_PIN_SFT 14 +#define RT5670_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5670_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5670_GP3_PIN_MASK (0x3 << 12) +#define RT5670_GP3_PIN_SFT 12 +#define RT5670_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5670_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5670_GP3_PIN_IRQ (0x2 << 12) +#define RT5670_GP4_PIN_MASK (0x1 << 11) +#define RT5670_GP4_PIN_SFT 11 +#define RT5670_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5670_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5670_DP_SIG_MASK (0x1 << 10) +#define RT5670_DP_SIG_SFT 10 +#define RT5670_DP_SIG_TEST (0x0 << 10) +#define RT5670_DP_SIG_AP (0x1 << 10) +#define RT5670_GPIO_M_MASK (0x1 << 9) +#define RT5670_GPIO_M_SFT 9 +#define RT5670_GPIO_M_FLT (0x0 << 9) +#define RT5670_GPIO_M_PH (0x1 << 9) +#define RT5670_I2S2_PIN_MASK (0x1 << 8) +#define RT5670_I2S2_PIN_SFT 8 +#define RT5670_I2S2_PIN_I2S (0x0 << 8) +#define RT5670_I2S2_PIN_GPIO (0x1 << 8) +#define RT5670_GP5_PIN_MASK (0x1 << 7) +#define RT5670_GP5_PIN_SFT 7 +#define RT5670_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5670_GP5_PIN_DMIC3_SDA (0x1 << 7) +#define RT5670_GP6_PIN_MASK (0x1 << 6) +#define RT5670_GP6_PIN_SFT 6 +#define RT5670_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5670_GP6_PIN_DMIC1_SDA (0x1 << 6) +#define RT5670_GP7_PIN_MASK (0x3 << 4) +#define RT5670_GP7_PIN_SFT 4 +#define RT5670_GP7_PIN_GPIO7 (0x0 << 4) +#define RT5670_GP7_PIN_DMIC1_SDA (0x1 << 4) +#define RT5670_GP7_PIN_PDM_SCL2 (0x2 << 4) +#define RT5670_GP8_PIN_MASK (0x1 << 3) +#define RT5670_GP8_PIN_SFT 3 +#define RT5670_GP8_PIN_GPIO8 (0x0 << 3) +#define RT5670_GP8_PIN_DMIC2_SDA (0x1 << 3) +#define RT5670_GP9_PIN_MASK (0x1 << 2) +#define RT5670_GP9_PIN_SFT 2 +#define RT5670_GP9_PIN_GPIO9 (0x0 << 2) +#define RT5670_GP9_PIN_DMIC3_SDA (0x1 << 2) +#define RT5670_GP10_PIN_MASK (0x3) +#define RT5670_GP10_PIN_SFT 0 +#define RT5670_GP10_PIN_GPIO9 (0x0) +#define RT5670_GP10_PIN_DMIC3_SDA (0x1) +#define RT5670_GP10_PIN_PDM_ADT2 (0x2) + +/* GPIO Control 2 (0xc1) */ +#define RT5670_GP4_PF_MASK (0x1 << 11) +#define RT5670_GP4_PF_SFT 11 +#define RT5670_GP4_PF_IN (0x0 << 11) +#define RT5670_GP4_PF_OUT (0x1 << 11) +#define RT5670_GP4_OUT_MASK (0x1 << 10) +#define RT5670_GP4_OUT_SFT 10 +#define RT5670_GP4_OUT_LO (0x0 << 10) +#define RT5670_GP4_OUT_HI (0x1 << 10) +#define RT5670_GP4_P_MASK (0x1 << 9) +#define RT5670_GP4_P_SFT 9 +#define RT5670_GP4_P_NOR (0x0 << 9) +#define RT5670_GP4_P_INV (0x1 << 9) +#define RT5670_GP3_PF_MASK (0x1 << 8) +#define RT5670_GP3_PF_SFT 8 +#define RT5670_GP3_PF_IN (0x0 << 8) +#define RT5670_GP3_PF_OUT (0x1 << 8) +#define RT5670_GP3_OUT_MASK (0x1 << 7) +#define RT5670_GP3_OUT_SFT 7 +#define RT5670_GP3_OUT_LO (0x0 << 7) +#define RT5670_GP3_OUT_HI (0x1 << 7) +#define RT5670_GP3_P_MASK (0x1 << 6) +#define RT5670_GP3_P_SFT 6 +#define RT5670_GP3_P_NOR (0x0 << 6) +#define RT5670_GP3_P_INV (0x1 << 6) +#define RT5670_GP2_PF_MASK (0x1 << 5) +#define RT5670_GP2_PF_SFT 5 +#define RT5670_GP2_PF_IN (0x0 << 5) +#define RT5670_GP2_PF_OUT (0x1 << 5) +#define RT5670_GP2_OUT_MASK (0x1 << 4) +#define RT5670_GP2_OUT_SFT 4 +#define RT5670_GP2_OUT_LO (0x0 << 4) +#define RT5670_GP2_OUT_HI (0x1 << 4) +#define RT5670_GP2_P_MASK (0x1 << 3) +#define RT5670_GP2_P_SFT 3 +#define RT5670_GP2_P_NOR (0x0 << 3) +#define RT5670_GP2_P_INV (0x1 << 3) +#define RT5670_GP1_PF_MASK (0x1 << 2) +#define RT5670_GP1_PF_SFT 2 +#define RT5670_GP1_PF_IN (0x0 << 2) +#define RT5670_GP1_PF_OUT (0x1 << 2) +#define RT5670_GP1_OUT_MASK (0x1 << 1) +#define RT5670_GP1_OUT_SFT 1 +#define RT5670_GP1_OUT_LO (0x0 << 1) +#define RT5670_GP1_OUT_HI (0x1 << 1) +#define RT5670_GP1_P_MASK (0x1) +#define RT5670_GP1_P_SFT 0 +#define RT5670_GP1_P_NOR (0x0) +#define RT5670_GP1_P_INV (0x1) + +/* Scramble Function (0xcd) */ +#define RT5670_SCB_KEY_MASK (0xff) +#define RT5670_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5670_SCB_SWAP_MASK (0x1 << 15) +#define RT5670_SCB_SWAP_SFT 15 +#define RT5670_SCB_SWAP_DIS (0x0 << 15) +#define RT5670_SCB_SWAP_EN (0x1 << 15) +#define RT5670_SCB_MASK (0x1 << 14) +#define RT5670_SCB_SFT 14 +#define RT5670_SCB_DIS (0x0 << 14) +#define RT5670_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5670_BB_MASK (0x1 << 15) +#define RT5670_BB_SFT 15 +#define RT5670_BB_DIS (0x0 << 15) +#define RT5670_BB_EN (0x1 << 15) +#define RT5670_BB_CT_MASK (0x7 << 12) +#define RT5670_BB_CT_SFT 12 +#define RT5670_BB_CT_A (0x0 << 12) +#define RT5670_BB_CT_B (0x1 << 12) +#define RT5670_BB_CT_C (0x2 << 12) +#define RT5670_BB_CT_D (0x3 << 12) +#define RT5670_M_BB_L_MASK (0x1 << 9) +#define RT5670_M_BB_L_SFT 9 +#define RT5670_M_BB_R_MASK (0x1 << 8) +#define RT5670_M_BB_R_SFT 8 +#define RT5670_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5670_M_BB_HPF_L_SFT 7 +#define RT5670_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5670_M_BB_HPF_R_SFT 6 +#define RT5670_G_BB_BST_MASK (0x3f) +#define RT5670_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5670_M_MP3_L_MASK (0x1 << 15) +#define RT5670_M_MP3_L_SFT 15 +#define RT5670_M_MP3_R_MASK (0x1 << 14) +#define RT5670_M_MP3_R_SFT 14 +#define RT5670_M_MP3_MASK (0x1 << 13) +#define RT5670_M_MP3_SFT 13 +#define RT5670_M_MP3_DIS (0x0 << 13) +#define RT5670_M_MP3_EN (0x1 << 13) +#define RT5670_EG_MP3_MASK (0x1f << 8) +#define RT5670_EG_MP3_SFT 8 +#define RT5670_MP3_HLP_MASK (0x1 << 7) +#define RT5670_MP3_HLP_SFT 7 +#define RT5670_MP3_HLP_DIS (0x0 << 7) +#define RT5670_MP3_HLP_EN (0x1 << 7) +#define RT5670_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5670_M_MP3_ORG_L_SFT 6 +#define RT5670_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5670_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5670_MP3_WT_MASK (0x1 << 13) +#define RT5670_MP3_WT_SFT 13 +#define RT5670_MP3_WT_1_4 (0x0 << 13) +#define RT5670_MP3_WT_1_2 (0x1 << 13) +#define RT5670_OG_MP3_MASK (0x1f << 8) +#define RT5670_OG_MP3_SFT 8 +#define RT5670_HG_MP3_MASK (0x3f) +#define RT5670_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5670_3D_CF_MASK (0x1 << 15) +#define RT5670_3D_CF_SFT 15 +#define RT5670_3D_CF_DIS (0x0 << 15) +#define RT5670_3D_CF_EN (0x1 << 15) +#define RT5670_3D_HP_MASK (0x1 << 14) +#define RT5670_3D_HP_SFT 14 +#define RT5670_3D_HP_DIS (0x0 << 14) +#define RT5670_3D_HP_EN (0x1 << 14) +#define RT5670_3D_BT_MASK (0x1 << 13) +#define RT5670_3D_BT_SFT 13 +#define RT5670_3D_BT_DIS (0x0 << 13) +#define RT5670_3D_BT_EN (0x1 << 13) +#define RT5670_3D_1F_MIX_MASK (0x3 << 11) +#define RT5670_3D_1F_MIX_SFT 11 +#define RT5670_3D_HP_M_MASK (0x1 << 10) +#define RT5670_3D_HP_M_SFT 10 +#define RT5670_3D_HP_M_SUR (0x0 << 10) +#define RT5670_3D_HP_M_FRO (0x1 << 10) +#define RT5670_M_3D_HRTF_MASK (0x1 << 9) +#define RT5670_M_3D_HRTF_SFT 9 +#define RT5670_M_3D_D2H_MASK (0x1 << 8) +#define RT5670_M_3D_D2H_SFT 8 +#define RT5670_M_3D_D2R_MASK (0x1 << 7) +#define RT5670_M_3D_D2R_SFT 7 +#define RT5670_M_3D_REVB_MASK (0x1 << 6) +#define RT5670_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5670_2ND_HPF_MASK (0x1 << 15) +#define RT5670_2ND_HPF_SFT 15 +#define RT5670_2ND_HPF_DIS (0x0 << 15) +#define RT5670_2ND_HPF_EN (0x1 << 15) +#define RT5670_HPF_CF_L_MASK (0x7 << 12) +#define RT5670_HPF_CF_L_SFT 12 +#define RT5670_1ST_HPF_MASK (0x1 << 11) +#define RT5670_1ST_HPF_SFT 11 +#define RT5670_1ST_HPF_DIS (0x0 << 11) +#define RT5670_1ST_HPF_EN (0x1 << 11) +#define RT5670_HPF_CF_R_MASK (0x7 << 8) +#define RT5670_HPF_CF_R_SFT 8 +#define RT5670_ZD_T_MASK (0x3 << 6) +#define RT5670_ZD_T_SFT 6 +#define RT5670_ZD_F_MASK (0x3 << 4) +#define RT5670_ZD_F_SFT 4 +#define RT5670_ZD_F_IM (0x0 << 4) +#define RT5670_ZD_F_ZC_IM (0x1 << 4) +#define RT5670_ZD_F_ZC_IOD (0x2 << 4) +#define RT5670_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5670_SI_DAC_MASK (0x1 << 11) +#define RT5670_SI_DAC_SFT 11 +#define RT5670_SI_DAC_AUTO (0x0 << 11) +#define RT5670_SI_DAC_TEST (0x1 << 11) +#define RT5670_DC_CAL_M_MASK (0x1 << 10) +#define RT5670_DC_CAL_M_SFT 10 +#define RT5670_DC_CAL_M_CAL (0x0 << 10) +#define RT5670_DC_CAL_M_NOR (0x1 << 10) +#define RT5670_DC_CAL_MASK (0x1 << 9) +#define RT5670_DC_CAL_SFT 9 +#define RT5670_DC_CAL_DIS (0x0 << 9) +#define RT5670_DC_CAL_EN (0x1 << 9) +#define RT5670_HPD_RCV_MASK (0x7 << 6) +#define RT5670_HPD_RCV_SFT 6 +#define RT5670_HPD_PS_MASK (0x1 << 5) +#define RT5670_HPD_PS_SFT 5 +#define RT5670_HPD_PS_DIS (0x0 << 5) +#define RT5670_HPD_PS_EN (0x1 << 5) +#define RT5670_CAL_M_MASK (0x1 << 4) +#define RT5670_CAL_M_SFT 4 +#define RT5670_CAL_M_DEP (0x0 << 4) +#define RT5670_CAL_M_CAL (0x1 << 4) +#define RT5670_CAL_MASK (0x1 << 3) +#define RT5670_CAL_SFT 3 +#define RT5670_CAL_DIS (0x0 << 3) +#define RT5670_CAL_EN (0x1 << 3) +#define RT5670_CAL_TEST_MASK (0x1 << 2) +#define RT5670_CAL_TEST_SFT 2 +#define RT5670_CAL_TEST_DIS (0x0 << 2) +#define RT5670_CAL_TEST_EN (0x1 << 2) +#define RT5670_CAL_P_MASK (0x3) +#define RT5670_CAL_P_SFT 0 +#define RT5670_CAL_P_NONE (0x0) +#define RT5670_CAL_P_CAL (0x1) +#define RT5670_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5670_SV_MASK (0x1 << 15) +#define RT5670_SV_SFT 15 +#define RT5670_SV_DIS (0x0 << 15) +#define RT5670_SV_EN (0x1 << 15) +#define RT5670_SPO_SV_MASK (0x1 << 14) +#define RT5670_SPO_SV_SFT 14 +#define RT5670_SPO_SV_DIS (0x0 << 14) +#define RT5670_SPO_SV_EN (0x1 << 14) +#define RT5670_OUT_SV_MASK (0x1 << 13) +#define RT5670_OUT_SV_SFT 13 +#define RT5670_OUT_SV_DIS (0x0 << 13) +#define RT5670_OUT_SV_EN (0x1 << 13) +#define RT5670_HP_SV_MASK (0x1 << 12) +#define RT5670_HP_SV_SFT 12 +#define RT5670_HP_SV_DIS (0x0 << 12) +#define RT5670_HP_SV_EN (0x1 << 12) +#define RT5670_ZCD_DIG_MASK (0x1 << 11) +#define RT5670_ZCD_DIG_SFT 11 +#define RT5670_ZCD_DIG_DIS (0x0 << 11) +#define RT5670_ZCD_DIG_EN (0x1 << 11) +#define RT5670_ZCD_MASK (0x1 << 10) +#define RT5670_ZCD_SFT 10 +#define RT5670_ZCD_PD (0x0 << 10) +#define RT5670_ZCD_PU (0x1 << 10) +#define RT5670_M_ZCD_MASK (0x3f << 4) +#define RT5670_M_ZCD_SFT 4 +#define RT5670_M_ZCD_RM_L (0x1 << 9) +#define RT5670_M_ZCD_RM_R (0x1 << 8) +#define RT5670_M_ZCD_SM_L (0x1 << 7) +#define RT5670_M_ZCD_SM_R (0x1 << 6) +#define RT5670_M_ZCD_OM_L (0x1 << 5) +#define RT5670_M_ZCD_OM_R (0x1 << 4) +#define RT5670_SV_DLY_MASK (0xf) +#define RT5670_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5670_ZCD_HP_MASK (0x1 << 15) +#define RT5670_ZCD_HP_SFT 15 +#define RT5670_ZCD_HP_DIS (0x0 << 15) +#define RT5670_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5670_3D_SPK_MASK (0x1 << 15) +#define RT5670_3D_SPK_SFT 15 +#define RT5670_3D_SPK_DIS (0x0 << 15) +#define RT5670_3D_SPK_EN (0x1 << 15) +#define RT5670_3D_SPK_M_MASK (0x3 << 13) +#define RT5670_3D_SPK_M_SFT 13 +#define RT5670_3D_SPK_CG_MASK (0x1f << 8) +#define RT5670_3D_SPK_CG_SFT 8 +#define RT5670_3D_SPK_SG_MASK (0x1f) +#define RT5670_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5670_WND_MASK (0x1 << 15) +#define RT5670_WND_SFT 15 +#define RT5670_WND_DIS (0x0 << 15) +#define RT5670_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5670_WND_FC_NW_MASK (0x3f << 10) +#define RT5670_WND_FC_NW_SFT 10 +#define RT5670_WND_FC_WK_MASK (0x3f << 4) +#define RT5670_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5670_HPF_FC_MASK (0x3f << 6) +#define RT5670_HPF_FC_SFT 6 +#define RT5670_WND_FC_ST_MASK (0x3f) +#define RT5670_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5670_WND_TH_LO_MASK (0x3ff) +#define RT5670_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5670_WND_TH_HI_MASK (0x3ff) +#define RT5670_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5670_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5670_WND_WIND_SFT 13 +#define RT5670_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5670_WND_STRONG_SFT 12 +enum { + RT5670_NO_WIND, + RT5670_BREEZE, + RT5670_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5670_DP_ATT_MASK (0x3 << 14) +#define RT5670_DP_ATT_SFT 14 +#define RT5670_DP_SPK_MASK (0x1 << 10) +#define RT5670_DP_SPK_SFT 10 +#define RT5670_DP_SPK_DIS (0x0 << 10) +#define RT5670_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5670_EQ_PRE_VOL_MASK (0xffff) +#define RT5670_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5670_EQ_PST_VOL_MASK (0xffff) +#define RT5670_EQ_PST_VOL_SFT 0 + +/* Jack Detect Control 3 (0xf8) */ +#define RT5670_CMP_MIC_IN_DET_MASK (0x7 << 12) +#define RT5670_JD_CBJ_EN (0x1 << 7) +#define RT5670_JD_CBJ_POL (0x1 << 6) +#define RT5670_JD_TRI_CBJ_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_CBJ_SEL_SFT (3) +#define RT5670_JD_CBJ_GPIO_JD1 (0x0 << 3) +#define RT5670_JD_CBJ_JD1_1 (0x1 << 3) +#define RT5670_JD_CBJ_JD1_2 (0x2 << 3) +#define RT5670_JD_CBJ_JD2 (0x3 << 3) +#define RT5670_JD_CBJ_JD3 (0x4 << 3) +#define RT5670_JD_CBJ_GPIO_JD2 (0x5 << 3) +#define RT5670_JD_CBJ_MX0B_12 (0x6 << 3) +#define RT5670_JD_TRI_HPO_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_HPO_SEL_SFT (0) +#define RT5670_JD_HPO_GPIO_JD1 (0x0) +#define RT5670_JD_HPO_JD1_1 (0x1) +#define RT5670_JD_HPO_JD1_2 (0x2) +#define RT5670_JD_HPO_JD2 (0x3) +#define RT5670_JD_HPO_JD3 (0x4) +#define RT5670_JD_HPO_GPIO_JD2 (0x5) +#define RT5670_JD_HPO_MX0B_12 (0x6) + +/* Digital Misc Control (0xfa) */ +#define RT5670_RST_DSP (0x1 << 13) +#define RT5670_IF1_ADC1_IN1_SEL (0x1 << 12) +#define RT5670_IF1_ADC1_IN1_SFT 12 +#define RT5670_IF1_ADC1_IN2_SEL (0x1 << 11) +#define RT5670_IF1_ADC1_IN2_SFT 11 +#define RT5670_IF1_ADC2_IN1_SEL (0x1 << 10) +#define RT5670_IF1_ADC2_IN1_SFT 10 + +/* General Control2 (0xfb) */ +#define RT5670_RXDC_SRC_MASK (0x1 << 7) +#define RT5670_RXDC_SRC_STO (0x0 << 7) +#define RT5670_RXDC_SRC_MONO (0x1 << 7) +#define RT5670_RXDC_SRC_SFT (7) +#define RT5670_RXDP2_SEL_MASK (0x1 << 3) +#define RT5670_RXDP2_SEL_IF2 (0x0 << 3) +#define RT5670_RXDP2_SEL_ADC (0x1 << 3) +#define RT5670_RXDP2_SEL_SFT (3) + +/* System Clock Source */ +enum { + RT5670_SCLK_S_MCLK, + RT5670_SCLK_S_PLL1, + RT5670_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5670_PLL1_S_MCLK, + RT5670_PLL1_S_BCLK1, + RT5670_PLL1_S_BCLK2, + RT5670_PLL1_S_BCLK3, + RT5670_PLL1_S_BCLK4, +}; + +enum { + RT5670_AIF1, + RT5670_AIF2, + RT5670_AIF3, + RT5670_AIF4, + RT5670_AIFS, +}; + +enum { + RT5670_DMIC_DATA_GPIO6, + RT5670_DMIC_DATA_IN2P, + RT5670_DMIC_DATA_GPIO7, +}; + +enum { + RT5670_DMIC_DATA_GPIO8, + RT5670_DMIC_DATA_IN3N, +}; + +enum { + RT5670_DMIC_DATA_GPIO9, + RT5670_DMIC_DATA_GPIO10, + RT5670_DMIC_DATA_GPIO5, +}; + +struct rt5670_priv { + struct snd_soc_codec *codec; + struct rt5670_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5670_AIFS]; + int bclk[RT5670_AIFS]; + int master[RT5670_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int dsp_sw; /* expected parameter setting */ + int dsp_rate; + int jack_type; +}; + +#endif /* __RT5670_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3] ASoC: tas2552: Support TI TAS2552 Amplifier
by Dan Murphy 02 Jul '14

02 Jul '14

Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v3 - Updated bindings doc per comments, rearranged probe pdata vs np check - https://patchwork.kernel.org/patch/4453481/ .../devicetree/bindings/sound/tas2552.txt | 22 + include/sound/tas2552-plat.h | 25 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2552.c | 463 ++++++++++++++++++++ sound/soc/codecs/tas2552.h | 75 ++++ 6 files changed, 592 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 include/sound/tas2552-plat.h create mode 100644 sound/soc/codecs/tas2552.c create mode 100644 sound/soc/codecs/tas2552.h diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 0000000..ada8fd4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,22 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - One of: + "ti,tas2552" - TAS2552 + +- reg - I2C slave address + +Optional properties: + +- power-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 0000000..65e7627 --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0b9571c..cc09261 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -99,6 +99,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TAS2552 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE @@ -754,4 +755,8 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1c..33bc7228 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -162,6 +162,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -325,3 +326,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 0000000..79b8212 --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,463 @@ +/* + * ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/tas2552-plat.h> + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x16}, + {TAS2552_CFG_3, 0x5E}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xC8}, + {TAS2552_PDM_CFG, 0x02}, + {TAS2552_PGA_GAIN, 0x10}, + {TAS2552_BOOST_PT_CTRL, 0x0F}, + {TAS2552_LIMIT_LVL_CTRL, 0x0C}, + {TAS2552_LIMIT_RATE_HYS, 0x20}, + {TAS2552_CFG_2, 0xEA}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x05}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +struct tas2552_data { + struct mutex mutex; + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + unsigned char regs[TAS2552_VBAT_DATA]; + int power_gpio; + u8 power_state:1; +}; + +static int tas2552_power(struct tas2552_data *data, u8 power) +{ + int ret = 0; + + BUG_ON(data->tas2552_client == NULL); + + mutex_lock(&data->mutex); + if (power == data->power_state) + goto exit; + + if (power) { + if (data->power_gpio >= 0) + gpio_set_value(data->power_gpio, 1); + + data->power_state = 1; + } else { + if (data->power_gpio >= 0) + gpio_set_value(data->power_gpio, 0); + + data->power_state = 0; + } + +exit: + mutex_unlock(&data->mutex); + return ret; +} + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg = 0x0; + + if (sw_shutdown) + cfg1_reg |= (sw_shutdown << 1); + else + cfg1_reg &= ~TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static void tas2552_init(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, TAS2552_CFG_1, 0x16); + snd_soc_write(codec, TAS2552_CFG_3, 0x5E); + snd_soc_write(codec, TAS2552_DOUT, 0x00); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, 0xC8); + snd_soc_write(codec, TAS2552_PDM_CFG, 0x02); + snd_soc_write(codec, TAS2552_PGA_GAIN, 0x10); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, 0x0F); + snd_soc_write(codec, TAS2552_LIMIT_LVL_CTRL, 0x0C); + snd_soc_write(codec, TAS2552_LIMIT_RATE_HYS, 0x20); + snd_soc_write(codec, TAS2552_CFG_2, 0xEA); +} + +static int tas2552_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u8 wclk_reg; + struct snd_soc_codec *codec = dai->codec; + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (params_rate(params)) { + case 8000: + wclk_reg = TAS2552_8KHZ; + break; + case 11025: + wclk_reg = TAS2552_11_12KHZ; + break; + case 16000: + wclk_reg = TAS2552_16KHZ; + break; + case 32000: + wclk_reg = TAS2552_32KHZ; + break; + case 22050: + case 24000: + wclk_reg = TAS2552_22_24KHZ; + break; + case 44100: + case 48000: + wclk_reg = TAS2552_44_48KHZ; + break; + case 96000: + wclk_reg = TAS2552_88_96KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, TAS2552_CFG_3, TAS2552_WCLK_MASK, wclk_reg); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + u8 serial_format; + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, + (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK), + serial_format); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format = 0x0; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format = TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format = TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format = TAS2552_DAIFMT_LEFT_J; + break; + + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, TAS2552_DATA_FORMAT_MASK, + serial_format); + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *data = dev_get_drvdata(dai->dev); + + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + + tas2552_sw_shutdown(data, 1); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + snd_soc_write(codec, TAS2552_PLL_CTRL_1, 0x10); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, 0x00); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, 0x00); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + tas2552_sw_shutdown(data, 0); + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg = 0x0; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg |= TAS2552_MUTE_MASK; + else + cfg1_reg &= ~TAS2552_MUTE_MASK; + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + return 0; +} + +static int tas2552_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_sw_shutdown(tas2552, 1); + tas2552_power(tas2552, 1); + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + tas2552_sw_shutdown(tas2552, 0); + + return 0; +} + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_sw_shutdown(tas2552, 1); + tas2552_power(tas2552, 0); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .startup = tas2552_startup, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552->codec = codec; + tas2552_power(tas2552, 1); + tas2552_init(codec); + + return 0; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_power(tas2552, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + struct tas2552_platform_data *pdata = client->dev.platform_data; + struct device_node *np = client->dev.of_node; + int ret; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + if (np) { + data->power_gpio = of_get_named_gpio(np, "enable-gpio", 0); + } else if (pdata) { + data->power_gpio = pdata->enable_gpio; + } else { + dev_err(dev, "Platform or dev tree data not set\n"); + return -ENODEV; + } + + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + dev_set_drvdata(&client->dev, data); + + mutex_init(&data->mutex); + + if (data->power_gpio >= 0) { + ret = devm_gpio_request(dev, data->power_gpio, + "tas2552 enable"); + if (ret < 0) { + dev_err(dev, "Failed to request power GPIO (%d)\n", + data->power_gpio); + goto err_gpio; + } + gpio_direction_output(data->power_gpio, 0); + } + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return 0; + +err_gpio: + data->tas2552_client = NULL; + return ret; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552-codec", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 0000000..174c64d --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,75 @@ +/* + * ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_LIMIT_LVL_CTRL 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) +#define TAS2552_CLASSD_EN 0x80 + +#define TAS2552_PLL_ENABLE (1 << 3) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK 0x80 +#define TAS2552_BIT_CLK_MASK 0x40 +#define TAS2552_DATA_FORMAT_MASK 0x0c + +#define TAS2552_DAIFMT_DSP 0x04 +#define TAS2552_DAIFMT_RIGHT_J 0x08 +#define TAS2552_DAIFMT_LEFT_J 0x0c + +/* WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ 0x01 +#define TAS2552_16KHZ 0x02 +#define TAS2552_22_24KHZ 0x03 +#define TAS2552_32KHZ 0x04 +#define TAS2552_44_48KHZ 0x05 +#define TAS2552_88_96KHZ 0x06 +#define TAS2552_176_192KHZ 0x07 + +#endif -- 1.7.9.5

3 9

[alsa-devel] [PATCH v2] ALSA: hda - restore BCLK M/N values when resuming HSW/BDW display controller
by mengdong.lin＠intel.com 02 Jul '14

02 Jul '14

From: Mengdong Lin <mengdong.lin(a)intel.com> For Intel Haswell/Broadwell display HD-A controller, the 24MHz HD-A link BCLK is converted from Core Display Clock (CDCLK): BCLK = CDCLK * M / N And there are two registers EM4 and EM5 to program M, N value respectively. The EM4/EM5 values will be lost and when the display power well is disabled. BIOS programs CDCLK selected by OEM and EM4/EM5, but BIOS has no idea about display power well on/off at runtime. So the M/N can be wrong if non-default CDCLK is used when the audio controller resumes, which results in an invalid BCLK and abnormal audio playback rate. So this patch saves and restores valid M/N values on controller suspend/resume. And 'struct hda_intel' is defined to contain standard HD-A 'struct azx' and Intel specific fields, as Takashi suggested. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bb65a124..ff9cacd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -288,6 +288,24 @@ static char *driver_short_names[] = { [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; + +/* Intel HSW/BDW display HDA controller Extended Mode registers. + * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display + * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled. + */ +#define ICH6_REG_EM4 0x100c +#define ICH6_REG_EM5 0x1010 + +struct hda_intel { + struct azx chip; + + /* HSW/BDW display HDA controller to restore BCLK from CDCLK */ + unsigned int bclk_m; + unsigned int bclk_n; +}; + + #ifdef CONFIG_X86 static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on) { @@ -580,6 +598,22 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ +static void haswell_save_bclk(struct azx *chip) +{ + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + + hda->bclk_m = azx_readw(chip, EM4); + hda->bclk_n = azx_readw(chip, EM5); +} + +static void haswell_restore_bclk(struct azx *chip) +{ + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + + azx_writew(chip, EM4, hda->bclk_m); + azx_writew(chip, EM5, hda->bclk_n); +} + #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -606,6 +640,13 @@ static int azx_suspend(struct device *dev) free_irq(chip->irq, chip); chip->irq = -1; } + + /* Save BCLK M/N values before they become invalid in D3. + * Will test if display power well can be released now. + */ + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + haswell_save_bclk(chip); + if (chip->msi) pci_disable_msi(chip->pci); pci_disable_device(pci); @@ -625,8 +666,10 @@ static int azx_resume(struct device *dev) if (chip->disabled) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); + haswell_restore_bclk(chip); + } pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); if (pci_enable_device(pci) < 0) { @@ -670,8 +713,10 @@ static int azx_runtime_suspend(struct device *dev) azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + haswell_save_bclk(chip); hda_display_power(false); + } return 0; } @@ -689,8 +734,10 @@ static int azx_runtime_resume(struct device *dev) if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); + haswell_restore_bclk(chip); + } /* Read STATESTS before controller reset */ status = azx_readw(chip, STATESTS); @@ -883,6 +930,8 @@ static int register_vga_switcheroo(struct azx *chip) static int azx_free(struct azx *chip) { struct pci_dev *pci = chip->pci; + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + int i; if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) @@ -930,7 +979,7 @@ static int azx_free(struct azx *chip) hda_display_power(false); hda_i915_exit(); } - kfree(chip); + kfree(hda); return 0; } @@ -1174,6 +1223,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, static struct snd_device_ops ops = { .dev_free = azx_dev_free, }; + struct hda_intel *hda; struct azx *chip; int err; @@ -1183,13 +1233,14 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, if (err < 0) return err; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (!chip) { - dev_err(card->dev, "Cannot allocate chip\n"); + hda = kzalloc(sizeof(*hda), GFP_KERNEL); + if (!hda) { + dev_err(card->dev, "Cannot allocate hda\n"); pci_disable_device(pci); return -ENOMEM; } + chip = &hda->chip; spin_lock_init(&chip->reg_lock); mutex_init(&chip->open_mutex); chip->card = card; -- 1.8.1.2

3 8

[alsa-devel] [PATCH v2 0/5] ALSA/ASoC/dmaengine: Fix 3 bytes physical sample support
by Peter Ujfalusi 02 Jul '14

02 Jul '14

Hi, Changes since v1: - the hw.formats mask preparation moved to dmaengine_pcm_set_runtime_hwparams() - we prepare the mask for all supported sample formats based on the physical width of the sample and beased on the slave_caps information. I have noticed while switching the daVinci audio to dmaengine_pcm that the _3LE and _3BE support is not working correctly via dmaengine_pcm: In case of _3LE/_3BE formats the samples are stored in 3 consecutive bytes without padding it to 4 bytes. This means that the DMA needs to be able to support 3 bytes word length in order to read/write the samples from memory correctly. Originally the code treated 24 bits physical length samples as they were 32 bits which leads to corruption when playing or recording audio. To fix the support for 3 bytes physical samples the dma driver also need to have support for such data arrangement. eDMA does have support for it in HW and the legacy davinci-pcm platform driver w/o dmaengine was able to support such formats. First step is to add DMA_SLAVE_BUSWIDTH_3_BYTES to dma_slave_buswidth for engines and users to select 3 bytes as bus width. Followed by: In dmaengine_pcm_set_runtime_hwparams() we should check the slave_caps of the dma if it supports the give sample physical width of the sample. Based on this information we initialize the hw.formats: masking out all non supported formats based on the physical width. In case the dma driver does not implement the slave_caps the default assumption is that it supports 1, 2 and 4 bytes widths. Regards, Peter --- Peter Ujfalusi (5): dma: Support for 3 bytes word size ALSA: pcm_dmaengine: Use the available wrapper to get physical width ASoC: generic-dmaengine-pcm: Prepare formats mask for valid physical sample sizes ALSA: pcm_dmaengine: Correct support for 24bits physical sample widths dma: edma: Declare DMA_SLAVE_BUSWIDTH_3_BYTES as supported buswidth drivers/dma/edma.c | 1 + include/linux/dmaengine.h | 1 + sound/core/pcm_dmaengine.c | 4 +++- sound/soc/soc-generic-dmaengine-pcm.c | 26 +++++++++++++++++++++++++- 4 files changed, 30 insertions(+), 2 deletions(-) -- 2.0.0

3 10

[alsa-devel] PCM Plugin Channel question
by Brandon Yates 02 Jul '14

02 Jul '14

Hello, I have developed a pcm ext plugin that applies some processing to the audio stream in the transfer callback. My plugin works for a mono (1 channel) stream. I'd now like to expand it to support stereo data in the form of 2 channels interleaved. I can't find much information on how this works. I have a couple questions that I'd greatly appreciate if anyone could help with. 1) Is there any way in the 'SND_PCM_PLUGIN_DEFINE_FUNC' function that I can determine if the user application has opened the stream with 1 channel vs 2 channel? 2) I assume in the case of two channels my transfer function would need to: a)deinterleave the data from source so my processing routine can access it in a linear buffer and b) reinterleave the data before I copy it to destination. Is this correct? Is there more to it or things I am overlooking? 3) For IO-plugin I see there is a `SND_PCM_IOPLUG_HW_ACCESS` value I could use to restrict the stream type to `SND_PCM_ACCESS_RW_INTERLEAVED`. Does there exist a similar function for External PCM Plugin? 4)Is there some fundamental design idea here I am not grasping? Thanks much! Brandon

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[alsa-devel] [PATCH] First attempt at a driver for the internal codec on the Allwinner A20.
by Jon Smirl 02 Jul '14

02 Jul '14

First attempt at a driver for the internal codec on the Allwinner A20. This codec is a combination codec and DAI, it is all on-chip. So why don't I have a DMA buffer? Any other hints to get this working would be appreciated. [ 31.910394] pcm_sanity_check (null) (null) [ 31.910415] ------------[ cut here ]------------ [ 31.910440] WARNING: CPU: 1 PID: 738 at sound/core/pcm_lib.c:2090 pcm_sanity_check+0x70/0x98() [ 31.910448] Modules linked in: brcmfmac brcmutil [ 31.910470] CPU: 1 PID: 738 Comm: aplay Not tainted 3.15.0+ #161 [ 31.910508] [<c001513c>] (unwind_backtrace) from [<c0011d08>] (show_stack+0x18/0x1c) [ 31.910528] [<c0011d08>] (show_stack) from [<c0566934>] (dump_stack+0x78/0x94) [ 31.910549] [<c0566934>] (dump_stack) from [<c00237c4>] (warn_slowpath_common+0x6c/0x90) [ 31.910567] [<c00237c4>] (warn_slowpath_common) from [<c0023890>] (warn_slowpath_null+0x20/0x28) [ 31.910585] [<c0023890>] (warn_slowpath_null) from [<c03f7e64>] (pcm_sanity_check+0x70/0x98) [ 31.910602] [<c03f7e64>] (pcm_sanity_check) from [<c03faa08>] (snd_pcm_lib_write+0x1c/0x74) [ 31.910619] [<c03faa08>] (snd_pcm_lib_write) from [<c03f6400>] (snd_pcm_playback_ioctl1+0x3c8/0x4a8) [ 31.910637] [<c03f6400>] (snd_pcm_playback_ioctl1) from [<c013762c>] (do_vfs_ioctl+0x4d4/0x594) [ 31.910653] [<c013762c>] (do_vfs_ioctl) from [<c0137740>] (SyS_ioctl+0x54/0x7c) [ 31.910668] [<c0137740>] (SyS_ioctl) from [<c000e420>] (ret_fast_syscall+0x0/0x30) [ 31.910677] ---[ end trace 86fcd83010b5b5b5 ]--- --- sound/soc/sunxi/sunxi_codec_main.c | 687 ++++++++++++++++++++++++++++++++++++ 1 file changed, 687 insertions(+) create mode 100644 sound/soc/sunxi/sunxi_codec_main.c diff --git a/sound/soc/sunxi/sunxi_codec_main.c b/sound/soc/sunxi/sunxi_codec_main.c new file mode 100644 index 0000000..4070bfe --- /dev/null +++ b/sound/soc/sunxi/sunxi_codec_main.c @@ -0,0 +1,687 @@ +/* + * + * Licensed under the GPL-2. + */ + +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <linux/of_address.h> +#include <linux/clk.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/dmaengine_pcm.h> + +//Codec Register +#define SUNXI_DAC_DPC (0x00) +#define SUNXI_DAC_FIFOC (0x04) +#define SUNXI_DAC_FIFOS (0x08) +#define SUNXI_DAC_TXDATA (0x0c) +#define SUNXI_DAC_ACTL (0x10) +#define SUNXI_DAC_TUNE (0x14) +#define SUNXI_DAC_DEBUG (0x18) +#define SUNXI_ADC_FIFOC (0x1c) +#define SUNXI_ADC_FIFOS (0x20) +#define SUNXI_ADC_RXDATA (0x24) +#define SUNXI_ADC_ACTL (0x28) +#define SUNXI_ADC_DEBUG (0x2c) +#define SUNXI_DAC_TXCNT (0x30) +#define SUNXI_ADC_RXCNT (0x34) +#define SUNXI_BIAS_CRT (0x38) +#define SUNXI_MIC_CRT (0x3c) +#define SUNXI_CODEC_REGS_NUM (13) + +#define DAIFMT_16BITS (16) +#define DAIFMT_20BITS (20) + +#define DAIFMT_BS_MASK (~(1<<16)) //FIFO big small mode mask +#define DAIFMT_BITS_MASK (~(1<<5)) //FIFO Bits select mask,not used yet. +#define SAMPLE_RATE_MASK (~(7<<29)) //Sample Rate slect mask + +#define DAC_EN (31) +#define DIGITAL_VOL (12) +//For CODEC OLD VERSION +#define DAC_VERSION (23) + +#define DAC_CHANNEL (6) +#define LAST_SE (26) +#define TX_FIFO_MODE (24) +#define DRA_LEVEL (21) +#define TX_TRI_LEVEL (8) +#define DAC_MODE (6) //not used yet +#define TASR (5) //not used yet +#define DAC_DRQ (4) +#define DAC_FIFO_FLUSH (0) + +#define VOLUME (0) +#define PA_MUTE (6) +#define MIXPAS (7) +#define DACPAS (8) +#define MIXEN (29) +#define DACAEN_L (30) +#define DACAEN_R (31) + +#define ADC_DIG_EN (28) +#define RX_FIFO_MODE (24) +#define RX_TRI_LEVEL (8) +#define ADC_MODE (7) +#define RASR (6) +#define ADC_DRQ (4) +#define ADC_FIFO_FLUSH (0) + +#define ADC_LF_EN (31) +#define ADC_RI_EN (30) +#define ADC_EN (30) +#define MIC1_EN (29) +#define MIC2_EN (28) +#define VMIC_EN (27) +#define MIC_GAIN (25) +#define ADC_SELECT (17) +#define PA_ENABLE (4) +#define HP_DIRECT (3) + + +enum sunxi_device_id {SUN4A, SUN4I, SUN5I, SUN7I}; + +struct sunxi_priv { + struct regmap *regmap; + int irq; + struct clk *clk_apb, *clk_pll2, *clk_module; + + enum sunxi_device_id id; + + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; +}; + +static int codec_play_start(struct sunxi_priv *priv) +{ +#ifdef JDS + if (gpio_pa_shutdown) + gpio_write_one_pin_value(gpio_pa_shutdown, 1, "audio_pa_ctrl"); +#endif + //flush TX FIFO + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0x1 << DAC_FIFO_FLUSH, 0x1 << DAC_FIFO_FLUSH); + //enable dac drq + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0x1 << DAC_DRQ, 0x1 << DAC_DRQ); + return 0; +} + +static int codec_play_stop(struct sunxi_priv *priv) +{ + //pa mute +#ifdef JDS + if (gpio_pa_shutdown) + gpio_write_one_pin_value(gpio_pa_shutdown, 0, "audio_pa_ctrl"); +#endif + regmap_update_bits(priv->regmap, SUNXI_DAC_ACTL, 0x1 << PA_MUTE, 0x0 << PA_MUTE); + mdelay(5); + //disable dac drq + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0x1 << DAC_DRQ, 0x0 << DAC_DRQ); + //pa mute + regmap_update_bits(priv->regmap, SUNXI_DAC_ACTL, 0x1 << PA_MUTE, 0x0 << PA_MUTE); + regmap_update_bits(priv->regmap, SUNXI_DAC_ACTL, 0x1 << DACAEN_L, 0x0 << DACAEN_L); + regmap_update_bits(priv->regmap, SUNXI_DAC_ACTL, 0x1 << DACAEN_R, 0x0 << DACAEN_R); + return 0; +} + +static int codec_capture_start(struct sunxi_priv *priv) +{ + //enable adc drq +#ifdef JDS + if (gpio_pa_shutdown) + gpio_write_one_pin_value(gpio_pa_shutdown, 1, "audio_pa_ctrl"); +#endif + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0x1 << ADC_DRQ, 0x1 << ADC_DRQ); + return 0; +} + +static int codec_capture_stop(struct sunxi_priv *priv) +{ + //disable adc drq + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0x1 << ADC_DRQ, 0x0 << ADC_DRQ); + //enable mic1 pa + regmap_update_bits(priv->regmap, SUNXI_ADC_ACTL, 0x1 << MIC1_EN, 0x0 << MIC1_EN); + //enable VMIC + regmap_update_bits(priv->regmap, SUNXI_ADC_ACTL, 0x1 << VMIC_EN, 0x0 << VMIC_EN); + if (priv->id == SUN7I) { + regmap_update_bits(priv->regmap, SUNXI_DAC_TUNE, 0x3 << 8, 0x0 << 8); + } + //enable adc digital + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0x1 << ADC_DIG_EN, 0x0 << ADC_DIG_EN); + //set RX FIFO mode + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0x1 << RX_FIFO_MODE, 0x0 << RX_FIFO_MODE); + //flush RX FIFO + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0x1 << ADC_FIFO_FLUSH, 0x0 << ADC_FIFO_FLUSH); + //enable adc1 analog + regmap_update_bits(priv->regmap, SUNXI_ADC_ACTL, 0x3 << ADC_EN, 0x0 << ADC_EN); + return 0; +} + +static int sunxi_codec_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct sunxi_priv *priv = snd_soc_card_get_drvdata(card); + + printk("JDS - sunxi_codec_trigger cmd %d\n", cmd); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + codec_capture_start(priv); + else + codec_play_start(priv); + break; break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + codec_capture_stop(priv); + else + codec_play_stop(priv); + break; + default: + return -EINVAL; + } + return 0; +} + +static int sunxi_codec_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct sunxi_priv *priv = snd_soc_card_get_drvdata(card); + + printk("JDS - sunxi_codec_prepare\n"); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK){ + regmap_update_bits(priv->regmap, SUNXI_DAC_DPC, 0x1 << DAC_EN, 0x1 << DAC_EN); + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0x1 << DAC_FIFO_FLUSH, 0x1 << DAC_FIFO_FLUSH); + //set TX FIFO send drq level + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0xf << TX_TRI_LEVEL, 0xf << TX_TRI_LEVEL); + if (substream->runtime->rate > 32000) { + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0x1 << 28, 0x0 << 28); + } else { + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0x1 << 28, 0x1 << 28); + } + //set TX FIFO MODE + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0x1 << TX_FIFO_MODE, 0x1 << TX_FIFO_MODE); + //send last sample when dac fifo under run + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 0x1 << LAST_SE, 0x0 << LAST_SE); + //enable dac analog + regmap_update_bits(priv->regmap, SUNXI_DAC_ACTL, 0x1 << DACAEN_L, 0x1 << DACAEN_L); + regmap_update_bits(priv->regmap, SUNXI_DAC_ACTL, 0x1 << DACAEN_R, 0x1 << DACAEN_R); + //enable dac to pa + regmap_update_bits(priv->regmap, SUNXI_DAC_ACTL, 0x1 << DACPAS, 0x1 << DACPAS); + } else { + //enable mic1 pa + regmap_update_bits(priv->regmap, SUNXI_ADC_ACTL, 0x1 << MIC1_EN, 0x1 << MIC1_EN); + //mic1 gain 32dB + regmap_update_bits(priv->regmap, SUNXI_ADC_ACTL, 0x3 << 25, 0x1 << 25); + //enable VMIC + regmap_update_bits(priv->regmap, SUNXI_ADC_ACTL, 0x1 << VMIC_EN, 0x1 << VMIC_EN); + + if (priv->id == SUN7I) { + /* boost up record effect */ + regmap_update_bits(priv->regmap, SUNXI_DAC_TUNE, 0x3 << 8, 0x1 << 8); + } + + //enable adc digital + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0x1 << ADC_DIG_EN, 0x1 << ADC_DIG_EN); + //set RX FIFO mode + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0x1 << RX_FIFO_MODE, 0x1 << RX_FIFO_MODE); + //flush RX FIFO + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0x1 << ADC_FIFO_FLUSH, 0x1 << ADC_FIFO_FLUSH); + //set RX FIFO rec drq level + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 0xf << RX_TRI_LEVEL, 0x7 << RX_TRI_LEVEL); + //enable adc1 analog + regmap_update_bits(priv->regmap, SUNXI_ADC_ACTL, 0x3 << ADC_EN, 0x3 << ADC_EN); + } + return 0; +} + +static int sunxi_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct sunxi_priv *priv = snd_soc_card_get_drvdata(card); + unsigned int rate = params_rate(params); + + printk("CLK - sunxi_codec_hw_params substream %p runtime %p\n", substream, rtd); + switch (params_rate(params)) { + case 44100: + case 22050: + case 11025: + default: + clk_set_rate(priv->clk_pll2, 22579200); + clk_set_rate(priv->clk_module, 22579200); + break; + case 192000: + case 96000: + case 48000: + case 32000: + case 24000: + case 16000: + case 12000: + case 8000: + clk_set_rate(priv->clk_pll2, 24576000); + clk_set_rate(priv->clk_module, 24576000); + break; + } + + switch (params_rate(params)) { + default: + case 44100: + rate = 0; + break; + case 22050: + rate = 2; + break; + case 11025: + rate = 4; + break; + case 192000: + rate = 6; + break; + case 96000: + rate = 7; + break; + case 48000: + rate = 0; + break; + case 32000: + rate = 1; + break; + case 24000: + rate = 2; + break; + case 16000: + rate = 3; + break; + case 12000: + rate = 4; + break; + case 8000: + rate = 5; + break; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 7 << 29, rate << 29); + if (substream->runtime->channels == 1) + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 1 << 6, 1 << 6); + else + regmap_update_bits(priv->regmap, SUNXI_DAC_FIFOC, 1 << 6, 0 << 6); + } else { + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 7 << 29, rate << 29); + if (substream->runtime->channels == 1) + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 1 << 7, 1 << 7); + else + regmap_update_bits(priv->regmap, SUNXI_ADC_FIFOC, 1 << 7, 0 << 7); + } + return 0; +} + +static const struct snd_kcontrol_new sun7i_dac_ctls[] = { + /*SUNXI_DAC_ACTL = 0x10,PAVOL*/ + SOC_SINGLE("Master Playback Volume", SUNXI_DAC_ACTL, 0, 0x3f, 0), + SOC_SINGLE("Playback Switch", SUNXI_DAC_ACTL, 6, 1, 0), //全局输出开关 + SOC_SINGLE("FmL Switch", SUNXI_DAC_ACTL, 17, 1, 0), //Fm左开关 + SOC_SINGLE("FmR Switch", SUNXI_DAC_ACTL, 16, 1, 0), //Fm右开关 + SOC_SINGLE("LineL Switch", SUNXI_DAC_ACTL, 19, 1, 0), //Line左开关 + SOC_SINGLE("LineR Switch", SUNXI_DAC_ACTL, 18, 1, 0), //Line右开关 + SOC_SINGLE("Ldac Left Mixer", SUNXI_DAC_ACTL, 15, 1, 0), + SOC_SINGLE("Rdac Right Mixer", SUNXI_DAC_ACTL, 14, 1, 0), + SOC_SINGLE("Ldac Right Mixer", SUNXI_DAC_ACTL, 13, 1, 0), + SOC_SINGLE("Mic Input Mux", SUNXI_DAC_ACTL, 9, 15, 0), //from bit 9 to bit 12.Mic（麦克风）输入静音 + SOC_SINGLE("MIC output volume", SUNXI_DAC_ACTL, 20, 7, 0), + /* FM Input to output mixer Gain Control + * From -4.5db to 6db,1.5db/step,default is 0db + * -4.5db:0x0,-3.0db:0x1,-1.5db:0x2,0db:0x3 + * 1.5db:0x4,3.0db:0x5,4.5db:0x6,6db:0x7 + */ + SOC_SINGLE("Fm output Volume", SUNXI_DAC_ACTL, 23, 7, 0), + /* Line-in gain stage to output mixer Gain Control + * 0:-1.5db,1:0db + */ + SOC_SINGLE("Line output Volume", SUNXI_DAC_ACTL, 26, 1, 0), + + SOC_SINGLE("Master Capture Mute", SUNXI_ADC_ACTL, 4, 1, 0), + SOC_SINGLE("Right Capture Mute", SUNXI_ADC_ACTL, 31, 1, 0), + SOC_SINGLE("Left Capture Mute", SUNXI_ADC_ACTL, 30, 1, 0), + SOC_SINGLE("Linein Pre-AMP", SUNXI_ADC_ACTL, 13, 7, 0), + SOC_SINGLE("LINEIN APM Volume", SUNXI_MIC_CRT, 13, 0x7, 0), + /* ADC Input Gain Control, capture volume + * 000:-4.5db,001:-3db,010:-1.5db,011:0db,100:1.5db,101:3db,110:4.5db,111:6db + */ + SOC_SINGLE("Capture Volume", SUNXI_ADC_ACTL, 20, 7, 0), + /* + * MIC2 pre-amplifier Gain Control + * 00:0db,01:35db,10:38db,11:41db + */ + SOC_SINGLE("MicL Volume", SUNXI_ADC_ACTL, 25, 3, 0), //mic左音量 + SOC_SINGLE("MicR Volume", SUNXI_ADC_ACTL, 23, 3, 0), //mic右音量 + SOC_SINGLE("Mic2 Boost", SUNXI_ADC_ACTL, 29, 1, 0), + SOC_SINGLE("Mic1 Boost", SUNXI_ADC_ACTL, 28, 1, 0), + SOC_SINGLE("Mic Power", SUNXI_ADC_ACTL, 27, 1, 0), + SOC_SINGLE("ADC Input Mux", SUNXI_ADC_ACTL, 17, 7, 0), //ADC输入静音 + SOC_SINGLE("Mic2 gain Volume", SUNXI_MIC_CRT, 26, 7, 0), + /* + * MIC1 pre-amplifier Gain Control + * 00:0db,01:35db,10:38db,11:41db + */ + SOC_SINGLE("Mic1 gain Volume", SUNXI_MIC_CRT, 29, 3, 0), +}; + + +static int sunxi_codec_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai); + struct sunxi_priv *priv = snd_soc_card_get_drvdata(card); + + printk("CLK - sunxi_codec_dai_probe %p\n", priv); + snd_soc_dai_init_dma_data(dai, &priv->playback_dma_data, &priv->capture_dma_data); + + return 0; +} + +static int sunxi_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct sunxi_priv *priv = snd_soc_card_get_drvdata(card); + + int ret; + printk("JDS - CLK sunxi_codec_startup %p %p\n", codec, priv->clk_module); + ret = clk_prepare_enable(priv->clk_module); + if (ret) + return ret; + + printk("JDS - sunxi_codec_startup - ok\n"); + return 0; +} + +static void sunxi_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct sunxi_priv *priv = snd_soc_card_get_drvdata(card); + + printk("JDS - sunxi_codec_shutdown\n"); + clk_disable_unprepare(priv->clk_module); +} + +static const struct snd_soc_dai_ops sunxi_codec_dai_ops = { + .startup = sunxi_codec_startup, + .shutdown = sunxi_codec_shutdown, + .trigger = sunxi_codec_trigger, + .hw_params = sunxi_codec_hw_params, + .prepare = sunxi_codec_prepare, +}; + +static struct snd_soc_dai_driver sunxi_codec_dai = { + .probe = sunxi_codec_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_22050| SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000 |\ + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 192000, + }, + .ops = &sunxi_codec_dai_ops, +}; + +static const struct snd_soc_component_driver sunxi_codec_component = { + .name = "sunxi-codec", +}; + +static const struct regmap_config sunxi_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUNXI_MIC_CRT, +}; + +static const struct snd_soc_dapm_widget codec_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("Mic Bias"), + SND_SOC_DAPM_OUTPUT("HP_OUT"), + SND_SOC_DAPM_INPUT("MIC_IN"), + SND_SOC_DAPM_INPUT("LINE_IN"), +}; + +static struct snd_soc_dai_link cdc_dai = { + .name = "cdc", + .stream_name = "CDC PCM", + .codec_dai_name = "sunxi-codec-dai", + .cpu_dai_name = "1c22c00.codec", + .codec_name = "1c22c00.codec", + //.init = tegra_wm8903_init, + //.ops = &tegra_wm8903_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S, +}; + +static struct snd_soc_card snd_soc_sunxi_codec = { + .name = "sunxi-codec", + .owner = THIS_MODULE, + .dai_link = &cdc_dai, + .num_links = 1, + + .fully_routed = true, +}; + +static struct snd_soc_codec_driver dummy_codec = { + .controls = sun7i_dac_ctls, + .num_controls = ARRAY_SIZE(sun7i_dac_ctls), + .dapm_widgets = codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(codec_dapm_widgets), +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) +static struct snd_soc_dai_driver dummy_dai = { + .name = "sunxi-codec-dai", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static const struct of_device_id sunxi_codec_of_match[] = { + { .compatible = "allwinner,sun4i-a10a-codec", .data = (void *)SUN4A}, + { .compatible = "allwinner,sun4i-a10-codec", .data = (void *)SUN4I}, + { .compatible = "allwinner,sun5i-a13-codec", .data = (void *)SUN5I}, + { .compatible = "allwinner,sun7i-a20-codec", .data = (void *)SUN7I}, + {} +}; +MODULE_DEVICE_TABLE(of, sunxi_codec_of_match); + +static int sunxi_codec_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_sunxi_codec; + const struct of_device_id *of_id; + struct device *dev = &pdev->dev; + struct sunxi_priv *priv; + struct resource *res; + void __iomem *base; + int ret; + + if (!of_device_is_available(np)) + return -ENODEV; + + of_id = of_match_device(sunxi_codec_of_match, dev); + if (!of_id) + return -EINVAL; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, priv); + + priv->id = (int)of_id->data; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + priv->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sunxi_codec_regmap_config); + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + priv->irq = irq_of_parse_and_map(np, 0); + if (!priv->irq) { + dev_err(dev, "no irq for node %s\n", np->full_name); + return -ENXIO; + } + + /* Clock */ + priv->clk_apb = devm_clk_get(dev, "apb"); + if (IS_ERR(priv->clk_apb)) { + dev_err(dev, "failed to get apb clock.\n"); + return PTR_ERR(priv->clk_apb); + } + priv->clk_pll2 = devm_clk_get(dev, "pll2"); + if (IS_ERR(priv->clk_pll2)) { + dev_err(dev, "failed to get pll2 clock.\n"); + return PTR_ERR(priv->clk_pll2); + } + priv->clk_module = devm_clk_get(dev, "codec"); + if (IS_ERR(priv->clk_module)) { + dev_err(dev, "failed to get codec clock.\n"); + return PTR_ERR(priv->clk_module); + } + printk("CLK = card %p priv %p priv->clk_module %p\n", card, priv, priv->clk_module); + + ret = clk_set_rate(priv->clk_pll2, 24576000); + if (ret) { + dev_err(dev, "set codec base clock rate failed!\n"); + return ret; + } + if (clk_prepare_enable(priv->clk_pll2)) { + dev_err(dev, "try to enable clk_pll2 failed\n"); + return -EINVAL; + } + if (clk_prepare_enable(priv->clk_apb)) { + dev_err(dev, "try to enable clk_apb failed\n"); + return -EINVAL; + } + + priv->playback_dma_data.addr = res->start + SUNXI_DAC_TXDATA; + priv->playback_dma_data.maxburst = 4; + priv->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + priv->capture_dma_data.addr = res->start + SUNXI_ADC_RXDATA; + priv->capture_dma_data.maxburst = 4; + priv->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + ret = snd_soc_register_codec(&pdev->dev, &dummy_codec, &dummy_dai, 1); + + ret = devm_snd_soc_register_component(&pdev->dev, &sunxi_codec_component, &sunxi_codec_dai, 1); + if (ret) + goto err_clk_disable; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_clk_disable; + + ret = snd_soc_of_parse_audio_routing(card, "routing"); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: +err: +err_clk_disable: + clk_disable_unprepare(priv->clk_module); + clk_disable_unprepare(priv->clk_apb); + clk_disable_unprepare(priv->clk_pll2); + return ret; +} + +static int sunxi_codec_dev_remove(struct platform_device *pdev) +{ + struct sunxi_priv *priv = platform_get_drvdata(pdev); + + clk_disable_unprepare(priv->clk_module); + clk_disable_unprepare(priv->clk_apb); + clk_disable_unprepare(priv->clk_pll2); + + return 0; +} + +static struct platform_driver sunxi_codec_driver = { + .driver = { + .name = "sunxi-codec", + .owner = THIS_MODULE, + .of_match_table = sunxi_codec_of_match, + }, + .probe = sunxi_codec_probe, + .remove = sunxi_codec_dev_remove, +}; +module_platform_driver(sunxi_codec_driver); + +MODULE_ALIAS("platform:sunxi-codec"); +MODULE_DESCRIPTION("sunxi CODEC ALSA codec driver"); +MODULE_AUTHOR("software"); +MODULE_LICENSE("GPL v2"); +

3 2

[alsa-devel] [PATCH 0/4] ALSA/dmaengine: Fix 3 bytes physical sample support
by Peter Ujfalusi 02 Jul '14

02 Jul '14

Hi, I have noticed while switching the daVinci audio to dmaengine_pcm that the _3LE and _3BE support is not working correctly via dmaengine_pcm: In case of _3LE/_3BE formats the samples are stored in 3 consecutive bytes without padding it to 4 bytes. This means that the DMA needs to be able to support 3 bytes word length in order to read/write the samples from memory correctly. Originally the code treated 24 bits physical length samples as they were 32 bits which leads to corruption when playing or recording audio. To fix the support for 3 bytes physical samples the dma driver also need to have support for such data arrangement. eDMA does have support for it in HW and the legacy davinci-pcm platform driver w/o dmaengine was able to support such formats. First step is to add DMA_SLAVE_BUSWIDTH_3_BYTES to dma_slave_buswidth for engines and users to select 3 bytes as bus width. Followed by: In snd_dmaengine_pcm_open() we should check the slave_caps of the dma if it supports DMA_SLAVE_BUSWIDTH_3_BYTES. Based on this information we initialize the runtime->hw.formats: if DMA_SLAVE_BUSWIDTH_3_BYTES is not supported or the slave_caps is not provided by the driver we mask out the 24 bits physical width sample formats so they will be not available for user space to pick. If the DMA_SLAVE_BUSWIDTH_3_BYTES is supported _3LE/_3BE formats will not be masked so later on they can be valid if both CPU and codec dai supports them. Regards, Peter --- Peter Ujfalusi (4): dma: Support for 3 bytes word size dma: edma: Declare DMA_SLAVE_BUSWIDTH_3_BYTES as supported buswidth ALSA: pcm_dmaengine: Use the available wrapper to get physical width ALSA: pcm_dmaengine: Correct support for 3 physical bytes samples drivers/dma/edma.c | 1 + include/linux/dmaengine.h | 1 + sound/core/pcm_dmaengine.c | 41 +++++++++++++++++++++++++++++++++++++++-- 3 files changed, 41 insertions(+), 2 deletions(-) -- 2.0.0

2 8

Re: [alsa-devel] [Alsa-user] Constant delay between starting a playback and a capture stream
by Max Schmidt 02 Jul '14

02 Jul '14

Hi, just for correction if someone will ever read my question again: of course it's microseconds and not milliseconds Sorry for the disturbance and regards

1 0

[alsa-devel] [PATCH 1/1] ice1712: Aligning definitions in header file
by Konstantinos Tsimpoukas 02 Jul '14

02 Jul '14

Aligning #defines in ice1712.h file. Signed-off-by: Konstantinos Tsimpoukas <kostaslinuxxx(a)gmail.com> --- sound/pci/ice1712/ice1712.h | 84 ++++++++++++++++++++++----------------------- 1 file changed, 42 insertions(+), 42 deletions(-) diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index b209fc3..f88800a 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -41,18 +41,18 @@ #define ICEREG(ice, x) ((ice)->port + ICE1712_REG_##x) #define ICE1712_REG_CONTROL 0x00 /* byte */ -#define ICE1712_RESET 0x80 /* reset whole chip */ -#define ICE1712_SERR_LEVEL 0x04 /* SERR# level otherwise edge */ -#define ICE1712_NATIVE 0x01 /* native mode otherwise SB */ +#define ICE1712_RESET 0x80 /* reset whole chip */ +#define ICE1712_SERR_LEVEL 0x04 /* SERR# level otherwise edge */ +#define ICE1712_NATIVE 0x01 /* native mode otherwise SB */ #define ICE1712_REG_IRQMASK 0x01 /* byte */ -#define ICE1712_IRQ_MPU1 0x80 -#define ICE1712_IRQ_TIMER 0x40 -#define ICE1712_IRQ_MPU2 0x20 -#define ICE1712_IRQ_PROPCM 0x10 -#define ICE1712_IRQ_FM 0x08 /* FM/MIDI - legacy */ -#define ICE1712_IRQ_PBKDS 0x04 /* playback DS channels */ -#define ICE1712_IRQ_CONCAP 0x02 /* consumer capture */ -#define ICE1712_IRQ_CONPBK 0x01 /* consumer playback */ +#define ICE1712_IRQ_MPU1 0x80 +#define ICE1712_IRQ_TIMER 0x40 +#define ICE1712_IRQ_MPU2 0x20 +#define ICE1712_IRQ_PROPCM 0x10 +#define ICE1712_IRQ_FM 0x08 /* FM/MIDI - legacy */ +#define ICE1712_IRQ_PBKDS 0x04 /* playback DS channels */ +#define ICE1712_IRQ_CONCAP 0x02 /* consumer capture */ +#define ICE1712_IRQ_CONPBK 0x01 /* consumer playback */ #define ICE1712_REG_IRQSTAT 0x02 /* byte */ /* look to ICE1712_IRQ_* */ #define ICE1712_REG_INDEX 0x03 /* byte - indirect CCIxx regs */ @@ -62,23 +62,23 @@ #define ICE1712_REG_NMI_INDEX 0x07 /* byte */ #define ICE1712_REG_AC97_INDEX 0x08 /* byte */ #define ICE1712_REG_AC97_CMD 0x09 /* byte */ -#define ICE1712_AC97_COLD 0x80 /* cold reset */ -#define ICE1712_AC97_WARM 0x40 /* warm reset */ -#define ICE1712_AC97_WRITE 0x20 /* W: write, R: write in progress */ -#define ICE1712_AC97_READ 0x10 /* W: read, R: read in progress */ -#define ICE1712_AC97_READY 0x08 /* codec ready status bit */ -#define ICE1712_AC97_PBK_VSR 0x02 /* playback VSR */ -#define ICE1712_AC97_CAP_VSR 0x01 /* capture VSR */ +#define ICE1712_AC97_COLD 0x80 /* cold reset */ +#define ICE1712_AC97_WARM 0x40 /* warm reset */ +#define ICE1712_AC97_WRITE 0x20 /* W: write, R: write in progress */ +#define ICE1712_AC97_READ 0x10 /* W: read, R: read in progress */ +#define ICE1712_AC97_READY 0x08 /* codec ready status bit */ +#define ICE1712_AC97_PBK_VSR 0x02 /* playback VSR */ +#define ICE1712_AC97_CAP_VSR 0x01 /* capture VSR */ #define ICE1712_REG_AC97_DATA 0x0a /* word (little endian) */ #define ICE1712_REG_MPU1_CTRL 0x0c /* byte */ #define ICE1712_REG_MPU1_DATA 0x0d /* byte */ #define ICE1712_REG_I2C_DEV_ADDR 0x10 /* byte */ -#define ICE1712_I2C_WRITE 0x01 /* write direction */ +#define ICE1712_I2C_WRITE 0x01 /* write direction */ #define ICE1712_REG_I2C_BYTE_ADDR 0x11 /* byte */ #define ICE1712_REG_I2C_DATA 0x12 /* byte */ #define ICE1712_REG_I2C_CTRL 0x13 /* byte */ -#define ICE1712_I2C_EEPROM 0x80 /* EEPROM exists */ -#define ICE1712_I2C_BUSY 0x01 /* busy bit */ +#define ICE1712_I2C_EEPROM 0x80 /* EEPROM exists */ +#define ICE1712_I2C_BUSY 0x01 /* busy bit */ #define ICE1712_REG_CONCAP_ADDR 0x14 /* dword - consumer capture */ #define ICE1712_REG_CONCAP_COUNT 0x18 /* word - current/base count */ #define ICE1712_REG_SERR_SHADOW 0x1b /* byte */ @@ -128,14 +128,14 @@ #define ICE1712_DSC_ADDR1 0x02 /* dword - base address 1 */ #define ICE1712_DSC_COUNT1 0x03 /* word - count 1 */ #define ICE1712_DSC_CONTROL 0x04 /* byte - control & status */ -#define ICE1712_BUFFER1 0x80 /* buffer1 is active */ -#define ICE1712_BUFFER1_AUTO 0x40 /* buffer1 auto init */ -#define ICE1712_BUFFER0_AUTO 0x20 /* buffer0 auto init */ -#define ICE1712_FLUSH 0x10 /* flush FIFO */ -#define ICE1712_STEREO 0x08 /* stereo */ -#define ICE1712_16BIT 0x04 /* 16-bit data */ -#define ICE1712_PAUSE 0x02 /* pause */ -#define ICE1712_START 0x01 /* start */ +#define ICE1712_BUFFER1 0x80 /* buffer1 is active */ +#define ICE1712_BUFFER1_AUTO 0x40 /* buffer1 auto init */ +#define ICE1712_BUFFER0_AUTO 0x20 /* buffer0 auto init */ +#define ICE1712_FLUSH 0x10 /* flush FIFO */ +#define ICE1712_STEREO 0x08 /* stereo */ +#define ICE1712_16BIT 0x04 /* 16-bit data */ +#define ICE1712_PAUSE 0x02 /* pause */ +#define ICE1712_START 0x01 /* start */ #define ICE1712_DSC_RATE 0x05 /* dword - rate */ #define ICE1712_DSC_VOLUME 0x06 /* word - volume control */ @@ -146,12 +146,12 @@ #define ICEMT(ice, x) ((ice)->profi_port + ICE1712_MT_##x) #define ICE1712_MT_IRQ 0x00 /* byte - interrupt mask */ -#define ICE1712_MULTI_CAPTURE 0x80 /* capture IRQ */ -#define ICE1712_MULTI_PLAYBACK 0x40 /* playback IRQ */ -#define ICE1712_MULTI_CAPSTATUS 0x02 /* capture IRQ status */ -#define ICE1712_MULTI_PBKSTATUS 0x01 /* playback IRQ status */ +#define ICE1712_MULTI_CAPTURE 0x80 /* capture IRQ */ +#define ICE1712_MULTI_PLAYBACK 0x40 /* playback IRQ */ +#define ICE1712_MULTI_CAPSTATUS 0x02 /* capture IRQ status */ +#define ICE1712_MULTI_PBKSTATUS 0x01 /* playback IRQ status */ #define ICE1712_MT_RATE 0x01 /* byte - sampling rate select */ -#define ICE1712_SPDIF_MASTER 0x10 /* S/PDIF input is master clock */ +#define ICE1712_SPDIF_MASTER 0x10 /* S/PDIF input is master clock */ #define ICE1712_MT_I2S_FORMAT 0x02 /* byte - I2S data format */ #define ICE1712_MT_AC97_INDEX 0x04 /* byte - AC'97 index */ #define ICE1712_MT_AC97_CMD 0x05 /* byte - AC'97 command & status */ @@ -161,14 +161,14 @@ #define ICE1712_MT_PLAYBACK_SIZE 0x14 /* word - playback size */ #define ICE1712_MT_PLAYBACK_COUNT 0x16 /* word - playback count */ #define ICE1712_MT_PLAYBACK_CONTROL 0x18 /* byte - control */ -#define ICE1712_CAPTURE_START_SHADOW 0x04 /* capture start */ -#define ICE1712_PLAYBACK_PAUSE 0x02 /* playback pause */ -#define ICE1712_PLAYBACK_START 0x01 /* playback start */ +#define ICE1712_CAPTURE_START_SHADOW 0x04 /* capture start */ +#define ICE1712_PLAYBACK_PAUSE 0x02 /* playback pause */ +#define ICE1712_PLAYBACK_START 0x01 /* playback start */ #define ICE1712_MT_CAPTURE_ADDR 0x20 /* dword - capture address */ #define ICE1712_MT_CAPTURE_SIZE 0x24 /* word - capture size */ #define ICE1712_MT_CAPTURE_COUNT 0x26 /* word - capture count */ #define ICE1712_MT_CAPTURE_CONTROL 0x28 /* byte - control */ -#define ICE1712_CAPTURE_START 0x01 /* capture start */ +#define ICE1712_CAPTURE_START 0x01 /* capture start */ #define ICE1712_MT_ROUTE_PSDOUT03 0x30 /* word */ #define ICE1712_MT_ROUTE_SPDOUT 0x32 /* word */ #define ICE1712_MT_ROUTE_CAPTURE 0x34 /* dword */ @@ -176,7 +176,7 @@ #define ICE1712_MT_MONITOR_INDEX 0x3a /* byte */ #define ICE1712_MT_MONITOR_RATE 0x3b /* byte */ #define ICE1712_MT_MONITOR_ROUTECTRL 0x3c /* byte */ -#define ICE1712_ROUTE_AC97 0x01 /* route digital mixer output to AC'97 */ +#define ICE1712_ROUTE_AC97 0x01 /* route digital mixer output to AC'97 */ #define ICE1712_MT_MONITOR_PEAKINDEX 0x3e /* byte */ #define ICE1712_MT_MONITOR_PEAKDATA 0x3f /* byte */ @@ -186,9 +186,9 @@ /* PCI[60] System Configuration */ #define ICE1712_CFG_CLOCK 0xc0 -#define ICE1712_CFG_CLOCK512 0x00 /* 22.5692Mhz, 44.1kHz*512 */ -#define ICE1712_CFG_CLOCK384 0x40 /* 16.9344Mhz, 44.1kHz*384 */ -#define ICE1712_CFG_EXT 0x80 /* external clock */ +#define ICE1712_CFG_CLOCK512 0x00 /* 22.5692Mhz, 44.1kHz*512 */ +#define ICE1712_CFG_CLOCK384 0x40 /* 16.9344Mhz, 44.1kHz*384 */ +#define ICE1712_CFG_EXT 0x80 /* external clock */ #define ICE1712_CFG_2xMPU401 0x20 /* two MPU401 UARTs */ #define ICE1712_CFG_NO_CON_AC97 0x10 /* consumer AC'97 codec is not present */ #define ICE1712_CFG_ADC_MASK 0x0c /* one, two, three, four stereo ADCs */ -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: add Wolfson WM8973 codec driver
by Xavier Hsu 02 Jul '14

02 Jul '14

This patch adds support for the wm8973 codec, based on the existing wm8971 codec driver. Any comments about improving the patch are welcome. Thanks. Signed-off-by: Xavier Hsu <xavier.hsu(a)linaro.org> --- Documentation/devicetree/bindings/sound/wm8973.txt | 26 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8973.c | 1028 ++++++++++++++++++++ sound/soc/codecs/wm8973.h | 57 ++ 5 files changed, 1117 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8973.txt create mode 100644 sound/soc/codecs/wm8973.c create mode 100644 sound/soc/codecs/wm8973.h diff --git a/Documentation/devicetree/bindings/sound/wm8973.txt b/Documentation/devicetree/bindings/sound/wm8973.txt new file mode 100644 index 0000000..2374873 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8973.txt @@ -0,0 +1,26 @@ +WM8973 audio CODEC + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8973". + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Optional properties: + + - mclk-div : Setting the CLKDIV2 bit for dividing MCLK. + mclk-div = <0> (Default & not divide). + mclk-div = <1> (Divide by 2). + +Example: + +codec: wm8973@1a { + compatible = "wlf,wm8973"; + reg = <0x1a>; + + mclk-div = <1>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 350878e..acf83bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -139,6 +139,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8961 if I2C select SND_SOC_WM8962 if I2C && INPUT select SND_SOC_WM8971 if I2C + select SND_SOC_WM8973 if I2C select SND_SOC_WM8974 if I2C select SND_SOC_WM8978 if I2C select SND_SOC_WM8983 if SND_SOC_I2C_AND_SPI @@ -692,6 +693,9 @@ config SND_SOC_WM8962 config SND_SOC_WM8971 tristate +config SND_SOC_WM8973 + tristate + config SND_SOC_WM8974 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1c..aad29a3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -141,6 +141,7 @@ snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o snd-soc-wm8962-objs := wm8962.o snd-soc-wm8971-objs := wm8971.o +snd-soc-wm8973-objs := wm8973.o snd-soc-wm8974-objs := wm8974.o snd-soc-wm8978-objs := wm8978.o snd-soc-wm8983-objs := wm8983.o @@ -303,6 +304,7 @@ obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8962) += snd-soc-wm8962.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8973) += snd-soc-wm8973.o obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8983) += snd-soc-wm8983.o diff --git a/sound/soc/codecs/wm8973.c b/sound/soc/codecs/wm8973.c new file mode 100644 index 0000000..fc03de3 --- /dev/null +++ b/sound/soc/codecs/wm8973.c @@ -0,0 +1,1028 @@ +/* + * wm8973.c -- WM8973 ALSA SoC Audio driver + * + * Copyright (C) 2013 -2014 Fujitsu Semiconductor, Ltd + * Copyright (C) 2014 Linaro, Ltd Xavier Hsu <xavier.hsu(a)linaro.org> + * + * Based on wm8971 driver Copyright 2005 Lab126, Inc. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8973.h" + +static int mclk_div; + +struct wm8973_priv { + struct regmap *regmap; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + int playback_fs; + bool deemph; +}; + +/* + * wm8973 register cache + * We can't read the WM8973 register space when we + * are using 2 wire for device control, so we cache them instead. + */ +static const struct reg_default wm8973_reg_defaults[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x0079 }, + { 3, 0x0079 }, + { 4, 0x0000 }, + { 5, 0x0008 }, + { 6, 0x0000 }, + { 7, 0x000a }, + { 8, 0x0000 }, + { 9, 0x0000 }, + { 10, 0x00ff }, + { 11, 0x00ff }, + { 12, 0x000f }, + { 13, 0x000f }, + { 14, 0x0000 }, + { 15, 0x0000 }, + { 16, 0x0000 }, + { 17, 0x007b }, + { 18, 0x0000 }, + { 19, 0x0032 }, + { 20, 0x0000 }, + { 21, 0x01c3 }, + { 22, 0x01c3 }, + { 23, 0x0040 }, + { 24, 0x0014 }, + { 25, 0x01c2 }, + { 26, 0x0060 }, + { 27, 0x0000 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0050 }, + { 35, 0x0050 }, + { 36, 0x0050 }, + { 37, 0x0050 }, + { 38, 0x0050 }, + { 39, 0x0050 }, + { 40, 0x0079 }, + { 41, 0x0079 }, + { 42, 0x0079 }, +}; + +static const char const *wm8973_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char const *wm8973_bass_filter[] = { "130Hz @ 48kHz", + "200Hz @ 48kHz" }; +static const char const *wm8973_treble[] = {"8kHz", "4kHz"}; +static const char const *wm8973_3d_lc[] = {"200Hz", "500Hz"}; +static const char const *wm8973_3d_uc[] = {"2.2kHz", "1.5kHz"}; +static const char const *wm8973_3d_func[] = {"Capture", "Playback"}; +static const char const *wm8973_alc_func[] = {"Off", "Right", "Left", + "Stereo"}; +static const char const *wm8973_ng_type[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const char const *wm8973_line_mux[] = {"Line 1", "Line 2", "Line 3", + "PGA", "Differential"}; +static const char const *wm8973_pga_sel[] = {"Line 1", "Line 2", "Line 3", + "Differential"}; +static const char const *wm8973_out3[] = {"VREF", "ROUT1 + Vol", "MonoOut", + "ROUT1"}; +static const char const *wm8973_diff_sel[] = {"Line 1", "Line 2"}; +static const char const *wm8973_adcpol[] = {"Normal", "L Invert", "R Invert", + "L + R Invert"}; +static const char const *wm8973_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; + +static const SOC_ENUM_SINGLE_DECL(bass_boost, WM8973_BASS, 7, wm8973_bass); +static const SOC_ENUM_SINGLE_DECL(bass_filter, WM8973_BASS, + 6, wm8973_bass_filter); +static const SOC_ENUM_SINGLE_DECL(treble_cutoff, WM8973_TREBLE, + 6, wm8973_treble); +static const SOC_ENUM_SINGLE_DECL(lower_cutoff, WM8973_3D, 5, wm8973_3d_lc); +static const SOC_ENUM_SINGLE_DECL(upper_cutoff, WM8973_3D, 6, wm8973_3d_uc); +static const SOC_ENUM_SINGLE_DECL(mode, WM8973_3D, 7, wm8973_3d_func); +static const SOC_ENUM_SINGLE_DECL(alc_capture_func, WM8973_ALC1, + 7, wm8973_alc_func); +static const SOC_ENUM_SINGLE_DECL(alc_capture_ngtype, WM8973_NGATE, + 1, wm8973_ng_type); +static const SOC_ENUM_SINGLE_DECL(left_line, WM8973_LOUTM1, + 0, wm8973_line_mux); +static const SOC_ENUM_SINGLE_DECL(right_line, WM8973_ROUTM1, + 0, wm8973_line_mux); +static const SOC_ENUM_SINGLE_DECL(left_pga, WM8973_LADCIN, 6, wm8973_pga_sel); +static const SOC_ENUM_SINGLE_DECL(right_pga, WM8973_RADCIN, 6, wm8973_pga_sel); +static const SOC_ENUM_SINGLE_DECL(out3, WM8973_ADCTL2, 7, wm8973_out3); +static const SOC_ENUM_SINGLE_DECL(diffmux, WM8973_ADCIN, 8, wm8973_diff_sel); +static const SOC_ENUM_SINGLE_DECL(capture_polarity, WM8973_ADCDAC, + 5, wm8973_adcpol); +static const SOC_ENUM_SINGLE_DECL(monomux, WM8973_ADCIN, 6, wm8973_mono_mux); + +static int wm8973_deemph[] = { 0, 32000, 44100, 48000 }; + +static int wm8973_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8973_priv *wm8973 = snd_soc_codec_get_drvdata(codec); + int val = 0, i, best = 0; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8973->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(wm8973_deemph); i++) { + if (abs(wm8973_deemph[i] - wm8973->playback_fs) < + abs(wm8973_deemph[best] - wm8973->playback_fs)) + best = i; + } + val = best << 1; + } + + dev_dbg(codec->dev, "Set deemphasis %d (%dHz)\n", + best, wm8973_deemph[best]); + + return snd_soc_update_bits(codec, WM8973_ADCDAC, 0x6, val); +} + +static int wm8973_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct wm8973_priv *wm8973 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = wm8973->deemph; + + return 0; +} + +static int wm8973_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct wm8973_priv *wm8973 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + int ret = 0; + + if (deemph > 1) + return -EINVAL; + + mutex_lock(&codec->mutex); + if (wm8973->deemph != deemph) { + wm8973->deemph = deemph; + wm8973_set_deemph(codec); + + ret = 1; + } + mutex_unlock(&codec->mutex); + + return ret; +} + +static const DECLARE_TLV_DB_SCALE(in_vol, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(out_vol, -6700, 91, 0); +static const DECLARE_TLV_DB_SCALE(attenuate_6db, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol, -12700, 50, 0); +static const DECLARE_TLV_DB_SCALE(tone_vol, -600, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_tar_vol, -2850, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_max_vol, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol, -9700, 50, 0); +static const DECLARE_TLV_DB_SCALE(bypass_out_vol, -1500, 300, 0); + +static const struct snd_kcontrol_new wm8973_snd_controls[] = { + /* Left & Right Input volume */ + SOC_DOUBLE_R_TLV("Capture Volume", WM8973_LINVOL, WM8973_RINVOL, + 0, 63, 0, in_vol), + SOC_DOUBLE_R("Capture ZC Switch", WM8973_LINVOL, WM8973_RINVOL, + 6, 1, 0), + SOC_DOUBLE_R("Capture Switch", WM8973_LINVOL, WM8973_RINVOL, 7, 1, 1), + + /* LOUT1 & ROUT1 volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8973_LOUT1V, + WM8973_ROUT1V, 0, 127, 0, out_vol), + SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8973_LOUT1V, + WM8973_ROUT1V, 7, 1, 0), + + /* ADC & DAC control */ + SOC_SINGLE("Capture Filter Switch", WM8973_ADCDAC, 0, 1, 1), + SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0, + wm8973_get_deemph, wm8973_put_deemph), + SOC_ENUM("Capture Polarity", capture_polarity), + SOC_SINGLE_TLV("Playback 6dB Attenuate", WM8973_ADCDAC, + 7, 1, 0, attenuate_6db), + SOC_SINGLE_TLV("Capture 6dB Attenuate", WM8973_ADCDAC, + 8, 1, 0, attenuate_6db), + /* ADCDAC Bit 4 - HPOR */ + + /* Left & Right Channel Digital Volume */ + SOC_DOUBLE_R_TLV("DAC Volume", WM8973_LDAC, WM8973_RDAC, + 0, 255, 0, dac_vol), + + /* Bass Control */ + SOC_SINGLE_TLV("Bass Volume", WM8973_BASS, 0, 15, 1, tone_vol), + SOC_ENUM("Bass Boost", bass_boost), + SOC_ENUM("Bass Filter", bass_filter), + + /* Treble Control */ + SOC_SINGLE_TLV("Treble Volume", WM8973_TREBLE, 0, 15, 0, tone_vol), + SOC_ENUM("Treble Cut-off", treble_cutoff), + + /* 3D Control */ + SOC_SINGLE("3D Switch", WM8973_3D, 0, 1, 0), + SOC_SINGLE("3D Volume", WM8973_3D, 1, 15, 0), + SOC_ENUM("3D Lower Cut-off", lower_cutoff), + SOC_ENUM("3D Upper Cut-off", upper_cutoff), + SOC_ENUM("3D Mode", mode), + + /* ALC1 & ALC2 & ALC3 Control */ + SOC_SINGLE_TLV("ALC Capture Target Volume", WM8973_ALC1, + 0, 15, 0, alc_tar_vol), + SOC_SINGLE_TLV("ALC Capture Max Volume", WM8973_ALC1, + 4, 7, 0, alc_max_vol), + SOC_ENUM("ALC Capture Function", alc_capture_func), + + SOC_SINGLE("ALC Capture Hold Time", WM8973_ALC2, 0, 15, 0), + SOC_SINGLE("ALC Capture ZC Switch", WM8973_ALC2, 7, 1, 0), + + SOC_SINGLE("ALC Capture Attack Time", WM8973_ALC3, 0, 15, 0), + SOC_SINGLE("ALC Capture Decay Time", WM8973_ALC3, 4, 15, 0), + + /* Noise Gate Control */ + SOC_SINGLE("ALC Capture NG Switch", WM8973_NGATE, 0, 1, 0), + SOC_ENUM("ALC Capture NG Type", alc_capture_ngtype), + SOC_SINGLE("ALC Capture NG Threshold", WM8973_NGATE, 3, 31, 0), + + /* Left & Right ADC Digital Volume*/ + SOC_DOUBLE_R_TLV("ADC Volume", WM8973_LADC, WM8973_RADC, + 0, 255, 0, adc_vol), + + /* Additional Control 1 */ + SOC_SINGLE("ZC Timeout Switch", WM8973_ADCTL1, 0, 1, 0), + SOC_SINGLE("Playback Invert Switch", WM8973_ADCTL1, 1, 1, 0), + SOC_SINGLE("Analogue Bias", WM8973_ADCTL1, 6, 3, 0), + /* ADCTL1 Bit 2,3 - DATSEL */ + /* ADCTL1 Bit 6,7 - VSEL */ + + /* Additional Control 2 */ + SOC_SINGLE("Right Speaker Playback Invert Switch", WM8973_ADCTL2, + 4, 1, 0), + /* ADCTL2 Bit 2 - LRCM */ + SOC_SINGLE("LRCLK Switch", WM8973_ADCTL2, 2, 1, 0), + /* ADCTL2 Bit 3 - TRI */ + SOC_SINGLE("Headphone Switch POL", WM8973_ADCTL2, 5, 1, 0), + SOC_SINGLE("Headphone Switch EN", WM8973_ADCTL2, 6, 1, 0), + + /* Additional Control 3 */ + /* ADCTL3 Bit 5 - HPFLREN */ + /* ADCTL3 Bit 6 - VROI */ + /* ADCTL3 Bit 7,8 - ADCLRM */ + + /* ADC input Mode */ + /* ADCIN Bit 4 - LDCM */ + /* ADCIN Bit 5 - RDCM */ + /* ADCIN Bit 6,7 - MONOMIX */ + + /* Left & Right ADC Signal Path Control*/ + SOC_DOUBLE_R("Mic Boost", WM8973_LADCIN, WM8973_RADCIN, 4, 3, 0), + + /* Left OUT Mixer Control */ + SOC_DOUBLE_R_TLV("Bypass Left Playback Volume", WM8973_LOUTM1, + WM8973_LOUTM2, 4, 7, 1, bypass_out_vol), + + /* Right OUT Mixer Control */ + SOC_DOUBLE_R_TLV("Bypass Right Playback Volume", WM8973_ROUTM1, + WM8973_ROUTM2, 4, 7, 1, bypass_out_vol), + + /* Mono OUT Mixer Control */ + SOC_DOUBLE_R_TLV("Bypass Mono Playback Volume", WM8973_MOUTM1, + WM8973_MOUTM2, 4, 7, 1, bypass_out_vol), + + /* LOUT2 & ROUT2 volume */ + SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8973_LOUT2V, + WM8973_ROUT2V, 0, 127, 0, out_vol), + SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8973_LOUT2V, + WM8973_ROUT2V, 7, 1, 0), + + /* MONOOUT volume */ + SOC_SINGLE_TLV("Mono Playback Volume", WM8973_MOUTV, + 0, 127, 0, out_vol), + SOC_SINGLE("Mono Playback ZC Switch", WM8973_MOUTV, 7, 1, 0), + + SOC_SINGLE("Right Out 2", WM8973_PWR2, 3, 1, 0), + SOC_SINGLE("Left Out 2", WM8973_PWR2, 4, 1, 0), +}; + +/* + * DAPM Controls + */ + +/* Left Mixer */ +static const struct snd_kcontrol_new wm8973_left_mixer_controls[] = { +SOC_DAPM_SINGLE("Playback Switch", WM8973_LOUTM1, 8, 1, 0), +SOC_DAPM_SINGLE("Left Bypass Switch", WM8973_LOUTM1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Playback Switch", WM8973_LOUTM2, 8, 1, 0), +SOC_DAPM_SINGLE("Right Bypass Switch", WM8973_LOUTM2, 7, 1, 0), +}; + +/* Right Mixer */ +static const struct snd_kcontrol_new wm8973_right_mixer_controls[] = { +SOC_DAPM_SINGLE("Left Playback Switch", WM8973_ROUTM1, 8, 1, 0), +SOC_DAPM_SINGLE("Left Bypass Switch", WM8973_ROUTM1, 7, 1, 0), +SOC_DAPM_SINGLE("Playback Switch", WM8973_ROUTM2, 8, 1, 0), +SOC_DAPM_SINGLE("Right Bypass Switch", WM8973_ROUTM2, 7, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new wm8973_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("Left Playback Switch", WM8973_MOUTM1, 8, 1, 0), +SOC_DAPM_SINGLE("Left Bypass Switch", WM8973_MOUTM1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Playback Switch", WM8973_MOUTM2, 8, 1, 0), +SOC_DAPM_SINGLE("Right Bypass Switch", WM8973_MOUTM2, 7, 1, 0), +}; + +/* Left Line Mux */ +static const struct snd_kcontrol_new wm8973_left_line_controls = +SOC_DAPM_ENUM("Route", left_line); + +/* Right Line Mux */ +static const struct snd_kcontrol_new wm8973_right_line_controls = +SOC_DAPM_ENUM("Route", right_line); + +/* Left PGA Mux */ +static const struct snd_kcontrol_new wm8973_left_pga_controls = +SOC_DAPM_ENUM("Route", left_pga); + +/* Right PGA Mux */ +static const struct snd_kcontrol_new wm8973_right_pga_controls = +SOC_DAPM_ENUM("Route", right_pga); + +/* Out 3 Mux */ +static const struct snd_kcontrol_new wm8973_out3_controls = +SOC_DAPM_ENUM("Route", out3); + +/* Differential Mux */ +static const struct snd_kcontrol_new wm8973_diffmux_controls = +SOC_DAPM_ENUM("Route", diffmux); + +/* Mono ADC Mux */ +static const struct snd_kcontrol_new wm8973_monomux_controls = +SOC_DAPM_ENUM("Route", monomux); + +static const struct snd_soc_dapm_widget wm8973_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + &wm8973_left_mixer_controls[0], + ARRAY_SIZE(wm8973_left_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + &wm8973_right_mixer_controls[0], + ARRAY_SIZE(wm8973_right_mixer_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", WM8973_PWR2, 2, 0, + &wm8973_mono_mixer_controls[0], + ARRAY_SIZE(wm8973_mono_mixer_controls)), + + SND_SOC_DAPM_PGA("Right Out 2", WM8973_PWR2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 2", WM8973_PWR2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Out 1", WM8973_PWR2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 1", WM8973_PWR2, 6, 0, NULL, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8973_PWR2, 7, 0), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8973_PWR2, 8, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8973_PWR1, 1, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8973_PWR1, 2, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8973_PWR1, 3, 0), + SND_SOC_DAPM_MUX("Right PGA Mux", WM8973_PWR1, 4, 0, + &wm8973_right_pga_controls), + SND_SOC_DAPM_MUX("Left PGA Mux", WM8973_PWR1, 5, 0, + &wm8973_left_pga_controls), + + SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, + &wm8973_left_line_controls), + SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, + &wm8973_right_line_controls), + + SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, + &wm8973_out3_controls), + SND_SOC_DAPM_PGA("Out 3", WM8973_PWR2, 1, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Mono Out 1", WM8973_PWR2, 2, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &wm8973_diffmux_controls), + + SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8973_monomux_controls), + SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8973_monomux_controls), + + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("MONO1"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("VREF"), + + SND_SOC_DAPM_INPUT("LINPUT1"), + SND_SOC_DAPM_INPUT("LINPUT2"), + SND_SOC_DAPM_INPUT("LINPUT3"), + SND_SOC_DAPM_INPUT("RINPUT1"), + SND_SOC_DAPM_INPUT("RINPUT2"), + SND_SOC_DAPM_INPUT("RINPUT3"), +}; + +static const struct snd_soc_dapm_route wm8973_dapm_routes[] = { + {"Left Mixer", "Playback Switch", "Left DAC"}, + {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, + {"Left Mixer", "Right Playback Switch", "Right DAC"}, + {"Left Mixer", "Right Bypass Switch", "Right Line Mux"}, + + {"Right Mixer", "Left Playback Switch", "Left DAC"}, + {"Right Mixer", "Left Bypass Switch", "Left Line Mux"}, + {"Right Mixer", "Playback Switch", "Right DAC"}, + {"Right Mixer", "Right Bypass Switch", "Right Line Mux"}, + + {"Left Out 1", NULL, "Left Mixer"}, + {"LOUT1", NULL, "Left Out 1"}, + + {"Left Out 2", NULL, "Left Mixer"}, + {"LOUT2", NULL, "Left Out 2"}, + + {"Right Out 1", NULL, "Right Mixer"}, + {"ROUT1", NULL, "Right Out 1"}, + + {"Right Out 2", NULL, "Right Mixer"}, + {"ROUT2", NULL, "Right Out 2"}, + + {"Mono Mixer", "Left Playback Switch", "Left DAC"}, + {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"}, + {"Mono Mixer", "Right Playback Switch", "Right DAC"}, + {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"}, + + {"Mono Out 1", NULL, "Mono Mixer"}, + {"MONO1", NULL, "Mono Out 1"}, + + {"Out3 Mux", "VREF", "VREF"}, + {"Out3 Mux", "ROUT1 + Vol", "ROUT1"}, + {"Out3 Mux", "ROUT1", "Right Mixer"}, + {"Out3 Mux", "MonoOut", "MONO1"}, + {"Out 3", NULL, "Out3 Mux"}, + {"OUT3", NULL, "Out 3"}, + + {"Left Line Mux", "Line 1", "LINPUT1"}, + {"Left Line Mux", "Line 2", "LINPUT2"}, + {"Left Line Mux", "Line 3", "LINPUT3"}, + {"Left Line Mux", "PGA", "Left PGA Mux"}, + {"Left Line Mux", "Differential", "Differential Mux"}, + + {"Right Line Mux", "Line 1", "RINPUT1"}, + {"Right Line Mux", "Line 2", "RINPUT2"}, + {"Right Line Mux", "Line 3", "RINPUT3"}, + /* {"Right Line Mux", "Mic", "MIC"}, */ + {"Right Line Mux", "PGA", "Right PGA Mux"}, + {"Right Line Mux", "Differential", "Differential Mux"}, + + {"Left PGA Mux", "Line 1", "LINPUT1"}, + {"Left PGA Mux", "Line 2", "LINPUT2"}, + {"Left PGA Mux", "Line 3", "LINPUT3"}, + {"Left PGA Mux", "Differential", "Differential Mux"}, + + {"Right PGA Mux", "Line 1", "RINPUT1"}, + {"Right PGA Mux", "Line 2", "RINPUT2"}, + {"Right PGA Mux", "Line 3", "RINPUT3"}, + {"Right PGA Mux", "Differential", "Differential Mux"}, + + {"Differential Mux", "Line 1", "LINPUT1"}, + {"Differential Mux", "Line 1", "RINPUT1"}, + {"Differential Mux", "Line 2", "LINPUT2"}, + {"Differential Mux", "Line 2", "RINPUT2"}, + + {"Left ADC Mux", "Stereo", "Left PGA Mux"}, + {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"}, + {"Left ADC Mux", "Digital Mono", "Left PGA Mux"}, + + {"Right ADC Mux", "Stereo", "Right PGA Mux"}, + {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"}, + {"Right ADC Mux", "Digital Mono", "Right PGA Mux"}, + + {"Left ADC", NULL, "Left ADC Mux"}, + {"Right ADC", NULL, "Right ADC Mux"}, +}; + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:5; + u8 usb:1; +}; + +/* codec hifi mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 8k */ + {12288000, 8000, 1536, 0x6, 0x0}, + {11289600, 8000, 1408, 0x16, 0x0}, + {18432000, 8000, 2304, 0x7, 0x0}, + {16934400, 8000, 2112, 0x17, 0x0}, + {12000000, 8000, 1500, 0x6, 0x1}, + + /* 11.025k */ + {11289600, 11025, 1024, 0x18, 0x0}, + {16934400, 11025, 1536, 0x19, 0x0}, + {12000000, 11025, 1088, 0x19, 0x1}, + + /* 16k */ + {12288000, 16000, 768, 0xa, 0x0}, + {18432000, 16000, 1152, 0xb, 0x0}, + {12000000, 16000, 750, 0xa, 0x1}, + + /* 22.05k */ + {11289600, 22050, 512, 0x1a, 0x0}, + {16934400, 22050, 768, 0x1b, 0x0}, + {12000000, 22050, 544, 0x1b, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0xc, 0x0}, + {18432000, 32000, 576, 0xd, 0x0}, + {12000000, 32000, 375, 0xa, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x10, 0x0}, + {16934400, 44100, 384, 0x11, 0x0}, + {12000000, 44100, 272, 0x11, 0x1}, + + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0}, + {18432000, 48000, 384, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0x1e, 0x0}, + {16934400, 88200, 192, 0x1f, 0x0}, + {12000000, 88200, 136, 0x1f, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0xe, 0x0}, + {18432000, 96000, 192, 0xf, 0x0}, + {12000000, 96000, 125, 0xe, 0x1}, +}; + +static int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +/* The set of rates we can generate from the above for each SYSCLK */ +static unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static unsigned int rates_112896[] = { + 8000, 11025, 22050, 44100, 88200 +}; + +static struct snd_pcm_hw_constraint_list constraints_112896 = { + .count = ARRAY_SIZE(rates_112896), + .list = rates_112896, +}; + +static unsigned int rates_18432[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list constraints_18432 = { + .count = ARRAY_SIZE(rates_18432), + .list = rates_18432, +}; + +static unsigned int rates_169344[] = { + 8000, 11025, 22050, 44100, 88200 +}; + +static struct snd_pcm_hw_constraint_list constraints_169344 = { + .count = ARRAY_SIZE(rates_169344), + .list = rates_169344, +}; + +static unsigned int rates_12[] = { + 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 48000, 88235, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12 = { + .count = ARRAY_SIZE(rates_12), + .list = rates_12, +}; + +static int wm8973_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8973_priv *wm8973 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 12288000: + case 24576000: + wm8973->sysclk_constraints = &constraints_12288; + wm8973->sysclk = freq; + return 0; + + case 11289600: + case 22579200: + wm8973->sysclk_constraints = &constraints_112896; + wm8973->sysclk = freq; + return 0; + + case 18432000: + case 36864000: + wm8973->sysclk_constraints = &constraints_18432; + wm8973->sysclk = freq; + return 0; + + case 16934400: + case 33868800: + wm8973->sysclk_constraints = &constraints_169344; + wm8973->sysclk = freq; + return 0; + + case 12000000: + case 24000000: + wm8973->sysclk_constraints = &constraints_12; + wm8973->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int wm8973_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, WM8973_IFACE, 0x0040, 0x0040); + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, WM8973_IFACE, 0x0002, 0x0002); + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, WM8973_IFACE, 0x0001, 0x0001); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, WM8973_IFACE, 0x0003, 0x0003); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, WM8973_IFACE, 0x0013, 0x0013); + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + snd_soc_update_bits(codec, WM8973_IFACE, 0x0090, 0x0090); + break; + case SND_SOC_DAIFMT_IB_NF: + snd_soc_update_bits(codec, WM8973_IFACE, 0x0080, 0x0080); + break; + case SND_SOC_DAIFMT_NB_IF: + snd_soc_update_bits(codec, WM8973_IFACE, 0x0010, 0x0010); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8973_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8973_priv *wm8973 = snd_soc_codec_get_drvdata(codec); + u16 iface = snd_soc_read(codec, WM8973_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8973_SRATE) & 0x1c0; + int coeff = get_coeff(wm8973->sysclk, params_rate(params)); + + wm8973->playback_fs = params_rate(params); + + /* bit size */ + switch (params_width(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + + wm8973_set_deemph(codec); + + /* set iface & srate */ + snd_soc_write(codec, WM8973_IFACE, iface); + if (coeff >= 0) { + snd_soc_write(codec, WM8973_SRATE, srate | + (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); + } + + return 0; +} + +static int wm8973_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, WM8973_ADCDAC) & 0xfff7; + + if (mute) + snd_soc_write(codec, WM8973_ADCDAC, mute_reg | 0x8); + else + snd_soc_write(codec, WM8973_ADCDAC, mute_reg); + return 0; +} + +static int wm8973_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8973_priv *wm8973 = snd_soc_codec_get_drvdata(codec); + u16 pwr_reg = snd_soc_read(codec, WM8973_PWR1) & 0x03e; + + switch (level) { + case SND_SOC_BIAS_ON: + /* set vmid to 50k and unmute dac */ + snd_soc_write(codec, WM8973_PWR1, pwr_reg | 0x00c2); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(wm8973->regmap); + + /* mute dac and set vmid to 500k, enable VREF */ + snd_soc_write(codec, WM8973_PWR1, pwr_reg | 0x0141); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, WM8973_PWR1, 0x0001); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +#define WM8973_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define WM8973_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops wm8973_dai_ops = { + .hw_params = wm8973_pcm_hw_params, + .digital_mute = wm8973_mute, + .set_fmt = wm8973_set_dai_fmt, + .set_sysclk = wm8973_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver wm8973_dai[] = { + { + .name = "wm8973-hifi-playback", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8973_RATES, + .formats = WM8973_FORMATS, + }, + .ops = &wm8973_dai_ops, + }, + { + .name = "wm8973-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8973_RATES, + .formats = WM8973_FORMATS, + }, + .ops = &wm8973_dai_ops, + }, +}; + +static int wm8973_suspend(struct snd_soc_codec *codec) +{ + struct wm8973_priv *wm8973 = snd_soc_codec_get_drvdata(codec); + + wm8973_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_mark_dirty(wm8973->regmap); + return 0; +} + +static int wm8973_resume(struct snd_soc_codec *codec) +{ + u16 reg; + + wm8973_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* charge wm8973 caps */ + if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { + reg = snd_soc_read(codec, WM8973_PWR1) & 0xfe3e; + snd_soc_write(codec, WM8973_PWR1, reg | 0x01c0); + codec->dapm.bias_level = SND_SOC_BIAS_ON; + msleep(100); + } + + return 0; +} + +static int wm8973_probe(struct snd_soc_codec *codec) +{ + struct wm8973_priv *wm8973 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + u16 reg; + const int *p; + + mclk_div = 0; + + codec->control_data = wm8973->regmap; + + snd_soc_write(codec, WM8973_RESET, 0); + + if (codec->dev->of_node) { + p = of_get_property(codec->dev->of_node, "mclk-div", NULL); + if (p) + mclk_div = be32_to_cpu(*p); + } + /* Master Clock Divide by 2 (0 = not div, 1 = div by 2) */ + if (mclk_div) + snd_soc_update_bits(codec, WM8973_SRATE, 0x0040, 0x0040); + + /* charge output caps - set vmid to 5k for quick power up */ + reg = snd_soc_read(codec, WM8973_PWR1) & 0x03e; + snd_soc_write(codec, WM8973_PWR1, reg | 0x1f0); + + codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; + + /* set the update bits */ + snd_soc_update_bits(codec, WM8973_LDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8973_RDAC, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8973_LOUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8973_ROUT1V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8973_LOUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8973_ROUT2V, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8973_LINVOL, 0x0100, 0x0100); + snd_soc_update_bits(codec, WM8973_RINVOL, 0x0100, 0x0100); + + return ret; +} + + +/* power down chip */ +static int wm8973_remove(struct snd_soc_codec *codec) +{ + wm8973_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +struct regmap *wm8973_get_regmap(struct device *dev) +{ + struct wm8973_priv *priv = dev_get_drvdata(dev); + + return priv->regmap; +} + +static struct snd_soc_codec_driver soc_codec_dev_wm8973 = { + .probe = wm8973_probe, + .remove = wm8973_remove, + .suspend = wm8973_suspend, + .resume = wm8973_resume, + .set_bias_level = wm8973_set_bias_level, + .get_regmap = wm8973_get_regmap, + .controls = wm8973_snd_controls, + .num_controls = ARRAY_SIZE(wm8973_snd_controls), + .dapm_widgets = wm8973_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8973_dapm_widgets), + .dapm_routes = wm8973_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8973_dapm_routes), +}; + +static const struct regmap_config wm8973_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8973_MOUTV, + .reg_defaults = wm8973_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8973_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static int wm8973_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8973_priv *wm8973; + int ret; + + wm8973 = devm_kzalloc(&i2c->dev, sizeof(struct wm8973_priv), + GFP_KERNEL); + if (wm8973 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, wm8973); + + wm8973->regmap = devm_regmap_init_i2c(i2c, &wm8973_regmap); + if (IS_ERR(wm8973->regmap)) { + ret = PTR_ERR(wm8973->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8973, + wm8973_dai, ARRAY_SIZE(wm8973_dai)); + + return ret; +} + +static int wm8973_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id wm8973_i2c_id[] = { + { "wm8973", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8973_i2c_id); + +static const struct of_device_id wm8973_dt_ids[] = { + { .compatible = "wlf,wm8973" }, + { /* sentinel */ } +}; + +static struct i2c_driver wm8973_i2c_driver = { + .driver = { + .name = "wm8973", + .owner = THIS_MODULE, + .of_match_table = wm8973_dt_ids, + }, + .probe = wm8973_i2c_probe, + .remove = wm8973_i2c_remove, + .id_table = wm8973_i2c_id, +}; + +module_i2c_driver(wm8973_i2c_driver); + +MODULE_DESCRIPTION("ASoC WM8973 driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("*wm8973*"); diff --git a/sound/soc/codecs/wm8973.h b/sound/soc/codecs/wm8973.h new file mode 100644 index 0000000..5e6a026 --- /dev/null +++ b/sound/soc/codecs/wm8973.h @@ -0,0 +1,57 @@ +/* + * sound/soc/codecs/wm8973.h -- audio driver for WM8973 + * + * Copyright (C) 2013 - 2014 Fujitsu Semiconductor, Ltd + * + * Author: Xavier Hsu <xavier.hsu(a)linaro.org> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _WM8973_H +#define _WM8973_H + +#define WM8973_LINVOL 0x00 +#define WM8973_RINVOL 0x01 +#define WM8973_LOUT1V 0x02 +#define WM8973_ROUT1V 0x03 +#define WM8973_ADCDAC 0x05 +#define WM8973_IFACE 0x07 +#define WM8973_SRATE 0x08 +#define WM8973_LDAC 0x0a +#define WM8973_RDAC 0x0b +#define WM8973_BASS 0x0c +#define WM8973_TREBLE 0x0d +#define WM8973_RESET 0x0f +#define WM8973_3D 0x10 +#define WM8973_ALC1 0x11 +#define WM8973_ALC2 0x12 +#define WM8973_ALC3 0x13 +#define WM8973_NGATE 0x14 +#define WM8973_LADC 0x15 +#define WM8973_RADC 0x16 +#define WM8973_ADCTL1 0x17 +#define WM8973_ADCTL2 0x18 +#define WM8973_PWR1 0x19 +#define WM8973_PWR2 0x1a +#define WM8973_ADCTL3 0x1b +#define WM8973_ADCIN 0x1f +#define WM8973_LADCIN 0x20 +#define WM8973_RADCIN 0x21 +#define WM8973_LOUTM1 0x22 +#define WM8973_LOUTM2 0x23 +#define WM8973_ROUTM1 0x24 +#define WM8973_ROUTM2 0x25 +#define WM8973_MOUTM1 0x26 +#define WM8973_MOUTM2 0x27 +#define WM8973_LOUT2V 0x28 +#define WM8973_ROUT2V 0x29 +#define WM8973_MOUTV 0x2A + +#define WM8973_SYSCLK 0 + +#endif -- 1.7.9.5

3 3

[alsa-devel] [PATCH 0/2] ASoC: add CSR SiRFSoC audio USP interface driver
by Rongjun Ying 01 Jul '14

01 Jul '14

This patchset adds the SiRF USP controller driver, this driver uses the USP as I2S or DSP_A mode interface. Rongjun Ying (2): ASoC: sirf: Add audio usp interface driver ASoC: sirf: Add device tree binding for the USP audio device .../devicetree/bindings/sound/sirf-usp.txt | 27 ++ sound/soc/sirf/Kconfig | 6 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-usp.c | 409 +++++++++++++++++++++ sound/soc/sirf/sirf-usp.h | 293 +++++++++++++++ 5 files changed, 737 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.9.3 Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom More information can be found at www.csr.com. Keep up to date with CSR on our technical blog, www.csr.com/blog, CSR people blog, www.csr.com/people, YouTube, www.youtube.com/user/CSRplc, Facebook, www.facebook.com/pages/CSR/191038434253534, or follow us on Twitter at www.twitter.com/CSR_plc. New for 2014, you can now access the wide range of products powered by aptX at www.aptx.com.

2 3

[alsa-devel] next-20140640 - fatal crash at boot time in sound drivers...
by Valdis Kletnieks 01 Jul '14

01 Jul '14

Seeing this on a Dell Latitude E6530 - kernel lives for just a few seconds before committing hari-kiri trying to initialize the sound chipset. lspci reports the following audio devices: 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) 01:00.1 Audio device: NVIDIA Corporation GF108 High Definition Audio Controller (rev a1) Is working fine in next-20140618 This ring any bells, before I start doing a git bisect? [ 2.467707] netconsole: network logging started [ 2.467790] rtc_cmos 00:01: setting system clock to 2014-06-30 15:09:43 UTC (1404140983) [ 2.468008] BUG: unable to handle kernel NULL pointer dereference at (null) [ 2.468077] IP: [<ffffffff8128c97d>] __list_add+0x6e/0x13c [ 2.468122] PGD 0 [ 2.468143] Oops: 0000 [#1] PREEMPT SMP [ 2.468185] Modules linked in: [ 2.468214] CPU: 2 PID: 84 Comm: kworker/2:2 Not tainted 3.16.0-rc3-next-20140630 #244 [ 2.468244] ALSA device list: [ 2.468245] No soundcards found. [ 2.468307] Hardware name: Dell Inc. Latitude E6530/07Y85M, BIOS A14 01/13/2014 [ 2.468311] Workqueue: events azx_probe_work [ 2.468313] task: ffff8802237f12d0 ti: ffff8802237f4000 task.ti: ffff8802237f4000 [ 2.468316] RIP: 0010:[<ffffffff8128c97d>] [<ffffffff8128c97d>] __list_add+0x6e/0x13c [ 2.468318] RSP: 0018:ffff8802237f7b60 EFLAGS: 00010206 [ 2.468319] RAX: ffff880223752dd0 RBX: 0000000000000000 RCX: 0000000000000000 [ 2.468320] RDX: 0000000000000000 RSI: 0000000000000000 RDI: ffffffff81d2a3e0 [ 2.468322] RBP: ffff8802237f7b88 R08: ffff880223752d00 R09: 000000000000ffff [ 2.468323] R10: ffff8802237f7ad8 R11: 00000000fffffffe R12: ffffffff81ea4e40 [ 2.468324] R13: ffff880223752d00 R14: 0000000000000000 R15: ffff88003f99fb00 [ 2.468326] FS: 0000000000000000(0000) GS:ffff88022dd00000(0000) knlGS:0000000000000000 [ 2.468327] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 2.468328] CR2: 0000000000000000 CR3: 0000000001c10000 CR4: 00000000001407e0 [ 2.468329] Stack: [ 2.468332] ffff88003f99f040 ffffffff81ea4e40 0000000000000000 ffffffff81b47254 [ 2.468336] ffff88003f99fb80 ffff8802237f7bb0 ffffffff814670a8 0000000000000000 [ 2.468339] ffff88003f8e9000 ffffffff818ba498 ffff8802237f7bc0 ffffffff8148c830 [ 2.468340] Call Trace: [ 2.468345] [<ffffffff814670a8>] _snd_ctl_add_slave+0xb6/0xbf [ 2.468349] [<ffffffff8148c830>] snd_ctl_add_slave+0xb/0xd [ 2.468352] [<ffffffff8148c71d>] map_slaves+0xb4/0xec [ 2.468354] [<ffffffff8148c825>] ? snd_hda_enum_helper_info+0x4b/0x4b [ 2.468358] [<ffffffff8148d9a0>] __snd_hda_add_vmaster+0x9b/0xfc [ 2.468361] [<ffffffff81499c9e>] snd_hda_gen_build_controls+0xff/0x1ab [ 2.468365] [<ffffffff81491a63>] snd_hda_codec_build_controls+0x38/0x196 [ 2.468368] [<ffffffff81491bed>] snd_hda_build_controls+0x2c/0x87 [ 2.468370] [<ffffffff81496c4e>] azx_mixer_create+0x10/0x12 [ 2.468373] [<ffffffff814a2c06>] azx_probe_work+0x3db/0x519 [ 2.468378] [<ffffffff81057523>] process_one_work+0x296/0x4b8 [ 2.468381] [<ffffffff81057f2c>] worker_thread+0x3fc/0x54e [ 2.468384] [<ffffffff81057b30>] ? cancel_delayed_work+0xb7/0xb7 [ 2.468387] [<ffffffff8105dbfc>] kthread+0xd6/0xde [ 2.468390] [<ffffffff8105db26>] ? __kthread_parkme+0x62/0x62 [ 2.468394] [<ffffffff8168bfac>] ret_from_fork+0x7c/0xb0 [ 2.468396] [<ffffffff8105db26>] ? __kthread_parkme+0x62/0x62 [ 2.468443] Code: be 1d 00 00 00 48 c7 c2 f4 51 b0 81 48 c7 c7 41 52 b0 81 31 c0 e8 d3 0d db ff 31 d2 44 89 f6 48 c7 c7 e0 a3 d2 81 e8 62 ca e4 ff <4c> 39 23 48 c7 c7 b8 a3 d2 81 41 0f 95 c7 31 d2 45 0f b6 f7 44 [ 2.468445] RIP [<ffffffff8128c97d>] __list_add+0x6e/0x13c [ 2.468446] RSP <ffff8802237f7b60> [ 2.468447] CR2: 0000000000000000 [ 2.468449] ---[ end trace 2ff29d6bc6a184d4 ]--- [ 2.605416] tsc: Refined TSC clocksource calibration: 2691.263 MHz [ 2.687337] usb 1-1: new high-speed USB device number 2 using ehci-pci [ 2.741253] ata2: SATA link up 1.5 Gbps (SStatus 113 SControl 300) [ 2.742914] ata2.00: ATAPI: MATSHITA DVD+/-RW UJ8C2, 1.02, max UDMA/133 [ 2.744350] ata2.00: configured for UDMA/133 [ 2.745510] scsi 1:0:0:0: CD-ROM MATSHITA DVD+-RW UJ8C2 1.02 PQ: 0 ANSI: 5 [ 2.804514] usb 1-1: New USB device found, idVendor=8087, idProduct=0024 [ 2.804545] usb 1-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 [ 2.804909] hub 1-1:1.0: USB hub found [ 2.805005] hub 1-1:1.0: 6 ports detected [ 2.908047] usb 2-1: new high-speed USB device number 2 using ehci-pci [ 3.024199] usb 2-1: New USB device found, idVendor=8087, idProduct=0024 [ 3.024253] usb 2-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 [ 3.024612] hub 2-1:1.0: USB hub found [ 3.024694] hub 2-1:1.0: 8 ports detected [ 3.179680] usb 3-1: new low-speed USB device number 2 using xhci_hcd [ 3.358523] usb 3-1: New USB device found, idVendor=045e, idProduct=0023 [ 3.358578] usb 3-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 3.358634] usb 3-1: Product: Microsoft Trackball Optical® [ 3.358678] usb 3-1: Manufacturer: Microsoft [ 3.358930] usb 3-1: ep 0x81 - rounding interval to 64 microframes, ep desc says 80 microframes [ 3.365460] input: Microsoft Microsoft Trackball Optical® as /devices/pci0000:00/0000:00:14.0/usb3/3-1/3-1:1.0/0003:045E:0023.0001/input/input9 [ 3.365884] hid-generic 0003:045E:0023.0001: input,hidraw0: USB HID v1.00 Mouse [Microsoft Microsoft Trackball Optical®] on usb-0000:00:14.0-1/input0 [ 3.518233] usb 3-4: new high-speed USB device number 3 using xhci_hcd [ 4.742601] Switched to clocksource tsc [ 4.742844] kworker/2:2 (84) used greatest stack depth: 11888 bytes left [ 4.743118] BUG: unable to handle kernel paging request at ffffffffffffffa8 [ 4.746083] IP: [<ffffffff8105dfd2>] kthread_data+0xc/0x11 [ 4.749966] PGD 1c11067 PUD 1c13067 PMD 0 [ 4.754807] Oops: 0000 [#2] PREEMPT SMP [ 4.759798] Modules linked in: [ 4.764668] CPU: 2 PID: 84 Comm: kworker/2:2 Tainted: G D 3.16.0-rc3-next-20140630 #244 [ 4.769503] Hardware name: Dell Inc. Latitude E6530/07Y85M, BIOS A14 01/13/2014 [ 4.774468] task: ffff8802237f12d0 ti: ffff8802237f4000 task.ti: ffff8802237f4000 [ 4.779351] RIP: 0010:[<ffffffff8105dfd2>] [<ffffffff8105dfd2>] kthread_data+0xc/0x11 [ 4.784250] RSP: 0018:ffff8802237f77a0 EFLAGS: 00010002 [ 4.789373] RAX: 0000000000000000 RBX: ffff88022dd13d00 RCX: 000000000000000f [ 4.794280] RDX: 0000000000000000 RSI: 0000000000000002 RDI: ffff8802237f12d0 [ 4.799157] RBP: ffff8802237f77c8 R08: ffff88022dd13da8 R09: ffffffff81ea3000 [ 4.804085] R10: ffff8802237f7650 R11: ffff8802249ac780 R12: 0000000000000000 [ 4.808950] R13: 0000000000000002 R14: 0000000000000040 R15: ffff8802237f7898 [ 4.813761] FS: 0000000000000000(0000) GS:ffff88022dd00000(0000) knlGS:0000000000000000 [ 4.818642] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 4.823588] CR2: 0000000000000028 CR3: 0000000001c10000 CR4: 00000000001407e0 [ 4.828539] Stack: [ 4.833335] ffffffff81058150 ffff88022dd13d00 0000000000000000 0000000000000002 [ 4.838307] 0000000000000040 ffff8802237f7838 ffffffff8168639f ffff8802237f7fd8 [ 4.843246] ffff8802237f12d0 ffff8802249ac780 0000000000013d00 ffff88022dd13d00 [ 4.848167] Call Trace: [ 4.853149] [<ffffffff81058150>] ? wq_worker_sleeping+0x13/0x105 [ 4.858028] [<ffffffff8168639f>] __schedule+0x2a3/0xa91 [ 4.862946] [<ffffffff81686bf1>] schedule+0x64/0x66 [ 4.867860] [<ffffffff8103f119>] do_exit+0xbbf/0xc0c [ 4.872616] [<ffffffff8107fc72>] ? arch_local_irq_save+0x9/0xc [ 4.877432] [<ffffffff8100577f>] oops_end+0x7c/0x81 [ 4.882219] [<ffffffff8102f336>] no_context+0x25c/0x2b0 [ 4.887043] [<ffffffff8102f3f9>] __bad_area_nosemaphore+0x6f/0x1fe [ 4.892019] [<ffffffff8102f596>] bad_area_nosemaphore+0xe/0x10 [ 4.897024] [<ffffffff8102f955>] __do_page_fault+0x3bd/0x724 [ 4.901834] [<ffffffff81081c77>] ? mark_lock+0x2a/0x203 [ 4.906643] [<ffffffff810829a6>] ? __lock_acquire+0x68c/0xed1 [ 4.911611] [<ffffffff810d89c4>] ? time_hardirqs_off+0x1b/0x2f [ 4.916428] [<ffffffff8107ff5a>] ? trace_hardirqs_off_caller+0x4c/0xb9 [ 4.921245] [<ffffffff81282e0a>] ? trace_hardirqs_off_thunk+0x3a/0x3c [ 4.926095] [<ffffffff8102fcc8>] do_page_fault+0xc/0xe [ 4.930911] [<ffffffff8168dc52>] page_fault+0x22/0x30 [ 4.935712] [<ffffffff8128c97d>] ? __list_add+0x6e/0x13c [ 4.940601] [<ffffffff814670a8>] _snd_ctl_add_slave+0xb6/0xbf [ 4.945404] [<ffffffff8148c830>] snd_ctl_add_slave+0xb/0xd [ 4.950219] [<ffffffff8148c71d>] map_slaves+0xb4/0xec [ 4.954877] [<ffffffff8148c825>] ? snd_hda_enum_helper_info+0x4b/0x4b [ 4.959733] [<ffffffff8148d9a0>] __snd_hda_add_vmaster+0x9b/0xfc [ 4.964346] [<ffffffff81499c9e>] snd_hda_gen_build_controls+0xff/0x1ab [ 4.968948] [<ffffffff81491a63>] snd_hda_codec_build_controls+0x38/0x196 [ 4.973562] [<ffffffff81491bed>] snd_hda_build_controls+0x2c/0x87 [ 4.978178] [<ffffffff81496c4e>] azx_mixer_create+0x10/0x12 [ 4.982693] [<ffffffff814a2c06>] azx_probe_work+0x3db/0x519 [ 4.987371] [<ffffffff81057523>] process_one_work+0x296/0x4b8 [ 4.991806] [<ffffffff81057f2c>] worker_thread+0x3fc/0x54e [ 4.996432] [<ffffffff81057b30>] ? cancel_delayed_work+0xb7/0xb7 [ 5.000828] [<ffffffff8105dbfc>] kthread+0xd6/0xde [ 5.005151] [<ffffffff8105db26>] ? __kthread_parkme+0x62/0x62 [ 5.009500] [<ffffffff8168bfac>] ret_from_fork+0x7c/0xb0 [ 5.013921] [<ffffffff8105db26>] ? __kthread_parkme+0x62/0x62 [ 5.018307] Code: 48 8b 04 25 80 b9 00 00 48 8b 80 d8 04 00 00 48 89 e5 5d 48 8b 40 98 48 c1 e8 02 83 e0 01 c3 48 8b 87 d8 04 00 00 55 48 89 e5 5d <48> 8b 40 a8 c3 55 ba 08 00 00 00 48 89 e5 50 48 8b b7 d8 04 00 [ 5.023570] RIP [<ffffffff8105dfd2>] kthread_data+0xc/0x11 [ 5.028173] RSP <ffff8802237f77a0> [ 5.032760] CR2: ffffffffffffffa8 [ 5.037505] ---[ end trace 2ff29d6bc6a184d5 ]--- [ 5.211674] Fixing recursive fault but reboot is needed! [ 5.216280] BUG: scheduling while atomic: kworker/2:2/84/0x00000004 [ 5.220852] INFO: lockdep is turned off. [ 5.225435] Modules linked in: [ 5.230060] irq event stamp: 4640 [ 5.234609] hardirqs last enabled at (4639): [<ffffffff8113286a>] __kmalloc+0x94/0x10d [ 5.239246] hardirqs last disabled at (4640): [<ffffffff8168de43>] error_sti+0x5/0x6 [ 5.244073] softirqs last enabled at (4294): [<ffffffff81041faa>] __do_softirq+0x2ab/0x403 [ 5.248509] softirqs last disabled at (4255): [<ffffffff810423a5>] irq_exit+0x4c/0xbb [ 5.253019] CPU: 2 PID: 84 Comm: kworker/2:2 Tainted: G D 3.16.0-rc3-next-20140630 #244 [ 5.257493] Hardware name: Dell Inc. Latitude E6530/07Y85M, BIOS A14 01/13/2014 [ 5.261861] 0000000000000000 ffff8802237f73f0 ffffffff81680ccb ffff8802237f12d0 [ 5.266268] 0000000000000001 ffff8802237f7408 ffffffff8106667f ffff88022dd13d00 [ 5.270961] ffff8802237f7478 ffffffff816861b6 ffff8802237f7fd8 ffff8802237f12d0 [ 5.275445] Call Trace: [ 5.279729] [<ffffffff81680ccb>] dump_stack+0x51/0xaa [ 5.284243] [<ffffffff8106667f>] __schedule_bug+0x5e/0x6d [ 5.288769] [<ffffffff816861b6>] __schedule+0xba/0xa91 [ 5.293159] [<ffffffff81686bf1>] schedule+0x64/0x66 [ 5.297348] [<ffffffff8103e700>] do_exit+0x1a6/0xc0c [ 5.301489] [<ffffffff8107fc72>] ? arch_local_irq_save+0x9/0xc [ 5.305667] [<ffffffff81090e0b>] ? __rcu_read_unlock+0xf2/0xf9 [ 5.309791] [<ffffffff8108ab84>] ? kmsg_dump+0x18b/0x194 [ 5.313868] [<ffffffff8100577f>] oops_end+0x7c/0x81 [ 5.317922] [<ffffffff8102f336>] no_context+0x25c/0x2b0 [ 5.321983] [<ffffffff8102f3f9>] __bad_area_nosemaphore+0x6f/0x1fe [ 5.326018] [<ffffffff8102f596>] bad_area_nosemaphore+0xe/0x10 [ 5.330032] [<ffffffff8102f955>] __do_page_fault+0x3bd/0x724 [ 5.334012] [<ffffffff81083554>] ? lock_acquire+0xc1/0x14e [ 5.337966] [<ffffffff810d89c4>] ? time_hardirqs_off+0x1b/0x2f [ 5.341895] [<ffffffff8168de43>] ? error_sti+0x5/0x6 [ 5.345767] [<ffffffff8107ff2d>] ? trace_hardirqs_off_caller+0x1f/0xb9 [ 5.349649] [<ffffffff81282e0a>] ? trace_hardirqs_off_thunk+0x3a/0x3c [ 5.353528] [<ffffffff8102fcc8>] do_page_fault+0xc/0xe [ 5.357389] [<ffffffff8168dc52>] page_fault+0x22/0x30 [ 5.361234] [<ffffffff8105dfd2>] ? kthread_data+0xc/0x11 [ 5.365089] [<ffffffff81058150>] ? wq_worker_sleeping+0x13/0x105 [ 5.368997] [<ffffffff8168639f>] __schedule+0x2a3/0xa91 [ 5.372809] [<ffffffff81686bf1>] schedule+0x64/0x66 [ 5.376729] [<ffffffff8103f119>] do_exit+0xbbf/0xc0c [ 5.380532] [<ffffffff8107fc72>] ? arch_local_irq_save+0x9/0xc [ 5.384415] [<ffffffff8100577f>] oops_end+0x7c/0x81 [ 5.388189] [<ffffffff8102f336>] no_context+0x25c/0x2b0 [ 5.392030] [<ffffffff8102f3f9>] __bad_area_nosemaphore+0x6f/0x1fe [ 5.396037] [<ffffffff8102f596>] bad_area_nosemaphore+0xe/0x10 [ 5.400019] [<ffffffff8102f955>] __do_page_fault+0x3bd/0x724 [ 5.403823] [<ffffffff81081c77>] ? mark_lock+0x2a/0x203 [ 5.407624] [<ffffffff810829a6>] ? __lock_acquire+0x68c/0xed1 [ 5.411405] [<ffffffff810d89c4>] ? time_hardirqs_off+0x1b/0x2f [ 5.415209] [<ffffffff8107ff5a>] ? trace_hardirqs_off_caller+0x4c/0xb9 [ 5.419107] [<ffffffff81282e0a>] ? trace_hardirqs_off_thunk+0x3a/0x3c [ 5.422894] [<ffffffff8102fcc8>] do_page_fault+0xc/0xe [ 5.426916] [<ffffffff8168dc52>] page_fault+0x22/0x30 [ 5.430708] [<ffffffff8128c97d>] ? __list_add+0x6e/0x13c [ 5.434489] [<ffffffff814670a8>] _snd_ctl_add_slave+0xb6/0xbf [ 5.438290] [<ffffffff8148c830>] snd_ctl_add_slave+0xb/0xd [ 5.442229] [<ffffffff8148c71d>] map_slaves+0xb4/0xec [ 5.446010] [<ffffffff8148c825>] ? snd_hda_enum_helper_info+0x4b/0x4b [ 5.450140] [<ffffffff8148d9a0>] __snd_hda_add_vmaster+0x9b/0xfc [ 5.453932] [<ffffffff81499c9e>] snd_hda_gen_build_controls+0xff/0x1ab [ 5.457821] [<ffffffff81491a63>] snd_hda_codec_build_controls+0x38/0x196 [ 5.461724] [<ffffffff81491bed>] snd_hda_build_controls+0x2c/0x87 [ 5.465524] [<ffffffff81496c4e>] azx_mixer_create+0x10/0x12 [ 5.469319] [<ffffffff814a2c06>] azx_probe_work+0x3db/0x519 [ 5.473161] [<ffffffff81057523>] process_one_work+0x296/0x4b8 [ 5.476952] [<ffffffff81057f2c>] worker_thread+0x3fc/0x54e [ 5.480734] [<ffffffff81057b30>] ? cancel_delayed_work+0xb7/0xb7 [ 5.484603] [<ffffffff8105dbfc>] kthread+0xd6/0xde [ 5.488415] [<ffffffff8105db26>] ? __kthread_parkme+0x62/0x62 [ 5.492334] [<ffffffff8168bfac>] ret_from_fork+0x7c/0xb0 [ 5.496116] [<ffffffff8105db26>] ? __kthread_parkme+0x62/0x62 [ 56.430656] Kernel panic - not syncing: Watchdog detected hard LOCKUP on cpu 2 [ 57.577582] Shutting down cpus with NMI [ 57.580830] Kernel Offset: 0x0 from 0xffffffff81000000 (relocation range: 0xffffffff80000000-0xffffffff9fffffff)

3 2

[alsa-devel] [PATCH 1/5] ALSA: hda: Remove unused variable
by Sachin Kamat 01 Jul '14

01 Jul '14

'status' is not used in the function. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Thierry Reding <thierry.reding(a)gmail.com> --- All patches in this series only compile tested. --- sound/pci/hda/hda_tegra.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index a366ba9293a8..8cd7b06eecef 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -272,13 +272,9 @@ static int hda_tegra_resume(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); - int status; hda_tegra_enable_clocks(hda); - /* Read STATESTS before controller reset */ - status = azx_readw(chip, STATESTS); - hda_tegra_init(hda); azx_init_chip(chip, 1); -- 1.7.9.5

4 16

Re: [alsa-devel] [PATCH 2/2] ASoC: add driver for Rockchip RK3xxx I2S controller
by Lars-Peter Clausen 01 Jul '14

01 Jul '14

On 07/01/2014 04:49 PM, Huang Tao wrote: > Dear Lars: > > 于 2014年07月01日 17:51, Lars-Peter Clausen 写道: >>> + /* Try to set the I2S Channel id from dt */ >>> >+ pdev->id = of_alias_get_id(np, "i2s"); >>> >+ dev_set_name(&pdev->dev, "%s.%d", >>> >+ pdev->dev.driver->name, >>> >+ pdev->id); >> A device should not change its id or name. >> > Could you tell me the reason? > When registering a device the Linux device core that the id and name are unique. If you change them behind the back of the core there is no guarantee that they are still unique and undefined behavior might occur. Also changing them can cause all kind of other weird behavior, e.g. the name no longer matches the name of the sysfs files, etc. - Lars

2 1

[alsa-devel] [RFC] question regarding codec DAI ops
by Arnaud Pouliquen 01 Jul '14

01 Jul '14

Hello, I have a platform with several audio IOs (DAC , HDMI...) and I2S In/out connectors. I'm using my codec driver but for this I2S output i used dummy codec driver with the snd-soc-dummy-dai dai. On platform side I have several instances of CPU DAI, and i'm using helpers function like snd_soc_dai_set_tdm_slot to configure associated codec DAIs. My issue is that these functions return error for my I2S output because ops functions are not implemented in dummy driver. I see 3 solutions: - implement ops in dummy drivers ( soc-utils) - return 0 instead of error when ops is not implemented ( soc-core) - Add a test in my driver on ops before calling it. First one seems the best compromise to avoid to accumulate tests in platform drivers. From your point of view, what should be best solution? BR Arnaud

2 5

[alsa-devel] [PATCH 2/2] ASoC: add driver for Rockchip RK3xxx I2S controller
by jianqun 01 Jul '14

01 Jul '14

From: Jianqun Xu <xjq(a)rock-chips.com> Add driver for I2S controller in Rockchip RK3xxx SoCs. This driver patch has been tested on the RK3288 SDK board. Signed-off-by: Jianqun Xu <xjq(a)rock-chips.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/rockchip/Kconfig | 16 + sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/i2s.h | 222 +++++++++++++ sound/soc/rockchip/pcm.h | 14 + sound/soc/rockchip/rockchip_i2s.c | 622 +++++++++++++++++++++++++++++++++++++ sound/soc/rockchip/rockchip_pcm.c | 64 ++++ 8 files changed, 946 insertions(+) create mode 100644 sound/soc/rockchip/Kconfig create mode 100644 sound/soc/rockchip/Makefile create mode 100644 sound/soc/rockchip/i2s.h create mode 100644 sound/soc/rockchip/pcm.h create mode 100644 sound/soc/rockchip/rockchip_i2s.c create mode 100644 sound/soc/rockchip/rockchip_pcm.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0060b31..0e96233 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -47,6 +47,7 @@ source "sound/soc/kirkwood/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/rockchip/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 5f1df02..534714a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -24,6 +24,7 @@ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ obj-$(CONFIG_SND_SOC) += pxa/ +obj-$(CONFIG_SND_SOC) += rockchip/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig new file mode 100644 index 0000000..946b60c --- /dev/null +++ b/sound/soc/rockchip/Kconfig @@ -0,0 +1,16 @@ +config SND_SOC_ROCKCHIP + tristate "ASoC support for Rockchip" + depends on SND_SOC && ARCH_ROCKCHIP + select SND_SOC_GENERIC_DMAENGINE_PCM + select SND_ROCKCHIP_PCM + select SND_ROCKCHIP_I2S + help + Say Y or M if you want to add support for codecs attached to + the Rockchip SoCs' Audio interfaces. You will also need to + select the audio interfaces to support below. + +config SND_ROCKCHIP_I2S + tristate + +config SND_ROCKCHIP_PCM + tristate diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile new file mode 100644 index 0000000..0b5d47e --- /dev/null +++ b/sound/soc/rockchip/Makefile @@ -0,0 +1,6 @@ +# ROCKCHIP Platform Support +snd-soc-i2s-objs := rockchip_i2s.o +snd-soc-pcm-objs := rockchip_pcm.o + +obj-$(CONFIG_SND_ROCKCHIP_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_ROCKCHIP_PCM) += snd-soc-pcm.o diff --git a/sound/soc/rockchip/i2s.h b/sound/soc/rockchip/i2s.h new file mode 100644 index 0000000..54a5a67 --- /dev/null +++ b/sound/soc/rockchip/i2s.h @@ -0,0 +1,222 @@ +/* + * i2s.h - ALSA IIS interface for the rockchip SoC + * + * Driver for rockchip iis audio + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/version.h> + +#ifndef _ROCKCHIP_IIS_H +#define _ROCKCHIP_IIS_H + +/* + * TXCR + * transmit operation control register +*/ +#define I2S_TXCR_RCNT_SHIFT 17 +#define I2S_TXCR_RCNT_MASK (0x3f << I2S_TXCR_RCNT_SHIFT) +#define I2S_TXCR_CSR_SHIFT 15 +#define I2S_TXCR_CSR(x) (x << I2S_TXCR_CSR_SHIFT) +#define I2S_TXCR_CSR_MASK (3 << I2S_TXCR_CSR_SHIFT) +#define I2S_TXCR_HWT BIT(14) +#define I2S_TXCR_SJM_SHIFT 12 +#define I2S_TXCR_SJM_R (0 << I2S_TXCR_SJM_SHIFT) +#define I2S_TXCR_SJM_L (1 << I2S_TXCR_SJM_SHIFT) +#define I2S_TXCR_FBM_SHIFT 11 +#define I2S_TXCR_FBM_MSB (0 << I2S_TXCR_FBM_SHIFT) +#define I2S_TXCR_FBM_LSB (1 << I2S_TXCR_FBM_SHIFT) +#define I2S_TXCR_IBM_SHIFT 9 +#define I2S_TXCR_IBM_NORMAL (0 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_LSJM (1 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_RSJM (2 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_MASK (3 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_PBM_SHIFT 7 +#define I2S_TXCR_PBM_MODE(x) (x << I2S_TXCR_PBM_SHIFT) +#define I2S_TXCR_PBM_MASK (3 << I2S_TXCR_PBM_SHIFT) +#define I2S_TXCR_TFS_SHIFT 5 +#define I2S_TXCR_TFS_I2S (0 << I2S_TXCR_TFS_SHIFT) +#define I2S_TXCR_TFS_PCM (1 << I2S_TXCR_TFS_SHIFT) +#define I2S_TXCR_VDW_SHIFT 0 +#define I2S_TXCR_VDW(x) ((x - 1) << I2S_TXCR_VDW_SHIFT) +#define I2S_TXCR_VDW_MASK (0x1f << I2S_TXCR_VDW_SHIFT) + +/* + * RXCR + * receive operation control register +*/ +#define I2S_RXCR_HWT BIT(14) +#define I2S_RXCR_SJM_SHIFT 12 +#define I2S_RXCR_SJM_R (0 << I2S_RXCR_SJM_SHIFT) +#define I2S_RXCR_SJM_L (1 << I2S_RXCR_SJM_SHIFT) +#define I2S_RXCR_FBM_SHIFT 11 +#define I2S_RXCR_FBM_MSB (0 << I2S_RXCR_FBM_SHIFT) +#define I2S_RXCR_FBM_LSB (1 << I2S_RXCR_FBM_SHIFT) +#define I2S_RXCR_IBM_SHIFT 9 +#define I2S_RXCR_IBM_NORMAL (0 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_LSJM (1 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_RSJM (2 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_MASK (3 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_PBM_SHIFT 7 +#define I2S_RXCR_PBM_MODE(x) (x << I2S_RXCR_PBM_SHIFT) +#define I2S_RXCR_PBM_MASK (3 << I2S_RXCR_PBM_SHIFT) +#define I2S_RXCR_TFS_SHIFT 5 +#define I2S_RXCR_TFS_I2S (0 << I2S_RXCR_TFS_SHIFT) +#define I2S_RXCR_TFS_PCM (1 << I2S_RXCR_TFS_SHIFT) +#define I2S_RXCR_VDW_SHIFT 0 +#define I2S_RXCR_VDW(x) ((x - 1) << I2S_RXCR_VDW_SHIFT) +#define I2S_RXCR_VDW_MASK (0x1f << I2S_RXCR_VDW_SHIFT) + +/* + * CKR + * clock generation register +*/ +#define I2S_CKR_MSS_SHIFT 27 +#define I2S_CKR_MSS_MASTER (0 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_MSS_SLAVE (1 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_MSS_MASK (1 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_CKP_SHIFT 26 +#define I2S_CKR_CKP_NEG (0 << I2S_CKR_CKP_SHIFT) +#define I2S_CKR_CKP_POS (1 << I2S_CKR_CKP_SHIFT) +#define I2S_CKR_RLP_SHIFT 25 +#define I2S_CKR_RLP_NORMAL (0 << I2S_CKR_RLP_SHIFT) +#define I2S_CKR_RLP_OPPSITE (1 << I2S_CKR_RLP_SHIFT) +#define I2S_CKR_TLP_SHIFT 24 +#define I2S_CKR_TLP_NORMAL (0 << I2S_CKR_TLP_SHIFT) +#define I2S_CKR_TLP_OPPSITE (1 << I2S_CKR_TLP_SHIFT) +#define I2S_CKR_MDIV_SHIFT 16 +#define I2S_CKR_MDIV(x) ((x - 1) << I2S_CKR_MDIV_SHIFT) +#define I2S_CKR_MDIV_MASK (0xff << I2S_CKR_MDIV_SHIFT) +#define I2S_CKR_RSD_SHIFT 8 +#define I2S_CKR_RSD(x) ((x - 1) << I2S_CKR_RSD_SHIFT) +#define I2S_CKR_RSD_MASK (0xff << I2S_CKR_RSD_SHIFT) +#define I2S_CKR_TSD_SHIFT 0 +#define I2S_CKR_TSD(x) ((x - 1) << I2S_CKR_TSD_SHIFT) +#define I2S_CKR_TSD_MASK (0xff << I2S_CKR_TSD_SHIFT) + +/* + * FIFOLR + * FIFO level register +*/ +#define I2S_FIFOLR_RFL_SHIFT 24 +#define I2S_FIFOLR_RFL_MASK (0x3f << I2S_FIFOLR_RFL_SHIFT) +#define I2S_FIFOLR_TFL3_SHIFT 18 +#define I2S_FIFOLR_TFL3_MASK (0x3f << I2S_FIFOLR_TFL3_SHIFT) +#define I2S_FIFOLR_TFL2_SHIFT 12 +#define I2S_FIFOLR_TFL2_MASK (0x3f << I2S_FIFOLR_TFL2_SHIFT) +#define I2S_FIFOLR_TFL1_SHIFT 6 +#define I2S_FIFOLR_TFL1_MASK (0x3f << I2S_FIFOLR_TFL1_SHIFT) +#define I2S_FIFOLR_TFL0_SHIFT 0 +#define I2S_FIFOLR_TFL0_MASK (0x3f << I2S_FIFOLR_TFL0_SHIFT) + +/* + * DMACR + * DMA control register +*/ +#define I2S_DMACR_RDE_SHIFT 24 +#define I2S_DMACR_RDE_DISABLE (0 << I2S_DMACR_RDE_SHIFT) +#define I2S_DMACR_RDE_ENABLE (1 << I2S_DMACR_RDE_SHIFT) +#define I2S_DMACR_RDL_SHIFT 16 +#define I2S_DMACR_RDL(x) ((x - 1) << I2S_DMACR_RDL_SHIFT) +#define I2S_DMACR_RDL_MASK (0x1f << I2S_DMACR_RDL_SHIFT) +#define I2S_DMACR_TDE_SHIFT 8 +#define I2S_DMACR_TDE_DISABLE (0 << I2S_DMACR_TDE_SHIFT) +#define I2S_DMACR_TDE_ENABLE (1 << I2S_DMACR_TDE_SHIFT) +#define I2S_DMACR_TDL_SHIFT 0 +#define I2S_DMACR_TDL(x) ((x - 1) << I2S_DMACR_TDL_SHIFT) +#define I2S_DMACR_TDL_MASK (0x1f << I2S_DMACR_TDL_SHIFT) + +/* + * INTCR + * interrupt control register +*/ +#define I2S_INTCR_RFT_SHIFT 20 +#define I2S_INTCR_RFT(x) ((x - 1) << I2S_INTCR_RFT_SHIFT) +#define I2S_INTCR_RXOIC BIT(18) +#define I2S_INTCR_RXOIE_SHIFT 17 +#define I2S_INTCR_RXOIE_DISABLE (0 << I2S_INTCR_RXOIE_SHIFT) +#define I2S_INTCR_RXOIE_ENABLE (1 << I2S_INTCR_RXOIE_SHIFT) +#define I2S_INTCR_RXFIE_SHIFT 16 +#define I2S_INTCR_RXFIE_DISABLE (0 << I2S_INTCR_RXFIE_SHIFT) +#define I2S_INTCR_RXFIE_ENABLE (1 << I2S_INTCR_RXFIE_SHIFT) +#define I2S_INTCR_TFT_SHIFT 4 +#define I2S_INTCR_TFT(x) ((x - 1) << I2S_INTCR_TFT_SHIFT) +#define I2S_INTCR_TFT_MASK (0x1f << I2S_INTCR_TFT_SHIFT) +#define I2S_INTCR_TXUIC BIT(2) +#define I2S_INTCR_TXUIE_SHIFT 1 +#define I2S_INTCR_TXUIE_DISABLE (0 << I2S_INTCR_TXUIE_SHIFT) +#define I2S_INTCR_TXUIE_ENABLE (1 << I2S_INTCR_TXUIE_SHIFT) + +/* + * INTSR + * interrupt status register +*/ +#define I2S_INTSR_TXEIE_SHIFT 0 +#define I2S_INTSR_TXEIE_DISABLE (0 << I2S_INTSR_TXEIE_SHIFT) +#define I2S_INTSR_TXEIE_ENABLE (1 << I2S_INTSR_TXEIE_SHIFT) +#define I2S_INTSR_RXOI_SHIFT 17 +#define I2S_INTSR_RXOI_INA (0 << I2S_INTSR_RXOI_SHIFT) +#define I2S_INTSR_RXOI_ACT (1 << I2S_INTSR_RXOI_SHIFT) +#define I2S_INTSR_RXFI_SHIFT 16 +#define I2S_INTSR_RXFI_INA (0 << I2S_INTSR_RXFI_SHIFT) +#define I2S_INTSR_RXFI_ACT (1 << I2S_INTSR_RXFI_SHIFT) +#define I2S_INTSR_TXUI_SHIFT 1 +#define I2S_INTSR_TXUI_INA (0 << I2S_INTSR_TXUI_SHIFT) +#define I2S_INTSR_TXUI_ACT (1 << I2S_INTSR_TXUI_SHIFT) +#define I2S_INTSR_TXEI_SHIFT 0 +#define I2S_INTSR_TXEI_INA (0 << I2S_INTSR_TXEI_SHIFT) +#define I2S_INTSR_TXEI_ACT (1 << I2S_INTSR_TXEI_SHIFT) + +/* + * XFER + * Transfer start register +*/ +#define I2S_XFER_RXS_SHIFT 1 +#define I2S_XFER_RXS_STOP (0 << I2S_XFER_RXS_SHIFT) +#define I2S_XFER_RXS_START (1 << I2S_XFER_RXS_SHIFT) +#define I2S_XFER_TXS_SHIFT 0 +#define I2S_XFER_TXS_STOP (0 << I2S_XFER_TXS_SHIFT) +#define I2S_XFER_TXS_START (1 << I2S_XFER_TXS_SHIFT) + +/* + * CLR + * clear SCLK domain logic register +*/ +#define I2S_CLR_RXC BIT(1) +#define I2S_CLR_TXC BIT(0) + +/* + * TXDR + * Transimt FIFO data register, write only. +*/ +#define I2S_TXDR_MASK (0xff) + +/* + * RXDR + * Receive FIFO data register, write only. +*/ +#define I2S_RXDR_MASK (0xff) + +/* Clock divider id */ +enum { + ROCKCHIP_DIV_MCLK = 0, + ROCKCHIP_DIV_BCLK, +}; + +/* I2S REGS */ +#define I2S_TXCR (0x0000) +#define I2S_RXCR (0x0004) +#define I2S_CKR (0x0008) +#define I2S_FIFOLR (0x000c) +#define I2S_DMACR (0x0010) +#define I2S_INTCR (0x0014) +#define I2S_INTSR (0x0018) +#define I2S_XFER (0x001c) +#define I2S_CLR (0x0020) +#define I2S_TXDR (0x0024) +#define I2S_RXDR (0x0028) + +#endif /* _ROCKCHIP_IIS_H */ diff --git a/sound/soc/rockchip/pcm.h b/sound/soc/rockchip/pcm.h new file mode 100644 index 0000000..c373058 --- /dev/null +++ b/sound/soc/rockchip/pcm.h @@ -0,0 +1,14 @@ +/* sound/soc/rockchip/pcm.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ROCKCHIP_PCM_H +#define _ROCKCHIP_PCM_H + +int rockchip_pcm_platform_register(struct device *dev); +void rockchip_pcm_platform_unregister(struct device *dev); + +#endif /* _ROCKCHIP_PCM_H */ diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c new file mode 100644 index 0000000..bb52ae3 --- /dev/null +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -0,0 +1,622 @@ +/* sound/soc/rockchip/i2s.c + * + * ALSA SoC Audio Layer - Rockchip I2S Controller driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/version.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/spinlock.h> +#include <linux/workqueue.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "pcm.h" +#include "i2s.h" + +#define DRV_NAME "rockchip-i2s" + +struct rk_i2s_dev { + void __iomem *regs; + struct device *dev; + + struct clk *hclk; + struct clk *clk; + + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; + + bool i2s_tx_status; + bool i2s_rx_status; + + spinlock_t lock; +}; + +static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) +{ + return snd_soc_dai_get_drvdata(dai); +} + +static inline void i2s_writel(struct rk_i2s_dev *i2s, u32 value, + unsigned int offset) +{ + writel_relaxed(value, i2s->regs + offset); +} + +static inline u32 i2s_readl(struct rk_i2s_dev *i2s, unsigned int offset) +{ + return readl_relaxed(i2s->regs + offset); +} + +static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +{ + u32 opr, xfer, clr; + unsigned long flags; + bool need_delay = false; + int try_cnt = 10; + + spin_lock_irqsave(&i2s->lock, flags); + + opr = i2s_readl(i2s, I2S_DMACR); + xfer = i2s_readl(i2s, I2S_XFER); + clr = i2s_readl(i2s, I2S_CLR); + + if (on) { + if ((opr & I2S_DMACR_TDE_ENABLE) == 0) { + opr |= I2S_DMACR_TDE_ENABLE; + i2s_writel(i2s, opr, I2S_DMACR); + } + + if ((xfer & I2S_XFER_TXS_START) == 0 || + (xfer & I2S_XFER_RXS_START) == 0) { + xfer |= I2S_XFER_TXS_START; + xfer |= I2S_XFER_RXS_START; + i2s_writel(i2s, xfer, I2S_XFER); + } + + i2s->i2s_tx_status = 1; + } else { + i2s->i2s_tx_status = 0; + + opr &= ~I2S_DMACR_TDE_ENABLE; + i2s_writel(i2s, opr, I2S_DMACR); + + if (i2s->i2s_rx_status == 0) { + xfer &= ~I2S_XFER_TXS_START; + xfer &= ~I2S_XFER_RXS_START; + i2s_writel(i2s, xfer, I2S_XFER); + + clr |= I2S_CLR_TXC; + clr |= I2S_CLR_RXC; + i2s_writel(i2s, clr, I2S_CLR); + + need_delay = true; + } + } + + spin_unlock_irqrestore(&i2s->lock, flags); + + if (need_delay) { + while (i2s_readl(i2s, I2S_CLR) && try_cnt) { + udelay(1); + try_cnt--; + } + } +} + +static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +{ + u32 opr, xfer, clr; + unsigned long flags; + bool need_delay = false; + int try_cnt = 10; + + spin_lock_irqsave(&i2s->lock, flags); + + opr = i2s_readl(i2s, I2S_DMACR); + xfer = i2s_readl(i2s, I2S_XFER); + clr = i2s_readl(i2s, I2S_CLR); + + if (on) { + if ((opr & I2S_DMACR_RDE_ENABLE) == 0) { + opr |= I2S_DMACR_RDE_ENABLE; + i2s_writel(i2s, opr, I2S_DMACR); + } + + if ((xfer & I2S_XFER_TXS_START) == 0 || + (xfer & I2S_XFER_RXS_START) == 0) { + xfer |= I2S_XFER_TXS_START; + xfer |= I2S_XFER_RXS_START; + i2s_writel(i2s, xfer, I2S_XFER); + } + + i2s->i2s_rx_status = 1; + } else { + i2s->i2s_rx_status = 0; + + opr &= ~I2S_DMACR_RDE_ENABLE; + i2s_writel(i2s, opr, I2S_DMACR); + + if (i2s->i2s_tx_status == 0) { + xfer &= ~I2S_XFER_RXS_START; + xfer &= ~I2S_XFER_TXS_START; + i2s_writel(i2s, xfer, I2S_XFER); + + clr |= I2S_CLR_TXC; + clr |= I2S_CLR_RXC; + i2s_writel(i2s, clr, I2S_CLR); + + need_delay = true; + } + } + + spin_unlock_irqrestore(&i2s->lock, flags); + + if (need_delay) { + while (i2s_readl(i2s, I2S_CLR) && try_cnt) { + udelay(1); + try_cnt--; + } + } +} + +static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct rk_i2s_dev *i2s = to_info(cpu_dai); + unsigned long flags; + u32 tx_ctl, rx_ctl; + u32 iis_ckr_value; + int ret = 0; + + spin_lock_irqsave(&i2s->lock, flags); + + iis_ckr_value = i2s_readl(i2s, I2S_CKR); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iis_ckr_value &= ~I2S_CKR_MSS_MASK; + iis_ckr_value |= I2S_CKR_MSS_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iis_ckr_value &= ~I2S_CKR_MSS_MASK; + iis_ckr_value |= I2S_CKR_MSS_MASTER; + break; + default: + ret = -EINVAL; + goto exit; + } + + i2s_writel(i2s, iis_ckr_value, I2S_CKR); + + tx_ctl = i2s_readl(i2s, I2S_TXCR); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + tx_ctl &= ~I2S_TXCR_IBM_MASK; + tx_ctl |= I2S_TXCR_IBM_RSJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + tx_ctl &= ~I2S_TXCR_IBM_MASK; + tx_ctl |= I2S_TXCR_IBM_LSJM; + break; + case SND_SOC_DAIFMT_I2S: + tx_ctl &= ~I2S_TXCR_IBM_MASK; + tx_ctl |= I2S_TXCR_IBM_NORMAL; + break; + default: + ret = -EINVAL; + goto exit; + } + + i2s_writel(i2s, tx_ctl, I2S_TXCR); + + rx_ctl = i2s_readl(i2s, I2S_RXCR); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + rx_ctl &= ~I2S_RXCR_IBM_MASK; + rx_ctl |= I2S_RXCR_IBM_RSJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + rx_ctl &= ~I2S_RXCR_IBM_MASK; + rx_ctl |= I2S_RXCR_IBM_LSJM; + break; + case SND_SOC_DAIFMT_I2S: + rx_ctl &= ~I2S_RXCR_IBM_MASK; + rx_ctl |= I2S_RXCR_IBM_NORMAL; + break; + default: + ret = -EINVAL; + goto exit; + } + + i2s_writel(i2s, rx_ctl, I2S_RXCR); + +exit: + spin_unlock_irqrestore(&i2s->lock, flags); + + return ret; +} + +static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct rk_i2s_dev *i2s = to_info(dai); + unsigned long flags; + u32 tx_ctl, rx_ctl; + u32 dmacr; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = &i2s->playback_dma_data; + else + dai->capture_dma_data = &i2s->capture_dma_data; + + spin_lock_irqsave(&i2s->lock, flags); + + tx_ctl = i2s_readl(i2s, I2S_TXCR); + tx_ctl &= ~I2S_TXCR_VDW_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + tx_ctl |= I2S_TXCR_VDW(8); + break; + case SNDRV_PCM_FORMAT_S16_LE: + tx_ctl |= I2S_TXCR_VDW(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + tx_ctl |= I2S_TXCR_VDW(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + tx_ctl |= I2S_TXCR_VDW(24); + break; + case SNDRV_PCM_FORMAT_S32_LE: + tx_ctl |= I2S_TXCR_VDW(32); + break; + } + + i2s_writel(i2s, tx_ctl, I2S_TXCR); + + rx_ctl = i2s_readl(i2s, I2S_TXCR); + rx_ctl &= ~I2S_TXCR_VDW_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + rx_ctl |= I2S_TXCR_VDW(8); + break; + case SNDRV_PCM_FORMAT_S16_LE: + rx_ctl |= I2S_TXCR_VDW(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + rx_ctl |= I2S_TXCR_VDW(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + rx_ctl |= I2S_TXCR_VDW(24); + break; + case SNDRV_PCM_FORMAT_S32_LE: + rx_ctl |= I2S_TXCR_VDW(32); + break; + } + + i2s_writel(i2s, rx_ctl, I2S_RXCR); + + dmacr = i2s_readl(i2s, I2S_DMACR); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dmacr = ((dmacr & ~I2S_DMACR_TDL_MASK) | + I2S_DMACR_TDL(1) | + I2S_DMACR_TDE_ENABLE); + else + dmacr = ((dmacr & ~I2S_DMACR_RDL_MASK) | + I2S_DMACR_RDL(1) | + I2S_DMACR_RDE_ENABLE); + + i2s_writel(i2s, dmacr, I2S_DMACR); + + spin_unlock_irqrestore(&i2s->lock, flags); + + return 0; +} + +static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct rk_i2s_dev *i2s = to_info(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_snd_rxctrl(i2s, 1); + else + rockchip_snd_txctrl(i2s, 1); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_snd_rxctrl(i2s, 0); + else + rockchip_snd_txctrl(i2s, 0); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct rk_i2s_dev *i2s = to_info(cpu_dai); + + clk_set_rate(i2s->clk, freq); + + return 0; +} + +static int rockchip_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct rk_i2s_dev *i2s = to_info(cpu_dai); + unsigned long flags; + u32 reg; + int ret; + + spin_lock_irqsave(&i2s->lock, flags); + + reg = i2s_readl(i2s, I2S_CKR); + switch (div_id) { + case ROCKCHIP_DIV_BCLK: + reg &= ~I2S_CKR_TSD_MASK; + reg |= I2S_CKR_TSD(div); + reg &= ~I2S_CKR_RSD_MASK; + reg |= I2S_CKR_RSD(div); + break; + case ROCKCHIP_DIV_MCLK: + reg &= ~I2S_CKR_MDIV_MASK; + reg |= I2S_CKR_MDIV(div); + break; + default: + ret = -EINVAL; + goto exit; + } + + i2s_writel(i2s, reg, I2S_CKR); + +exit: + spin_unlock_irqrestore(&i2s->lock, flags); + + return ret; +} + +static struct snd_soc_dai_ops rockchip_i2s_dai_ops = { + .trigger = rockchip_i2s_trigger, + .hw_params = rockchip_i2s_hw_params, + .set_fmt = rockchip_i2s_set_fmt, + .set_clkdiv = rockchip_i2s_set_clkdiv, + .set_sysclk = rockchip_i2s_set_sysclk, +}; + +#define ROCKCHIP_I2S_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define ROCKCHIP_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_driver rockchip_i2s_dai[] = { + { + .name = "rockchip-i2s.0", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = ROCKCHIP_I2S_STEREO_RATES, + .formats = ROCKCHIP_I2S_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = ROCKCHIP_I2S_STEREO_RATES, + .formats = ROCKCHIP_I2S_FORMATS, + }, + .ops = &rockchip_i2s_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rockchip-i2s.1", + .id = 1, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = ROCKCHIP_I2S_STEREO_RATES, + .formats = ROCKCHIP_I2S_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = ROCKCHIP_I2S_STEREO_RATES, + .formats = ROCKCHIP_I2S_FORMATS, + }, + .ops = &rockchip_i2s_dai_ops, + .symmetric_rates = 1, + }, +}; + +static const struct snd_soc_component_driver rockchip_i2s_component = { + .name = DRV_NAME, +}; + +#ifdef CONFIG_PM_RUNTIME +static int i2s_runtime_suspend(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + + return 0; +} + +static int i2s_runtime_resume(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + + return clk_prepare_enable(i2s->clk); +} +#else +#define i2s_runtime_suspend NULL +#define i2s_runtime_resume NULL +#endif + +static int rockchip_i2s_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct rk_i2s_dev *i2s; + struct resource *res; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(struct rk_i2s_dev), + GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(i2s->regs)) { + dev_err(&pdev->dev, "Could not get I2S source, %p\n", + i2s->regs); + return PTR_ERR(i2s->regs); + } + + i2s->playback_dma_data.addr = res->start + I2S_TXDR; + i2s->playback_dma_data.addr_width = 4; + i2s->playback_dma_data.maxburst = 1; + + i2s->capture_dma_data.addr = res->start + I2S_RXDR; + i2s->capture_dma_data.addr_width = 4; + i2s->capture_dma_data.maxburst = 1; + + i2s->i2s_tx_status = false; + i2s->i2s_rx_status = false; + + spin_lock_init(&i2s->lock); + + ret = rockchip_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM\n"); + goto err_pcm_register; + } + + /* Must success to get some clocks from dt */ + i2s->hclk = devm_clk_get(&pdev->dev, "i2s_hclk"); + if (IS_ERR(i2s->hclk)) { + dev_err(&pdev->dev, "Could not get i2s_hclk\n"); + return PTR_ERR(i2s->hclk); + } + + i2s->clk = devm_clk_get(&pdev->dev, "i2s_clk"); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "Could not get i2s_clk\n"); + return PTR_ERR(i2s->clk); + } + + ret = clk_prepare_enable(i2s->hclk); + if (ret) { + dev_err(&pdev->dev, "Could not enable i2s_hclk\n"); + return ret; + } + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(&pdev->dev, "Could not enable i2s_clk\n"); + goto err_hclk_disable; + } + + /* Try to set the I2S Channel id from dt */ + pdev->id = of_alias_get_id(np, "i2s"); + dev_set_name(&pdev->dev, "%s.%d", + pdev->dev.driver->name, + pdev->id); + + i2s->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, i2s); + + ret = devm_snd_soc_register_component(&pdev->dev, + &rockchip_i2s_component, + &rockchip_i2s_dai[pdev->id], 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI\n"); + goto err_clk_disable; + } + + pm_runtime_enable(&pdev->dev); + + return 0; + +err_clk_disable: + clk_disable_unprepare(i2s->clk); +err_hclk_disable: + clk_disable_unprepare(i2s->hclk); +err_pcm_register: + rockchip_pcm_platform_unregister(&pdev->dev); + + return ret; +} + +static int rockchip_i2s_remove(struct platform_device *pdev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(&pdev->dev); + + clk_disable_unprepare(i2s->clk); + clk_disable_unprepare(i2s->hclk); + rockchip_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static const struct of_device_id rockchip_i2s_match[] = { + { + .compatible = "rockchip,rk3066-i2s" + }, +}; +MODULE_DEVICE_TABLE(of, rockchip_i2s_match); + +static const struct dev_pm_ops rockchip_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(i2s_runtime_suspend, + i2s_runtime_resume, NULL) +}; + +static struct platform_driver rockchip_i2s_driver = { + .probe = rockchip_i2s_probe, + .remove = rockchip_i2s_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rockchip_i2s_match), + .pm = &rockchip_i2s_pm_ops, + }, +}; +module_platform_driver(rockchip_i2s_driver); + +MODULE_DESCRIPTION("ROCKCHIP IIS ASoC Interface"); +MODULE_AUTHOR("jianqun <jay.xu(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c new file mode 100644 index 0000000..454c609 --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -0,0 +1,64 @@ +/* sound/soc/rockchip/pcm.c + * + * ALSA SoC Audio Layer - Rockchip I2S Controller driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "pcm.h" + +static const struct snd_pcm_hardware rockchip_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = 128*1024, + .period_bytes_min = 64, + .period_bytes_max = 2048*4, + .periods_min = 3, + .periods_max = 128, + .fifo_size = 16, +}; + +static const struct snd_dmaengine_pcm_config rockchip_dmaengine_pcm_config = { + .pcm_hardware = &rockchip_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = NULL, + .prealloc_buffer_size = PAGE_SIZE * 8, +}; + +int rockchip_pcm_platform_register(struct device *dev) +{ + return snd_dmaengine_pcm_register(dev, &rockchip_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT| + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); +} +EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); + +void rockchip_pcm_platform_unregister(struct device *dev) +{ + snd_dmaengine_pcm_unregister(dev); +} +EXPORT_SYMBOL_GPL(rockchip_pcm_platform_unregister); + +MODULE_DESCRIPTION("ROCKCHIP PCM ASoC Interface"); +MODULE_AUTHOR("jianqun <jay.xu(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ICE1712: add surround71 pcm definition
by Alexander E. Patrakov 01 Jul '14

01 Jul '14

The M-Audio Delta 1010 card has 7.1 analog output, but no ready-made pcm definition to use it. Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> Reported-and-tested-by: Matt Zagrabelny <mzagrabe(a)d.umn.edu> --- src/conf/cards/ICE1712.conf | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/src/conf/cards/ICE1712.conf b/src/conf/cards/ICE1712.conf index 398fa7a..db62684 100644 --- a/src/conf/cards/ICE1712.conf +++ b/src/conf/cards/ICE1712.conf @@ -78,6 +78,7 @@ ICE1712.pcm.surround40.0 { <confdir:pcm/surround41.conf> <confdir:pcm/surround50.conf> <confdir:pcm/surround51.conf> +<confdir:pcm/surround71.conf> ICE1712.pcm.surround51.0 { @args [ CARD ] @@ -98,6 +99,27 @@ ICE1712.pcm.surround51.0 { slave.channels 10 } +ICE1712.pcm.surround71.0 { + @args [ CARD ] + @args.CARD { + type string + } + type route + ttable.0.0 1 + ttable.1.1 1 + ttable.2.2 1 + ttable.3.3 1 + ttable.4.4 1 + ttable.5.5 1 + ttable.6.6 1 + ttable.7.7 1 + slave.pcm { + type hw + card $CARD + } + slave.channels 10 +} + <confdir:pcm/iec958.conf> ICE1712.pcm.iec958.0 { -- 2.0.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: trident: Remove unused variable in trident_main.c
by Sachin Kamat 01 Jul '14

01 Jul '14

'private_data' is not used in the function. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/pci/trident/trident_main.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 1272c18a2544..da875dced2ef 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3880,14 +3880,12 @@ void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voi { unsigned long flags; void (*private_free)(struct snd_trident_voice *); - void *private_data; if (voice == NULL || !voice->use) return; snd_trident_clear_voices(trident, voice->number, voice->number); spin_lock_irqsave(&trident->voice_alloc, flags); private_free = voice->private_free; - private_data = voice->private_data; voice->private_free = NULL; voice->private_data = NULL; if (voice->pcm) -- 1.7.9.5

2 2

[alsa-devel] [PATCH (hda-emu)] Adapt to new snd_hda_calc_stream_format
by David Henningsson 01 Jul '14

01 Jul '14

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 23 +++++++++++++++++++++++ hda-emu.c | 6 +++++- 2 files changed, 28 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 31b31cd..fe5c0e2 100644 --- a/configure.ac +++ b/configure.ac @@ -488,6 +488,29 @@ if test "$STREAM_FORMAT_WITH_SPDIF" = "1"; then fi fi +AC_MSG_CHECKING(for snd_hda_calc_stream_format with codec) +ac_save_CFLAGS="$CFLAGS" +CFLAGS="$CFLAGS -Iinclude" +STREAM_FORMAT_WITH_CODEC= +AC_TRY_COMPILE([ +#include "hda-types.h" +#include "hda-log.h" +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include "hda/hda_codec.h" +],[ + snd_hda_calc_stream_format(0, 0, 0, 0, 0, 0); +], + [AC_MSG_RESULT(yes); STREAM_FORMAT_WITH_CODEC="1"], + [AC_MSG_RESULT(no)], + [AC_MSG_RESULT(unknown)] +) +CFLAGS=$ac_save_CFLAGS +if test "$STREAM_FORMAT_WITH_CODEC" = "1"; then + AC_DEFINE(STREAM_FORMAT_WITH_CODEC) +fi + HAVE_COMMON_PREPARE= AC_MSG_CHECKING(for common snd_hda_codec_prepare/cleanup) if grep -q snd_hda_codec_prepare hda/hda_codec.h; then diff --git a/hda-emu.c b/hda-emu.c index b564e34..e8309d5 100644 --- a/hda-emu.c +++ b/hda-emu.c @@ -798,7 +798,11 @@ void hda_test_pcm(int id, int op, int subid, hda_log(HDA_LOG_INFO, "Prepare PCM, rate=%d, channels=%d, " "format=%d bits\n", rate, channels, format); - format_val = snd_hda_calc_stream_format(rate, channels, + format_val = snd_hda_calc_stream_format( +#if defined(STREAM_FORMAT_WITH_CODEC) + _codec, +#endif + rate, channels, get_alsa_format(format), format #if defined(INDIVIDUAL_SPDIF_CTLS) || defined(STREAM_FORMAT_WITH_SPDIF) -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: tas2552: Support TI TAS2552 Amplifier
by Dan Murphy 01 Jul '14

01 Jul '14

Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Address RFC comments- Added regmap, and snd_soc calls removed debug code, address checkpatch errors -https://patchwork.kernel.org/patch/4378281/ .../devicetree/bindings/sound/tas2552.txt | 22 + include/sound/tas2552-plat.h | 25 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2552.c | 462 ++++++++++++++++++++ sound/soc/codecs/tas2552.h | 75 ++++ 6 files changed, 591 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 include/sound/tas2552-plat.h create mode 100644 sound/soc/codecs/tas2552.c create mode 100644 sound/soc/codecs/tas2552.h diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 0000000..58e931b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,22 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tas2552" - TAS2552 + +- reg - <int> - I2C slave address + +Optional properties: + +- power-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 0000000..c8120f2 --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int power_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0b9571c..cc09261 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -99,6 +99,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TAS2552 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE @@ -754,4 +755,8 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1c..33bc7228 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -162,6 +162,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -325,3 +326,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 0000000..228d5bb --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,462 @@ +/* + * ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/tas2552-plat.h> + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x16}, + {TAS2552_CFG_3, 0x5E}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xC8}, + {TAS2552_PDM_CFG, 0x02}, + {TAS2552_PGA_GAIN, 0x10}, + {TAS2552_BOOST_PT_CTRL, 0x0F}, + {TAS2552_LIMIT_LVL_CTRL, 0x0C}, + {TAS2552_LIMIT_RATE_HYS, 0x20}, + {TAS2552_CFG_2, 0xEA}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x05}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +struct tas2552_data { + struct mutex mutex; + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + unsigned char regs[TAS2552_VBAT_DATA]; + int power_gpio; + u8 power_state:1; +}; + +static int tas2552_power(struct tas2552_data *data, u8 power) +{ + int ret = 0; + + BUG_ON(data->tas2552_client == NULL); + + mutex_lock(&data->mutex); + if (power == data->power_state) + goto exit; + + if (power) { + if (data->power_gpio >= 0) + gpio_set_value(data->power_gpio, 1); + + data->power_state = 1; + } else { + if (data->power_gpio >= 0) + gpio_set_value(data->power_gpio, 0); + + data->power_state = 0; + } + +exit: + mutex_unlock(&data->mutex); + return ret; +} + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg; + + if (sw_shutdown) + cfg1_reg |= (sw_shutdown << 1); + else + cfg1_reg &= ~TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static void tas2552_init(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, TAS2552_CFG_1, 0x16); + snd_soc_write(codec, TAS2552_CFG_3, 0x5E); + snd_soc_write(codec, TAS2552_DOUT, 0x00); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, 0xC8); + snd_soc_write(codec, TAS2552_PDM_CFG, 0x02); + snd_soc_write(codec, TAS2552_PGA_GAIN, 0x10); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, 0x0F); + snd_soc_write(codec, TAS2552_LIMIT_LVL_CTRL, 0x0C); + snd_soc_write(codec, TAS2552_LIMIT_RATE_HYS, 0x20); + snd_soc_write(codec, TAS2552_CFG_2, 0xEA); +} + +static int tas2552_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u8 wclk_reg; + struct snd_soc_codec *codec = dai->codec; + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (params_rate(params)) { + case 8000: + wclk_reg = TAS2552_8KHZ; + break; + case 11025: + wclk_reg = TAS2552_11_12KHZ; + break; + case 16000: + wclk_reg = TAS2552_16KHZ; + break; + case 32000: + wclk_reg = TAS2552_32KHZ; + break; + case 22050: + case 24000: + wclk_reg = TAS2552_22_24KHZ; + break; + case 44100: + case 48000: + wclk_reg = TAS2552_44_48KHZ; + break; + case 96000: + wclk_reg = TAS2552_88_96KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, TAS2552_CFG_3, TAS2552_WCLK_MASK, wclk_reg); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + u8 serial_format; + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, + (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK), + serial_format); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format = 0x0; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format = TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format = TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format = TAS2552_DAIFMT_LEFT_J; + break; + + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, TAS2552_DATA_FORMAT_MASK, + serial_format); + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *data = dev_get_drvdata(dai->dev); + + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + + tas2552_sw_shutdown(data, 1); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + snd_soc_write(codec, TAS2552_PLL_CTRL_1, 0x10); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, 0x00); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, 0x00); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + tas2552_sw_shutdown(data, 0); + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg |= TAS2552_MUTE_MASK; + else + cfg1_reg &= ~TAS2552_MUTE_MASK; + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + return 0; +} + +static int tas2552_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_sw_shutdown(tas2552, 1); + tas2552_power(tas2552, 1); + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + tas2552_sw_shutdown(tas2552, 0); + + return 0; +} + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_sw_shutdown(tas2552, 1); + tas2552_power(tas2552, 0); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .startup = tas2552_startup, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552->codec = codec; + tas2552_power(tas2552, 1); + tas2552_init(codec); + + return 0; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_power(tas2552, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + struct tas2552_platform_data *pdata = client->dev.platform_data; + struct device_node *np = client->dev.of_node; + int ret; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + if (pdata) { + data->power_gpio = pdata->power_gpio; + } else if (np) { + data->power_gpio = of_get_named_gpio(np, "enable-gpio", 0); + } else { + dev_err(dev, "Platform data not set\n"); + return -ENODEV; + } + + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + dev_set_drvdata(&client->dev, data); + + mutex_init(&data->mutex); + + if (data->power_gpio >= 0) { + ret = devm_gpio_request(dev, data->power_gpio, "tas2552 enable"); + if (ret < 0) { + dev_err(dev, "Failed to request power GPIO (%d)\n", + data->power_gpio); + goto err_gpio; + } + gpio_direction_output(data->power_gpio, 0); + } + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return 0; + +err_gpio: + data->tas2552_client = NULL; + return ret; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552-codec", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 0000000..174c64d --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,75 @@ +/* + * ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_LIMIT_LVL_CTRL 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) +#define TAS2552_CLASSD_EN 0x80 + +#define TAS2552_PLL_ENABLE (1 << 3) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK 0x80 +#define TAS2552_BIT_CLK_MASK 0x40 +#define TAS2552_DATA_FORMAT_MASK 0x0c + +#define TAS2552_DAIFMT_DSP 0x04 +#define TAS2552_DAIFMT_RIGHT_J 0x08 +#define TAS2552_DAIFMT_LEFT_J 0x0c + +/* WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ 0x01 +#define TAS2552_16KHZ 0x02 +#define TAS2552_22_24KHZ 0x03 +#define TAS2552_32KHZ 0x04 +#define TAS2552_44_48KHZ 0x05 +#define TAS2552_88_96KHZ 0x06 +#define TAS2552_176_192KHZ 0x07 + +#endif -- 1.7.9.5

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[alsa-devel] [PATCH 0/5] ASoC: Fix trival sparse warnings
by Lars-Peter Clausen 01 Jul '14

01 Jul '14

Hi, This series fixes a couple of trivial issues that are reported when running the sparse codechecker on the ASoC tree. - Lars Lars-Peter Clausen (5): ASoC: tlv320aic31xx: Remove duplicate const ASoC: cs42xx8: Make of match table static ASoC: omap-pcm: Include omap-pcm.h ASoC: rcar: Fix dma direction type ASoC: sh/fsi: Make one-bit bitfields unsigned sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 21 ++++++++++----------- sound/soc/omap/omap-pcm.c | 1 + sound/soc/sh/fsi.c | 12 ++++++------ sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/rsnd.h | 2 +- 6 files changed, 20 insertions(+), 20 deletions(-) -- 1.8.0

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Re: [alsa-devel] Enable HP machine to enable capture mute led and output mute led
by Takashi Iwai 01 Jul '14

01 Jul '14

At Fri, 20 Jun 2014 15:55:58 +0800, Kailang wrote: > > Hi Takashi, > > Maybe you don't receive. > I repost again. Don't worry, it's been applied now. I've been off for a few days. thanks, Takashi > > BR, > Kailang > > > > Attach again. > > It include below infomations. > > Thanks. > > > >> At Wed, 18 Jun 2014 17:13:57 +0800, > >> Kailang wrote: > >>> > >>> Hi Takashi, > >>> > >>> Attach was HP mute led support patch. > >>> It use Line1 vref and Mic1 vref to control mute led for ALC282. > >>> Line1 vref to control output mute led. > >>> Mic1 vref to control input mute led. > >>> > >>> It use GPIO3 and Mic1 vref to control mute led for ALC280. > >>> GPIO3 to control output mute led. > >>> Mic1 vref to control input mute led. > >> > >> Could you include this kind of information in the patch changelog? > >> The changelog text is way too short. > >> > >>> For the DMI string for output mute led application. > >>> I already request from HP to support it in next release. > >>> > >>> HP_Mute_LED_0_A => use Mic1 vref to control mute led. > >>> Is the string right? > >>> I have tested upper string for Mic1 mute led. > >> > >> Yes, this has been working for some machines, AFAIK. > >> > >> > >> thanks, > >> > >> Takashi > >> > >> ------Please consider the environment before printing this e-mail. > [2 0001-new-hp-mute-led.patch <application/octet-stream (quoted-printable)>] >

2 1

[alsa-devel] Constant delay between starting a playback and a capture stream
by Max Schmidt 01 Jul '14

01 Jul '14

Hi all, I first wrote this email to the ALSA user mailing list since I didn't know if it's appropriate to write to the ALSA dev list as I'm not directly involved in the project. I was suggested to better try it here. So, I'm sorry if this email is an inconvenience. See below my original post. Thanks and greetings! Gesendet: Montag, 23. Juni 2014 um 18:20 Uhr Von: "Max Schmidt" <schmidti444(a)web.de> An: alsa-user(a)lists.sourceforge.net Betreff: Constant delay between starting a playback and a capture stream Hey all, first a big hello to everyone since I'm new to this mailing-list. I've had a look regarding this issue here and in generell in the internet but didn't find anything relating. So, sorry if I overlooked something. I've got a BeagleBone Black (ARM Cortex A 8 µP) with an Audio-Cape (using McAsp and ALSA-Davinci-Drivers) and wrote an mmap based playback-capture application (both devices "hw:0,0"). The important thing is that the application needs a constant delay (not necessarily small) between the start of the playback and the capture stream. So when looping back the signal to the microfone the first played sample e.g. always (even after a restart of the BeagleBone or the app) has a delay in the capture stream of exactly 80 samples (when e.g. played with 48 kHz), and once measured can be seen as constant. To realize that I use the ALSA API function "snd_pcm_link(c_handle, p_handle)". When starting the playback stream (and therefore linked capture stream as welll) manually its buffer already is filled. There is no buffer underrun/overrun recovery. It already works quite fine but I still have some questions and it would be really nice if someone can help: 1) Looking at a plot of multiple measurements of looped back and captured square waves (or sines) there still is a jitter of about two to four samples (at 48 kHz; which is about 40 to 80 ms), no matter if I run it as RT-app (energy save modes etc. disabled, just important Processes have a higher Priority e.g. EDMA) or normal app. Please correct me if I'm wrong, but as far as I understand when the streams are linked, at start the processor is going through a linked list triggering all linked streams. And the trigger start is an atomic process so shouldn't get interrupted. Shouldn't it always take the same time between starting the playback and capture stream then? And if yes, where could the variable start delay come from? 2) Looking at the time stamp of both streams it tells that the difference between the start trigger normally is between 2 to 7 ms. What, I think, does not really fit to the observation written above since then there normally should be no big sample jitter at 48 kHz. Are the time stamp values not precise enough (actually how precise are the time stamps, just that they can show microseconds does not imply that the resolution really is microseconds, or am I wrong?) or are they correct and the difference in latency seen in measured plots comes from somewhere else? Delay due to Hardware should be constant so the issue must have something to do with ALSA or Linux. Would be nice if someone can help or has an idea! Tipps for an improvement are wlecome as well! Many thanks and cheers!

2 2

[alsa-devel] [PATCH 1/2] ASoC: max98090: Add max98091 compatible string
by Tushar Behera 30 Jun '14

30 Jun '14

From: Wonjoon Lee <woojoo.lee(a)samsung.com> The MAX98091 CODEC is the same as MAX98090 CODEC, but with an extra microphone. Existing driver for MAX98090 CODEC already has support for MAX98091 CODEC. Adding proper compatible string so that MAX98091 CODEC can be specified from device tree. Signed-off-by: Wonjoon Lee <woojoo.lee(a)samsung.com> Signed-off-by: Doug Anderson <dianders(a)chromium.org> Signed-off-by: Tushar Behera <tushar.b(a)samsung.com> --- Picked from https://chromium-review.googlesource.com/#/c/184091/ .../devicetree/bindings/sound/max98090.txt | 2 +- sound/soc/codecs/max98090.c | 2 ++ 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index a5e63fa..c454e67 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "maxim,max98090". +- compatible : "maxim,max98090" or "maxim,max98091". - reg : The I2C address of the device. diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7..4f5534d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2460,12 +2460,14 @@ static const struct dev_pm_ops max98090_pm = { static const struct i2c_device_id max98090_i2c_id[] = { { "max98090", MAX98090 }, + { "max98091", MAX98091 }, { } }; MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); static const struct of_device_id max98090_of_match[] = { { .compatible = "maxim,max98090", }, + { .compatible = "maxim,max98091", }, { } }; MODULE_DEVICE_TABLE(of, max98090_of_match); -- 1.7.9.5

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[alsa-devel] [PATCH RFC/RFT] ASoC: rt5677: Convert to use rl6231_pll_calc
by Axel Lin 30 Jun '14

30 Jun '14

The implementation of rt5677_pll_calc() has the same logic of rl6231_pll_calc(). The only difference is the lower boundary checking for freq_in. This patch calls rl6231_pll_calc() instead of open-coded. The k_bp of struct rt5677_pll_code is always false, thus also remove the code to check pll_code.k_bp. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Oder, This patch removes the usage of k_bp from struct rt5677_pll_code because it is always false. I think if k_bp is necessay in the further development, maybe it's better to add it to rl6231_pll_code. How do you think? Regards, Axel sound/soc/codecs/rt5677.c | 69 ++++++----------------------------------------- sound/soc/codecs/rt5677.h | 8 ------ 2 files changed, 8 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 45f99a8..e7c401f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2988,62 +2988,12 @@ static int rt5677_set_dai_sysclk(struct snd_soc_dai *dai, * Returns 0 for success or negative error code. */ static int rt5677_pll_calc(const unsigned int freq_in, - const unsigned int freq_out, struct rt5677_pll_code *pll_code) + const unsigned int freq_out, struct rl6231_pll_code *pll_code) { - int max_n = RT5677_PLL_N_MAX, max_m = RT5677_PLL_M_MAX; - int k, red, n_t, pll_out, in_t; - int n = 0, m = 0, m_t = 0; - int out_t, red_t = abs(freq_out - freq_in); - bool m_bp = false, k_bp = false; - - if (RT5677_PLL_INP_MAX < freq_in || RT5677_PLL_INP_MIN > freq_in) + if (RT5677_PLL_INP_MIN > freq_in) return -EINVAL; - k = 100000000 / freq_out - 2; - if (k > RT5677_PLL_K_MAX) - k = RT5677_PLL_K_MAX; - for (n_t = 0; n_t <= max_n; n_t++) { - in_t = freq_in / (k + 2); - pll_out = freq_out / (n_t + 2); - if (in_t < 0) - continue; - if (in_t == pll_out) { - m_bp = true; - n = n_t; - goto code_find; - } - red = abs(in_t - pll_out); - if (red < red_t) { - m_bp = true; - n = n_t; - m = m_t; - if (red == 0) - goto code_find; - red_t = red; - } - for (m_t = 0; m_t <= max_m; m_t++) { - out_t = in_t / (m_t + 2); - red = abs(out_t - pll_out); - if (red < red_t) { - m_bp = false; - n = n_t; - m = m_t; - if (red == 0) - goto code_find; - red_t = red; - } - } - } - pr_debug("Only get approximation about PLL\n"); - -code_find: - - pll_code->m_bp = m_bp; - pll_code->k_bp = k_bp; - pll_code->m_code = m; - pll_code->n_code = n; - pll_code->k_code = k; - return 0; + return rl6231_pll_calc(freq_in, freq_out, pll_code); } static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, @@ -3051,7 +3001,7 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, { struct snd_soc_codec *codec = dai->codec; struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - struct rt5677_pll_code pll_code; + struct rl6231_pll_code pll_code; int ret; if (source == rt5677->pll_src && freq_in == rt5677->pll_in && @@ -3109,15 +3059,12 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, return ret; } - dev_dbg(codec->dev, "m_bypass=%d k_bypass=%d m=%d n=%d k=%d\n", - pll_code.m_bp, pll_code.k_bp, - (pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code, - (pll_code.k_bp ? 0 : pll_code.k_code)); + dev_dbg(codec->dev, "m_bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); regmap_write(rt5677->regmap, RT5677_PLL1_CTRL1, - pll_code.n_code << RT5677_PLL_N_SFT | - pll_code.k_bp << RT5677_PLL_K_BP_SFT | - (pll_code.k_bp ? 0 : pll_code.k_code)); + pll_code.n_code << RT5677_PLL_N_SFT | pll_code.k_code); regmap_write(rt5677->regmap, RT5677_PLL1_CTRL2, (pll_code.m_bp ? 0 : pll_code.m_code) << RT5677_PLL_M_SFT | pll_code.m_bp << RT5677_PLL_M_BP_SFT); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index af4e9c7..08252e7 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1425,14 +1425,6 @@ enum { RT5677_AIFS, }; -struct rt5677_pll_code { - bool m_bp; /* Indicates bypass m code or not. */ - bool k_bp; /* Indicates bypass k code or not. */ - int m_code; - int n_code; - int k_code; -}; - struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; -- 1.8.3.2

3 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix S24_LE and U24_LE support
by Peter Ujfalusi 30 Jun '14

30 Jun '14

In case of S24_LE/U24_LE modes we expect 24bits on the bus while the samples are stored and transferred in memory on 32bits (lower 3 bytes of the 4 bytes). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9afb14629a17..6d5de26eae2f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -720,6 +720,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: + dma_params->data_type = 4; + word_length = 24; + break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; -- 2.0.0

3 2

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Correct S24_3LE support
by Peter Ujfalusi 30 Jun '14

30 Jun '14

Correct the hw_params callback to configure the codec correctly in case of S24_3LE format since in case of S24_3LE the codec has been configured to 16bit format mode. S24_LE is not defined as supported format for the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e12fafbb1e09..5360772bc1ad 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: -- 2.0.0

2 2

[alsa-devel] [PATCH] ASoC: rt5640: add ASRC support
by bardliao＠realtek.com 30 Jun '14

30 Jun '14

From: Bard Liao <bardliao(a)realtek.com> This patch add ASRC support for rt5640 series codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 79 ++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5640.h | 2 ++ 2 files changed, 80 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6bc6efd..3245ec4 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -341,6 +341,43 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } } +static const char * const rt5640_asrc_mode[] = { + "disable", "enable" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5640_asrc_enum, SND_SOC_NOPM, + 0, rt5640_asrc_mode); + +static int rt5640_asrc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + rt5640->asrc_en = ucontrol->value.integer.value[0]; + + if (rt5640->asrc_en) { + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x70, 0x70); + snd_soc_update_bits(codec, RT5640_JD_CTRL, 0x3, 0x3); + } else { + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x70, 0x0); + snd_soc_update_bits(codec, RT5640_JD_CTRL, 0x3, 0x0); + } + + return 0; +} + +static int rt5640_asrc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = rt5640->asrc_en; + + return 0; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -436,6 +473,10 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_ENUM("DAC IF1 Data Switch", rt5640_if1_dac_enum), SOC_ENUM("ADC IF2 Data Switch", rt5640_if2_adc_enum), SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), + + /* ASRC */ + SOC_ENUM_EXT("ASRC Switch", rt5640_asrc_enum, + rt5640_asrc_get, rt5640_asrc_put), }; static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { @@ -1020,9 +1061,37 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_asrc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(w->codec); + + if (!rt5640->asrc_en) + return 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, RT5640_ASRC_1, 0x9b00); + snd_soc_write(w->codec, RT5640_ASRC_2, 0xf800); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, RT5640_ASRC_1, 0x0); + snd_soc_write(w->codec, RT5640_ASRC_2, 0x0); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY_S("ASRC Function", 1, SND_SOC_NOPM, + 0, 0, rt5640_asrc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), /* Input Side */ /* micbias */ SND_SOC_DAPM_SUPPLY("LDO2", RT5640_PWR_ANLG1, @@ -1276,6 +1345,9 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { + {"I2S1", NULL, "ASRC Function"}, + {"I2S2", NULL, "ASRC Function"}, + {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1635,7 +1707,12 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, int dai_sel, pre_div, bclk_ms, frame_size; rt5640->lrck[dai->id] = params_rate(params); - pre_div = rl6231_get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); + if (rt5640->asrc_en) + pre_div = 0; + else + pre_div = rl6231_get_clk_info(rt5640->sysclk, + rt5640->lrck[dai->id]); + if (pre_div < 0) { dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", rt5640->lrck[dai->id], dai->id); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 58ebe96..0b39178 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2095,6 +2095,8 @@ struct rt5640_priv { int pll_out; bool hp_mute; + + int asrc_en; }; #endif -- 1.8.1.1.439.g50a6b54

4 4

[alsa-devel] [PATCH] ASoC: sgtl5000: fix devres handling
by Russell King 30 Jun '14

30 Jun '14

The sgtl5000 uses a two-stage initialisation process. The first stage is when the platform driver is probed, where some resources are found and initialised. The second stage is via the codec driver's probe function, where regulators are found and initialised using the managed resource support. The problem here is that this works fine until the codec driver is unbound. When this occurs, sgtl5000_remove() is called which is a no-op as far as the managed resource code is concerned. The regulators remain allocated, and their pointers in sgtl5000_priv remain valid. If the codec is now re-probed, it will again try and find the regulators, which will now be busy. This will fail. That's not the only problem - if using the LDO regulator, the regulator is unregistered from the regulator core code, but it still has a user. When the user is cleaned up (eg, by removing the module) it hits the free'd regulator, and this can oops the kernel. This bug was originally introduced by 63e54cd9caa3ce ("ASoC: sgtl5000: Use devm_regulator_bulk_get()"). Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- You may wish to give some consideration to whether ASoC core code should be doing this instead, this is probably a very common source of brokenness where two-stage driver initialisation exists. These kinds of issues are why I consider (as a general rule) that two-stage driver initialisation is really bad. sound/soc/codecs/sgtl5000.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 96ce061c1f10..3bbd89065313 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1305,6 +1305,9 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) int ret; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + if (!devres_open_group(codec->dev, NULL, GFP_KERNEL)) + return -ENOMEM; + ret = sgtl5000_enable_regulators(codec); if (ret) return ret; @@ -1362,6 +1365,9 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + + devres_release_group(codec->dev, NULL); + ldo_regulator_remove(codec); return ret; @@ -1375,6 +1381,9 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + + devres_release_group(codec->dev, NULL); + ldo_regulator_remove(codec); return 0; -- 1.8.3.1

3 2

[alsa-devel] [PATCH 0/7] HD-audio cleanup patches for 3.17
by Takashi Iwai 30 Jun '14

30 Jun '14

Hi, below is a series of cleanup patches for HD-audio. Most of changes are to move the PCI-controller specific code from the common code to hda_intel.c. A slight intesive change is the introduction of get_position and get_delay callbacks that replace the existing position_fix[] quirks although there should be no functional changes. Takashi

1 7

[alsa-devel] [PATCH] ALSA: hda - Fix invalid function call in snd_hda_add_vmaster()
by Takashi Iwai 30 Jun '14

30 Jun '14

The recent commit [6194b99d: ALSA: hda - Kill the rest of snd_print*() usages] changed the callback map_slaves(), but one call was forgotten to be replaced due to the cast, which leads to kernel Oops due to invalid function. This patch replaces it with a proper function. Fixes: 6194b99de9f5 ('ALSA: hda - Kill the rest of snd_print*() usages') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d91cb7f5961a..47a617786fca 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2831,6 +2831,12 @@ static int init_slave_unmute(struct hda_codec *codec, return put_kctl_with_value(slave, 1); } +static int add_slave(struct hda_codec *codec, + void *data, struct snd_kcontrol *slave) +{ + return snd_ctl_add_slave(data, slave); +} + /** * snd_hda_add_vmaster - create a virtual master control and add slaves * @codec: HD-audio codec @@ -2873,8 +2879,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, if (err < 0) return err; - err = map_slaves(codec, slaves, suffix, - (map_slave_func_t)snd_ctl_add_slave, kctl); + err = map_slaves(codec, slaves, suffix, add_slave, kctl); if (err < 0) return err; -- 2.0.0

1 0

[alsa-devel] [PATCH] ASoC: cs4265: Change return values to boolean.
by Paul Handrigan 30 Jun '14

30 Jun '14

The cs4265_volatile_register reutrns a bool. The function now returns true or false vs 1 and 0. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index c9c04d2..4c4bf21 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -87,9 +87,9 @@ static bool cs4265_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case CS4265_INT_STATUS: - return 1; + return true; default: - return 0; + return false; } } -- 1.9.1

2 1

[alsa-devel] [PATCHv2 0/9] DT support for N900 soundcard (rx51-audio)
by Sebastian Reichel 30 Jun '14

30 Jun '14

Hi, This patchset adds DT support in rx51-audio. I tested it on my Nokia N900 and was able to play a wav file using aplay. I have not tested the whole functionality, but output via speakers, headphones and the related enable controls seem to work. Changes since PATCHv1 [0]: * drop "ASoC: tlv320aic3x: fix shared reset pin for DT" (already applied) * drop "ASoC: RX-51: Convert to table based DAPM setup" (similar patch was applied to -next) * remove useless "card->dev = NULL;" from first patch * split patches from into smaller chunks * introduce new patch, which fixes a checkpatch warning * use gpiod API, which reduces the DT specific bits of the driver * rebase to newer -next checkout [0] https://lkml.org/lkml/2014/4/5/92 -- Sebastian Sebastian Reichel (9): ASoC: omap: rx51: Use static const char * const arrays ASoC: omap: rx51: Add module alias ASoC: omap: rx51: Use devm_snd_soc_register_card ASoC: Allow Aux Codecs to be specified using DT ASoC: omap: rx51: omap_mcbsp_st_add_controls: add id parameter ASoC: omap: rx51: get GPIO numbers via gpiod API ASoC: omap: rx51: Add some error messages ASoC: omap: rx51: Add DT support DTS: OMAP3-N900: Add sound support .../devicetree/bindings/sound/nokia,rx51.txt | 27 +++ arch/arm/boot/dts/omap3-n900.dts | 17 ++ include/sound/soc.h | 13 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 226 +++++++++++++++------ sound/soc/soc-core.c | 68 ++++--- 7 files changed, 271 insertions(+), 87 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nokia,rx51.txt -- 1.9.2

4 24

[alsa-devel] [PATCH] ASoC: add RT5670 CODEC driver
by bardliao＠realtek.com 30 Jun '14

30 Jun '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds a minimum support of Realtek ALC5670 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Currently, only playback with headphone or line out and capture with digital mic are tested. We will add other patches for other functions. --- include/sound/rt5670.h | 27 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5670-dsp.h | 54 + sound/soc/codecs/rt5670.c | 2720 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670.h | 2000 ++++++++++++++++++++++++++++++ 6 files changed, 4809 insertions(+) create mode 100644 include/sound/rt5670.h create mode 100644 sound/soc/codecs/rt5670-dsp.h create mode 100644 sound/soc/codecs/rt5670.c create mode 100644 sound/soc/codecs/rt5670.h diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h new file mode 100644 index 0000000..bd31119 --- /dev/null +++ b/include/sound/rt5670.h @@ -0,0 +1,27 @@ +/* + * linux/sound/rt5670.h -- Platform data for RT5670 + * + * Copyright 2014 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5670_H +#define __LINUX_SND_RT5670_H + +struct rt5670_platform_data { + int jd_mode; + bool in2_diff; + + bool dmic_en; + unsigned int dmic1_data_pin; + /* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/ + unsigned int dmic2_data_pin; + /* 0 = GPIO8; 1 = IN3N; */ + unsigned int dmic3_data_pin; + /* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e1..379a0f3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,6 +78,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C + select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE @@ -445,9 +446,11 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5640=y default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y + default y if SND_SOC_RT5670=y default m if SND_SOC_RT5640=m default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m + default m if SND_SOC_RT5670=m config SND_SOC_RT5631 tristate @@ -461,6 +464,9 @@ config SND_SOC_RT5645 config SND_SOC_RT5651 tristate +config SND_SOC_RT5670 + tristate + config SND_SOC_RT5677 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b..c664878 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -72,6 +72,7 @@ snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o +snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o +obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o diff --git a/sound/soc/codecs/rt5670-dsp.h b/sound/soc/codecs/rt5670-dsp.h new file mode 100644 index 0000000..a34d0cd --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.h @@ -0,0 +1,54 @@ +/* + * rt5670-dsp.h -- RT5670 ALSA SoC DSP driver + * + * Copyright 2014 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5670_DSP_H__ +#define __RT5670_DSP_H__ + +#define RT5670_DSP_CTRL1 0xe0 +#define RT5670_DSP_CTRL2 0xe1 +#define RT5670_DSP_CTRL3 0xe2 +#define RT5670_DSP_CTRL4 0xe3 +#define RT5670_DSP_CTRL5 0xe4 + +/* DSP Control 1 (0xe0) */ +#define RT5670_DSP_CMD_MASK (0xff << 8) +#define RT5670_DSP_CMD_PE (0x0d << 8) /* Patch Entry */ +#define RT5670_DSP_CMD_MW (0x3b << 8) /* Memory Write */ +#define RT5670_DSP_CMD_MR (0x37 << 8) /* Memory Read */ +#define RT5670_DSP_CMD_RR (0x60 << 8) /* Register Read */ +#define RT5670_DSP_CMD_RW (0x68 << 8) /* Register Write */ +#define RT5670_DSP_REG_DATHI (0x26 << 8) /* High Data Addr */ +#define RT5670_DSP_REG_DATLO (0x25 << 8) /* Low Data Addr */ +#define RT5670_DSP_CLK_MASK (0x3 << 6) +#define RT5670_DSP_CLK_SFT 6 +#define RT5670_DSP_CLK_768K (0x0 << 6) +#define RT5670_DSP_CLK_384K (0x1 << 6) +#define RT5670_DSP_CLK_192K (0x2 << 6) +#define RT5670_DSP_CLK_96K (0x3 << 6) +#define RT5670_DSP_BUSY_MASK (0x1 << 5) +#define RT5670_DSP_RW_MASK (0x1 << 4) +#define RT5670_DSP_DL_MASK (0x3 << 2) +#define RT5670_DSP_DL_0 (0x0 << 2) +#define RT5670_DSP_DL_1 (0x1 << 2) +#define RT5670_DSP_DL_2 (0x2 << 2) +#define RT5670_DSP_DL_3 (0x3 << 2) +#define RT5670_DSP_I2C_AL_16 (0x1 << 1) +#define RT5670_DSP_CMD_EN (0x1) + +struct rt5670_dsp_param { + u16 cmd_fmt; + u16 addr; + u16 data; + u8 cmd; +}; + +#endif /* __RT5670_DSP_H__ */ + diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c new file mode 100644 index 0000000..b30f9f0 --- /dev/null +++ b/sound/soc/codecs/rt5670.c @@ -0,0 +1,2720 @@ +/* + * rt5670.c -- RT5670 ALSA SoC audio codec driver + * + * Copyright 2014 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5670.h> + +#include "rl6231.h" +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define RT5670_DEVICE_ID 0x6271 + +#define RT5670_PR_RANGE_BASE (0xff + 1) +#define RT5670_PR_SPACING 0x100 + +#define RT5670_PR_BASE (RT5670_PR_RANGE_BASE + (0 * RT5670_PR_SPACING)) + +static const struct regmap_range_cfg rt5670_ranges[] = { + { .name = "PR", .range_min = RT5670_PR_BASE, + .range_max = RT5670_PR_BASE + 0xf8, + .selector_reg = RT5670_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5670_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + { RT5670_PR_BASE + 0x14, 0x9a8a }, + { RT5670_PR_BASE + 0x38, 0x3ba1 }, + { RT5670_PR_BASE + 0x3d, 0x3640 }, +}; +#define RT5670_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5670_reg[] = { + { 0x00, 0x0000 }, + { 0x02, 0x8888 }, + { 0x03, 0x8888 }, + { 0x0a, 0x0001 }, + { 0x0b, 0x0827 }, + { 0x0c, 0x0000 }, + { 0x0d, 0x0008 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0011 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x1f, 0x2f2f }, + { 0x20, 0x0000 }, + { 0x26, 0x7860 }, + { 0x27, 0x7860 }, + { 0x28, 0x7871 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5656 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, + { 0x2e, 0x2f2f }, + { 0x2f, 0x1002 }, + { 0x30, 0x0000 }, + { 0x31, 0x5f00 }, + { 0x32, 0x0000 }, + { 0x33, 0x0000 }, + { 0x34, 0x0000 }, + { 0x35, 0x0000 }, + { 0x36, 0x0000 }, + { 0x37, 0x0000 }, + { 0x38, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe00f }, + { 0x4c, 0x5380 }, + { 0x4f, 0x0073 }, + { 0x52, 0x00d3 }, + { 0x53, 0xf0f0 }, + { 0x61, 0x0000 }, + { 0x62, 0x0001 }, + { 0x63, 0x00c3 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6f, 0x8000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1110 }, + { 0x74, 0x0e00 }, + { 0x75, 0x1505 }, + { 0x76, 0x0015 }, + { 0x77, 0x0c00 }, + { 0x78, 0x4000 }, + { 0x79, 0x0123 }, + { 0x7f, 0x1100 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0000 }, + { 0x86, 0x0008 }, + { 0x87, 0x0000 }, + { 0x88, 0x0000 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0000 }, + { 0x8c, 0x0007 }, + { 0x8d, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c06 }, + { 0x93, 0x0000 }, + { 0x94, 0x0000 }, + { 0x95, 0x0000 }, + { 0x97, 0x0000 }, + { 0x98, 0x0000 }, + { 0x99, 0x0000 }, + { 0x9a, 0x2184 }, + { 0x9b, 0x010a }, + { 0x9c, 0x0aea }, + { 0x9d, 0x000c }, + { 0x9e, 0x0400 }, + { 0xae, 0x7000 }, + { 0xaf, 0x0000 }, + { 0xb0, 0x6000 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0000 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x1813 }, + { 0xd0, 0x0690 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0001 }, + { 0xdc, 0x0049 }, + { 0xdd, 0x0009 }, + { 0xe6, 0x8000 }, + { 0xe7, 0x0000 }, + { 0xec, 0xb300 }, + { 0xed, 0x0000 }, + { 0xee, 0xb300 }, + { 0xef, 0x0000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x8010 }, + { 0xfb, 0x0033 }, + { 0xfc, 0x0080 }, +}; + +static bool rt5670_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) { + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5670_RESET: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_PRIV_DATA: + case RT5670_ASRC_5: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_EQ_CTRL1: + case RT5670_ALC_CTRL_1: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_INT_IRQ_ST: + case RT5670_IL_CMD: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5670_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) { + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5670_RESET: + case RT5670_HP_VOL: + case RT5670_LOUT1: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_IN2: + case RT5670_INL1_INR1_VOL: + case RT5670_DAC1_DIG_VOL: + case RT5670_DAC2_DIG_VOL: + case RT5670_DAC_CTRL: + case RT5670_STO1_ADC_DIG_VOL: + case RT5670_MONO_ADC_DIG_VOL: + case RT5670_STO2_ADC_DIG_VOL: + case RT5670_ADC_BST_VOL1: + case RT5670_ADC_BST_VOL2: + case RT5670_STO2_ADC_MIXER: + case RT5670_STO1_ADC_MIXER: + case RT5670_MONO_ADC_MIXER: + case RT5670_AD_DA_MIXER: + case RT5670_STO_DAC_MIXER: + case RT5670_DD_MIXER: + case RT5670_DIG_MIXER: + case RT5670_DSP_PATH1: + case RT5670_DSP_PATH2: + case RT5670_DIG_INF1_DATA: + case RT5670_DIG_INF2_DATA: + case RT5670_PDM_OUT_CTRL: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL2: + case RT5670_PDM1_DATA_CTRL3: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL2: + case RT5670_PDM2_DATA_CTRL3: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_REC_L1_MIXER: + case RT5670_REC_L2_MIXER: + case RT5670_REC_R1_MIXER: + case RT5670_REC_R2_MIXER: + case RT5670_HPO_MIXER: + case RT5670_MONO_MIXER: + case RT5670_OUT_L1_MIXER: + case RT5670_OUT_R1_MIXER: + case RT5670_LOUT_MIXER: + case RT5670_PWR_DIG1: + case RT5670_PWR_DIG2: + case RT5670_PWR_ANLG1: + case RT5670_PWR_ANLG2: + case RT5670_PWR_MIXER: + case RT5670_PWR_VOL: + case RT5670_PRIV_INDEX: + case RT5670_PRIV_DATA: + case RT5670_I2S4_SDP: + case RT5670_I2S1_SDP: + case RT5670_I2S2_SDP: + case RT5670_I2S3_SDP: + case RT5670_ADDA_CLK1: + case RT5670_ADDA_CLK2: + case RT5670_DMIC_CTRL1: + case RT5670_DMIC_CTRL2: + case RT5670_TDM_CTRL_1: + case RT5670_TDM_CTRL_2: + case RT5670_TDM_CTRL_3: + case RT5670_DSP_CLK: + case RT5670_GLB_CLK: + case RT5670_PLL_CTRL1: + case RT5670_PLL_CTRL2: + case RT5670_ASRC_1: + case RT5670_ASRC_2: + case RT5670_ASRC_3: + case RT5670_ASRC_4: + case RT5670_ASRC_5: + case RT5670_ASRC_7: + case RT5670_ASRC_8: + case RT5670_ASRC_9: + case RT5670_ASRC_10: + case RT5670_ASRC_11: + case RT5670_ASRC_12: + case RT5670_ASRC_13: + case RT5670_ASRC_14: + case RT5670_DEPOP_M1: + case RT5670_DEPOP_M2: + case RT5670_DEPOP_M3: + case RT5670_CHARGE_PUMP: + case RT5670_MICBIAS: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL1: + case RT5670_VAD_CTRL2: + case RT5670_VAD_CTRL3: + case RT5670_VAD_CTRL4: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_ADC_EQ_CTRL2: + case RT5670_EQ_CTRL1: + case RT5670_EQ_CTRL2: + case RT5670_ALC_DRC_CTRL1: + case RT5670_ALC_DRC_CTRL2: + case RT5670_ALC_CTRL_1: + case RT5670_ALC_CTRL_2: + case RT5670_ALC_CTRL_3: + case RT5670_JD_CTRL: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_INT_IRQ_ST: + case RT5670_GPIO_CTRL1: + case RT5670_GPIO_CTRL2: + case RT5670_GPIO_CTRL3: + case RT5670_SCRABBLE_FUN: + case RT5670_SCRABBLE_CTRL: + case RT5670_BASE_BACK: + case RT5670_MP3_PLUS1: + case RT5670_MP3_PLUS2: + case RT5670_ADJ_HPF1: + case RT5670_ADJ_HPF2: + case RT5670_HP_CALIB_AMP_DET: + case RT5670_SV_ZCD1: + case RT5670_SV_ZCD2: + case RT5670_IL_CMD: + case RT5670_IL_CMD2: + case RT5670_IL_CMD3: + case RT5670_DRC_HL_CTRL1: + case RT5670_DRC_HL_CTRL2: + case RT5670_ADC_MONO_HP_CTRL1: + case RT5670_ADC_MONO_HP_CTRL2: + case RT5670_ADC_STO2_HP_CTRL1: + case RT5670_ADC_STO2_HP_CTRL2: + case RT5670_JD_CTRL3: + case RT5670_JD_CTRL4: + case RT5670_DIG_MISC: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_GEN_CTRL2: + case RT5670_GEN_CTRL3: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5670_data_select[] = { + "Normal", "Swap", "left copy to right", "right copy to left" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_DAC_SEL_SFT, rt5670_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); + +static const struct snd_kcontrol_new rt5670_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 39, 0, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, + RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL, + RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5670_DAC2_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5670_CJ_CTRL1, + RT5670_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5670_IN2, + RT5670_BST_SFT1, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5670_INL1_INR1_VOL, + RT5670_INL_VOL_SFT, RT5670_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5670_ADC_BST_VOL1, + RT5670_STO1_ADC_L_BST_SFT, RT5670_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5670_ADC_BST_VOL1, + RT5670_STO2_ADC_L_BST_SFT, RT5670_STO2_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int idx = -EINVAL; + + idx = rl6231_calc_dmic_clk(rt5670->sysclk); + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5670_DMIC_CTRL1, + RT5670_DMIC_CLK_MASK, idx << RT5670_DMIC_CLK_SFT); + return idx; +} + +static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5670_GLB_CLK); + val &= RT5670_SCLK_SRC_MASK; + if (val == RT5670_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_codec *codec, + unsigned int reg, unsigned int shift) +{ + unsigned int val; + + val = (snd_soc_read(codec, reg) >> shift) & 0xf; + + switch (val) { + case 1 ... 4: + return 1; + default: + return 0; + } +} + +static int check_adc_sto1_asrc_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + + return is_using_asrc(codec, RT5670_ASRC_2, 0); +} + +static int check_adc_sto2_asrc_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + + return is_using_asrc(codec, RT5670_ASRC_5, 12); +} + +static int check_adc_monol_asrc_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + + return is_using_asrc(codec, RT5670_ASRC_3, 4); +} + +static int check_adc_monor_asrc_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + + return is_using_asrc(codec, RT5670_ASRC_3, 0); +} + +static int check_dac_sto_asrc_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + + return is_using_asrc(codec, RT5670_ASRC_2, 12); +} + +static int check_dac_monol_asrc_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + + return is_using_asrc(codec, RT5670_ASRC_2, 8); +} + +static int check_dac_monor_asrc_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + + return is_using_asrc(codec, RT5670_ASRC_2, 4); +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5670_DIG_MIXER, + RT5670_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5670_DIG_MIXER, + RT5670_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5670_rec_l_mix[] = { + SOC_DAPM_SINGLE("INL Switch", RT5670_REC_L2_MIXER, + RT5670_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST1_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_rec_r_mix[] = { + SOC_DAPM_SINGLE("INR Switch", RT5670_REC_R2_MIXER, + RT5670_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST1_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_OUT_L1_MIXER, + RT5670_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_OUT_R1_MIXER, + RT5670_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5670_HPO_MIXER, + RT5670_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvoll_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvolr_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX L Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX R Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpl_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL1 Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpr_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR1 Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new lout_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1, + RT5670_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1, + RT5670_R_MUTE_SFT, 1, 1); + +/* DAC1 L/R source */ /* MX-29 [9:8] [11:10] */ +static const char * const rt5670_dac1_src[] = { + "IF1 DAC", "IF2 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1l_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_L_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1l_mux = + SOC_DAPM_ENUM("DAC1 L source", rt5670_dac1l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1r_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_R_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1r_mux = + SOC_DAPM_ENUM("DAC1 R source", rt5670_dac1r_enum); + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +/* TODO Use SOC_VALUE_ENUM_SINGLE_DECL */ +static const char * const rt5670_dac12_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", + "Bass", "VAD_ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2l_enum, RT5670_DAC_CTRL, + RT5670_DAC2_L_SEL_SFT, rt5670_dac12_src); + +static const struct snd_kcontrol_new rt5670_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_dac2l_enum); + +static const char * const rt5670_dacr2_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", "TxDP ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2r_enum, RT5670_DAC_CTRL, + RT5670_DAC2_R_SEL_SFT, rt5670_dacr2_src); + +static const struct snd_kcontrol_new rt5670_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5670_dac2r_enum); + +/*RxDP source*/ /* MX-2D [15:13] */ +static const char * const rt5670_rxdp_src[] = { + "IF2 DAC", "IF1 DAC", "STO1 ADC Mixer", "STO2 ADC Mixer", + "Mono ADC Mixer L", "Mono ADC Mixer R", "DAC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SEL_SFT, rt5670_rxdp_src); + +static const struct snd_kcontrol_new rt5670_rxdp_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_rxdp_enum); + +/* MX-2D [1] [0] */ +static const char * const rt5670_dsp_bypass_src[] = { + "DSP", "Bypass" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_ul_enum, RT5670_DSP_PATH1, + RT5670_DSP_UL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_ul_mux = + SOC_DAPM_ENUM("DSP UL source", rt5670_dsp_ul_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_dl_enum, RT5670_DSP_PATH1, + RT5670_DSP_DL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_dl_mux = + SOC_DAPM_ENUM("DSP DL source", rt5670_dsp_dl_enum); + +/* Stereo2 ADC source */ +/* MX-26 [15] */ +static const char * const rt5670_stereo2_adc_lr_src[] = { + "L", "LR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER, + RT5670_STO2_ADC_SRC_SFT, rt5670_stereo2_adc_lr_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_lr_mux = + SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5670_stereo2_adc_lr_enum); + +/* Stereo1 ADC source */ +/* MX-27 MX-26 [12] */ +static const char * const rt5670_stereo_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = + SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = + SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); + +/* MX-27 MX-26 [11] */ +static const char * const rt5670_stereo_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = + SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = + SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = + SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = + SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); + +/* MX-27 MX26 [10] */ +static const char * const rt5670_stereo_adc_src[] = { + "ADC1L ADC2R", "ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_mux = + SOC_DAPM_ENUM("Stereo1 ADC source", rt5670_stereo1_adc_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_mux = + SOC_DAPM_ENUM("Stereo2 ADC source", rt5670_stereo2_adc_enum); + +/* MX-27 MX-26 [9:8] */ +static const char * const rt5670_stereo_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5670_stereo1_dmic_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto2_dmic_mux = + SOC_DAPM_ENUM("Stereo2 DMIC source", rt5670_stereo2_dmic_enum); + +/* MX-27 [0] */ +static const char * const rt5670_stereo_dmic3_src[] = { + "DMIC3", "PDM ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC3_SRC_SFT, rt5670_stereo_dmic3_src); + +static const struct snd_kcontrol_new rt5670_sto_dmic3_mux = + SOC_DAPM_ENUM("Stereo DMIC3 source", rt5670_stereo_dmic3_enum); + +/* Mono ADC source */ +/* MX-28 [12] */ +static const char * const rt5670_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L1_SRC_SFT, rt5670_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5670_mono_adc_l1_enum); +/* MX-28 [11] */ +static const char * const rt5670_mono_adc_l2_src[] = { + "Mono DAC MIXL", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L2_SRC_SFT, rt5670_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5670_mono_adc_l2_enum); + +/* MX-28 [9:8] */ +static const char * const rt5670_mono_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_L_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC left source", rt5670_mono_dmic_l_enum); +/* MX-28 [1:0] */ +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_R_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC Right source", rt5670_mono_dmic_r_enum); +/* MX-28 [4] */ +static const char * const rt5670_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADC2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R1_SRC_SFT, rt5670_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5670_mono_adc_r1_enum); +/* MX-28 [3] */ +static const char * const rt5670_mono_adc_r2_src[] = { + "Mono DAC MIXR", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R2_SRC_SFT, rt5670_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5670_mono_adc_r2_enum); + +/* MX-2D [3:2] */ +static const char * const rt5670_txdp_slot_src[] = { + "Slot 0-1", "Slot 2-3", "Slot 4-5", "Slot 6-7" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_txdp_slot_enum, RT5670_DSP_PATH1, + RT5670_TXDP_SLOT_SEL_SFT, rt5670_txdp_slot_src); + +static const struct snd_kcontrol_new rt5670_txdp_slot_mux = + SOC_DAPM_ENUM("TxDP Slot source", rt5670_txdp_slot_enum); + +/* MX-2F [15] */ +static const char * const rt5670_if1_adc2_in_src[] = { + "IF_ADC2", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF1_ADC2_IN_SFT, rt5670_if1_adc2_in_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN source", rt5670_if1_adc2_in_enum); + +/* MX-2F [14:12] */ +static const char * const rt5670_if2_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3", "TxDC_DAC", "TxDP_ADC", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_IN_SFT, rt5670_if2_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if2_adc_in_mux = + SOC_DAPM_ENUM("IF2 ADC IN source", rt5670_if2_adc_in_enum); + +/* MX-30 [5:4] */ +static const char * const rt5670_if4_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA, + RT5670_IF4_ADC_IN_SFT, rt5670_if4_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if4_adc_in_mux = + SOC_DAPM_ENUM("IF4 ADC IN source", rt5670_if4_adc_in_enum); + +/* MX-31 [15] [13] [11] [9] */ +static const char * const rt5670_pdm_src[] = { + "Mono DAC", "Stereo DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 L source", rt5670_pdm1_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 R source", rt5670_pdm1_r_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_l_mux = + SOC_DAPM_ENUM("PDM2 L source", rt5670_pdm2_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_r_mux = + SOC_DAPM_ENUM("PDM2 R source", rt5670_pdm2_r_enum); + +/* MX-FA [12] */ +static const char * const rt5670_if1_adc1_in1_src[] = { + "IF_ADC1", "IF1_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN1_SFT, rt5670_if1_adc1_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in1_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN1 source", rt5670_if1_adc1_in1_enum); + +/* MX-FA [11] */ +static const char * const rt5670_if1_adc1_in2_src[] = { + "IF1_ADC1_IN1", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN2_SFT, rt5670_if1_adc1_in2_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in2_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN2 source", rt5670_if1_adc1_in2_enum); + +/* MX-FA [10] */ +static const char * const rt5670_if1_adc2_in1_src[] = { + "IF1_ADC2_IN", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC2_IN1_SFT, rt5670_if1_adc2_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in1_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN1 source", rt5670_if1_adc2_in1_enum); + +/* MX-9D [9:8] */ +static const char * const rt5670_vad_adc_src[] = { + "Sto1 ADC L", "Mono ADC L", "Mono ADC R", "Sto2 ADC L" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_vad_adc_enum, RT5670_VAD_CTRL4, + RT5670_VAD_SEL_SFT, rt5670_vad_adc_src); + +static const struct snd_kcontrol_new rt5670_vad_adc_mux = + SOC_DAPM_ENUM("VAD ADC source", rt5670_vad_adc_enum); + +static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5670->regmap, RT5670_CHARGE_PUMP, + RT5670_PM_HP_MASK, RT5670_PM_HP_HV); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0400, 0x0400); + /* headphone amp power on */ + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HA | RT5670_PWR_FV1 | + RT5670_PWR_FV2, RT5670_PWR_HA | + RT5670_PWR_FV1 | RT5670_PWR_FV2); + /* depop parameters */ + regmap_write(rt5670->regmap, RT5670_DEPOP_M2, 0x3100); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8009); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_HP_DCC_INT1, 0x9f00); + mdelay(20); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x0004); + msleep(30); + break; + default: + return 0; + } + + return 0; +} + +static int rt5670_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* headphone unmute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x805d); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0300, 0x0300); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, 0); + msleep(80); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* headphone mute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x803d); + mdelay(10); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + mdelay(10); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, + RT5670_L_MUTE | RT5670_R_MUTE); + msleep(20); + regmap_update_bits(rt5670->regmap, + RT5670_GEN_CTRL2, 0x0300, 0x0); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0707); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xfc00); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, RT5670_PWR_BST1_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, RT5670_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5670_PWR_ANLG2, + RT5670_PWR_PLL_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S DSP", RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5670_PWR_VOL, + RT5670_PWR_MIC_DET_BIT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5670_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5670_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5670_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5670_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5670_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5670_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5670_ASRC_1, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5670_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5670_ASRC_1, + 0, 0, NULL, 0), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_MICBIAS("MICBIAS1", RT5670_PWR_ANLG2, + RT5670_PWR_MB1_BIT, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_INPUT("DMIC L2"), + SND_SOC_DAPM_INPUT("DMIC R2"), + SND_SOC_DAPM_INPUT("DMIC L3"), + SND_SOC_DAPM_INPUT("DMIC R3"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_3_EN_SFT, 0, NULL, 0), + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5670_PWR_ANLG2, RT5670_PWR_BST1_BIT, + 0, NULL, 0, rt5670_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5670_PWR_ANLG2, RT5670_PWR_BST2_BIT, + 0, NULL, 0, rt5670_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_R_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5670_PWR_MIXER, RT5670_PWR_RM_L_BIT, 0, + rt5670_rec_l_mix, ARRAY_SIZE(rt5670_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5670_PWR_MIXER, RT5670_PWR_RM_R_BIT, 0, + rt5670_rec_r_mix, ARRAY_SIZE(rt5670_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 2", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PGA("ADC 1_2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC 1 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC 2 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", RT5670_PR_BASE + + RT5670_CHOP_DAC_ADC, 12, 0, NULL, 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_dmic_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_lr_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("adc stereo2 filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix, + ARRAY_SIZE(rt5670_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL, + RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix, + ARRAY_SIZE(rt5670_sto1_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_l_mix, + ARRAY_SIZE(rt5670_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_r_mix, + ARRAY_SIZE(rt5670_sto2_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("adc mono left filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, 1, rt5670_mono_adc_l_mix, + ARRAY_SIZE(rt5670_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("adc mono right filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", RT5670_MONO_ADC_DIG_VOL, + RT5670_R_MUTE_SFT, 1, rt5670_mono_adc_r_mix, + ARRAY_SIZE(rt5670_mono_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VAD_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DSP */ + SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDC_DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("TDM Data Mux", SND_SOC_NOPM, 0, 0, + &rt5670_txdp_slot_mux), + + SND_SOC_DAPM_MUX("DSP UL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_ul_mux), + SND_SOC_DAPM_MUX("DSP DL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_dl_mux), + + SND_SOC_DAPM_MUX("RxDP Mux", SND_SOC_NOPM, 0, 0, + &rt5670_rxdp_mux), + + /* IF2 Mux */ + SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if2_adc_in_mux), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5670_PWR_DIG1, + RT5670_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5670_PWR_DIG1, + RT5670_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 ADC1 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in1_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 IN2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in2_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in1_mux), + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_vad_adc_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5670_GPIO_CTRL1, RT5670_I2S2_PIN_SFT, 1), + + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dac_l_mix, ARRAY_SIZE(rt5670_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dac_r_mix, ARRAY_SIZE(rt5670_dac_r_mix)), + SND_SOC_DAPM_PGA("DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1l_mux), + SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_l_mix, + ARRAY_SIZE(rt5670_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_r_mix, + ARRAY_SIZE(rt5670_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_l_mix, + ARRAY_SIZE(rt5670_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_r_mix, + ARRAY_SIZE(rt5670_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dig_l_mix, + ARRAY_SIZE(rt5670_dig_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dig_r_mix, + ARRAY_SIZE(rt5670_dig_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0), + /* OUT Mixer */ + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5670_PWR_MIXER, RT5670_PWR_OM_L_BIT, + 0, rt5670_out_l_mix, ARRAY_SIZE(rt5670_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5670_PWR_MIXER, RT5670_PWR_OM_R_BIT, + 0, rt5670_out_r_mix, ARRAY_SIZE(rt5670_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5670_PWR_VOL, + RT5670_PWR_HV_L_BIT, 0, + rt5670_hpvoll_mix, ARRAY_SIZE(rt5670_hpvoll_mix)), + SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5670_PWR_VOL, + RT5670_PWR_HV_R_BIT, 0, + rt5670_hpvolr_mix, ARRAY_SIZE(rt5670_hpvolr_mix)), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, + rt5670_hpo_mix, ARRAY_SIZE(rt5670_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5670_PWR_ANLG1, RT5670_PWR_LM_BIT, + 0, rt5670_lout_mix, ARRAY_SIZE(rt5670_lout_mix)), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0, + &lout_l_enable_control), + SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0, + &lout_r_enable_control), + SND_SOC_DAPM_PGA("LOUT Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux), + SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("PDM1L"), + SND_SOC_DAPM_OUTPUT("PDM1R"), + SND_SOC_DAPM_OUTPUT("PDM2L"), + SND_SOC_DAPM_OUTPUT("PDM2R"), +}; + +static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { + { "adc stereo1 filter", NULL, "ADC STO1 ASRC", + check_adc_sto1_asrc_source }, + { "adc stereo2 filter", NULL, "ADC STO2 ASRC", + check_adc_sto2_asrc_source }, + { "adc mono left filter", NULL, "ADC MONO L ASRC", + check_adc_monol_asrc_source }, + { "adc mono right filter", NULL, "ADC MONO R ASRC", + check_adc_monor_asrc_source }, + { "dac mono left filter", NULL, "DAC MONO L ASRC", + check_dac_monol_asrc_source }, + { "dac mono right filter", NULL, "DAC MONO R ASRC", + check_dac_monor_asrc_source }, + { "dac stereo1 filter", NULL, "DAC STO ASRC", + check_dac_sto_asrc_source }, + + {"I2S1", NULL, "I2S1 ASRC"}, + {"I2S2", NULL, "I2S2 ASRC"}, + + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + { "DMIC3", NULL, "DMIC L3" }, + { "DMIC3", NULL, "DMIC R3" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "Mic Det Power" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "INL VOL", NULL, "IN2P" }, + { "INR VOL", NULL, "IN2N" }, + + { "RECMIXL", "INL Switch", "INL VOL" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + + { "RECMIXR", "INR Switch", "INR VOL" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + + { "ADC 1", NULL, "RECMIXL" }, + { "ADC 1", NULL, "ADC 1 power" }, + { "ADC 1", NULL, "ADC clock" }, + { "ADC 2", NULL, "RECMIXR" }, + { "ADC 2", NULL, "ADC 2 power" }, + { "ADC 2", NULL, "ADC clock" }, + + { "DMIC L1", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 Power" }, + { "DMIC R1", NULL, "DMIC CLK" }, + { "DMIC R1", NULL, "DMIC1 Power" }, + { "DMIC L2", NULL, "DMIC CLK" }, + { "DMIC L2", NULL, "DMIC2 Power" }, + { "DMIC R2", NULL, "DMIC CLK" }, + { "DMIC R2", NULL, "DMIC2 Power" }, + { "DMIC L3", NULL, "DMIC CLK" }, + { "DMIC L3", NULL, "DMIC3 Power" }, + { "DMIC R3", NULL, "DMIC CLK" }, + { "DMIC R3", NULL, "DMIC3 Power" }, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo2 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", "DMIC3", "DMIC L3" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", "DMIC3", "DMIC R3" }, + + { "ADC 1_2", NULL, "ADC 1" }, + { "ADC 1_2", NULL, "ADC 2" }, + + { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo1 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo1 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo1 ADC R2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Mono ADC L2 Mux", "DMIC", "Mono DMIC L Mux" }, + { "Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "ADC1", "ADC 1" }, + + { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + { "Mono ADC R1 Mux", "ADC2", "ADC 2" }, + { "Mono ADC R2 Mux", "DMIC", "Mono DMIC R Mux" }, + { "Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + + { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux" }, + { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux" }, + { "Mono ADC MIXL", NULL, "adc mono left filter" }, + { "adc mono left filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux" }, + { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux" }, + { "Mono ADC MIXR", NULL, "adc mono right filter" }, + { "adc mono right filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC L2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo2 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo2 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo2 ADC R2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Sto2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC L1 Mux" }, + { "Sto2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC L2 Mux" }, + { "Sto2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC R1 Mux" }, + { "Sto2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC R2 Mux" }, + + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXL" }, + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXR" }, + + { "Stereo2 ADC LR Mux", "L", "Sto2 ADC MIXL" }, + { "Stereo2 ADC LR Mux", "LR", "Sto2 ADC LR MIX" }, + + { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, + { "Stereo2 ADC MIXL", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, + { "Stereo2 ADC MIXR", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "VAD ADC Mux", "Sto1 ADC L", "Stereo1 ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC L", "Mono ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC R", "Mono ADC MIXR" }, + { "VAD ADC Mux", "Sto2 ADC L", "Sto2 ADC MIXL" }, + + { "VAD_ADC", NULL, "VAD ADC Mux" }, + + { "IF_ADC1", NULL, "Stereo1 ADC MIXL" }, + { "IF_ADC1", NULL, "Stereo1 ADC MIXR" }, + { "IF_ADC2", NULL, "Mono ADC MIXL" }, + { "IF_ADC2", NULL, "Mono ADC MIXR" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXL" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXR" }, + + { "IF1 ADC1 IN1 Mux", "IF_ADC1", "IF_ADC1" }, + { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, + + { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, + { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, + { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, + + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" }, + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXL" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXR" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXL" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXR" }, + + { "RxDP Mux", "IF2 DAC", "IF2 DAC" }, + { "RxDP Mux", "IF1 DAC", "IF1 DAC2" }, + { "RxDP Mux", "STO1 ADC Mixer", "Stereo1 ADC MIX" }, + { "RxDP Mux", "STO2 ADC Mixer", "Stereo2 ADC MIX" }, + { "RxDP Mux", "Mono ADC Mixer L", "Mono ADC MIXL" }, + { "RxDP Mux", "Mono ADC Mixer R", "Mono ADC MIXR" }, + { "RxDP Mux", "DAC1", "DAC MIX" }, + + { "TDM Data Mux", "Slot 0-1", "Stereo1 ADC MIX" }, + { "TDM Data Mux", "Slot 2-3", "Mono ADC MIX" }, + { "TDM Data Mux", "Slot 4-5", "Stereo2 ADC MIX" }, + { "TDM Data Mux", "Slot 6-7", "IF2 DAC" }, + + { "DSP UL Mux", "Bypass", "TDM Data Mux" }, + { "DSP UL Mux", NULL, "I2S DSP" }, + { "DSP DL Mux", "Bypass", "RxDP Mux" }, + { "DSP DL Mux", NULL, "I2S DSP" }, + + { "TxDP_ADC_L", NULL, "DSP UL Mux" }, + { "TxDP_ADC_R", NULL, "DSP UL Mux" }, + { "TxDC_DAC", NULL, "DSP DL Mux" }, + + { "TxDP_ADC", NULL, "TxDP_ADC_L" }, + { "TxDP_ADC", NULL, "TxDP_ADC_R" }, + + { "IF1 ADC", NULL, "I2S1" }, + { "IF1 ADC", NULL, "IF1_ADC1" }, + { "IF1 ADC", NULL, "IF1_ADC2" }, + { "IF1 ADC", NULL, "IF_ADC3" }, + { "IF1 ADC", NULL, "TxDP_ADC" }, + + { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF2 ADC Mux", "IF_ADC3", "IF_ADC3" }, + { "IF2 ADC Mux", "TxDC_DAC", "TxDC_DAC" }, + { "IF2 ADC Mux", "TxDP_ADC", "TxDP_ADC" }, + { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF2 ADC L", NULL, "IF2 ADC Mux" }, + { "IF2 ADC R", NULL, "IF2 ADC Mux" }, + + { "IF2 ADC", NULL, "I2S2" }, + { "IF2 ADC", NULL, "IF2 ADC L" }, + { "IF2 ADC", NULL, "IF2 ADC R" }, + + { "AIF1TX", NULL, "IF1 ADC" }, + { "AIF2TX", NULL, "IF2 ADC" }, + + { "IF1 DAC1", NULL, "AIF1RX" }, + { "IF1 DAC2", NULL, "AIF1RX" }, + { "IF2 DAC", NULL, "AIF2RX" }, + + { "IF1 DAC1", NULL, "I2S1" }, + { "IF1 DAC2", NULL, "I2S1" }, + { "IF2 DAC", NULL, "I2S2" }, + + { "IF1 DAC2 L", NULL, "IF1 DAC2" }, + { "IF1 DAC2 R", NULL, "IF1 DAC2" }, + { "IF1 DAC1 L", NULL, "IF1 DAC1" }, + { "IF1 DAC1 R", NULL, "IF1 DAC1" }, + { "IF2 DAC L", NULL, "IF2 DAC" }, + { "IF2 DAC R", NULL, "IF2 DAC" }, + + { "DAC1 L Mux", "IF1 DAC", "IF1 DAC1 L" }, + { "DAC1 L Mux", "IF2 DAC", "IF2 DAC L" }, + + { "DAC1 R Mux", "IF1 DAC", "IF1 DAC1 R" }, + { "DAC1 R Mux", "IF2 DAC", "IF2 DAC R" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "DAC1 L Mux" }, + { "DAC1 MIXL", NULL, "dac stereo1 filter" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "DAC1 R Mux" }, + { "DAC1 MIXR", NULL, "dac stereo1 filter" }, + + { "DAC MIX", NULL, "DAC1 MIXL" }, + { "DAC MIX", NULL, "DAC1 MIXR" }, + + { "Audio DSP", NULL, "DAC1 MIXL" }, + { "Audio DSP", NULL, "DAC1 MIXR" }, + + { "DAC L2 Mux", "IF1 DAC", "IF1 DAC2 L" }, + { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" }, + { "DAC L2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC L2 Mux", "VAD_ADC", "VAD_ADC" }, + { "DAC L2 Volume", NULL, "DAC L2 Mux" }, + { "DAC L2 Volume", NULL, "dac mono left filter" }, + + { "DAC R2 Mux", "IF1 DAC", "IF1 DAC2 R" }, + { "DAC R2 Mux", "IF2 DAC", "IF2 DAC R" }, + { "DAC R2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC R2 Mux", "TxDP ADC", "TxDP_ADC" }, + { "DAC R2 Volume", NULL, "DAC R2 Mux" }, + { "DAC R2 Volume", NULL, "dac mono right filter" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "Stereo DAC MIXL", NULL, "DAC L1 Power" }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + { "Stereo DAC MIXR", NULL, "DAC R1 Power" }, + + { "Mono DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXR", NULL, "dac mono right filter" }, + + { "DAC MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + + { "DAC L1", NULL, "DAC L1 Power" }, + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R1", NULL, "DAC R1 Power" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R2", NULL, "Mono DAC MIXR" }, + { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, + + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "INL Switch", "INL VOL" }, + { "OUT MIXL", "DAC L2 Switch", "DAC L2" }, + { "OUT MIXL", "DAC L1 Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "INR Switch", "INR VOL" }, + { "OUT MIXR", "DAC R2 Switch", "DAC R2" }, + { "OUT MIXR", "DAC R1 Switch", "DAC R1" }, + + { "HPOVOL MIXL", "DAC1 Switch", "DAC L1" }, + { "HPOVOL MIXL", "INL Switch", "INL VOL" }, + { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" }, + { "HPOVOL MIXR", "INR Switch", "INR VOL" }, + + { "DAC 2", NULL, "DAC L2" }, + { "DAC 2", NULL, "DAC R2" }, + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + { "HPOVOL", NULL, "HPOVOL MIXL" }, + { "HPOVOL", NULL, "HPOVOL MIXR" }, + { "HPO MIX", "DAC1 Switch", "DAC 1" }, + { "HPO MIX", "HPVOL Switch", "HPOVOL" }, + + { "LOUT MIX", "DAC L1 Switch", "DAC L1" }, + { "LOUT MIX", "DAC R1 Switch", "DAC R1" }, + { "LOUT MIX", "OUTMIX L Switch", "OUT MIXL" }, + { "LOUT MIX", "OUTMIX R Switch", "OUT MIXR" }, + + { "PDM1 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM1 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM1 L Mux", NULL, "PDM1 Power" }, + { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM1 R Mux", NULL, "PDM1 Power" }, + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, + + { "HP Amp", NULL, "HPO MIX" }, + { "HP Amp", NULL, "Mic Det Power" }, + { "HPOL", NULL, "HP Amp" }, + { "HPOL", NULL, "HP L Amp" }, + { "HPOL", NULL, "Improve HP Amp Drv" }, + { "HPOR", NULL, "HP Amp" }, + { "HPOR", NULL, "HP R Amp" }, + { "HPOR", NULL, "Improve HP Amp Drv" }, + + { "LOUT Amp", NULL, "LOUT MIX" }, + { "LOUT L Playback", "Switch", "LOUT Amp" }, + { "LOUT R Playback", "Switch", "LOUT Amp" }, + { "LOUTL", NULL, "LOUT L Playback" }, + { "LOUTR", NULL, "LOUT R Playback" }, + { "LOUTL", NULL, "Improve HP Amp Drv" }, + { "LOUTR", NULL, "Improve HP Amp Drv" }, + + { "PDM1L", NULL, "PDM1 L Mux" }, + { "PDM1R", NULL, "PDM1 R Mux" }, + { "PDM2L", NULL, "PDM2 L Mux" }, + { "PDM2R", NULL, "PDM2 R Mux" }, +}; + +static int rt5670_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5670->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5670->sysclk, rt5670->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5670->lrck[dai->id], dai->id); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32; + rt5670->bclk[dai->id] = rt5670->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5670->bclk[dai->id], rt5670->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= RT5670_I2S_DL_20; + break; + case 24: + val_len |= RT5670_I2S_DL_24; + break; + case 8: + val_len |= RT5670_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + mask_clk = RT5670_I2S_BCLK_MS1_MASK | RT5670_I2S_PD1_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS1_SFT | + pre_div << RT5670_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5670_AIF2: + mask_clk = RT5670_I2S_BCLK_MS2_MASK | RT5670_I2S_PD2_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS2_SFT | + pre_div << RT5670_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5670->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5670_I2S_MS_S; + rt5670->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5670_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5670_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5670_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5670_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) + return 0; + + switch (clk_id) { + case RT5670_SCLK_S_MCLK: + reg_val |= RT5670_SCLK_SRC_MCLK; + break; + case RT5670_SCLK_S_PLL1: + reg_val |= RT5670_SCLK_SRC_PLL1; + break; + case RT5670_SCLK_S_RCCLK: + reg_val |= RT5670_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, reg_val); + rt5670->sysclk = freq; + rt5670->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5670->pll_src && freq_in == rt5670->pll_in && + freq_out == rt5670->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5670->pll_in = 0; + rt5670->pll_out = 0; + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5670_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_MCLK); + break; + case RT5670_PLL1_S_BCLK1: + case RT5670_PLL1_S_BCLK2: + case RT5670_PLL1_S_BCLK3: + case RT5670_PLL1_S_BCLK4: + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK1); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK2); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5670_PLL_CTRL1, + pll_code.n_code << RT5670_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5670_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5670_PLL_M_SFT | + pll_code.m_bp << RT5670_PLL_M_BP_SFT); + + rt5670->pll_in = freq_in; + rt5670->pll_out = freq_out; + rt5670->pll_src = source; + + return 0; +} + +static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 14); + + switch (slots) { + case 4: + val |= (1 << 12); + break; + case 6: + val |= (2 << 12); + break; + case 8: + val |= (3 << 12); + break; + case 2: + default: + break; + } + + switch (slot_width) { + case 20: + val |= (1 << 10); + break; + case 24: + val |= (2 << 10); + break; + case 32: + val |= (3 << 10); + break; + case 16: + default: + break; + } + + snd_soc_update_bits(codec, RT5670_TDM_CTRL_1, 0x7c00, val); + + return 0; +} + +static int rt5670_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_FV1 | RT5670_PWR_FV2); + snd_soc_update_bits(codec, RT5670_CHARGE_PUMP, + RT5670_OSW_L_MASK | RT5670_OSW_R_MASK, + RT5670_OSW_L_DIS | RT5670_OSW_R_DIS); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x1); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x3); + } + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); + snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); + snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); + snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x1); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5670_probe(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + rt5670->codec = codec; + + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int rt5670_remove(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + return 0; +} + +#ifdef CONFIG_PM +static int rt5670_suspend(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, true); + regcache_mark_dirty(rt5670->regmap); + return 0; +} + +static int rt5670_resume(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, false); + regcache_sync(rt5670->regmap); + + return 0; +} +#else +#define rt5670_suspend NULL +#define rt5670_resume NULL +#endif + +#define RT5670_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5670_aif_dai_ops = { + .hw_params = rt5670_hw_params, + .set_fmt = rt5670_set_dai_fmt, + .set_sysclk = rt5670_set_dai_sysclk, + .set_tdm_slot = rt5670_set_tdm_slot, + .set_pll = rt5670_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5670_dai[] = { + { + .name = "rt5670-aif1", + .id = RT5670_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, + { + .name = "rt5670-aif2", + .id = RT5670_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { + .probe = rt5670_probe, + .remove = rt5670_remove, + .suspend = rt5670_suspend, + .resume = rt5670_resume, + .set_bias_level = rt5670_set_bias_level, + .idle_bias_off = true, + .controls = rt5670_snd_controls, + .num_controls = ARRAY_SIZE(rt5670_snd_controls), + .dapm_widgets = rt5670_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5670_dapm_widgets), + .dapm_routes = rt5670_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes), +}; + +static const struct regmap_config rt5670_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * + RT5670_PR_SPACING), + .volatile_reg = rt5670_volatile_register, + .readable_reg = rt5670_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5670_reg, + .num_reg_defaults = ARRAY_SIZE(rt5670_reg), + .ranges = rt5670_ranges, + .num_ranges = ARRAY_SIZE(rt5670_ranges), +}; + +static const struct i2c_device_id rt5670_i2c_id[] = { + { "rt5670", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); + +static int rt5670_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5670_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5670_priv *rt5670; + int ret; + unsigned int val; + + rt5670 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5670_priv), + GFP_KERNEL); + if (NULL == rt5670) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5670); + + if (pdata) + rt5670->pdata = *pdata; + + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); + if (IS_ERR(rt5670->regmap)) { + ret = PTR_ERR(rt5670->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5670->regmap, RT5670_VENDOR_ID2, &val); + if (val != RT5670_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5670/72\n", val); + return -ENODEV; + } + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HP_L | RT5670_PWR_HP_R | + RT5670_PWR_VREF2, RT5670_PWR_VREF2); + msleep(100); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + + ret = regmap_register_patch(rt5670->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5670->pdata.in2_diff) + regmap_update_bits(rt5670->regmap, RT5670_IN2, + RT5670_IN_DF2, RT5670_IN_DF2); + + if (i2c->irq) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); + + } + + if (rt5670->pdata.jd_mode) { + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_MB, RT5670_PWR_MB); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, + RT5670_PWR_JD1, RT5670_PWR_JD1); + regmap_update_bits(rt5670->regmap, RT5670_IRQ_CTRL1, + RT5670_JD1_1_EN_MASK, RT5670_JD1_1_EN); + regmap_update_bits(rt5670->regmap, RT5670_JD_CTRL3, + RT5670_JD_TRI_CBJ_SEL_MASK | + RT5670_JD_TRI_HPO_SEL_MASK, + RT5670_JD_CBJ_JD1_1 | RT5670_JD_HPO_JD1_1); + switch (rt5670->pdata.jd_mode) { + case 1: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_0); + break; + case 2: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_1); + break; + case 3: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_2); + break; + default: + break; + } + } + + if (rt5670->pdata.dmic_en) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP2_PIN_MASK, + RT5670_GP2_PIN_DMIC1_SCL); + + switch (rt5670->pdata.dmic1_data_pin) { + case RT5670_DMIC_DATA_IN2P: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_IN2P); + break; + + case RT5670_DMIC_DATA_GPIO6: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_GPIO6); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP6_PIN_MASK, + RT5670_GP6_PIN_DMIC1_SDA); + break; + + case RT5670_DMIC_DATA_GPIO7: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_GPIO7); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP7_PIN_MASK, + RT5670_GP7_PIN_DMIC1_SDA); + break; + + default: + break; + } + + switch (rt5670->pdata.dmic2_data_pin) { + case RT5670_DMIC_DATA_IN3N: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_2_DP_MASK, + RT5670_DMIC_2_DP_IN3N); + break; + + case RT5670_DMIC_DATA_GPIO8: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_2_DP_MASK, + RT5670_DMIC_2_DP_GPIO8); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP8_PIN_MASK, + RT5670_GP8_PIN_DMIC2_SDA); + break; + + default: + break; + } + + switch (rt5670->pdata.dmic3_data_pin) { + case RT5670_DMIC_DATA_GPIO5: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL2, + RT5670_DMIC_3_DP_MASK, + RT5670_DMIC_3_DP_GPIO5); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP5_PIN_MASK, + RT5670_GP5_PIN_DMIC3_SDA); + break; + + case RT5670_DMIC_DATA_GPIO9: + case RT5670_DMIC_DATA_GPIO10: + dev_err(&i2c->dev, + "Always use GPIO5 as DMIC3 data pin\n"); + break; + + default: + break; + } + + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, + rt5670_dai, ARRAY_SIZE(rt5670_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5670_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt5670_i2c_driver = { + .driver = { + .name = "rt5670", + .owner = THIS_MODULE, + }, + .probe = rt5670_i2c_probe, + .remove = rt5670_i2c_remove, + .id_table = rt5670_i2c_id, +}; + +module_i2c_driver(rt5670_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5670 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h new file mode 100644 index 0000000..a0b5c85 --- /dev/null +++ b/sound/soc/codecs/rt5670.h @@ -0,0 +1,2000 @@ +/* + * rt5670.h -- RT5670 ALSA SoC audio driver + * + * Copyright 2014 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5670_H__ +#define __RT5670_H__ + +#include <sound/rt5670.h> + +/* Info */ +#define RT5670_RESET 0x00 +#define RT5670_VENDOR_ID 0xfd +#define RT5670_VENDOR_ID1 0xfe +#define RT5670_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5670_HP_VOL 0x02 +#define RT5670_LOUT1 0x03 +/* I/O - Input */ +#define RT5670_CJ_CTRL1 0x0a +#define RT5670_CJ_CTRL2 0x0b +#define RT5670_CJ_CTRL3 0x0c +#define RT5670_IN2 0x0e +#define RT5670_INL1_INR1_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5670_DAC1_DIG_VOL 0x19 +#define RT5670_DAC2_DIG_VOL 0x1a +#define RT5670_DAC_CTRL 0x1b +#define RT5670_STO1_ADC_DIG_VOL 0x1c +#define RT5670_MONO_ADC_DIG_VOL 0x1d +#define RT5670_ADC_BST_VOL1 0x1e +#define RT5670_STO2_ADC_DIG_VOL 0x1f +/* Mixer - D-D */ +#define RT5670_ADC_BST_VOL2 0x20 +#define RT5670_STO2_ADC_MIXER 0x26 +#define RT5670_STO1_ADC_MIXER 0x27 +#define RT5670_MONO_ADC_MIXER 0x28 +#define RT5670_AD_DA_MIXER 0x29 +#define RT5670_STO_DAC_MIXER 0x2a +#define RT5670_DD_MIXER 0x2b +#define RT5670_DIG_MIXER 0x2c +#define RT5670_DSP_PATH1 0x2d +#define RT5670_DSP_PATH2 0x2e +#define RT5670_DIG_INF1_DATA 0x2f +#define RT5670_DIG_INF2_DATA 0x30 +/* Mixer - PDM */ +#define RT5670_PDM_OUT_CTRL 0x31 +#define RT5670_PDM_DATA_CTRL1 0x32 +#define RT5670_PDM1_DATA_CTRL2 0x33 +#define RT5670_PDM1_DATA_CTRL3 0x34 +#define RT5670_PDM1_DATA_CTRL4 0x35 +#define RT5670_PDM2_DATA_CTRL2 0x36 +#define RT5670_PDM2_DATA_CTRL3 0x37 +#define RT5670_PDM2_DATA_CTRL4 0x38 +/* Mixer - ADC */ +#define RT5670_REC_L1_MIXER 0x3b +#define RT5670_REC_L2_MIXER 0x3c +#define RT5670_REC_R1_MIXER 0x3d +#define RT5670_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5670_HPO_MIXER 0x45 +#define RT5670_MONO_MIXER 0x4c +#define RT5670_OUT_L1_MIXER 0x4f +#define RT5670_OUT_R1_MIXER 0x52 +#define RT5670_LOUT_MIXER 0x53 +/* Power */ +#define RT5670_PWR_DIG1 0x61 +#define RT5670_PWR_DIG2 0x62 +#define RT5670_PWR_ANLG1 0x63 +#define RT5670_PWR_ANLG2 0x64 +#define RT5670_PWR_MIXER 0x65 +#define RT5670_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5670_PRIV_INDEX 0x6a +#define RT5670_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5670_I2S4_SDP 0x6f +#define RT5670_I2S1_SDP 0x70 +#define RT5670_I2S2_SDP 0x71 +#define RT5670_I2S3_SDP 0x72 +#define RT5670_ADDA_CLK1 0x73 +#define RT5670_ADDA_CLK2 0x74 +#define RT5670_DMIC_CTRL1 0x75 +#define RT5670_DMIC_CTRL2 0x76 +/* Format - TDM Control */ +#define RT5670_TDM_CTRL_1 0x77 +#define RT5670_TDM_CTRL_2 0x78 +#define RT5670_TDM_CTRL_3 0x79 + +/* Function - Analog */ +#define RT5670_DSP_CLK 0x7f +#define RT5670_GLB_CLK 0x80 +#define RT5670_PLL_CTRL1 0x81 +#define RT5670_PLL_CTRL2 0x82 +#define RT5670_ASRC_1 0x83 +#define RT5670_ASRC_2 0x84 +#define RT5670_ASRC_3 0x85 +#define RT5670_ASRC_4 0x86 +#define RT5670_ASRC_5 0x87 +#define RT5670_ASRC_7 0x89 +#define RT5670_ASRC_8 0x8a +#define RT5670_ASRC_9 0x8b +#define RT5670_ASRC_10 0x8c +#define RT5670_ASRC_11 0x8d +#define RT5670_DEPOP_M1 0x8e +#define RT5670_DEPOP_M2 0x8f +#define RT5670_DEPOP_M3 0x90 +#define RT5670_CHARGE_PUMP 0x91 +#define RT5670_MICBIAS 0x93 +#define RT5670_A_JD_CTRL1 0x94 +#define RT5670_A_JD_CTRL2 0x95 +#define RT5670_ASRC_12 0x97 +#define RT5670_ASRC_13 0x98 +#define RT5670_ASRC_14 0x99 +#define RT5670_VAD_CTRL1 0x9a +#define RT5670_VAD_CTRL2 0x9b +#define RT5670_VAD_CTRL3 0x9c +#define RT5670_VAD_CTRL4 0x9d +#define RT5670_VAD_CTRL5 0x9e +/* Function - Digital */ +#define RT5670_ADC_EQ_CTRL1 0xae +#define RT5670_ADC_EQ_CTRL2 0xaf +#define RT5670_EQ_CTRL1 0xb0 +#define RT5670_EQ_CTRL2 0xb1 +#define RT5670_ALC_DRC_CTRL1 0xb2 +#define RT5670_ALC_DRC_CTRL2 0xb3 +#define RT5670_ALC_CTRL_1 0xb4 +#define RT5670_ALC_CTRL_2 0xb5 +#define RT5670_ALC_CTRL_3 0xb6 +#define RT5670_ALC_CTRL_4 0xb7 +#define RT5670_JD_CTRL 0xbb +#define RT5670_IRQ_CTRL1 0xbd +#define RT5670_IRQ_CTRL2 0xbe +#define RT5670_INT_IRQ_ST 0xbf +#define RT5670_GPIO_CTRL1 0xc0 +#define RT5670_GPIO_CTRL2 0xc1 +#define RT5670_GPIO_CTRL3 0xc2 +#define RT5670_SCRABBLE_FUN 0xcd +#define RT5670_SCRABBLE_CTRL 0xce +#define RT5670_BASE_BACK 0xcf +#define RT5670_MP3_PLUS1 0xd0 +#define RT5670_MP3_PLUS2 0xd1 +#define RT5670_ADJ_HPF1 0xd3 +#define RT5670_ADJ_HPF2 0xd4 +#define RT5670_HP_CALIB_AMP_DET 0xd6 +#define RT5670_SV_ZCD1 0xd9 +#define RT5670_SV_ZCD2 0xda +#define RT5670_IL_CMD 0xdb +#define RT5670_IL_CMD2 0xdc +#define RT5670_IL_CMD3 0xdd +#define RT5670_DRC_HL_CTRL1 0xe6 +#define RT5670_DRC_HL_CTRL2 0xe7 +#define RT5670_ADC_MONO_HP_CTRL1 0xec +#define RT5670_ADC_MONO_HP_CTRL2 0xed +#define RT5670_ADC_STO2_HP_CTRL1 0xee +#define RT5670_ADC_STO2_HP_CTRL2 0xef +#define RT5670_JD_CTRL3 0xf8 +#define RT5670_JD_CTRL4 0xf9 +/* General Control */ +#define RT5670_DIG_MISC 0xfa +#define RT5670_GEN_CTRL2 0xfb +#define RT5670_GEN_CTRL3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5670_DIG_VOL 0x00 +#define RT5670_PR_ALC_CTRL_1 0x01 +#define RT5670_PR_ALC_CTRL_2 0x02 +#define RT5670_PR_ALC_CTRL_3 0x03 +#define RT5670_PR_ALC_CTRL_4 0x04 +#define RT5670_PR_ALC_CTRL_5 0x05 +#define RT5670_PR_ALC_CTRL_6 0x06 +#define RT5670_BIAS_CUR1 0x12 +#define RT5670_BIAS_CUR3 0x14 +#define RT5670_CLSD_INT_REG1 0x1c +#define RT5670_MAMP_INT_REG2 0x37 +#define RT5670_CHOP_DAC_ADC 0x3d +#define RT5670_MIXER_INT_REG 0x3f +#define RT5670_3D_SPK 0x63 +#define RT5670_WND_1 0x6c +#define RT5670_WND_2 0x6d +#define RT5670_WND_3 0x6e +#define RT5670_WND_4 0x6f +#define RT5670_WND_5 0x70 +#define RT5670_WND_8 0x73 +#define RT5670_DIP_SPK_INF 0x75 +#define RT5670_HP_DCC_INT1 0x77 +#define RT5670_EQ_BW_LOP 0xa0 +#define RT5670_EQ_GN_LOP 0xa1 +#define RT5670_EQ_FC_BP1 0xa2 +#define RT5670_EQ_BW_BP1 0xa3 +#define RT5670_EQ_GN_BP1 0xa4 +#define RT5670_EQ_FC_BP2 0xa5 +#define RT5670_EQ_BW_BP2 0xa6 +#define RT5670_EQ_GN_BP2 0xa7 +#define RT5670_EQ_FC_BP3 0xa8 +#define RT5670_EQ_BW_BP3 0xa9 +#define RT5670_EQ_GN_BP3 0xaa +#define RT5670_EQ_FC_BP4 0xab +#define RT5670_EQ_BW_BP4 0xac +#define RT5670_EQ_GN_BP4 0xad +#define RT5670_EQ_FC_HIP1 0xae +#define RT5670_EQ_GN_HIP1 0xaf +#define RT5670_EQ_FC_HIP2 0xb0 +#define RT5670_EQ_BW_HIP2 0xb1 +#define RT5670_EQ_GN_HIP2 0xb2 +#define RT5670_EQ_PRE_VOL 0xb3 +#define RT5670_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5670_L_MUTE (0x1 << 15) +#define RT5670_L_MUTE_SFT 15 +#define RT5670_VOL_L_MUTE (0x1 << 14) +#define RT5670_VOL_L_SFT 14 +#define RT5670_R_MUTE (0x1 << 7) +#define RT5670_R_MUTE_SFT 7 +#define RT5670_VOL_R_MUTE (0x1 << 6) +#define RT5670_VOL_R_SFT 6 +#define RT5670_L_VOL_MASK (0x3f << 8) +#define RT5670_L_VOL_SFT 8 +#define RT5670_R_VOL_MASK (0x3f) +#define RT5670_R_VOL_SFT 0 + +/* Combo Jack Control 1 (0x0a) */ +#define RT5670_CBJ_BST1_MASK (0xf << 12) +#define RT5670_CBJ_BST1_SFT (12) +#define RT5670_CBJ_JD_HP_EN (0x1 << 9) +#define RT5670_CBJ_JD_MIC_EN (0x1 << 8) +#define RT5670_CBJ_BST1_EN (0x1 << 2) + +/* Combo Jack Control 1 (0x0b) */ +#define RT5670_CBJ_MN_JD (0x1 << 12) +#define RT5670_CAPLESS_EN (0x1 << 11) +#define RT5670_CBJ_DET_MODE (0x1 << 7) + +/* IN2 Control (0x0e) */ +#define RT5670_BST_MASK1 (0xf<<12) +#define RT5670_BST_SFT1 12 +#define RT5670_BST_MASK2 (0xf<<8) +#define RT5670_BST_SFT2 8 +#define RT5670_IN_DF1 (0x1 << 7) +#define RT5670_IN_SFT1 7 +#define RT5670_IN_DF2 (0x1 << 6) +#define RT5670_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5670_INL_SEL_MASK (0x1 << 15) +#define RT5670_INL_SEL_SFT 15 +#define RT5670_INL_SEL_IN4P (0x0 << 15) +#define RT5670_INL_SEL_MONOP (0x1 << 15) +#define RT5670_INL_VOL_MASK (0x1f << 8) +#define RT5670_INL_VOL_SFT 8 +#define RT5670_INR_SEL_MASK (0x1 << 7) +#define RT5670_INR_SEL_SFT 7 +#define RT5670_INR_SEL_IN4N (0x0 << 7) +#define RT5670_INR_SEL_MONON (0x1 << 7) +#define RT5670_INR_VOL_MASK (0x1f) +#define RT5670_INR_VOL_SFT 0 + +/* Sidetone Control (0x18) */ +#define RT5670_ST_SEL_MASK (0x7 << 9) +#define RT5670_ST_SEL_SFT 9 +#define RT5670_M_ST_DACR2 (0x1 << 8) +#define RT5670_M_ST_DACR2_SFT 8 +#define RT5670_M_ST_DACL2 (0x1 << 7) +#define RT5670_M_ST_DACL2_SFT 7 +#define RT5670_ST_EN (0x1 << 6) +#define RT5670_ST_EN_SFT 6 + +/* DAC1 Digital Volume (0x19) */ +#define RT5670_DAC_L1_VOL_MASK (0xff << 8) +#define RT5670_DAC_L1_VOL_SFT 8 +#define RT5670_DAC_R1_VOL_MASK (0xff) +#define RT5670_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5670_DAC_L2_VOL_MASK (0xff << 8) +#define RT5670_DAC_L2_VOL_SFT 8 +#define RT5670_DAC_R2_VOL_MASK (0xff) +#define RT5670_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5670_M_DAC_L2_VOL (0x1 << 13) +#define RT5670_M_DAC_L2_VOL_SFT 13 +#define RT5670_M_DAC_R2_VOL (0x1 << 12) +#define RT5670_M_DAC_R2_VOL_SFT 12 +#define RT5670_DAC2_L_SEL_MASK (0x7 << 4) +#define RT5670_DAC2_L_SEL_SFT 4 +#define RT5670_DAC2_R_SEL_MASK (0x7 << 0) +#define RT5670_DAC2_R_SEL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5670_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_ADC_L_VOL_SFT 8 +#define RT5670_ADC_R_VOL_MASK (0x7f) +#define RT5670_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5670_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_MONO_ADC_L_VOL_SFT 8 +#define RT5670_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5670_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5670_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5670_STO1_ADC_L_BST_SFT 14 +#define RT5670_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5670_STO1_ADC_R_BST_SFT 12 +#define RT5670_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5670_STO1_ADC_COMP_SFT 10 +#define RT5670_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5670_STO2_ADC_L_BST_SFT 8 +#define RT5670_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5670_STO2_ADC_R_BST_SFT 6 +#define RT5670_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5670_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5670_STO2_ADC_SRC_MASK (0x1 << 15) +#define RT5670_STO2_ADC_SRC_SFT 15 + +/* Stereo ADC Mixer Control (0x26 0x27) */ +#define RT5670_M_ADC_L1 (0x1 << 14) +#define RT5670_M_ADC_L1_SFT 14 +#define RT5670_M_ADC_L2 (0x1 << 13) +#define RT5670_M_ADC_L2_SFT 13 +#define RT5670_ADC_1_SRC_MASK (0x1 << 12) +#define RT5670_ADC_1_SRC_SFT 12 +#define RT5670_ADC_1_SRC_ADC (0x1 << 12) +#define RT5670_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5670_ADC_2_SRC_MASK (0x1 << 11) +#define RT5670_ADC_2_SRC_SFT 11 +#define RT5670_ADC_SRC_MASK (0x1 << 10) +#define RT5670_ADC_SRC_SFT 10 +#define RT5670_DMIC_SRC_MASK (0x3 << 8) +#define RT5670_DMIC_SRC_SFT 8 +#define RT5670_M_ADC_R1 (0x1 << 6) +#define RT5670_M_ADC_R1_SFT 6 +#define RT5670_M_ADC_R2 (0x1 << 5) +#define RT5670_M_ADC_R2_SFT 5 +#define RT5670_DMIC3_SRC_MASK (0x1 << 1) +#define RT5670_DMIC3_SRC_SFT 0 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5670_M_MONO_ADC_L1 (0x1 << 14) +#define RT5670_M_MONO_ADC_L1_SFT 14 +#define RT5670_M_MONO_ADC_L2 (0x1 << 13) +#define RT5670_M_MONO_ADC_L2_SFT 13 +#define RT5670_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5670_MONO_ADC_L1_SRC_SFT 12 +#define RT5670_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5670_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5670_MONO_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5670_MONO_ADC_L2_SRC_SFT 11 +#define RT5670_MONO_ADC_L_SRC_MASK (0x1 << 10) +#define RT5670_MONO_ADC_L_SRC_SFT 10 +#define RT5670_MONO_DMIC_L_SRC_MASK (0x3 << 8) +#define RT5670_MONO_DMIC_L_SRC_SFT 8 +#define RT5670_M_MONO_ADC_R1 (0x1 << 6) +#define RT5670_M_MONO_ADC_R1_SFT 6 +#define RT5670_M_MONO_ADC_R2 (0x1 << 5) +#define RT5670_M_MONO_ADC_R2_SFT 5 +#define RT5670_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_SFT 4 +#define RT5670_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5670_MONO_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5670_MONO_ADC_R2_SRC_SFT 3 +#define RT5670_MONO_DMIC_R_SRC_MASK (0x3) +#define RT5670_MONO_DMIC_R_SRC_SFT 0 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5670_M_ADCMIX_L (0x1 << 15) +#define RT5670_M_ADCMIX_L_SFT 15 +#define RT5670_M_DAC1_L (0x1 << 14) +#define RT5670_M_DAC1_L_SFT 14 +#define RT5670_DAC1_R_SEL_MASK (0x3 << 10) +#define RT5670_DAC1_R_SEL_SFT 10 +#define RT5670_DAC1_R_SEL_IF1 (0x0 << 10) +#define RT5670_DAC1_R_SEL_IF2 (0x1 << 10) +#define RT5670_DAC1_R_SEL_IF3 (0x2 << 10) +#define RT5670_DAC1_R_SEL_IF4 (0x3 << 10) +#define RT5670_DAC1_L_SEL_MASK (0x3 << 8) +#define RT5670_DAC1_L_SEL_SFT 8 +#define RT5670_DAC1_L_SEL_IF1 (0x0 << 8) +#define RT5670_DAC1_L_SEL_IF2 (0x1 << 8) +#define RT5670_DAC1_L_SEL_IF3 (0x2 << 8) +#define RT5670_DAC1_L_SEL_IF4 (0x3 << 8) +#define RT5670_M_ADCMIX_R (0x1 << 7) +#define RT5670_M_ADCMIX_R_SFT 7 +#define RT5670_M_DAC1_R (0x1 << 6) +#define RT5670_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5670_M_DAC_L1 (0x1 << 14) +#define RT5670_M_DAC_L1_SFT 14 +#define RT5670_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_STO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2 (0x1 << 12) +#define RT5670_M_DAC_L2_SFT 12 +#define RT5670_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_STO_L_VOL_SFT 11 +#define RT5670_M_DAC_R1_STO_L (0x1 << 9) +#define RT5670_M_DAC_R1_STO_L_SFT 9 +#define RT5670_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R1_STO_L_VOL_SFT 8 +#define RT5670_M_DAC_R1 (0x1 << 6) +#define RT5670_M_DAC_R1_SFT 6 +#define RT5670_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_STO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2 (0x1 << 4) +#define RT5670_M_DAC_R2_SFT 4 +#define RT5670_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_STO_R_VOL_SFT 3 +#define RT5670_M_DAC_L1_STO_R (0x1 << 1) +#define RT5670_M_DAC_L1_STO_R_SFT 1 +#define RT5670_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5670_DAC_L1_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5670_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5670_M_DAC_L1_MONO_L_SFT 14 +#define RT5670_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5670_M_DAC_L2_MONO_L_SFT 12 +#define RT5670_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5670_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5670_M_DAC_R2_MONO_L_SFT 10 +#define RT5670_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5670_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5670_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5670_M_DAC_R1_MONO_R_SFT 6 +#define RT5670_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5670_M_DAC_R2_MONO_R_SFT 4 +#define RT5670_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5670_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5670_M_DAC_L2_MONO_R_SFT 2 +#define RT5670_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5670_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5670_M_STO_L_DAC_L (0x1 << 15) +#define RT5670_M_STO_L_DAC_L_SFT 15 +#define RT5670_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5670_STO_L_DAC_L_VOL_SFT 14 +#define RT5670_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5670_M_DAC_L2_DAC_L_SFT 13 +#define RT5670_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5670_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5670_M_STO_R_DAC_R (0x1 << 11) +#define RT5670_M_STO_R_DAC_R_SFT 11 +#define RT5670_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5670_STO_R_DAC_R_VOL_SFT 10 +#define RT5670_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5670_M_DAC_R2_DAC_R_SFT 9 +#define RT5670_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R2_DAC_R_VOL_SFT 8 +#define RT5670_M_DAC_R2_DAC_L (0x1 << 7) +#define RT5670_M_DAC_R2_DAC_L_SFT 7 +#define RT5670_DAC_R2_DAC_L_VOL_MASK (0x1 << 6) +#define RT5670_DAC_R2_DAC_L_VOL_SFT 6 +#define RT5670_M_DAC_L2_DAC_R (0x1 << 5) +#define RT5670_M_DAC_L2_DAC_R_SFT 5 +#define RT5670_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) +#define RT5670_DAC_L2_DAC_R_VOL_SFT 4 + +/* DSP Path Control 1 (0x2d) */ +#define RT5670_RXDP_SEL_MASK (0x7 << 13) +#define RT5670_RXDP_SEL_SFT 13 +#define RT5670_RXDP_SRC_MASK (0x3 << 11) +#define RT5670_RXDP_SRC_SFT 11 +#define RT5670_RXDP_SRC_NOR (0x0 << 11) +#define RT5670_RXDP_SRC_DIV2 (0x1 << 11) +#define RT5670_RXDP_SRC_DIV3 (0x2 << 11) +#define RT5670_TXDP_SRC_MASK (0x3 << 4) +#define RT5670_TXDP_SRC_SFT 4 +#define RT5670_TXDP_SRC_NOR (0x0 << 4) +#define RT5670_TXDP_SRC_DIV2 (0x1 << 4) +#define RT5670_TXDP_SRC_DIV3 (0x2 << 4) +#define RT5670_TXDP_SLOT_SEL_MASK (0x3 << 2) +#define RT5670_TXDP_SLOT_SEL_SFT 2 +#define RT5670_DSP_UL_SEL (0x1 << 1) +#define RT5670_DSP_UL_SFT 1 +#define RT5670_DSP_DL_SEL 0x1 +#define RT5670_DSP_DL_SFT 0 + +/* DSP Path Control 2 (0x2e) */ +#define RT5670_TXDP_L_VOL_MASK (0x7f << 8) +#define RT5670_TXDP_L_VOL_SFT 8 +#define RT5670_TXDP_R_VOL_MASK (0x7f) +#define RT5670_TXDP_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5670_IF1_ADC2_IN_SEL (0x1 << 15) +#define RT5670_IF1_ADC2_IN_SFT 15 +#define RT5670_IF2_ADC_IN_MASK (0x7 << 12) +#define RT5670_IF2_ADC_IN_SFT 12 +#define RT5670_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5670_IF2_DAC_SEL_SFT 10 +#define RT5670_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5670_IF2_ADC_SEL_SFT 8 + +/* Digital Interface Data Control (0x30) */ +#define RT5670_IF4_ADC_IN_MASK (0x3 << 4) +#define RT5670_IF4_ADC_IN_SFT 4 + +/* PDM Output Control (0x31) */ +#define RT5670_PDM1_L_MASK (0x1 << 15) +#define RT5670_PDM1_L_SFT 15 +#define RT5670_M_PDM1_L (0x1 << 14) +#define RT5670_M_PDM1_L_SFT 14 +#define RT5670_PDM1_R_MASK (0x1 << 13) +#define RT5670_PDM1_R_SFT 13 +#define RT5670_M_PDM1_R (0x1 << 12) +#define RT5670_M_PDM1_R_SFT 12 +#define RT5670_PDM2_L_MASK (0x1 << 11) +#define RT5670_PDM2_L_SFT 11 +#define RT5670_M_PDM2_L (0x1 << 10) +#define RT5670_M_PDM2_L_SFT 10 +#define RT5670_PDM2_R_MASK (0x1 << 9) +#define RT5670_PDM2_R_SFT 9 +#define RT5670_M_PDM2_R (0x1 << 8) +#define RT5670_M_PDM2_R_SFT 8 +#define RT5670_PDM2_BUSY (0x1 << 7) +#define RT5670_PDM1_BUSY (0x1 << 6) +#define RT5670_PDM_PATTERN (0x1 << 5) +#define RT5670_PDM_GAIN (0x1 << 4) +#define RT5670_PDM_DIV_MASK (0x3) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5670_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5670_G_HP_L_RM_L_SFT 13 +#define RT5670_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5670_G_IN_L_RM_L_SFT 10 +#define RT5670_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_L_SFT 7 +#define RT5670_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_L_SFT 4 +#define RT5670_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5670_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_L_SFT 13 +#define RT5670_M_IN_L_RM_L (0x1 << 5) +#define RT5670_M_IN_L_RM_L_SFT 5 +#define RT5670_M_BST2_RM_L (0x1 << 3) +#define RT5670_M_BST2_RM_L_SFT 3 +#define RT5670_M_BST1_RM_L (0x1 << 1) +#define RT5670_M_BST1_RM_L_SFT 1 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5670_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5670_G_HP_R_RM_R_SFT 13 +#define RT5670_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5670_G_IN_R_RM_R_SFT 10 +#define RT5670_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_R_SFT 7 +#define RT5670_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_R_SFT 4 +#define RT5670_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5670_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_R_SFT 13 +#define RT5670_M_IN_R_RM_R (0x1 << 5) +#define RT5670_M_IN_R_RM_R_SFT 5 +#define RT5670_M_BST2_RM_R (0x1 << 3) +#define RT5670_M_BST2_RM_R_SFT 3 +#define RT5670_M_BST1_RM_R (0x1 << 1) +#define RT5670_M_BST1_RM_R_SFT 1 + +/* HPMIX Control (0x45) */ +#define RT5670_M_DAC2_HM (0x1 << 15) +#define RT5670_M_DAC2_HM_SFT 15 +#define RT5670_M_HPVOL_HM (0x1 << 14) +#define RT5670_M_HPVOL_HM_SFT 14 +#define RT5670_M_DAC1_HM (0x1 << 13) +#define RT5670_M_DAC1_HM_SFT 13 +#define RT5670_G_HPOMIX_MASK (0x1 << 12) +#define RT5670_G_HPOMIX_SFT 12 +#define RT5670_M_INR1_HMR (0x1 << 3) +#define RT5670_M_INR1_HMR_SFT 3 +#define RT5670_M_DACR1_HMR (0x1 << 2) +#define RT5670_M_DACR1_HMR_SFT 2 +#define RT5670_M_INL1_HML (0x1 << 1) +#define RT5670_M_INL1_HML_SFT 1 +#define RT5670_M_DACL1_HML (0x1) +#define RT5670_M_DACL1_HML_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5670_M_DAC_R2_MA (0x1 << 15) +#define RT5670_M_DAC_R2_MA_SFT 15 +#define RT5670_M_DAC_L2_MA (0x1 << 14) +#define RT5670_M_DAC_L2_MA_SFT 14 +#define RT5670_M_OV_R_MM (0x1 << 13) +#define RT5670_M_OV_R_MM_SFT 13 +#define RT5670_M_OV_L_MM (0x1 << 12) +#define RT5670_M_OV_L_MM_SFT 12 +#define RT5670_G_MONOMIX_MASK (0x1 << 10) +#define RT5670_G_MONOMIX_SFT 10 +#define RT5670_M_DAC_R2_MM (0x1 << 9) +#define RT5670_M_DAC_R2_MM_SFT 9 +#define RT5670_M_DAC_L2_MM (0x1 << 8) +#define RT5670_M_DAC_L2_MM_SFT 8 +#define RT5670_M_BST4_MM (0x1 << 7) +#define RT5670_M_BST4_MM_SFT 7 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5670_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5670_G_BST3_OM_L_SFT 13 +#define RT5670_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_L_SFT 10 +#define RT5670_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_L_SFT 7 +#define RT5670_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5670_G_IN_L_OM_L_SFT 4 +#define RT5670_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5670_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5670_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5670_G_DAC_R2_OM_L_SFT 13 +#define RT5670_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5670_G_DAC_L2_OM_L_SFT 10 +#define RT5670_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5670_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5670_M_BST1_OM_L (0x1 << 5) +#define RT5670_M_BST1_OM_L_SFT 5 +#define RT5670_M_IN_L_OM_L (0x1 << 4) +#define RT5670_M_IN_L_OM_L_SFT 4 +#define RT5670_M_DAC_L2_OM_L (0x1 << 1) +#define RT5670_M_DAC_L2_OM_L_SFT 1 +#define RT5670_M_DAC_L1_OM_L (0x1) +#define RT5670_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5670_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5670_G_BST4_OM_R_SFT 13 +#define RT5670_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_R_SFT 10 +#define RT5670_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_R_SFT 7 +#define RT5670_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5670_G_IN_R_OM_R_SFT 4 +#define RT5670_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5670_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5670_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5670_G_DAC_L2_OM_R_SFT 13 +#define RT5670_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5670_G_DAC_R2_OM_R_SFT 10 +#define RT5670_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5670_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5670_M_BST2_OM_R (0x1 << 6) +#define RT5670_M_BST2_OM_R_SFT 6 +#define RT5670_M_IN_R_OM_R (0x1 << 4) +#define RT5670_M_IN_R_OM_R_SFT 4 +#define RT5670_M_DAC_R2_OM_R (0x1 << 1) +#define RT5670_M_DAC_R2_OM_R_SFT 1 +#define RT5670_M_DAC_R1_OM_R (0x1) +#define RT5670_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5670_M_DAC_L1_LM (0x1 << 15) +#define RT5670_M_DAC_L1_LM_SFT 15 +#define RT5670_M_DAC_R1_LM (0x1 << 14) +#define RT5670_M_DAC_R1_LM_SFT 14 +#define RT5670_M_OV_L_LM (0x1 << 13) +#define RT5670_M_OV_L_LM_SFT 13 +#define RT5670_M_OV_R_LM (0x1 << 12) +#define RT5670_M_OV_R_LM_SFT 12 +#define RT5670_G_LOUTMIX_MASK (0x1 << 11) +#define RT5670_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5670_PWR_I2S1 (0x1 << 15) +#define RT5670_PWR_I2S1_BIT 15 +#define RT5670_PWR_I2S2 (0x1 << 14) +#define RT5670_PWR_I2S2_BIT 14 +#define RT5670_PWR_DAC_L1 (0x1 << 12) +#define RT5670_PWR_DAC_L1_BIT 12 +#define RT5670_PWR_DAC_R1 (0x1 << 11) +#define RT5670_PWR_DAC_R1_BIT 11 +#define RT5670_PWR_DAC_L2 (0x1 << 7) +#define RT5670_PWR_DAC_L2_BIT 7 +#define RT5670_PWR_DAC_R2 (0x1 << 6) +#define RT5670_PWR_DAC_R2_BIT 6 +#define RT5670_PWR_ADC_L (0x1 << 2) +#define RT5670_PWR_ADC_L_BIT 2 +#define RT5670_PWR_ADC_R (0x1 << 1) +#define RT5670_PWR_ADC_R_BIT 1 +#define RT5670_PWR_CLS_D (0x1) +#define RT5670_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5670_PWR_ADC_S1F (0x1 << 15) +#define RT5670_PWR_ADC_S1F_BIT 15 +#define RT5670_PWR_ADC_MF_L (0x1 << 14) +#define RT5670_PWR_ADC_MF_L_BIT 14 +#define RT5670_PWR_ADC_MF_R (0x1 << 13) +#define RT5670_PWR_ADC_MF_R_BIT 13 +#define RT5670_PWR_I2S_DSP (0x1 << 12) +#define RT5670_PWR_I2S_DSP_BIT 12 +#define RT5670_PWR_DAC_S1F (0x1 << 11) +#define RT5670_PWR_DAC_S1F_BIT 11 +#define RT5670_PWR_DAC_MF_L (0x1 << 10) +#define RT5670_PWR_DAC_MF_L_BIT 10 +#define RT5670_PWR_DAC_MF_R (0x1 << 9) +#define RT5670_PWR_DAC_MF_R_BIT 9 +#define RT5670_PWR_ADC_S2F (0x1 << 8) +#define RT5670_PWR_ADC_S2F_BIT 8 +#define RT5670_PWR_PDM1 (0x1 << 7) +#define RT5670_PWR_PDM1_BIT 7 +#define RT5670_PWR_PDM2 (0x1 << 6) +#define RT5670_PWR_PDM2_BIT 6 + +/* Power Management for Analog 1 (0x63) */ +#define RT5670_PWR_VREF1 (0x1 << 15) +#define RT5670_PWR_VREF1_BIT 15 +#define RT5670_PWR_FV1 (0x1 << 14) +#define RT5670_PWR_FV1_BIT 14 +#define RT5670_PWR_MB (0x1 << 13) +#define RT5670_PWR_MB_BIT 13 +#define RT5670_PWR_LM (0x1 << 12) +#define RT5670_PWR_LM_BIT 12 +#define RT5670_PWR_BG (0x1 << 11) +#define RT5670_PWR_BG_BIT 11 +#define RT5670_PWR_HP_L (0x1 << 7) +#define RT5670_PWR_HP_L_BIT 7 +#define RT5670_PWR_HP_R (0x1 << 6) +#define RT5670_PWR_HP_R_BIT 6 +#define RT5670_PWR_HA (0x1 << 5) +#define RT5670_PWR_HA_BIT 5 +#define RT5670_PWR_VREF2 (0x1 << 4) +#define RT5670_PWR_VREF2_BIT 4 +#define RT5670_PWR_FV2 (0x1 << 3) +#define RT5670_PWR_FV2_BIT 3 +#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5670_PWR_BST1 (0x1 << 15) +#define RT5670_PWR_BST1_BIT 15 +#define RT5670_PWR_BST2 (0x1 << 13) +#define RT5670_PWR_BST2_BIT 13 +#define RT5670_PWR_MB1 (0x1 << 11) +#define RT5670_PWR_MB1_BIT 11 +#define RT5670_PWR_MB2 (0x1 << 10) +#define RT5670_PWR_MB2_BIT 10 +#define RT5670_PWR_PLL (0x1 << 9) +#define RT5670_PWR_PLL_BIT 9 +#define RT5670_PWR_BST1_P (0x1 << 6) +#define RT5670_PWR_BST1_P_BIT 6 +#define RT5670_PWR_BST2_P (0x1 << 4) +#define RT5670_PWR_BST2_P_BIT 4 +#define RT5670_PWR_JD1 (0x1 << 2) +#define RT5670_PWR_JD1_BIT 2 +#define RT5670_PWR_JD (0x1 << 1) +#define RT5670_PWR_JD_BIT 1 + +/* Power Management for Mixer (0x65) */ +#define RT5670_PWR_OM_L (0x1 << 15) +#define RT5670_PWR_OM_L_BIT 15 +#define RT5670_PWR_OM_R (0x1 << 14) +#define RT5670_PWR_OM_R_BIT 14 +#define RT5670_PWR_RM_L (0x1 << 11) +#define RT5670_PWR_RM_L_BIT 11 +#define RT5670_PWR_RM_R (0x1 << 10) +#define RT5670_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5670_PWR_HV_L (0x1 << 11) +#define RT5670_PWR_HV_L_BIT 11 +#define RT5670_PWR_HV_R (0x1 << 10) +#define RT5670_PWR_HV_R_BIT 10 +#define RT5670_PWR_IN_L (0x1 << 9) +#define RT5670_PWR_IN_L_BIT 9 +#define RT5670_PWR_IN_R (0x1 << 8) +#define RT5670_PWR_IN_R_BIT 8 +#define RT5670_PWR_MIC_DET (0x1 << 5) +#define RT5670_PWR_MIC_DET_BIT 5 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5670_I2S_MS_MASK (0x1 << 15) +#define RT5670_I2S_MS_SFT 15 +#define RT5670_I2S_MS_M (0x0 << 15) +#define RT5670_I2S_MS_S (0x1 << 15) +#define RT5670_I2S_IF_MASK (0x7 << 12) +#define RT5670_I2S_IF_SFT 12 +#define RT5670_I2S_O_CP_MASK (0x3 << 10) +#define RT5670_I2S_O_CP_SFT 10 +#define RT5670_I2S_O_CP_OFF (0x0 << 10) +#define RT5670_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5670_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5670_I2S_I_CP_MASK (0x3 << 8) +#define RT5670_I2S_I_CP_SFT 8 +#define RT5670_I2S_I_CP_OFF (0x0 << 8) +#define RT5670_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5670_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5670_I2S_BP_MASK (0x1 << 7) +#define RT5670_I2S_BP_SFT 7 +#define RT5670_I2S_BP_NOR (0x0 << 7) +#define RT5670_I2S_BP_INV (0x1 << 7) +#define RT5670_I2S_DL_MASK (0x3 << 2) +#define RT5670_I2S_DL_SFT 2 +#define RT5670_I2S_DL_16 (0x0 << 2) +#define RT5670_I2S_DL_20 (0x1 << 2) +#define RT5670_I2S_DL_24 (0x2 << 2) +#define RT5670_I2S_DL_8 (0x3 << 2) +#define RT5670_I2S_DF_MASK (0x3) +#define RT5670_I2S_DF_SFT 0 +#define RT5670_I2S_DF_I2S (0x0) +#define RT5670_I2S_DF_LEFT (0x1) +#define RT5670_I2S_DF_PCM_A (0x2) +#define RT5670_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5670_I2S2_SDI_MASK (0x1 << 6) +#define RT5670_I2S2_SDI_SFT 6 +#define RT5670_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5670_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5670_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5670_I2S_BCLK_MS1_SFT 15 +#define RT5670_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5670_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5670_I2S_PD1_MASK (0x7 << 12) +#define RT5670_I2S_PD1_SFT 12 +#define RT5670_I2S_PD1_1 (0x0 << 12) +#define RT5670_I2S_PD1_2 (0x1 << 12) +#define RT5670_I2S_PD1_3 (0x2 << 12) +#define RT5670_I2S_PD1_4 (0x3 << 12) +#define RT5670_I2S_PD1_6 (0x4 << 12) +#define RT5670_I2S_PD1_8 (0x5 << 12) +#define RT5670_I2S_PD1_12 (0x6 << 12) +#define RT5670_I2S_PD1_16 (0x7 << 12) +#define RT5670_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5670_I2S_BCLK_MS2_SFT 11 +#define RT5670_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5670_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5670_I2S_PD2_MASK (0x7 << 8) +#define RT5670_I2S_PD2_SFT 8 +#define RT5670_I2S_PD2_1 (0x0 << 8) +#define RT5670_I2S_PD2_2 (0x1 << 8) +#define RT5670_I2S_PD2_3 (0x2 << 8) +#define RT5670_I2S_PD2_4 (0x3 << 8) +#define RT5670_I2S_PD2_6 (0x4 << 8) +#define RT5670_I2S_PD2_8 (0x5 << 8) +#define RT5670_I2S_PD2_12 (0x6 << 8) +#define RT5670_I2S_PD2_16 (0x7 << 8) +#define RT5670_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5670_I2S_BCLK_MS3_SFT 7 +#define RT5670_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5670_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5670_I2S_PD3_MASK (0x7 << 4) +#define RT5670_I2S_PD3_SFT 4 +#define RT5670_I2S_PD3_1 (0x0 << 4) +#define RT5670_I2S_PD3_2 (0x1 << 4) +#define RT5670_I2S_PD3_3 (0x2 << 4) +#define RT5670_I2S_PD3_4 (0x3 << 4) +#define RT5670_I2S_PD3_6 (0x4 << 4) +#define RT5670_I2S_PD3_8 (0x5 << 4) +#define RT5670_I2S_PD3_12 (0x6 << 4) +#define RT5670_I2S_PD3_16 (0x7 << 4) +#define RT5670_DAC_OSR_MASK (0x3 << 2) +#define RT5670_DAC_OSR_SFT 2 +#define RT5670_DAC_OSR_128 (0x0 << 2) +#define RT5670_DAC_OSR_64 (0x1 << 2) +#define RT5670_DAC_OSR_32 (0x2 << 2) +#define RT5670_DAC_OSR_16 (0x3 << 2) +#define RT5670_ADC_OSR_MASK (0x3) +#define RT5670_ADC_OSR_SFT 0 +#define RT5670_ADC_OSR_128 (0x0) +#define RT5670_ADC_OSR_64 (0x1) +#define RT5670_ADC_OSR_32 (0x2) +#define RT5670_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5670_DAC_L_OSR_MASK (0x3 << 14) +#define RT5670_DAC_L_OSR_SFT 14 +#define RT5670_DAC_L_OSR_128 (0x0 << 14) +#define RT5670_DAC_L_OSR_64 (0x1 << 14) +#define RT5670_DAC_L_OSR_32 (0x2 << 14) +#define RT5670_DAC_L_OSR_16 (0x3 << 14) +#define RT5670_ADC_R_OSR_MASK (0x3 << 12) +#define RT5670_ADC_R_OSR_SFT 12 +#define RT5670_ADC_R_OSR_128 (0x0 << 12) +#define RT5670_ADC_R_OSR_64 (0x1 << 12) +#define RT5670_ADC_R_OSR_32 (0x2 << 12) +#define RT5670_ADC_R_OSR_16 (0x3 << 12) +#define RT5670_DAHPF_EN (0x1 << 11) +#define RT5670_DAHPF_EN_SFT 11 +#define RT5670_ADHPF_EN (0x1 << 10) +#define RT5670_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5670_DMIC_1_EN_MASK (0x1 << 15) +#define RT5670_DMIC_1_EN_SFT 15 +#define RT5670_DMIC_1_DIS (0x0 << 15) +#define RT5670_DMIC_1_EN (0x1 << 15) +#define RT5670_DMIC_2_EN_MASK (0x1 << 14) +#define RT5670_DMIC_2_EN_SFT 14 +#define RT5670_DMIC_2_DIS (0x0 << 14) +#define RT5670_DMIC_2_EN (0x1 << 14) +#define RT5670_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5670_DMIC_1L_LH_SFT 13 +#define RT5670_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5670_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5670_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5670_DMIC_1R_LH_SFT 12 +#define RT5670_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5670_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5670_DMIC_2_DP_MASK (0x1 << 10) +#define RT5670_DMIC_2_DP_SFT 10 +#define RT5670_DMIC_2_DP_GPIO8 (0x0 << 10) +#define RT5670_DMIC_2_DP_IN3N (0x1 << 10) +#define RT5670_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5670_DMIC_2L_LH_SFT 9 +#define RT5670_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5670_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5670_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5670_DMIC_2R_LH_SFT 8 +#define RT5670_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5670_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5670_DMIC_CLK_MASK (0x7 << 5) +#define RT5670_DMIC_CLK_SFT 5 +#define RT5670_DMIC_3_EN_MASK (0x1 << 4) +#define RT5670_DMIC_3_EN_SFT 4 +#define RT5670_DMIC_3_DIS (0x0 << 4) +#define RT5670_DMIC_3_EN (0x1 << 4) +#define RT5670_DMIC_1_DP_MASK (0x3 << 0) +#define RT5670_DMIC_1_DP_SFT 0 +#define RT5670_DMIC_1_DP_GPIO6 (0x0 << 0) +#define RT5670_DMIC_1_DP_IN2P (0x1 << 0) +#define RT5670_DMIC_1_DP_GPIO7 (0x2 << 0) + +/* Digital Microphone Control2 (0x76) */ +#define RT5670_DMIC_3_DP_MASK (0x3 << 6) +#define RT5670_DMIC_3_DP_SFT 6 +#define RT5670_DMIC_3_DP_GPIO9 (0x0 << 6) +#define RT5670_DMIC_3_DP_GPIO10 (0x1 << 6) +#define RT5670_DMIC_3_DP_GPIO5 (0x2 << 6) + +/* Global Clock Control (0x80) */ +#define RT5670_SCLK_SRC_MASK (0x3 << 14) +#define RT5670_SCLK_SRC_SFT 14 +#define RT5670_SCLK_SRC_MCLK (0x0 << 14) +#define RT5670_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ +#define RT5670_PLL1_SRC_MASK (0x3 << 12) +#define RT5670_PLL1_SRC_SFT 12 +#define RT5670_PLL1_SRC_MCLK (0x0 << 12) +#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5670_PLL1_PD_MASK (0x1 << 3) +#define RT5670_PLL1_PD_SFT 3 +#define RT5670_PLL1_PD_1 (0x0 << 3) +#define RT5670_PLL1_PD_2 (0x1 << 3) + +#define RT5670_PLL_INP_MAX 40000000 +#define RT5670_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5670_PLL_N_MAX 0x1ff +#define RT5670_PLL_N_MASK (RT5670_PLL_N_MAX << 7) +#define RT5670_PLL_N_SFT 7 +#define RT5670_PLL_K_MAX 0x1f +#define RT5670_PLL_K_MASK (RT5670_PLL_K_MAX) +#define RT5670_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5670_PLL_M_MAX 0xf +#define RT5670_PLL_M_MASK (RT5670_PLL_M_MAX << 12) +#define RT5670_PLL_M_SFT 12 +#define RT5670_PLL_M_BP (0x1 << 11) +#define RT5670_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5670_STO_T_MASK (0x1 << 15) +#define RT5670_STO_T_SFT 15 +#define RT5670_STO_T_SCLK (0x0 << 15) +#define RT5670_STO_T_LRCK1 (0x1 << 15) +#define RT5670_M1_T_MASK (0x1 << 14) +#define RT5670_M1_T_SFT 14 +#define RT5670_M1_T_I2S2 (0x0 << 14) +#define RT5670_M1_T_I2S2_D3 (0x1 << 14) +#define RT5670_I2S2_F_MASK (0x1 << 12) +#define RT5670_I2S2_F_SFT 12 +#define RT5670_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5670_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5670_DMIC_1_M_MASK (0x1 << 9) +#define RT5670_DMIC_1_M_SFT 9 +#define RT5670_DMIC_1_M_NOR (0x0 << 9) +#define RT5670_DMIC_1_M_ASYN (0x1 << 9) +#define RT5670_DMIC_2_M_MASK (0x1 << 8) +#define RT5670_DMIC_2_M_SFT 8 +#define RT5670_DMIC_2_M_NOR (0x0 << 8) +#define RT5670_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5670_MDA_L_M_MASK (0x1 << 15) +#define RT5670_MDA_L_M_SFT 15 +#define RT5670_MDA_L_M_NOR (0x0 << 15) +#define RT5670_MDA_L_M_ASYN (0x1 << 15) +#define RT5670_MDA_R_M_MASK (0x1 << 14) +#define RT5670_MDA_R_M_SFT 14 +#define RT5670_MDA_R_M_NOR (0x0 << 14) +#define RT5670_MDA_R_M_ASYN (0x1 << 14) +#define RT5670_MAD_L_M_MASK (0x1 << 13) +#define RT5670_MAD_L_M_SFT 13 +#define RT5670_MAD_L_M_NOR (0x0 << 13) +#define RT5670_MAD_L_M_ASYN (0x1 << 13) +#define RT5670_MAD_R_M_MASK (0x1 << 12) +#define RT5670_MAD_R_M_SFT 12 +#define RT5670_MAD_R_M_NOR (0x0 << 12) +#define RT5670_MAD_R_M_ASYN (0x1 << 12) +#define RT5670_ADC_M_MASK (0x1 << 11) +#define RT5670_ADC_M_SFT 11 +#define RT5670_ADC_M_NOR (0x0 << 11) +#define RT5670_ADC_M_ASYN (0x1 << 11) +#define RT5670_STO_DAC_M_MASK (0x1 << 5) +#define RT5670_STO_DAC_M_SFT 5 +#define RT5670_STO_DAC_M_NOR (0x0 << 5) +#define RT5670_STO_DAC_M_ASYN (0x1 << 5) +#define RT5670_I2S1_R_D_MASK (0x1 << 4) +#define RT5670_I2S1_R_D_SFT 4 +#define RT5670_I2S1_R_D_DIS (0x0 << 4) +#define RT5670_I2S1_R_D_EN (0x1 << 4) +#define RT5670_I2S2_R_D_MASK (0x1 << 3) +#define RT5670_I2S2_R_D_SFT 3 +#define RT5670_I2S2_R_D_DIS (0x0 << 3) +#define RT5670_I2S2_R_D_EN (0x1 << 3) +#define RT5670_PRE_SCLK_MASK (0x3) +#define RT5670_PRE_SCLK_SFT 0 +#define RT5670_PRE_SCLK_512 (0x0) +#define RT5670_PRE_SCLK_1024 (0x1) +#define RT5670_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5670_I2S1_RATE_MASK (0xf << 12) +#define RT5670_I2S1_RATE_SFT 12 +#define RT5670_I2S2_RATE_MASK (0xf << 8) +#define RT5670_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5670_I2S1_PD_MASK (0x7 << 12) +#define RT5670_I2S1_PD_SFT 12 +#define RT5670_I2S2_PD_MASK (0x7 << 8) +#define RT5670_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5670_HP_OVCD_MASK (0x1 << 10) +#define RT5670_HP_OVCD_SFT 10 +#define RT5670_HP_OVCD_DIS (0x0 << 10) +#define RT5670_HP_OVCD_EN (0x1 << 10) +#define RT5670_HP_OC_TH_MASK (0x3 << 8) +#define RT5670_HP_OC_TH_SFT 8 +#define RT5670_HP_OC_TH_90 (0x0 << 8) +#define RT5670_HP_OC_TH_105 (0x1 << 8) +#define RT5670_HP_OC_TH_120 (0x2 << 8) +#define RT5670_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5670_CLSD_OC_MASK (0x1 << 9) +#define RT5670_CLSD_OC_SFT 9 +#define RT5670_CLSD_OC_PU (0x0 << 9) +#define RT5670_CLSD_OC_PD (0x1 << 9) +#define RT5670_AUTO_PD_MASK (0x1 << 8) +#define RT5670_AUTO_PD_SFT 8 +#define RT5670_AUTO_PD_DIS (0x0 << 8) +#define RT5670_AUTO_PD_EN (0x1 << 8) +#define RT5670_CLSD_OC_TH_MASK (0x3f) +#define RT5670_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5670_CLSD_RATIO_MASK (0xf << 12) +#define RT5670_CLSD_RATIO_SFT 12 +#define RT5670_CLSD_OM_MASK (0x1 << 11) +#define RT5670_CLSD_OM_SFT 11 +#define RT5670_CLSD_OM_MONO (0x0 << 11) +#define RT5670_CLSD_OM_STO (0x1 << 11) +#define RT5670_CLSD_SCH_MASK (0x1 << 10) +#define RT5670_CLSD_SCH_SFT 10 +#define RT5670_CLSD_SCH_L (0x0 << 10) +#define RT5670_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5670_SMT_TRIG_MASK (0x1 << 15) +#define RT5670_SMT_TRIG_SFT 15 +#define RT5670_SMT_TRIG_DIS (0x0 << 15) +#define RT5670_SMT_TRIG_EN (0x1 << 15) +#define RT5670_HP_L_SMT_MASK (0x1 << 9) +#define RT5670_HP_L_SMT_SFT 9 +#define RT5670_HP_L_SMT_DIS (0x0 << 9) +#define RT5670_HP_L_SMT_EN (0x1 << 9) +#define RT5670_HP_R_SMT_MASK (0x1 << 8) +#define RT5670_HP_R_SMT_SFT 8 +#define RT5670_HP_R_SMT_DIS (0x0 << 8) +#define RT5670_HP_R_SMT_EN (0x1 << 8) +#define RT5670_HP_CD_PD_MASK (0x1 << 7) +#define RT5670_HP_CD_PD_SFT 7 +#define RT5670_HP_CD_PD_DIS (0x0 << 7) +#define RT5670_HP_CD_PD_EN (0x1 << 7) +#define RT5670_RSTN_MASK (0x1 << 6) +#define RT5670_RSTN_SFT 6 +#define RT5670_RSTN_DIS (0x0 << 6) +#define RT5670_RSTN_EN (0x1 << 6) +#define RT5670_RSTP_MASK (0x1 << 5) +#define RT5670_RSTP_SFT 5 +#define RT5670_RSTP_DIS (0x0 << 5) +#define RT5670_RSTP_EN (0x1 << 5) +#define RT5670_HP_CO_MASK (0x1 << 4) +#define RT5670_HP_CO_SFT 4 +#define RT5670_HP_CO_DIS (0x0 << 4) +#define RT5670_HP_CO_EN (0x1 << 4) +#define RT5670_HP_CP_MASK (0x1 << 3) +#define RT5670_HP_CP_SFT 3 +#define RT5670_HP_CP_PD (0x0 << 3) +#define RT5670_HP_CP_PU (0x1 << 3) +#define RT5670_HP_SG_MASK (0x1 << 2) +#define RT5670_HP_SG_SFT 2 +#define RT5670_HP_SG_DIS (0x0 << 2) +#define RT5670_HP_SG_EN (0x1 << 2) +#define RT5670_HP_DP_MASK (0x1 << 1) +#define RT5670_HP_DP_SFT 1 +#define RT5670_HP_DP_PD (0x0 << 1) +#define RT5670_HP_DP_PU (0x1 << 1) +#define RT5670_HP_CB_MASK (0x1) +#define RT5670_HP_CB_SFT 0 +#define RT5670_HP_CB_PD (0x0) +#define RT5670_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5670_DEPOP_MASK (0x1 << 13) +#define RT5670_DEPOP_SFT 13 +#define RT5670_DEPOP_AUTO (0x0 << 13) +#define RT5670_DEPOP_MAN (0x1 << 13) +#define RT5670_RAMP_MASK (0x1 << 12) +#define RT5670_RAMP_SFT 12 +#define RT5670_RAMP_DIS (0x0 << 12) +#define RT5670_RAMP_EN (0x1 << 12) +#define RT5670_BPS_MASK (0x1 << 11) +#define RT5670_BPS_SFT 11 +#define RT5670_BPS_DIS (0x0 << 11) +#define RT5670_BPS_EN (0x1 << 11) +#define RT5670_FAST_UPDN_MASK (0x1 << 10) +#define RT5670_FAST_UPDN_SFT 10 +#define RT5670_FAST_UPDN_DIS (0x0 << 10) +#define RT5670_FAST_UPDN_EN (0x1 << 10) +#define RT5670_MRES_MASK (0x3 << 8) +#define RT5670_MRES_SFT 8 +#define RT5670_MRES_15MO (0x0 << 8) +#define RT5670_MRES_25MO (0x1 << 8) +#define RT5670_MRES_35MO (0x2 << 8) +#define RT5670_MRES_45MO (0x3 << 8) +#define RT5670_VLO_MASK (0x1 << 7) +#define RT5670_VLO_SFT 7 +#define RT5670_VLO_3V (0x0 << 7) +#define RT5670_VLO_32V (0x1 << 7) +#define RT5670_DIG_DP_MASK (0x1 << 6) +#define RT5670_DIG_DP_SFT 6 +#define RT5670_DIG_DP_DIS (0x0 << 6) +#define RT5670_DIG_DP_EN (0x1 << 6) +#define RT5670_DP_TH_MASK (0x3 << 4) +#define RT5670_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5670_CP_SYS_MASK (0x7 << 12) +#define RT5670_CP_SYS_SFT 12 +#define RT5670_CP_FQ1_MASK (0x7 << 8) +#define RT5670_CP_FQ1_SFT 8 +#define RT5670_CP_FQ2_MASK (0x7 << 4) +#define RT5670_CP_FQ2_SFT 4 +#define RT5670_CP_FQ3_MASK (0x7) +#define RT5670_CP_FQ3_SFT 0 +#define RT5670_CP_FQ_1_5_KHZ 0 +#define RT5670_CP_FQ_3_KHZ 1 +#define RT5670_CP_FQ_6_KHZ 2 +#define RT5670_CP_FQ_12_KHZ 3 +#define RT5670_CP_FQ_24_KHZ 4 +#define RT5670_CP_FQ_48_KHZ 5 +#define RT5670_CP_FQ_96_KHZ 6 +#define RT5670_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5670_OSW_L_MASK (0x1 << 11) +#define RT5670_OSW_L_SFT 11 +#define RT5670_OSW_L_DIS (0x0 << 11) +#define RT5670_OSW_L_EN (0x1 << 11) +#define RT5670_OSW_R_MASK (0x1 << 10) +#define RT5670_OSW_R_SFT 10 +#define RT5670_OSW_R_DIS (0x0 << 10) +#define RT5670_OSW_R_EN (0x1 << 10) +#define RT5670_PM_HP_MASK (0x3 << 8) +#define RT5670_PM_HP_SFT 8 +#define RT5670_PM_HP_LV (0x0 << 8) +#define RT5670_PM_HP_MV (0x1 << 8) +#define RT5670_PM_HP_HV (0x2 << 8) +#define RT5670_IB_HP_MASK (0x3 << 6) +#define RT5670_IB_HP_SFT 6 +#define RT5670_IB_HP_125IL (0x0 << 6) +#define RT5670_IB_HP_25IL (0x1 << 6) +#define RT5670_IB_HP_5IL (0x2 << 6) +#define RT5670_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5670_PVDD_DET_MASK (0x1 << 15) +#define RT5670_PVDD_DET_SFT 15 +#define RT5670_PVDD_DET_DIS (0x0 << 15) +#define RT5670_PVDD_DET_EN (0x1 << 15) +#define RT5670_SPK_AG_MASK (0x1 << 14) +#define RT5670_SPK_AG_SFT 14 +#define RT5670_SPK_AG_DIS (0x0 << 14) +#define RT5670_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5670_MIC1_BS_MASK (0x1 << 15) +#define RT5670_MIC1_BS_SFT 15 +#define RT5670_MIC1_BS_9AV (0x0 << 15) +#define RT5670_MIC1_BS_75AV (0x1 << 15) +#define RT5670_MIC2_BS_MASK (0x1 << 14) +#define RT5670_MIC2_BS_SFT 14 +#define RT5670_MIC2_BS_9AV (0x0 << 14) +#define RT5670_MIC2_BS_75AV (0x1 << 14) +#define RT5670_MIC1_CLK_MASK (0x1 << 13) +#define RT5670_MIC1_CLK_SFT 13 +#define RT5670_MIC1_CLK_DIS (0x0 << 13) +#define RT5670_MIC1_CLK_EN (0x1 << 13) +#define RT5670_MIC2_CLK_MASK (0x1 << 12) +#define RT5670_MIC2_CLK_SFT 12 +#define RT5670_MIC2_CLK_DIS (0x0 << 12) +#define RT5670_MIC2_CLK_EN (0x1 << 12) +#define RT5670_MIC1_OVCD_MASK (0x1 << 11) +#define RT5670_MIC1_OVCD_SFT 11 +#define RT5670_MIC1_OVCD_DIS (0x0 << 11) +#define RT5670_MIC1_OVCD_EN (0x1 << 11) +#define RT5670_MIC1_OVTH_MASK (0x3 << 9) +#define RT5670_MIC1_OVTH_SFT 9 +#define RT5670_MIC1_OVTH_600UA (0x0 << 9) +#define RT5670_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5670_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5670_MIC2_OVCD_MASK (0x1 << 8) +#define RT5670_MIC2_OVCD_SFT 8 +#define RT5670_MIC2_OVCD_DIS (0x0 << 8) +#define RT5670_MIC2_OVCD_EN (0x1 << 8) +#define RT5670_MIC2_OVTH_MASK (0x3 << 6) +#define RT5670_MIC2_OVTH_SFT 6 +#define RT5670_MIC2_OVTH_600UA (0x0 << 6) +#define RT5670_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5670_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5670_PWR_MB_MASK (0x1 << 5) +#define RT5670_PWR_MB_SFT 5 +#define RT5670_PWR_MB_PD (0x0 << 5) +#define RT5670_PWR_MB_PU (0x1 << 5) +#define RT5670_PWR_CLK25M_MASK (0x1 << 4) +#define RT5670_PWR_CLK25M_SFT 4 +#define RT5670_PWR_CLK25M_PD (0x0 << 4) +#define RT5670_PWR_CLK25M_PU (0x1 << 4) + +/* Analog JD Control 1 (0x94) */ +#define RT5670_JD1_MODE_MASK (0x3 << 0) +#define RT5670_JD1_MODE_0 (0x0 << 0) +#define RT5670_JD1_MODE_1 (0x1 << 0) +#define RT5670_JD1_MODE_2 (0x2 << 0) + +/* VAD Control 4 (0x9d) */ +#define RT5670_VAD_SEL_MASK (0x3 << 8) +#define RT5670_VAD_SEL_SFT 8 + +/* EQ Control 1 (0xb0) */ +#define RT5670_EQ_SRC_MASK (0x1 << 15) +#define RT5670_EQ_SRC_SFT 15 +#define RT5670_EQ_SRC_DAC (0x0 << 15) +#define RT5670_EQ_SRC_ADC (0x1 << 15) +#define RT5670_EQ_UPD (0x1 << 14) +#define RT5670_EQ_UPD_BIT 14 +#define RT5670_EQ_CD_MASK (0x1 << 13) +#define RT5670_EQ_CD_SFT 13 +#define RT5670_EQ_CD_DIS (0x0 << 13) +#define RT5670_EQ_CD_EN (0x1 << 13) +#define RT5670_EQ_DITH_MASK (0x3 << 8) +#define RT5670_EQ_DITH_SFT 8 +#define RT5670_EQ_DITH_NOR (0x0 << 8) +#define RT5670_EQ_DITH_LSB (0x1 << 8) +#define RT5670_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5670_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5670_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5670_EQ_HPF1_M_SFT 8 +#define RT5670_EQ_HPF1_M_HI (0x0 << 8) +#define RT5670_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5670_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5670_EQ_LPF1_M_SFT 7 +#define RT5670_EQ_LPF1_M_LO (0x0 << 7) +#define RT5670_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5670_EQ_HPF2_MASK (0x1 << 6) +#define RT5670_EQ_HPF2_SFT 6 +#define RT5670_EQ_HPF2_DIS (0x0 << 6) +#define RT5670_EQ_HPF2_EN (0x1 << 6) +#define RT5670_EQ_HPF1_MASK (0x1 << 5) +#define RT5670_EQ_HPF1_SFT 5 +#define RT5670_EQ_HPF1_DIS (0x0 << 5) +#define RT5670_EQ_HPF1_EN (0x1 << 5) +#define RT5670_EQ_BPF4_MASK (0x1 << 4) +#define RT5670_EQ_BPF4_SFT 4 +#define RT5670_EQ_BPF4_DIS (0x0 << 4) +#define RT5670_EQ_BPF4_EN (0x1 << 4) +#define RT5670_EQ_BPF3_MASK (0x1 << 3) +#define RT5670_EQ_BPF3_SFT 3 +#define RT5670_EQ_BPF3_DIS (0x0 << 3) +#define RT5670_EQ_BPF3_EN (0x1 << 3) +#define RT5670_EQ_BPF2_MASK (0x1 << 2) +#define RT5670_EQ_BPF2_SFT 2 +#define RT5670_EQ_BPF2_DIS (0x0 << 2) +#define RT5670_EQ_BPF2_EN (0x1 << 2) +#define RT5670_EQ_BPF1_MASK (0x1 << 1) +#define RT5670_EQ_BPF1_SFT 1 +#define RT5670_EQ_BPF1_DIS (0x0 << 1) +#define RT5670_EQ_BPF1_EN (0x1 << 1) +#define RT5670_EQ_LPF_MASK (0x1) +#define RT5670_EQ_LPF_SFT 0 +#define RT5670_EQ_LPF_DIS (0x0) +#define RT5670_EQ_LPF_EN (0x1) +#define RT5670_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5670_MT_MASK (0x1 << 15) +#define RT5670_MT_SFT 15 +#define RT5670_MT_DIS (0x0 << 15) +#define RT5670_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5670_DRC_AGC_P_MASK (0x1 << 15) +#define RT5670_DRC_AGC_P_SFT 15 +#define RT5670_DRC_AGC_P_DAC (0x0 << 15) +#define RT5670_DRC_AGC_P_ADC (0x1 << 15) +#define RT5670_DRC_AGC_MASK (0x1 << 14) +#define RT5670_DRC_AGC_SFT 14 +#define RT5670_DRC_AGC_DIS (0x0 << 14) +#define RT5670_DRC_AGC_EN (0x1 << 14) +#define RT5670_DRC_AGC_UPD (0x1 << 13) +#define RT5670_DRC_AGC_UPD_BIT 13 +#define RT5670_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5670_DRC_AGC_AR_SFT 8 +#define RT5670_DRC_AGC_R_MASK (0x7 << 5) +#define RT5670_DRC_AGC_R_SFT 5 +#define RT5670_DRC_AGC_R_48K (0x1 << 5) +#define RT5670_DRC_AGC_R_96K (0x2 << 5) +#define RT5670_DRC_AGC_R_192K (0x3 << 5) +#define RT5670_DRC_AGC_R_441K (0x5 << 5) +#define RT5670_DRC_AGC_R_882K (0x6 << 5) +#define RT5670_DRC_AGC_R_1764K (0x7 << 5) +#define RT5670_DRC_AGC_RC_MASK (0x1f) +#define RT5670_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5670_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5670_DRC_AGC_POB_SFT 8 +#define RT5670_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5670_DRC_AGC_CP_SFT 7 +#define RT5670_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5670_DRC_AGC_CP_EN (0x1 << 7) +#define RT5670_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5670_DRC_AGC_CPR_SFT 5 +#define RT5670_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5670_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5670_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5670_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5670_DRC_AGC_PRB_MASK (0x1f) +#define RT5670_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5670_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5670_DRC_AGC_NGB_SFT 12 +#define RT5670_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5670_DRC_AGC_TAR_SFT 7 +#define RT5670_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5670_DRC_AGC_NG_SFT 6 +#define RT5670_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5670_DRC_AGC_NG_EN (0x1 << 6) +#define RT5670_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5670_DRC_AGC_NGH_SFT 5 +#define RT5670_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5670_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5670_DRC_AGC_NGT_MASK (0x1f) +#define RT5670_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5670_JD_MASK (0x7 << 13) +#define RT5670_JD_SFT 13 +#define RT5670_JD_DIS (0x0 << 13) +#define RT5670_JD_GPIO1 (0x1 << 13) +#define RT5670_JD_JD1_IN4P (0x2 << 13) +#define RT5670_JD_JD2_IN4N (0x3 << 13) +#define RT5670_JD_GPIO2 (0x4 << 13) +#define RT5670_JD_GPIO3 (0x5 << 13) +#define RT5670_JD_GPIO4 (0x6 << 13) +#define RT5670_JD_HP_MASK (0x1 << 11) +#define RT5670_JD_HP_SFT 11 +#define RT5670_JD_HP_DIS (0x0 << 11) +#define RT5670_JD_HP_EN (0x1 << 11) +#define RT5670_JD_HP_TRG_MASK (0x1 << 10) +#define RT5670_JD_HP_TRG_SFT 10 +#define RT5670_JD_HP_TRG_LO (0x0 << 10) +#define RT5670_JD_HP_TRG_HI (0x1 << 10) +#define RT5670_JD_SPL_MASK (0x1 << 9) +#define RT5670_JD_SPL_SFT 9 +#define RT5670_JD_SPL_DIS (0x0 << 9) +#define RT5670_JD_SPL_EN (0x1 << 9) +#define RT5670_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5670_JD_SPL_TRG_SFT 8 +#define RT5670_JD_SPL_TRG_LO (0x0 << 8) +#define RT5670_JD_SPL_TRG_HI (0x1 << 8) +#define RT5670_JD_SPR_MASK (0x1 << 7) +#define RT5670_JD_SPR_SFT 7 +#define RT5670_JD_SPR_DIS (0x0 << 7) +#define RT5670_JD_SPR_EN (0x1 << 7) +#define RT5670_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5670_JD_SPR_TRG_SFT 6 +#define RT5670_JD_SPR_TRG_LO (0x0 << 6) +#define RT5670_JD_SPR_TRG_HI (0x1 << 6) +#define RT5670_JD_MO_MASK (0x1 << 5) +#define RT5670_JD_MO_SFT 5 +#define RT5670_JD_MO_DIS (0x0 << 5) +#define RT5670_JD_MO_EN (0x1 << 5) +#define RT5670_JD_MO_TRG_MASK (0x1 << 4) +#define RT5670_JD_MO_TRG_SFT 4 +#define RT5670_JD_MO_TRG_LO (0x0 << 4) +#define RT5670_JD_MO_TRG_HI (0x1 << 4) +#define RT5670_JD_LO_MASK (0x1 << 3) +#define RT5670_JD_LO_SFT 3 +#define RT5670_JD_LO_DIS (0x0 << 3) +#define RT5670_JD_LO_EN (0x1 << 3) +#define RT5670_JD_LO_TRG_MASK (0x1 << 2) +#define RT5670_JD_LO_TRG_SFT 2 +#define RT5670_JD_LO_TRG_LO (0x0 << 2) +#define RT5670_JD_LO_TRG_HI (0x1 << 2) +#define RT5670_JD1_IN4P_MASK (0x1 << 1) +#define RT5670_JD1_IN4P_SFT 1 +#define RT5670_JD1_IN4P_DIS (0x0 << 1) +#define RT5670_JD1_IN4P_EN (0x1 << 1) +#define RT5670_JD2_IN4N_MASK (0x1) +#define RT5670_JD2_IN4N_SFT 0 +#define RT5670_JD2_IN4N_DIS (0x0) +#define RT5670_JD2_IN4N_EN (0x1) + +/* IRQ Control 1 (0xbd) */ +#define RT5670_IRQ_JD_MASK (0x1 << 15) +#define RT5670_IRQ_JD_SFT 15 +#define RT5670_IRQ_JD_BP (0x0 << 15) +#define RT5670_IRQ_JD_NOR (0x1 << 15) +#define RT5670_IRQ_OT_MASK (0x1 << 14) +#define RT5670_IRQ_OT_SFT 14 +#define RT5670_IRQ_OT_BP (0x0 << 14) +#define RT5670_IRQ_OT_NOR (0x1 << 14) +#define RT5670_JD_STKY_MASK (0x1 << 13) +#define RT5670_JD_STKY_SFT 13 +#define RT5670_JD_STKY_DIS (0x0 << 13) +#define RT5670_JD_STKY_EN (0x1 << 13) +#define RT5670_OT_STKY_MASK (0x1 << 12) +#define RT5670_OT_STKY_SFT 12 +#define RT5670_OT_STKY_DIS (0x0 << 12) +#define RT5670_OT_STKY_EN (0x1 << 12) +#define RT5670_JD_P_MASK (0x1 << 11) +#define RT5670_JD_P_SFT 11 +#define RT5670_JD_P_NOR (0x0 << 11) +#define RT5670_JD_P_INV (0x1 << 11) +#define RT5670_OT_P_MASK (0x1 << 10) +#define RT5670_OT_P_SFT 10 +#define RT5670_OT_P_NOR (0x0 << 10) +#define RT5670_OT_P_INV (0x1 << 10) +#define RT5670_JD1_1_EN_MASK (0x1 << 9) +#define RT5670_JD1_1_EN_SFT 9 +#define RT5670_JD1_1_DIS (0x0 << 9) +#define RT5670_JD1_1_EN (0x1 << 9) + +/* IRQ Control 2 (0xbe) */ +#define RT5670_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5670_IRQ_MB1_OC_SFT 15 +#define RT5670_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5670_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5670_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5670_IRQ_MB2_OC_SFT 14 +#define RT5670_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5670_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5670_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5670_MB1_OC_STKY_SFT 11 +#define RT5670_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5670_MB1_OC_STKY_EN (0x1 << 11) +#define RT5670_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5670_MB2_OC_STKY_SFT 10 +#define RT5670_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5670_MB2_OC_STKY_EN (0x1 << 10) +#define RT5670_MB1_OC_P_MASK (0x1 << 7) +#define RT5670_MB1_OC_P_SFT 7 +#define RT5670_MB1_OC_P_NOR (0x0 << 7) +#define RT5670_MB1_OC_P_INV (0x1 << 7) +#define RT5670_MB2_OC_P_MASK (0x1 << 6) +#define RT5670_MB2_OC_P_SFT 6 +#define RT5670_MB2_OC_P_NOR (0x0 << 6) +#define RT5670_MB2_OC_P_INV (0x1 << 6) +#define RT5670_MB1_OC_CLR (0x1 << 3) +#define RT5670_MB1_OC_CLR_SFT 3 +#define RT5670_MB2_OC_CLR (0x1 << 2) +#define RT5670_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5670_GP1_PIN_MASK (0x1 << 15) +#define RT5670_GP1_PIN_SFT 15 +#define RT5670_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5670_GP1_PIN_IRQ (0x1 << 15) +#define RT5670_GP2_PIN_MASK (0x1 << 14) +#define RT5670_GP2_PIN_SFT 14 +#define RT5670_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5670_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5670_GP3_PIN_MASK (0x3 << 12) +#define RT5670_GP3_PIN_SFT 12 +#define RT5670_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5670_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5670_GP3_PIN_IRQ (0x2 << 12) +#define RT5670_GP4_PIN_MASK (0x1 << 11) +#define RT5670_GP4_PIN_SFT 11 +#define RT5670_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5670_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5670_DP_SIG_MASK (0x1 << 10) +#define RT5670_DP_SIG_SFT 10 +#define RT5670_DP_SIG_TEST (0x0 << 10) +#define RT5670_DP_SIG_AP (0x1 << 10) +#define RT5670_GPIO_M_MASK (0x1 << 9) +#define RT5670_GPIO_M_SFT 9 +#define RT5670_GPIO_M_FLT (0x0 << 9) +#define RT5670_GPIO_M_PH (0x1 << 9) +#define RT5670_I2S2_PIN_MASK (0x1 << 8) +#define RT5670_I2S2_PIN_SFT 8 +#define RT5670_I2S2_PIN_I2S (0x0 << 8) +#define RT5670_I2S2_PIN_GPIO (0x1 << 8) +#define RT5670_GP5_PIN_MASK (0x1 << 7) +#define RT5670_GP5_PIN_SFT 7 +#define RT5670_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5670_GP5_PIN_DMIC3_SDA (0x1 << 7) +#define RT5670_GP6_PIN_MASK (0x1 << 6) +#define RT5670_GP6_PIN_SFT 6 +#define RT5670_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5670_GP6_PIN_DMIC1_SDA (0x1 << 6) +#define RT5670_GP7_PIN_MASK (0x3 << 4) +#define RT5670_GP7_PIN_SFT 4 +#define RT5670_GP7_PIN_GPIO7 (0x0 << 4) +#define RT5670_GP7_PIN_DMIC1_SDA (0x1 << 4) +#define RT5670_GP7_PIN_PDM_SCL2 (0x2 << 4) +#define RT5670_GP8_PIN_MASK (0x1 << 3) +#define RT5670_GP8_PIN_SFT 3 +#define RT5670_GP8_PIN_GPIO8 (0x0 << 3) +#define RT5670_GP8_PIN_DMIC2_SDA (0x1 << 3) +#define RT5670_GP9_PIN_MASK (0x1 << 2) +#define RT5670_GP9_PIN_SFT 2 +#define RT5670_GP9_PIN_GPIO9 (0x0 << 2) +#define RT5670_GP9_PIN_DMIC3_SDA (0x1 << 2) +#define RT5670_GP10_PIN_MASK (0x3) +#define RT5670_GP10_PIN_SFT 0 +#define RT5670_GP10_PIN_GPIO9 (0x0) +#define RT5670_GP10_PIN_DMIC3_SDA (0x1) +#define RT5670_GP10_PIN_PDM_ADT2 (0x2) + +/* GPIO Control 2 (0xc1) */ +#define RT5670_GP4_PF_MASK (0x1 << 11) +#define RT5670_GP4_PF_SFT 11 +#define RT5670_GP4_PF_IN (0x0 << 11) +#define RT5670_GP4_PF_OUT (0x1 << 11) +#define RT5670_GP4_OUT_MASK (0x1 << 10) +#define RT5670_GP4_OUT_SFT 10 +#define RT5670_GP4_OUT_LO (0x0 << 10) +#define RT5670_GP4_OUT_HI (0x1 << 10) +#define RT5670_GP4_P_MASK (0x1 << 9) +#define RT5670_GP4_P_SFT 9 +#define RT5670_GP4_P_NOR (0x0 << 9) +#define RT5670_GP4_P_INV (0x1 << 9) +#define RT5670_GP3_PF_MASK (0x1 << 8) +#define RT5670_GP3_PF_SFT 8 +#define RT5670_GP3_PF_IN (0x0 << 8) +#define RT5670_GP3_PF_OUT (0x1 << 8) +#define RT5670_GP3_OUT_MASK (0x1 << 7) +#define RT5670_GP3_OUT_SFT 7 +#define RT5670_GP3_OUT_LO (0x0 << 7) +#define RT5670_GP3_OUT_HI (0x1 << 7) +#define RT5670_GP3_P_MASK (0x1 << 6) +#define RT5670_GP3_P_SFT 6 +#define RT5670_GP3_P_NOR (0x0 << 6) +#define RT5670_GP3_P_INV (0x1 << 6) +#define RT5670_GP2_PF_MASK (0x1 << 5) +#define RT5670_GP2_PF_SFT 5 +#define RT5670_GP2_PF_IN (0x0 << 5) +#define RT5670_GP2_PF_OUT (0x1 << 5) +#define RT5670_GP2_OUT_MASK (0x1 << 4) +#define RT5670_GP2_OUT_SFT 4 +#define RT5670_GP2_OUT_LO (0x0 << 4) +#define RT5670_GP2_OUT_HI (0x1 << 4) +#define RT5670_GP2_P_MASK (0x1 << 3) +#define RT5670_GP2_P_SFT 3 +#define RT5670_GP2_P_NOR (0x0 << 3) +#define RT5670_GP2_P_INV (0x1 << 3) +#define RT5670_GP1_PF_MASK (0x1 << 2) +#define RT5670_GP1_PF_SFT 2 +#define RT5670_GP1_PF_IN (0x0 << 2) +#define RT5670_GP1_PF_OUT (0x1 << 2) +#define RT5670_GP1_OUT_MASK (0x1 << 1) +#define RT5670_GP1_OUT_SFT 1 +#define RT5670_GP1_OUT_LO (0x0 << 1) +#define RT5670_GP1_OUT_HI (0x1 << 1) +#define RT5670_GP1_P_MASK (0x1) +#define RT5670_GP1_P_SFT 0 +#define RT5670_GP1_P_NOR (0x0) +#define RT5670_GP1_P_INV (0x1) + +/* Scramble Function (0xcd) */ +#define RT5670_SCB_KEY_MASK (0xff) +#define RT5670_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5670_SCB_SWAP_MASK (0x1 << 15) +#define RT5670_SCB_SWAP_SFT 15 +#define RT5670_SCB_SWAP_DIS (0x0 << 15) +#define RT5670_SCB_SWAP_EN (0x1 << 15) +#define RT5670_SCB_MASK (0x1 << 14) +#define RT5670_SCB_SFT 14 +#define RT5670_SCB_DIS (0x0 << 14) +#define RT5670_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5670_BB_MASK (0x1 << 15) +#define RT5670_BB_SFT 15 +#define RT5670_BB_DIS (0x0 << 15) +#define RT5670_BB_EN (0x1 << 15) +#define RT5670_BB_CT_MASK (0x7 << 12) +#define RT5670_BB_CT_SFT 12 +#define RT5670_BB_CT_A (0x0 << 12) +#define RT5670_BB_CT_B (0x1 << 12) +#define RT5670_BB_CT_C (0x2 << 12) +#define RT5670_BB_CT_D (0x3 << 12) +#define RT5670_M_BB_L_MASK (0x1 << 9) +#define RT5670_M_BB_L_SFT 9 +#define RT5670_M_BB_R_MASK (0x1 << 8) +#define RT5670_M_BB_R_SFT 8 +#define RT5670_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5670_M_BB_HPF_L_SFT 7 +#define RT5670_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5670_M_BB_HPF_R_SFT 6 +#define RT5670_G_BB_BST_MASK (0x3f) +#define RT5670_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5670_M_MP3_L_MASK (0x1 << 15) +#define RT5670_M_MP3_L_SFT 15 +#define RT5670_M_MP3_R_MASK (0x1 << 14) +#define RT5670_M_MP3_R_SFT 14 +#define RT5670_M_MP3_MASK (0x1 << 13) +#define RT5670_M_MP3_SFT 13 +#define RT5670_M_MP3_DIS (0x0 << 13) +#define RT5670_M_MP3_EN (0x1 << 13) +#define RT5670_EG_MP3_MASK (0x1f << 8) +#define RT5670_EG_MP3_SFT 8 +#define RT5670_MP3_HLP_MASK (0x1 << 7) +#define RT5670_MP3_HLP_SFT 7 +#define RT5670_MP3_HLP_DIS (0x0 << 7) +#define RT5670_MP3_HLP_EN (0x1 << 7) +#define RT5670_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5670_M_MP3_ORG_L_SFT 6 +#define RT5670_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5670_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5670_MP3_WT_MASK (0x1 << 13) +#define RT5670_MP3_WT_SFT 13 +#define RT5670_MP3_WT_1_4 (0x0 << 13) +#define RT5670_MP3_WT_1_2 (0x1 << 13) +#define RT5670_OG_MP3_MASK (0x1f << 8) +#define RT5670_OG_MP3_SFT 8 +#define RT5670_HG_MP3_MASK (0x3f) +#define RT5670_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5670_3D_CF_MASK (0x1 << 15) +#define RT5670_3D_CF_SFT 15 +#define RT5670_3D_CF_DIS (0x0 << 15) +#define RT5670_3D_CF_EN (0x1 << 15) +#define RT5670_3D_HP_MASK (0x1 << 14) +#define RT5670_3D_HP_SFT 14 +#define RT5670_3D_HP_DIS (0x0 << 14) +#define RT5670_3D_HP_EN (0x1 << 14) +#define RT5670_3D_BT_MASK (0x1 << 13) +#define RT5670_3D_BT_SFT 13 +#define RT5670_3D_BT_DIS (0x0 << 13) +#define RT5670_3D_BT_EN (0x1 << 13) +#define RT5670_3D_1F_MIX_MASK (0x3 << 11) +#define RT5670_3D_1F_MIX_SFT 11 +#define RT5670_3D_HP_M_MASK (0x1 << 10) +#define RT5670_3D_HP_M_SFT 10 +#define RT5670_3D_HP_M_SUR (0x0 << 10) +#define RT5670_3D_HP_M_FRO (0x1 << 10) +#define RT5670_M_3D_HRTF_MASK (0x1 << 9) +#define RT5670_M_3D_HRTF_SFT 9 +#define RT5670_M_3D_D2H_MASK (0x1 << 8) +#define RT5670_M_3D_D2H_SFT 8 +#define RT5670_M_3D_D2R_MASK (0x1 << 7) +#define RT5670_M_3D_D2R_SFT 7 +#define RT5670_M_3D_REVB_MASK (0x1 << 6) +#define RT5670_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5670_2ND_HPF_MASK (0x1 << 15) +#define RT5670_2ND_HPF_SFT 15 +#define RT5670_2ND_HPF_DIS (0x0 << 15) +#define RT5670_2ND_HPF_EN (0x1 << 15) +#define RT5670_HPF_CF_L_MASK (0x7 << 12) +#define RT5670_HPF_CF_L_SFT 12 +#define RT5670_1ST_HPF_MASK (0x1 << 11) +#define RT5670_1ST_HPF_SFT 11 +#define RT5670_1ST_HPF_DIS (0x0 << 11) +#define RT5670_1ST_HPF_EN (0x1 << 11) +#define RT5670_HPF_CF_R_MASK (0x7 << 8) +#define RT5670_HPF_CF_R_SFT 8 +#define RT5670_ZD_T_MASK (0x3 << 6) +#define RT5670_ZD_T_SFT 6 +#define RT5670_ZD_F_MASK (0x3 << 4) +#define RT5670_ZD_F_SFT 4 +#define RT5670_ZD_F_IM (0x0 << 4) +#define RT5670_ZD_F_ZC_IM (0x1 << 4) +#define RT5670_ZD_F_ZC_IOD (0x2 << 4) +#define RT5670_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5670_SI_DAC_MASK (0x1 << 11) +#define RT5670_SI_DAC_SFT 11 +#define RT5670_SI_DAC_AUTO (0x0 << 11) +#define RT5670_SI_DAC_TEST (0x1 << 11) +#define RT5670_DC_CAL_M_MASK (0x1 << 10) +#define RT5670_DC_CAL_M_SFT 10 +#define RT5670_DC_CAL_M_CAL (0x0 << 10) +#define RT5670_DC_CAL_M_NOR (0x1 << 10) +#define RT5670_DC_CAL_MASK (0x1 << 9) +#define RT5670_DC_CAL_SFT 9 +#define RT5670_DC_CAL_DIS (0x0 << 9) +#define RT5670_DC_CAL_EN (0x1 << 9) +#define RT5670_HPD_RCV_MASK (0x7 << 6) +#define RT5670_HPD_RCV_SFT 6 +#define RT5670_HPD_PS_MASK (0x1 << 5) +#define RT5670_HPD_PS_SFT 5 +#define RT5670_HPD_PS_DIS (0x0 << 5) +#define RT5670_HPD_PS_EN (0x1 << 5) +#define RT5670_CAL_M_MASK (0x1 << 4) +#define RT5670_CAL_M_SFT 4 +#define RT5670_CAL_M_DEP (0x0 << 4) +#define RT5670_CAL_M_CAL (0x1 << 4) +#define RT5670_CAL_MASK (0x1 << 3) +#define RT5670_CAL_SFT 3 +#define RT5670_CAL_DIS (0x0 << 3) +#define RT5670_CAL_EN (0x1 << 3) +#define RT5670_CAL_TEST_MASK (0x1 << 2) +#define RT5670_CAL_TEST_SFT 2 +#define RT5670_CAL_TEST_DIS (0x0 << 2) +#define RT5670_CAL_TEST_EN (0x1 << 2) +#define RT5670_CAL_P_MASK (0x3) +#define RT5670_CAL_P_SFT 0 +#define RT5670_CAL_P_NONE (0x0) +#define RT5670_CAL_P_CAL (0x1) +#define RT5670_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5670_SV_MASK (0x1 << 15) +#define RT5670_SV_SFT 15 +#define RT5670_SV_DIS (0x0 << 15) +#define RT5670_SV_EN (0x1 << 15) +#define RT5670_SPO_SV_MASK (0x1 << 14) +#define RT5670_SPO_SV_SFT 14 +#define RT5670_SPO_SV_DIS (0x0 << 14) +#define RT5670_SPO_SV_EN (0x1 << 14) +#define RT5670_OUT_SV_MASK (0x1 << 13) +#define RT5670_OUT_SV_SFT 13 +#define RT5670_OUT_SV_DIS (0x0 << 13) +#define RT5670_OUT_SV_EN (0x1 << 13) +#define RT5670_HP_SV_MASK (0x1 << 12) +#define RT5670_HP_SV_SFT 12 +#define RT5670_HP_SV_DIS (0x0 << 12) +#define RT5670_HP_SV_EN (0x1 << 12) +#define RT5670_ZCD_DIG_MASK (0x1 << 11) +#define RT5670_ZCD_DIG_SFT 11 +#define RT5670_ZCD_DIG_DIS (0x0 << 11) +#define RT5670_ZCD_DIG_EN (0x1 << 11) +#define RT5670_ZCD_MASK (0x1 << 10) +#define RT5670_ZCD_SFT 10 +#define RT5670_ZCD_PD (0x0 << 10) +#define RT5670_ZCD_PU (0x1 << 10) +#define RT5670_M_ZCD_MASK (0x3f << 4) +#define RT5670_M_ZCD_SFT 4 +#define RT5670_M_ZCD_RM_L (0x1 << 9) +#define RT5670_M_ZCD_RM_R (0x1 << 8) +#define RT5670_M_ZCD_SM_L (0x1 << 7) +#define RT5670_M_ZCD_SM_R (0x1 << 6) +#define RT5670_M_ZCD_OM_L (0x1 << 5) +#define RT5670_M_ZCD_OM_R (0x1 << 4) +#define RT5670_SV_DLY_MASK (0xf) +#define RT5670_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5670_ZCD_HP_MASK (0x1 << 15) +#define RT5670_ZCD_HP_SFT 15 +#define RT5670_ZCD_HP_DIS (0x0 << 15) +#define RT5670_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5670_3D_SPK_MASK (0x1 << 15) +#define RT5670_3D_SPK_SFT 15 +#define RT5670_3D_SPK_DIS (0x0 << 15) +#define RT5670_3D_SPK_EN (0x1 << 15) +#define RT5670_3D_SPK_M_MASK (0x3 << 13) +#define RT5670_3D_SPK_M_SFT 13 +#define RT5670_3D_SPK_CG_MASK (0x1f << 8) +#define RT5670_3D_SPK_CG_SFT 8 +#define RT5670_3D_SPK_SG_MASK (0x1f) +#define RT5670_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5670_WND_MASK (0x1 << 15) +#define RT5670_WND_SFT 15 +#define RT5670_WND_DIS (0x0 << 15) +#define RT5670_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5670_WND_FC_NW_MASK (0x3f << 10) +#define RT5670_WND_FC_NW_SFT 10 +#define RT5670_WND_FC_WK_MASK (0x3f << 4) +#define RT5670_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5670_HPF_FC_MASK (0x3f << 6) +#define RT5670_HPF_FC_SFT 6 +#define RT5670_WND_FC_ST_MASK (0x3f) +#define RT5670_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5670_WND_TH_LO_MASK (0x3ff) +#define RT5670_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5670_WND_TH_HI_MASK (0x3ff) +#define RT5670_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5670_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5670_WND_WIND_SFT 13 +#define RT5670_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5670_WND_STRONG_SFT 12 +enum { + RT5670_NO_WIND, + RT5670_BREEZE, + RT5670_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5670_DP_ATT_MASK (0x3 << 14) +#define RT5670_DP_ATT_SFT 14 +#define RT5670_DP_SPK_MASK (0x1 << 10) +#define RT5670_DP_SPK_SFT 10 +#define RT5670_DP_SPK_DIS (0x0 << 10) +#define RT5670_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5670_EQ_PRE_VOL_MASK (0xffff) +#define RT5670_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5670_EQ_PST_VOL_MASK (0xffff) +#define RT5670_EQ_PST_VOL_SFT 0 + +/* Jack Detect Control 3 (0xf8) */ +#define RT5670_CMP_MIC_IN_DET_MASK (0x7 << 12) +#define RT5670_JD_CBJ_EN (0x1 << 7) +#define RT5670_JD_CBJ_POL (0x1 << 6) +#define RT5670_JD_TRI_CBJ_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_CBJ_SEL_SFT (3) +#define RT5670_JD_CBJ_GPIO_JD1 (0x0 << 3) +#define RT5670_JD_CBJ_JD1_1 (0x1 << 3) +#define RT5670_JD_CBJ_JD1_2 (0x2 << 3) +#define RT5670_JD_CBJ_JD2 (0x3 << 3) +#define RT5670_JD_CBJ_JD3 (0x4 << 3) +#define RT5670_JD_CBJ_GPIO_JD2 (0x5 << 3) +#define RT5670_JD_CBJ_MX0B_12 (0x6 << 3) +#define RT5670_JD_TRI_HPO_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_HPO_SEL_SFT (0) +#define RT5670_JD_HPO_GPIO_JD1 (0x0) +#define RT5670_JD_HPO_JD1_1 (0x1) +#define RT5670_JD_HPO_JD1_2 (0x2) +#define RT5670_JD_HPO_JD2 (0x3) +#define RT5670_JD_HPO_JD3 (0x4) +#define RT5670_JD_HPO_GPIO_JD2 (0x5) +#define RT5670_JD_HPO_MX0B_12 (0x6) + +/* Digital Misc Control (0xfa) */ +#define RT5670_RST_DSP (0x1 << 13) +#define RT5670_IF1_ADC1_IN1_SEL (0x1 << 12) +#define RT5670_IF1_ADC1_IN1_SFT 12 +#define RT5670_IF1_ADC1_IN2_SEL (0x1 << 11) +#define RT5670_IF1_ADC1_IN2_SFT 11 +#define RT5670_IF1_ADC2_IN1_SEL (0x1 << 10) +#define RT5670_IF1_ADC2_IN1_SFT 10 + +/* General Control2 (0xfb) */ +#define RT5670_RXDC_SRC_MASK (0x1 << 7) +#define RT5670_RXDC_SRC_STO (0x0 << 7) +#define RT5670_RXDC_SRC_MONO (0x1 << 7) +#define RT5670_RXDC_SRC_SFT (7) +#define RT5670_RXDP2_SEL_MASK (0x1 << 3) +#define RT5670_RXDP2_SEL_IF2 (0x0 << 3) +#define RT5670_RXDP2_SEL_ADC (0x1 << 3) +#define RT5670_RXDP2_SEL_SFT (3) + +/* System Clock Source */ +enum { + RT5670_SCLK_S_MCLK, + RT5670_SCLK_S_PLL1, + RT5670_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5670_PLL1_S_MCLK, + RT5670_PLL1_S_BCLK1, + RT5670_PLL1_S_BCLK2, + RT5670_PLL1_S_BCLK3, + RT5670_PLL1_S_BCLK4, +}; + +enum { + RT5670_AIF1, + RT5670_AIF2, + RT5670_AIF3, + RT5670_AIF4, + RT5670_AIFS, +}; + +enum { + RT5670_DMIC_DATA_GPIO6, + RT5670_DMIC_DATA_IN2P, + RT5670_DMIC_DATA_GPIO7, +}; + +enum { + RT5670_DMIC_DATA_GPIO8, + RT5670_DMIC_DATA_IN3N, +}; + +enum { + RT5670_DMIC_DATA_GPIO9, + RT5670_DMIC_DATA_GPIO10, + RT5670_DMIC_DATA_GPIO5, +}; + +struct rt5670_priv { + struct snd_soc_codec *codec; + struct rt5670_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5670_AIFS]; + int bclk[RT5670_AIFS]; + int master[RT5670_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int dsp_sw; /* expected parameter setting */ + int dsp_rate; + int jack_type; +}; + +#endif /* __RT5670_H__ */ -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] ALC286 recording delay
by Kailang 30 Jun '14

30 Jun '14

Hi Takashi, It will be recording voice delay for resume back recording for Headset Mic. This patch doesn't disable pin for suspend state. The issue was fixed. This is follow ALC286 programing guide. Thanks. BR, Kailang

1 0

[alsa-devel] Why SIGABORT
by Joël Krähemann 29 Jun '14

29 Jun '14

http://ags.sf.net 7ffe9f5d8000-7ffe9fdd8000 rw-p 00000000 00:00 0 7ffe9fdd8000-7ffe9fdd9000 ---p 00000000 00:00 0 7ffe9fdd9000-7ffea05d9000 rw-p 00000000 00:00 0 7ffea05d9000-7ffea05da000 ---p 00000000 00:00 0 7ffea05da000-7ffea0dda000 rw-p 00000000 00:00 0 7ffea0dda000-7ffea0ddb000 ---p 00000000 00:00 0 7ffea0ddb000-7ffea15db000 rw-p 00000000 00:00 0 7ffea15db000-7ffea15dc000 ---p 00000000 00:00 0 7ffea15dc000-7ffea1ddc000 rw-p 00000000 00:00 0 7ffea1ddc000-7ffea1ddd000 ---p 00000000 00:00 0 7ffea1ddd000-7ffea25dd000 rw-p 00000000 00:00 0 7ffea25dd000-7ffea25de000 ---p 00000000 00:00 0 7ffea25de000-7ffea2dde000 rw-p 00000000 00:00 0 7ffea2dde000-7ffea2ddf000 ---p 00000000 00:00 0 7ffea2ddf000-7ffea35df000 rw-p 00000000 00:00 0 7ffea35df000-7ffea35e0000 ---p 00000000 00:00 0 7ffea35e0000-7ffea3de0000 rw-p 00000000 00:00 0 7ffea3de0000-7ffea3de1000 ---p 00000000 00:00 0 7ffea3de1000-7ffea45e1000 rw-p 00000000 00:00 0 7ffea45e1000-7ffea45e2000 ---p 00000000 00:00 0 7ffea45e2000-7ffea4de2000 rw-p 00000000 00:00 0 7ffea4de2000-7ffea4de3000 ---p 00000000 00:00 0 7ffea4de3000-7ffea55e3000 rw-p 00000000 00:00 0 7ffea55e3000-7ffea55e4000 ---p 00000000 00:00 0 7ffea55e4000-7ffea5de4000 rw-p 00000000 00:00 0 7ffea5de4000-7ffea5de5000 ---p 00000000 00:00 0 7ffea5de5000-7ffea65e5000 rw-p 00000000 00:00 0 7ffea65e5000-7ffea65e6000 ---p 00000000 00:00 0 7ffea65e6000-7ffea6de6000 rw-p 00000000 00:00 0 7ffea6de6000-7ffea6de7000 ---p 00000000 00:00 0 7ffea6de7000-7ffea75e7000 rw-p 00000000 00:00 0 7ffea75e7000-7ffea75e8000 ---p 00000000 00:00 0 7ffea75e8000-7ffea7de8000 rw-p 00000000 00:00 0 7ffea7de8000-7ffea7de9000 ---p 00000000 00:00 0 7ffea7de9000-7ffea85e9000 rw-p 00000000 00:00 0 7ffea85e9000-7ffea85ea000 ---p 00000000 00:00 0 7ffea85ea000-7ffea8dea000 rw-p 00000000 00:00 0 7ffea8dea000-7ffea8deb000 ---p 00000000 00:00 0 7ffea8deb000-7ffea95eb000 rw-p 00000000 00:00 0 7ffea95eb000-7ffea95ec000 ---p 00000000 00:00 0 7ffea95ec000-7ffea9dec000 rw-p 00000000 00:00 0 7ffea9dec000-7ffea9ded000 ---p 00000000 00:00 0 7ffea9ded000-7ffeaa5ed000 rw-p 00000000 00:00 0 7ffeaa5ed000-7ffeaa5ee000 ---p 00000000 00:00 0 7ffeaa5ee000-7ffeaadee000 rw-p 00000000 00:00 0 7ffeaadee000-7ffeaadef000 ---p 00000000 00:00 0 7ffeaadef000-7ffeab5ef000 rw-p 00000000 00:00 0 [stack:13073] 7ffeab5ef000-7ffeab5f0000 ---p 00000000 00:00 0 7ffeab5f0000-7ffeabdf0000 rw-p 00000000 00:00 0 7ffeabdf0000-7ffeabdf1000 ---p 00000000 00:00 0 7ffeabdf1000-7ffeac5f1000 rw-p 00000000 00:00 0 7ffeac5f1000-7ffeac5f2000 ---p 00000000 00:00 0 7ffeac5f2000-7ffeacdf2000 rw-p 00000000 00:00 0 7ffeacdf2000-7ffeacdf3000 ---p 00000000 00:00 0 7ffeacdf3000-7ffead5f3000 rw-p 00000000 00:00 0 7ffead5f3000-7ffead5f4000 ---p 00000000 00:00 0 7ffead5f4000-7ffeaddf4000 rw-p 00000000 00:00 0 7ffeaddf4000-7ffeaddf5000 ---p 00000000 00:00 0 7ffeaddf5000-7ffeae5f5000 rw-p 00000000 00:00 0 7ffeae5f5000-7ffeae5f6000 ---p 00000000 00:00 0 7ffeae5f6000-7ffeaedf6000 rw-p 00000000 00:00 0 7ffeaedf6000-7ffeaedf7000 ---p 00000000 00:00 0 7ffeaedf7000-7ffeaf5f7000 rw-p 00000000 00:00 0 7ffeaf5f7000-7ffeaf5f8000 ---p 00000000 00:00 0 7ffeaf5f8000-7ffeafdf8000 rw-p 00000000 00:00 0 7ffeafdf8000-7ffeafdf9000 ---p 00000000 00:00 0 7ffeafdf9000-7ffeb05f9000 rw-p 00000000 00:00 0 7ffeb05f9000-7ffeb05fa000 ---p 00000000 00:00 0 7ffeb05fa000-7ffeb0dfa000 rw-p 00000000 00:00 0 7ffeb0dfa000-7ffeb0dfb000 ---p 00000000 00:00 0 7ffeb0dfb000-7ffeb15fb000 rw-p 00000000 00:00 0 7ffeb15fb000-7ffeb15fc000 ---p 00000000 00:00 0 7ffeb15fc000-7ffeb1dfc000 rw-p 00000000 00:00 0 7ffeb1dfc000-7ffeb1dfd000 ---p 00000000 00:00 0 7ffeb1dfd000-7ffeb25fd000 rw-p 00000000 00:00 0 7ffeb25fd000-7ffeb25fe000 ---p 00000000 00:00 0 7ffeb25fe000-7ffeb2dfe000 rw-p 00000000 00:00 0 7ffeb2dfe000-7ffeb2dff000 ---p 00000000 00:00 0 7ffeb2dff000-7ffeb35ff000 rw-p 00000000 00:00 0 7ffeb35ff000-7ffeb3600000 ---p 00000000 00:00 0 7ffeb3600000-7ffeb3e00000 rw-p 00000000 00:00 0 7ffeb3e00000-7ffeb3e01000 ---p 00000000 00:00 0 7ffeb3e01000-7ffeb4601000 rw-p 00000000 00:00 0 7ffeb4601000-7ffeb4602000 ---p 00000000 00:00 0 7ffeb4602000-7ffeb4e02000 rw-p 00000000 00:00 0 7ffeb4e02000-7ffeb4e03000 ---p 00000000 00:00 0 7ffeb4e03000-7ffeb5603000 rw-p 00000000 00:00 0 7ffeb5603000-7ffeb5604000 ---p 00000000 00:00 0 7ffeb5604000-7ffeb5e04000 rw-p 00000000 00:00 0 7ffeb5e04000-7ffeb5e05000 ---p 00000000 00:00 0 7ffeb5e05000-7ffeb6605000 rw-p 00000000 00:00 0 7ffeb6605000-7ffeb6606000 ---p 00000000 00:00 0 7ffeb6606000-7ffeb6e06000 rw-p 00000000 00:00 0 7ffeb6e06000-7ffeb6e07000 ---p 00000000 00:00 0 7ffeb6e07000-7ffeb7607000 rw-p 00000000 00:00 0 7ffeb7607000-7ffeb7608000 ---p 00000000 00:00 0 7ffeb7608000-7ffeb7e08000 rw-p 00000000 00:00 0 7ffeb7e08000-7ffeb7e09000 ---p 00000000 00:00 0 7ffeb7e09000-7ffeb8609000 rw-p 00000000 00:00 0 7ffeb8609000-7ffeb860a000 ---p 00000000 00:00 0 7ffeb860a000-7ffeb8e0a000 rw-p 00000000 00:00 0 7ffeb8e0a000-7ffeb8e0b000 ---p 00000000 00:00 0 7ffeb8e0b000-7ffeb960b000 rw-p 00000000 00:00 0 7ffeb960b000-7ffeb960c000 ---p 00000000 00:00 0 7ffeb960c000-7ffeb9e0c000 rw-p 00000000 00:00 0 7ffeb9e0c000-7ffeb9e0d000 ---p 00000000 00:00 0 7ffeb9e0d000-7ffeba60d000 rw-p 00000000 00:00 0 7ffeba60d000-7ffeba60e000 ---p 00000000 00:00 0 7ffeba60e000-7ffebae0e000 rw-p 00000000 00:00 0 7ffebae0e000-7ffebae0f000 ---p 00000000 00:00 0 7ffebae0f000-7ffebb60f000 rw-p 00000000 00:00 0 7ffebb60f000-7ffebb610000 ---p 00000000 00:00 0 7ffebb610000-7ffebbe10000 rw-p 00000000 00:00 0 7ffebbe10000-7ffebbe11000 ---p 00000000 00:00 0 7ffebbe11000-7ffebc611000 rw-p 00000000 00:00 0 7ffebc611000-7ffebc612000 ---p 00000000 00:00 0 7ffebc612000-7ffebce12000 rw-p 00000000 00:00 0 7ffebce12000-7ffebce13000 ---p 00000000 00:00 0 7ffebce13000-7ffebd613000 rw-p 00000000 00:00 0 7ffebd613000-7ffebd614000 ---p 00000000 00:00 0 7ffebd614000-7ffebde14000 rw-p 00000000 00:00 0 7ffebde14000-7ffebde15000 ---p 00000000 00:00 0 7ffebde15000-7ffebe615000 rw-p 00000000 00:00 0 7ffebe615000-7ffebe616000 ---p 00000000 00:00 0 7ffebe616000-7ffebee16000 rw-p 00000000 00:00 0 7ffebee16000-7ffebee17000 ---p 00000000 00:00 0 7ffebee17000-7ffebf617000 rw-p 00000000 00:00 0 7ffebf617000-7ffebf618000 ---p 00000000 00:00 0 7ffebf618000-7ffebfe18000 rw-p 00000000 00:00 0 7ffebfe18000-7ffebfe19000 ---p 00000000 00:00 0 7ffebfe19000-7ffec0619000 rw-p 00000000 00:00 0 7ffec0619000-7ffec061a000 ---p 00000000 00:00 0 7ffec061a000-7ffec0e1a000 rw-p 00000000 00:00 0 7ffec0e1a000-7ffec0e1b000 ---p 00000000 00:00 0 7ffec0e1b000-7ffec161b000 rw-p 00000000 00:00 0 7ffec161b000-7ffec161c000 ---p 00000000 00:00 0 7ffec161c000-7ffec1e1c000 rw-p 00000000 00:00 0 7ffec1e1c000-7ffec1e1d000 ---p 00000000 00:00 0 7ffec1e1d000-7ffec261d000 rw-p 00000000 00:00 0 7ffec261d000-7ffec261e000 ---p 00000000 00:00 0 7ffec261e000-7ffec2e1e000 rw-p 00000000 00:00 0 7ffec2e1e000-7ffec2e1f000 ---p 00000000 00:00 0 7ffec2e1f000-7ffec361f000 rw-p 00000000 00:00 0 7ffec361f000-7ffec3620000 ---p 00000000 00:00 0 7ffec3620000-7ffec3e20000 rw-p 00000000 00:00 0 7ffec3e20000-7ffec3e21000 ---p 00000000 00:00 0 7ffec3e21000-7ffec4621000 rw-p 00000000 00:00 0 7ffec4621000-7ffec4622000 ---p 00000000 00:00 0 7ffec4622000-7ffec4e22000 rw-p 00000000 00:00 0 7ffec4e22000-7ffec4e23000 ---p 00000000 00:00 0 7ffec4e23000-7ffec5623000 rw-p 00000000 00:00 0 7ffec5623000-7ffec5624000 ---p 00000000 00:00 0 7ffec5624000-7ffec5e24000 rw-p 00000000 00:00 0 7ffec5e24000-7ffec5e25000 ---p 00000000 00:00 0 7ffec5e25000-7ffec6625000 rw-p 00000000 00:00 0 7ffec6625000-7ffec6626000 ---p 00000000 00:00 0 7ffec6626000-7ffec6e26000 rw-p 00000000 00:00 0 7ffec6e26000-7ffec6e27000 ---p 00000000 00:00 0 7ffec6e27000-7ffec7627000 rw-p 00000000 00:00 0 7ffec7627000-7ffec7628000 ---p 00000000 00:00 0 7ffec7628000-7ffec7e28000 rw-p 00000000 00:00 0 7ffec7e28000-7ffec7e29000 ---p 00000000 00:00 0 7ffec7e29000-7ffec8629000 rw-p 00000000 00:00 0 7ffec8629000-7ffec862a000 ---p 00000000 00:00 0 7ffec862a000-7ffec8e2a000 rw-p 00000000 00:00 0 7ffec8e2a000-7ffec8e2b000 ---p 00000000 00:00 0 7ffec8e2b000-7ffec962b000 rw-p 00000000 00:00 0 7ffec962b000-7ffec962c000 ---p 00000000 00:00 0 7ffec962c000-7ffec9e2c000 rw-p 00000000 00:00 0 7ffec9e2c000-7ffec9e2d000 ---p 00000000 00:00 0 7ffec9e2d000-7ffeca62d000 rw-p 00000000 00:00 0 7ffeca62d000-7ffeca62e000 ---p 00000000 00:00 0 7ffeca62e000-7ffecae2e000 rw-p 00000000 00:00 0 7ffecae2e000-7ffecae2f000 ---p 00000000 00:00 0 7ffecae2f000-7ffecb62f000 rw-p 00000000 00:00 0 7ffecb62f000-7ffecb630000 ---p 00000000 00:00 0 7ffecb630000-7ffecbe30000 rw-p 00000000 00:00 0 7ffecbe30000-7ffecbe31000 ---p 00000000 00:00 0 7ffecbe31000-7ffecc631000 rw-p 00000000 00:00 0 7ffecc631000-7ffecc632000 ---p 00000000 00:00 0 7ffecc632000-7ffecce32000 rw-p 00000000 00:00 0 7ffecce32000-7ffecce33000 ---p 00000000 00:00 0 7ffecce33000-7ffecd633000 rw-p 00000000 00:00 0 7ffecd633000-7ffecd634000 ---p 00000000 00:00 0 7ffecd634000-7ffecde34000 rw-p 00000000 00:00 0 7ffecde34000-7ffecde35000 ---p 00000000 00:00 0 7ffecde35000-7ffece635000 rw-p 00000000 00:00 0 7ffece635000-7ffece636000 ---p 00000000 00:00 0 7ffece636000-7ffecee36000 rw-p 00000000 00:00 0 7ffecee36000-7ffecee37000 ---p 00000000 00:00 0 7ffecee37000-7ffecf637000 rw-p 00000000 00:00 0 7ffecf637000-7ffecf638000 ---p 00000000 00:00 0 7ffecf638000-7ffecfe38000 rw-p 00000000 00:00 0 7ffecfe38000-7ffecfe39000 ---p 00000000 00:00 0 7ffecfe39000-7ffed0639000 rw-p 00000000 00:00 0 7ffed0639000-7ffed063a000 ---p 00000000 00:00 0 7ffed063a000-7ffed0e3a000 rw-p 00000000 00:00 0 7ffed0e3a000-7ffed0e3b000 ---p 00000000 00:00 0 7ffed0e3b000-7ffed163b000 rw-p 00000000 00:00 0 7ffed163b000-7ffed163c000 ---p 00000000 00:00 0 7ffed163c000-7ffed1e3c000 rw-p 00000000 00:00 0 7ffed1e3c000-7ffed1e3d000 ---p 00000000 00:00 0 7ffed1e3d000-7ffed263d000 rw-p 00000000 00:00 0 [stack:12994] 7ffed263d000-7ffed263e000 ---p 00000000 00:00 0 7ffed263e000-7ffed2e3e000 rw-p 00000000 00:00 0 7ffed2e3e000-7ffed2e3f000 ---p 00000000 00:00 0 7ffed2e3f000-7ffed363f000 rw-p 00000000 00:00 0 7ffed363f000-7ffed3640000 ---p 00000000 00:00 0 7ffed3640000-7ffed3e40000 rw-p 00000000 00:00 0 7ffed3e40000-7ffed3e41000 ---p 00000000 00:00 0 7ffed3e41000-7ffed4641000 rw-p 00000000 00:00 0 7ffed4641000-7ffed4642000 ---p 00000000 00:00 0 7ffed4642000-7ffed4e42000 rw-p 00000000 00:00 0 7ffed4e42000-7ffed4e43000 ---p 00000000 00:00 0 7ffed4e43000-7ffed5643000 rw-p 00000000 00:00 0 7ffed5643000-7ffed5644000 ---p 00000000 00:00 0 7ffed5644000-7ffed5e44000 rw-p 00000000 00:00 0 [stack:12987] 7ffed5e44000-7ffed5e45000 ---p 00000000 00:00 0 7ffed5e45000-7ffed6645000 rw-p 00000000 00:00 0 7ffed6645000-7ffed6646000 ---p 00000000 00:00 0 7ffed6646000-7ffed6e46000 rw-p 00000000 00:00 0 7ffed6e46000-7ffed6e47000 ---p 00000000 00:00 0 7ffed6e47000-7ffed7647000 rw-p 00000000 00:00 0 7ffed7647000-7ffed7648000 ---p 00000000 00:00 0 7ffed7648000-7ffed7e48000 rw-p 00000000 00:00 0 7ffed7e48000-7ffed7e49000 ---p 00000000 00:00 0 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00:00 0 7ffedd653000-7ffedd654000 ---p 00000000 00:00 0 7ffedd654000-7ffedde54000 rw-p 00000000 00:00 0 7ffedde54000-7ffedde55000 ---p 00000000 00:00 0 7ffedde55000-7ffede655000 rw-p 00000000 00:00 0 7ffede655000-7ffede656000 ---p 00000000 00:00 0 7ffede656000-7ffedee56000 rw-p 00000000 00:00 0 7ffedee56000-7ffedee57000 ---p 00000000 00:00 0 7ffedee57000-7ffedf657000 rw-p 00000000 00:00 0 7ffedf657000-7ffedf658000 ---p 00000000 00:00 0 7ffedf658000-7ffedfe58000 rw-p 00000000 00:00 0 7ffedfe58000-7ffedfe59000 ---p 00000000 00:00 0 7ffedfe59000-7ffee0659000 rw-p 00000000 00:00 0 7ffee0659000-7ffee065a000 ---p 00000000 00:00 0 7ffee065a000-7ffee0e5a000 rw-p 00000000 00:00 0 7ffee0e5a000-7ffee0e5b000 ---p 00000000 00:00 0 7ffee0e5b000-7ffee165b000 rw-p 00000000 00:00 0 7ffee165b000-7ffee165c000 ---p 00000000 00:00 0 7ffee165c000-7ffee1e5c000 rw-p 00000000 00:00 0 7ffee1e5c000-7ffee1e5d000 ---p 00000000 00:00 0 7ffee1e5d000-7ffee265d000 rw-p 00000000 00:00 0 7ffee265d000-7ffee265e000 ---p 00000000 00:00 0 7ffee265e000-7ffee2e5e000 rw-p 00000000 00:00 0 7ffee2e5e000-7ffee2e5f000 ---p 00000000 00:00 0 7ffee2e5f000-7ffee365f000 rw-p 00000000 00:00 0 7ffee365f000-7ffee3660000 ---p 00000000 00:00 0 7ffee3660000-7ffee3e60000 rw-p 00000000 00:00 0 7ffee3e60000-7ffee3e61000 ---p 00000000 00:00 0 7ffee3e61000-7ffee4661000 rw-p 00000000 00:00 0 7ffee4661000-7ffee4662000 ---p 00000000 00:00 0 7ffee4662000-7ffee4e62000 rw-p 00000000 00:00 0 7ffee4e62000-7ffee4e63000 ---p 00000000 00:00 0 7ffee4e63000-7ffee5663000 rw-p 00000000 00:00 0 7ffee5663000-7ffee5664000 ---p 00000000 00:00 0 7ffee5664000-7ffee5e64000 rw-p 00000000 00:00 0 7ffee5e64000-7ffee5e65000 ---p 00000000 00:00 0 7ffee5e65000-7ffee6665000 rw-p 00000000 00:00 0 7ffee6665000-7ffee6666000 ---p 00000000 00:00 0 7ffee6666000-7ffee6e66000 rw-p 00000000 00:00 0 7ffee6e66000-7ffee6e67000 ---p 00000000 00:00 0 7ffee6e67000-7ffee7667000 rw-p 00000000 00:00 0 7ffee7667000-7ffee7668000 ---p 00000000 00:00 0 7ffee7668000-7ffee7e68000 rw-p 00000000 00:00 0 7ffee7e68000-7ffee7e69000 ---p 00000000 00:00 0 7ffee7e69000-7ffee8669000 rw-p 00000000 00:00 0 7ffee8669000-7ffee866a000 ---p 00000000 00:00 0 7ffee866a000-7ffee8e6a000 rw-p 00000000 00:00 0 7ffee8e6a000-7ffee8e6b000 ---p 00000000 00:00 0 7ffee8e6b000-7ffee966b000 rw-p 00000000 00:00 0 7ffee966b000-7ffee966c000 ---p 00000000 00:00 0 7ffee966c000-7ffee9e6c000 rw-p 00000000 00:00 0 7ffee9e6c000-7ffee9e6d000 ---p 00000000 00:00 0 7ffee9e6d000-7ffeea66d000 rw-p 00000000 00:00 0 7ffeea66d000-7ffeea66e000 ---p 00000000 00:00 0 7ffeea66e000-7ffeeae6e000 rw-p 00000000 00:00 0 7ffeeae6e000-7ffeeae6f000 ---p 00000000 00:00 0 7ffeeae6f000-7ffeeb66f000 rw-p 00000000 00:00 0 7ffeeb66f000-7ffeeb670000 ---p 00000000 00:00 0 7ffeeb670000-7ffeebe70000 rw-p 00000000 00:00 0 7ffeebe70000-7ffeebe71000 ---p 00000000 00:00 0 7ffeebe71000-7ffeec671000 rw-p 00000000 00:00 0 7ffeec671000-7ffeec672000 ---p 00000000 00:00 0 7ffeec672000-7ffeece72000 rw-p 00000000 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00000000 00:00 0 7ffef1e7d000-7ffef267d000 rw-p 00000000 00:00 0 7ffef267d000-7ffef267e000 ---p 00000000 00:00 0 7ffef267e000-7ffef2e7e000 rw-p 00000000 00:00 0 7ffef2e7e000-7ffef2e7f000 ---p 00000000 00:00 0 7ffef2e7f000-7ffef367f000 rw-p 00000000 00:00 0 7ffef367f000-7ffef3680000 ---p 00000000 00:00 0 7ffef3680000-7ffef3e80000 rw-p 00000000 00:00 0 7ffef3e80000-7ffef3e81000 ---p 00000000 00:00 0 7ffef3e81000-7ffef4681000 rw-p 00000000 00:00 0 7ffef4681000-7ffef4682000 ---p 00000000 00:00 0 7ffef4682000-7ffef4e82000 rw-p 00000000 00:00 0 7ffef4e82000-7ffef4e83000 ---p 00000000 00:00 0 7ffef4e83000-7ffef5683000 rw-p 00000000 00:00 0 7ffef5683000-7ffef5684000 ---p 00000000 00:00 0 7ffef5684000-7ffef5e84000 rw-p 00000000 00:00 0 7ffef5e84000-7ffef5e85000 ---p 00000000 00:00 0 7ffef5e85000-7ffef6685000 rw-p 00000000 00:00 0 7ffef6685000-7ffef6686000 ---p 00000000 00:00 0 7ffef6686000-7ffef6e86000 rw-p 00000000 00:00 0 7ffef6e86000-7ffef6e87000 ---p 00000000 00:00 0 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00000000 00:00 0 7fff0169c000-7fff01e9c000 rw-p 00000000 00:00 0 7fff01e9c000-7fff01e9d000 ---p 00000000 00:00 0 7fff01e9d000-7fff0269d000 rw-p 00000000 00:00 0 7fff0269d000-7fff0269e000 ---p 00000000 00:00 0 7fff0269e000-7fff02e9e000 rw-p 00000000 00:00 0 7fff02e9e000-7fff02e9f000 ---p 00000000 00:00 0 7fff02e9f000-7fff0369f000 rw-p 00000000 00:00 0 7fff0369f000-7fff036a0000 ---p 00000000 00:00 0 7fff036a0000-7fff03ea0000 rw-p 00000000 00:00 0 7fff03ea0000-7fff03ea1000 ---p 00000000 00:00 0 7fff03ea1000-7fff046a1000 rw-p 00000000 00:00 0 7fff046a1000-7fff046a2000 ---p 00000000 00:00 0 7fff046a2000-7fff04ea2000 rw-p 00000000 00:00 0 7fff04ea2000-7fff04ea3000 ---p 00000000 00:00 0 7fff04ea3000-7fff056a3000 rw-p 00000000 00:00 0 7fff056a3000-7fff056a4000 ---p 00000000 00:00 0 7fff056a4000-7fff05ea4000 rw-p 00000000 00:00 0 7fff05ea4000-7fff05ea5000 ---p 00000000 00:00 0 7fff05ea5000-7fff066a5000 rw-p 00000000 00:00 0 7fff066a5000-7fff066a6000 ---p 00000000 00:00 0 7fff066a6000-7fff06ea6000 rw-p 00000000 00:00 0 7fff06ea6000-7fff06ea7000 ---p 00000000 00:00 0 7fff06ea7000-7fff076a7000 rw-p 00000000 00:00 0 7fff076a7000-7fff076a8000 ---p 00000000 00:00 0 7fff076a8000-7fff07ea8000 rw-p 00000000 00:00 0 7fff07ea8000-7fff07ea9000 ---p 00000000 00:00 0 7fff07ea9000-7fff086a9000 rw-p 00000000 00:00 0 7fff086a9000-7fff086aa000 ---p 00000000 00:00 0 7fff086aa000-7fff08eaa000 rw-p 00000000 00:00 0 7fff08eaa000-7fff08eab000 ---p 00000000 00:00 0 7fff08eab000-7fff096ab000 rw-p 00000000 00:00 0 7fff096ab000-7fff096ac000 ---p 00000000 00:00 0 7fff096ac000-7fff09eac000 rw-p 00000000 00:00 0 7fff09eac000-7fff09ead000 ---p 00000000 00:00 0 7fff09ead000-7fff0a6ad000 rw-p 00000000 00:00 0 7fff0a6ad000-7fff0a6ae000 ---p 00000000 00:00 0 7fff0a6ae000-7fff0aeae000 rw-p 00000000 00:00 0 7fff0aeae000-7fff0aeaf000 ---p 00000000 00:00 0 7fff0aeaf000-7fff0b6af000 rw-p 00000000 00:00 0 7fff0b6af000-7fff0b6b0000 ---p 00000000 00:00 0 7fff0b6b0000-7fff0beb0000 rw-p 00000000 00:00 0 7fff0beb0000-7fff0beb1000 ---p 00000000 00:00 0 7fff0beb1000-7fff0c6b1000 rw-p 00000000 00:00 0 7fff0c6b1000-7fff0c6b2000 ---p 00000000 00:00 0 7fff0c6b2000-7fff0ceb2000 rw-p 00000000 00:00 0 7fff0ceb2000-7fff0ceb3000 ---p 00000000 00:00 0 7fff0ceb3000-7fff0d6b3000 rw-p 00000000 00:00 0 7fff0d6b3000-7fff0d6b4000 ---p 00000000 00:00 0 7fff0d6b4000-7fff0deb4000 rw-p 00000000 00:00 0 7fff0deb4000-7fff0deb5000 ---p 00000000 00:00 0 7fff0deb5000-7fff0e6b5000 rw-p 00000000 00:00 0 7fff0e6b5000-7fff0e6b6000 ---p 00000000 00:00 0 7fff0e6b6000-7fff0eeb6000 rw-p 00000000 00:00 0 7fff0eeb6000-7fff0eeb7000 ---p 00000000 00:00 0 7fff0eeb7000-7fff0f6b7000 rw-p 00000000 00:00 0 7fff0f6b7000-7fff0f6b8000 ---p 00000000 00:00 0 7fff0f6b8000-7fff0feb8000 rw-p 00000000 00:00 0 7fff0feb8000-7fff0feb9000 ---p 00000000 00:00 0 7fff0feb9000-7fff106b9000 rw-p 00000000 00:00 0 7fff106b9000-7fff106ba000 ---p 00000000 00:00 0 7fff106ba000-7fff10eba000 rw-p 00000000 00:00 0 7fff10eba000-7fff10ebb000 ---p 00000000 00:00 0 7fff10ebb000-7fff116bb000 rw-p 00000000 00:00 0 7fff116bb000-7fff116bc000 ---p 00000000 00:00 0 7fff116bc000-7fff11ebc000 rw-p 00000000 00:00 0 7fff11ebc000-7fff11ebd000 ---p 00000000 00:00 0 7fff11ebd000-7fff126bd000 rw-p 00000000 00:00 0 7fff126bd000-7fff126be000 ---p 00000000 00:00 0 7fff126be000-7fff12ebe000 rw-p 00000000 00:00 0 7fff12ebe000-7fff12ebf000 ---p 00000000 00:00 0 7fff12ebf000-7fff136bf000 rw-p 00000000 00:00 0 7fff136bf000-7fff136c0000 ---p 00000000 00:00 0 7fff136c0000-7fff13ec0000 rw-p 00000000 00:00 0 7fff13ec0000-7fff13ec1000 ---p 00000000 00:00 0 7fff13ec1000-7fff146c1000 rw-p 00000000 00:00 0 7fff146c1000-7fff146c2000 ---p 00000000 00:00 0 7fff146c2000-7fff14ec2000 rw-p 00000000 00:00 0 7fff14ec2000-7fff14ec3000 ---p 00000000 00:00 0 7fff14ec3000-7fff156c3000 rw-p 00000000 00:00 0 7fff156c3000-7fff156c4000 ---p 00000000 00:00 0 7fff156c4000-7fff15ec4000 rw-p 00000000 00:00 0 7fff15ec4000-7fff15ec5000 ---p 00000000 00:00 0 7fff15ec5000-7fff166c5000 rw-p 00000000 00:00 0 7fff166c5000-7fff166c6000 ---p 00000000 00:00 0 7fff166c6000-7fff16ec6000 rw-p 00000000 00:00 0 7fff16ec6000-7fff16ec7000 ---p 00000000 00:00 0 7fff16ec7000-7fff176c7000 rw-p 00000000 00:00 0 7fff176c7000-7fff176c8000 ---p 00000000 00:00 0 7fff176c8000-7fff17ec8000 rw-p 00000000 00:00 0 7fff17ec8000-7fff17ec9000 ---p 00000000 00:00 0 7fff17ec9000-7fff186c9000 rw-p 00000000 00:00 0 7fff186c9000-7fff186ca000 ---p 00000000 00:00 0 7fff186ca000-7fff18eca000 rw-p 00000000 00:00 0 7fff18eca000-7fff18ecb000 ---p 00000000 00:00 0 7fff18ecb000-7fff196cb000 rw-p 00000000 00:00 0 7fff196cb000-7fff196cc000 ---p 00000000 00:00 0 7fff196cc000-7fff19ecc000 rw-p 00000000 00:00 0 7fff19ecc000-7fff19ecd000 ---p 00000000 00:00 0 7fff19ecd000-7fff1a6cd000 rw-p 00000000 00:00 0 7fff1a6cd000-7fff1a6ce000 ---p 00000000 00:00 0 7fff1a6ce000-7fff1aece000 rw-p 00000000 00:00 0 7fff1aece000-7fff1aecf000 ---p 00000000 00:00 0 7fff1aecf000-7fff1b6cf000 rw-p 00000000 00:00 0 7fff1b6cf000-7fff1b6d0000 ---p 00000000 00:00 0 7fff1b6d0000-7fff1bed0000 rw-p 00000000 00:00 0 7fff1bed0000-7fff1bed1000 ---p 00000000 00:00 0 7fff1bed1000-7fff1c6d1000 rw-p 00000000 00:00 0 7fff1c6d1000-7fff1c6d2000 ---p 00000000 00:00 0 7fff1c6d2000-7fff1ced2000 rw-p 00000000 00:00 0 7fff1ced2000-7fff1ced3000 ---p 00000000 00:00 0 7fff1ced3000-7fff1d6d3000 rw-p 00000000 00:00 0 7fff1d6d3000-7fff1d6d4000 ---p 00000000 00:00 0 7fff1d6d4000-7fff1ded4000 rw-p 00000000 00:00 0 7fff1ded4000-7fff1ded5000 ---p 00000000 00:00 0 7fff1ded5000-7fff1e6d5000 rw-p 00000000 00:00 0 7fff1e6d5000-7fff1e6d6000 ---p 00000000 00:00 0 7fff1e6d6000-7fff1eed6000 rw-p 00000000 00:00 0 7fff1eed6000-7fff1eed7000 ---p 00000000 00:00 0 7fff1eed7000-7fff1f6d7000 rw-p 00000000 00:00 0 7fff1f6d7000-7fff1f6d8000 ---p 00000000 00:00 0 7fff1f6d8000-7fff1fed8000 rw-p 00000000 00:00 0 7fff1fed8000-7fff1fed9000 ---p 00000000 00:00 0 7fff1fed9000-7fff206d9000 rw-p 00000000 00:00 0 7fff206d9000-7fff206da000 ---p 00000000 00:00 0 7fff206da000-7fff20eda000 rw-p 00000000 00:00 0 7fff20eda000-7fff20edb000 ---p 00000000 00:00 0 7fff20edb000-7fff216db000 rw-p 00000000 00:00 0 7fff216db000-7fff216dc000 ---p 00000000 00:00 0 7fff216dc000-7fff21edc000 rw-p 00000000 00:00 0 7fff21edc000-7fff21edd000 ---p 00000000 00:00 0 7fff21edd000-7fff226dd000 rw-p 00000000 00:00 0 7fff226dd000-7fff226de000 ---p 00000000 00:00 0 7fff226de000-7fff22ede000 rw-p 00000000 00:00 0 7fff22ede000-7fff22edf000 ---p 00000000 00:00 0 7fff22edf000-7fff236df000 rw-p 00000000 00:00 0 7fff236df000-7fff236e0000 ---p 00000000 00:00 0 7fff236e0000-7fff23ee0000 rw-p 00000000 00:00 0 7fff23ee0000-7fff23ee1000 ---p 00000000 00:00 0 7fff23ee1000-7fff246e1000 rw-p 00000000 00:00 0 7fff246e1000-7fff246e2000 ---p 00000000 00:00 0 7fff246e2000-7fff24ee2000 rw-p 00000000 00:00 0 7fff24ee2000-7fff24ee3000 ---p 00000000 00:00 0 7fff24ee3000-7fff256e3000 rw-p 00000000 00:00 0 7fff256e3000-7fff256e4000 ---p 00000000 00:00 0 7fff256e4000-7fff25ee4000 rw-p 00000000 00:00 0 7fff25ee4000-7fff25ee5000 ---p 00000000 00:00 0 7fff25ee5000-7fff266e5000 rw-p 00000000 00:00 0 7fff266e5000-7fff266e6000 ---p 00000000 00:00 0 7fff266e6000-7fff26ee6000 rw-p 00000000 00:00 0 7fff26ee6000-7fff26ee7000 ---p 00000000 00:00 0 7fff26ee7000-7fff276e7000 rw-p 00000000 00:00 0 7fff276e7000-7fff276e8000 ---p 00000000 00:00 0 7fff276e8000-7fff27ee8000 rw-p 00000000 00:00 0 7fff27ee8000-7fff27ee9000 ---p 00000000 00:00 0 7fff27ee9000-7fff286e9000 rw-p 00000000 00:00 0 7fff286e9000-7fff286ea000 ---p 00000000 00:00 0 7fff286ea000-7fff28eea000 rw-p 00000000 00:00 0 Program received signal SIGABRT, Aborted. [Switching to Thread 0x7fffeec86700 (LWP 12501)] 0x00007ffff0303349 in __GI_raise (sig=sig@entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56 56 return INLINE_SYSCALL (tgkill, 3, pid, selftid, sig); (gdb) bt #0 0x00007ffff0303349 in __GI_raise (sig=sig@entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56 #1 0x00007ffff0304748 in __GI_abort () at abort.c:89 #2 0x00007ffff03411d4 in __libc_message (do_abort=do_abort@entry=2, fmt=fmt@entry=0x7ffff04334b8 "*** Error in `%s': %s: 0x%s ***\n") at ../sysdeps/posix/libc_fatal.c:175 #3 0x00007ffff034695e in malloc_printerr (action=3, str=0x7ffff042f608 "free(): invalid size", ptr=<optimized out>) at malloc.c:4991 #4 0x00007ffff0347666 in _int_free (av=<optimized out>, p=<optimized out>, have_lock=0) at malloc.c:3837 #5 0x00007ffff290dec8 in g_free () from /usr/lib/../lib64/libglib-2.0.so.0 #6 0x00000000004bce3f in ags_list_free_and_free_link (list=0x7fffb438b000) at ./src/ags/lib/ags_list.c:29 #7 0x000000000046a6fc in ags_audio_signal_finalize (gobject=0x7fffb4006800) at ./src/ags/audio/ags_audio_signal.c:329 #8 0x00007ffff2df1282 in g_object_unref () from /usr/lib/../lib64/libgobject-2.0.so.0 #9 0x000000000046c702 in ags_recall_recycling_set_property (gobject=0x7fffd07954f0, prop_id=<optimized out>, value=0x7fffeec83d50, param_spec=0x7fffc4002190) at ./src/ags/audio/ags_recall_recycling.c:367 #10 0x00007ffff2dec876 in object_set_property () from /usr/lib/../lib64/libgobject-2.0.so.0 #11 0x00007ffff2dee774 in g_object_set_valist () from /usr/lib/../lib64/libgobject-2.0.so.0 #12 0x00007ffff2deef54 in g_object_set () from /usr/lib/../lib64/libgobject-2.0.so.0 #13 0x000000000046c286 in ags_recall_recycling_destination_add_audio_signal_callback (destination=<optimized out>, audio_signal=0x7fffb4006900, recall_recycling=0x7fffd07954f0) at ./src/ags/audio/ags_recall_recycling.c:728 #14 0x00007ffff2de986a in g_cclosure_marshal_VOID__OBJECT () from /usr/lib/../lib64/libgobject-2.0.so.0 #15 0x00007ffff2de4d5f in g_closure_invoke () from /usr/lib/../lib64/libgobject-2.0.so.0 #16 0x00007ffff2e01f24 in signal_emit_unlocked_R () from /usr/lib/../lib64/libgobject-2.0.so.0 #17 0x00007ffff2e00f7c in g_signal_emit_valist () from /usr/lib/../lib64/libgobject-2.0.so.0 #18 0x00007ffff2e014cf in g_signal_emit () from /usr/lib/../lib64/libgobject-2.0.so.0 #19 0x00000000004686cb in ags_recycling_add_audio_signal (recycling=0x7fffe00abcf0, audio_signal=0x7fffb4006900) at ./src/ags/audio/ags_recycling.c:278 #20 0x00000000004815e0 in ags_loop_channel_run_create_audio_signals (loop_channel_run=0x7fffe000aae0) at ./src/ags/audio/recall/ags_loop_channel_run.c:607 #21 0x00007ffff2de8607 in g_cclosure_marshal_VOID(unsigned int0_t, void) () from /usr/lib/../lib64/libgobject-2.0.so.0 #22 0x00007ffff2de4ff7 in _g_closure_invoke_va () from /usr/lib/../lib64/libgobject-2.0.so.0 #23 0x00007ffff2e002f9 in g_signal_emit_valist () from /usr/lib/../lib64/libgobject-2.0.so.0 #24 0x00007ffff2e014cf in g_signal_emit () from /usr/lib/../lib64/libgobject-2.0.so.0 #25 0x000000000047d6ea in ags_count_beats_audio_run_sequencer_loop (count_beats_audio_run=0x7fffd005ace0, run_order=0) at ./src/ags/audio/recall/ags_count_beats_audio_run.c:861 #26 0x000000000047d8eb in ags_count_beats_audio_run_sequencer_alloc_output_callback (delay_audio_run=<optimized out>, run_order=0, delay=<optimized out>, attack=<optimized out>, count_beats_audio_run=0x7fffd005ace0) at ./src/ags/audio/recall/ags_count_beats_audio_run.c:948 #27 0x00007ffff2de4d5f in g_closure_invoke () from /usr/lib/../lib64/libgobject-2.0.so.0 #28 0x00007ffff2e01c62 in signal_emit_unlocked_R () from /usr/lib/../lib64/libgobject-2.0.so.0 #29 0x00007ffff2e00f7c in g_signal_emit_valist () from /usr/lib/../lib64/libgobject-2.0.so.0 #30 0x00007ffff2e014cf in g_signal_emit () from /usr/lib/../lib64/libgobject-2.0.so.0 #31 0x0000000000476ff5 in ags_delay_audio_run_sequencer_alloc_output (delay_audio_run=0x7fffd078c010, run_order=0, delay=<optimized out>, attack=684) at ./src/ags/audio/recall/ags_delay_audio_run.c:574 #32 0x000000000047736a in ags_delay_audio_run_run_pre (recall=<optimized out>) at ./src/ags/audio/recall/ags_delay_audio_run.c:420 #33 0x00007ffff2de8607 in g_cclosure_marshal_VOID(unsigned int0_t, void) () from /usr/lib/../lib64/libgobject-2.0.so.0 #34 0x00007ffff2de5434 in g_type_class_meta_marshalv () from /usr/lib/../lib64/libgobject-2.0.so.0 #35 0x00007ffff2de4ff7 in _g_closure_invoke_va () from /usr/lib/../lib64/libgobject-2.0.so.0 #36 0x00007ffff2e002f9 in g_signal_emit_valist () from /usr/lib/../lib64/libgobject-2.0.so.0 #37 0x00007ffff2e014cf in g_signal_emit () from /usr/lib/../lib64/libgobject-2.0.so.0 #38 0x0000000000471ce6 in ags_recall_run_pre (recall=0x7fffd078c010) at ./src/ags/audio/ags_recall.c:1239 #39 0x000000000046fd69 in ags_audio_play (audio=audio@entry=0x7fffe00039f0, recall_id=recall_id@entry=0x7fffd00ed2c0, stage=0) at ./src/ags/audio/ags_audio.c:2252 #40 0x000000000048ca3b in ags_channel_recursive_play_output (output=output@entry=0x7fffe00c2470, output_recall_id=output_recall_id@entry=0x7fffd00ed2c0) at ./src/ags/audio/ags_channel.c:2011 #41 0x000000000048cfee in ags_channel_recursive_play (channel=channel@entry=0x7fffe00c2470, recall_id=0x7fffd00ed2c0, stage=stage@entry=0) at ./src/ags/audio/ags_channel.c:2088 #42 0x00000000004c324f in ags_audio_loop_play_audio (audio_loop=audio_loop@entry=0x902fd0) at ./src/ags/thread/ags_audio_loop.c:779 #43 0x00000000004c3458 in ags_audio_loop_run (thread=<optimized out>) at ./src/ags/thread/ags_audio_loop.c:540 #44 0x00007ffff2de7dd0 in g_cclosure_marshal_VOID__VOIDv () from /usr/lib/../lib64/libgobject-2.0.so.0 #45 0x00007ffff2de5434 in g_type_class_meta_marshalv () from /usr/lib/../lib64/libgobject-2.0.so.0 #46 0x00007ffff2de4ff7 in _g_closure_invoke_va () from /usr/lib/../lib64/libgobject-2.0.so.0 #47 0x00007ffff2e002f9 in g_signal_emit_valist () from /usr/lib/../lib64/libgobject-2.0.so.0 #48 0x00007ffff2e014cf in g_signal_emit () from /usr/lib/../lib64/libgobject-2.0.so.0 #49 0x00000000004c16e6 in ags_thread_run (thread=0x902fd0) at ./src/ags/thread/ags_thread.c:1914 #50 0x00000000004c19c7 in ags_thread_loop (ptr=<optimized out>) at ./src/ags/thread/ags_thread.c:1804 #51 0x00007ffff245f032 in start_thread (arg=0x7fffeec86700) at pthread_create.c:309 #52 0x00007ffff03b3d0d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111

1 0

[alsa-devel] [PATCH 0/7] Kirkwood ASoC updates
by Russell King - ARM Linux 28 Jun '14

28 Jun '14

Mark, Here are a set of patches I've had for a while as part of my DPCM work. I think it's worth getting these merged. The first patch fixes a bug at ASoC level with debugfs handling - it is not permitted to create more than one debugfs directory with the same name in the same parent directory - and this can happen when the dummy platform and codec are used. The second patch adds a helper definition to handle both I2S and SPDIF bits in the RECCTL register in the same way that we already do with the PLAYCTL register. Patch 3 fixes the RECCTL enable masking - we would accumulate enable bits but never clear any of them. Patch 4 makes the driver's handling of the PLAYCTL register slightly more conformant to the documentation - which requires that the appropriate mute bits are set while the respective output interface is disabled. Patch 5 updates my previous attempt to properly fix this. The previous attempt did stop things going wrong from the hardware perspective, but we are left with the occasional stuck-busy. Although that was harmless, this patch stops this occuring in addition to the previous fix. Patch 6 lifts some of the driver restrictions on the period size and number of periods - there's no reason to restrict these values, so let's give userspace more flexibility. Patch 7 adds NO_PERIOD_WAKEUP support for kirkwood, permitting interrupt-less operation. -- FTTC broadband for 0.8mile line: now at 9.7Mbps down 460kbps up... slowly improving, and getting towards what was expected from it.

5 12

[alsa-devel] [PATCH] ALSA: Remove the runtime local variable in snd_pcm_period_elapsed
by JongHo Kim 27 Jun '14

27 Jun '14

>From a46df32a4fc184364be45ffe8957d5a20818d979 Mon Sep 17 00:00:00 2001 From: JongHo Kim <furmuwon(a)gmail.com> Date: Sat, 28 Jun 2014 02:49:13 +0900 Subject: [PATCH] ALSA: Remove the runtime local variable in snd_pcm_period_elapsed The local runtime variable in snd_pcm_period_elapsed has the value of substream->runtime. If the substream->runtime memory was freed, The local runtime variable can point the invalid memory. If do not fix this, can refer to freed memory. This patch remove local runtime variable and check the NULL directly. Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> --- sound/core/pcm_lib.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9acc77e..c3de2d3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1836,15 +1836,13 @@ EXPORT_SYMBOL(snd_pcm_lib_ioctl); */ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime; unsigned long flags; if (PCM_RUNTIME_CHECK(substream)) return; - runtime = substream->runtime; - if (runtime->transfer_ack_begin) - runtime->transfer_ack_begin(substream); + if (substream->runtime && substream->runtime->transfer_ack_begin) + substream->runtime->transfer_ack_begin(substream); snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || @@ -1855,9 +1853,11 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_timer_interrupt(substream->timer, 1); _end: snd_pcm_stream_unlock_irqrestore(substream, flags); - if (runtime->transfer_ack_end) - runtime->transfer_ack_end(substream); - kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + if (substream->runtime && substream->runtime->transfer_ack_end) + substream->runtime->transfer_ack_end(substream); + + if (substream->runtime) + kill_fasync(&substream->runtime->fasync, SIGIO, POLL_IN); } EXPORT_SYMBOL(snd_pcm_period_elapsed); -- 1.8.1.2

2 1

[alsa-devel] alsa-tools 1.0.28 regression with sblive/emu10k1: init_live shell script broken.
by Roc Vallès 27 Jun '14

27 Jun '14

I've recently upgraded to alsa-tools 1.0.28. There's apparently a regression: live_init, the script that is bundled for sound blaster live soundcard initialization, has been broken. Without this, the soundcard can only be used at a very basic level (just 2 speakers, no equalization or other effects, most inputs unavailable). Reporting here as the bugtracker is down. Correct run (1.0.17): # sh /usr/bin/init_live # Broken run (1.0.28): # /usr/bin/init_live /usr/bin/init_live: line 49: log: command not found /usr/bin/init_live: line 51: local: can only be used in a function /usr/bin/init_live: line 55: [: 0: unary operator expected /usr/bin/init_live: line 60: runlo10k1: command not found /usr/bin/init_live: line 62: log: command not found rename: not enough arguments Usage: rename [options] expression replacement file... Options: -v, --verbose explain what is being done -s, --symlink act on symlink target -h, --help display this help and exit -V, --version output version information and exit For more details see rename(1). /usr/bin/init_live: line 79: log: command not found /usr/bin/init_live: line 81: runlo10k1: command not found /usr/bin/init_live: line 84: connect: command not found /usr/bin/init_live: line 90: log: command not found /usr/bin/init_live: line 92: runlo10k1: command not found /usr/bin/init_live: line 95: copy: command not found /usr/bin/init_live: line 97: local: can only be used in a function /usr/bin/init_live: line 98: local: can only be used in a function /usr/bin/init_live: line 100: log: command not found /usr/bin/init_live: line 103: runlo10k1: command not found /usr/bin/init_live: line 105: connect: command not found /usr/bin/init_live: line 108: volume: command not found /usr/bin/init_live: line 110: local: can only be used in a function /usr/bin/init_live: line 111: local: can only be used in a function /usr/bin/init_live: line 112: local: can only be used in a function /usr/bin/init_live: line 113: local: can only be used in a function /usr/bin/init_live: line 115: log: command not found /usr/bin/init_live: line 118: runlo10k1: command not found /usr/bin/init_live: line 120: connect: command not found /usr/bin/init_live: line 123: master_volume: command not found /usr/bin/init_live: line 125: local: can only be used in a function /usr/bin/init_live: line 126: local: can only be used in a function /usr/bin/init_live: line 127: local: can only be used in a function /usr/bin/init_live: line 128: local: can only be used in a function /usr/bin/init_live: line 130: log: command not found /usr/bin/init_live: line 133: runlo10k1: command not found /usr/bin/init_live: line 135: connect: command not found /usr/bin/init_live: line 138: switch: command not found /usr/bin/init_live: line 140: local: can only be used in a function /usr/bin/init_live: line 141: local: can only be used in a function /usr/bin/init_live: line 142: local: can only be used in a function /usr/bin/init_live: line 144: log: command not found /usr/bin/init_live: line 148: runlo10k1: command not found /usr/bin/init_live: line 150: connect: command not found /usr/bin/init_live: line 153: switch_2x2: command not found /usr/bin/init_live: line 155: local: can only be used in a function /usr/bin/init_live: line 156: local: can only be used in a function /usr/bin/init_live: line 157: local: can only be used in a function /usr/bin/init_live: line 158: local: can only be used in a function /usr/bin/init_live: line 160: log: command not found /usr/bin/init_live: line 163: runlo10k1: command not found /usr/bin/init_live: line 165: connect: command not found /usr/bin/init_live: line 166: connect: command not found /usr/bin/init_live: line 169: switch_5x2: command not found /usr/bin/init_live: line 171: local: can only be used in a function /usr/bin/init_live: line 172: local: can only be used in a function /usr/bin/init_live: line 173: local: can only be used in a function /usr/bin/init_live: line 174: local: can only be used in a function /usr/bin/init_live: line 176: log: command not found /usr/bin/init_live: line 179: runlo10k1: command not found /usr/bin/init_live: line 181: connect: command not found /usr/bin/init_live: line 182: connect: command not found /usr/bin/init_live: line 185: master_switch: command not found /usr/bin/init_live: line 187: local: can only be used in a function /usr/bin/init_live: line 188: local: can only be used in a function /usr/bin/init_live: line 189: local: can only be used in a function /usr/bin/init_live: line 190: local: can only be used in a function /usr/bin/init_live: line 192: log: command not found /usr/bin/init_live: line 195: runlo10k1: command not found /usr/bin/init_live: line 197: connect: command not found /usr/bin/init_live: line 200: playback: command not found /usr/bin/init_live: line 205: log: command not found /usr/bin/init_live: line 208: switch: command not found /usr/bin/init_live: line 209: volume: command not found /usr/bin/init_live: line 211: add_output: command not found /usr/bin/init_live: line 214: playback_noswitch: command not found /usr/bin/init_live: line 219: log: command not found /usr/bin/init_live: line 222: volume: command not found /usr/bin/init_live: line 224: add_output: command not found /usr/bin/init_live: line 227: capture: command not found /usr/bin/init_live: line 232: log: command not found /usr/bin/init_live: line 235: switch: command not found /usr/bin/init_live: line 236: volume: command not found /usr/bin/init_live: line 238: connect: command not found /usr/bin/init_live: line 241: capture_noswitch: command not found /usr/bin/init_live: line 246: log: command not found /usr/bin/init_live: line 249: volume: command not found /usr/bin/init_live: line 251: connect: command not found /usr/bin/init_live: line 254: master: command not found /usr/bin/init_live: line 256: local: can only be used in a function /usr/bin/init_live: line 257: local: can only be used in a function /usr/bin/init_live: line 258: local: can only be used in a function /usr/bin/init_live: line 260: log: command not found /usr/bin/init_live: line 263: master_volume: command not found /usr/bin/init_live: line 264: master_switch: command not found /usr/bin/init_live: line 267: dump_patches: command not found

3 2

[alsa-devel] [RFC] Add DummyDevice property to UCM
by Felipe Tonello 27 Jun '14

27 Jun '14

Hi, I would like to see what you guys think about adding a DummyDevice official property to a UCM device. This is very useful for users to configure properly when, let's say, the PCM is a dummy device. It is a common practice on embedded systems to have dummy PCMs. BR Felipe

3 4

[alsa-devel] [PATCH] ucm: Document PlaybackPCMIsDummy and CapturePCMIsDummy values
by Tanu Kaskinen 27 Jun '14

27 Jun '14

At least PulseAudio needs special handling for dummy devices. To allow that to happen automatically, the UCM configuration should contain the information about which PCMs are dummy. Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> --- include/use-case.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/include/use-case.h b/include/use-case.h index 4e13249..f30168f 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -258,7 +258,17 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * Recommended names for values: * TQ - Tone Quality * PlaybackPCM - full PCM playback device name + * PlaybackPCMIsDummy - Valid values: "yes" and "no". If set to "yes", the + * PCM named by the PlaybackPCM value is a dummy device, + * meaning that opening it enables an audio path in the + * hardware, but writing to the PCM device has no + * effect. * CapturePCM - full PCM capture device name + * CapturePCMIsDummy - Valid values: "yes" and "no". If set to "yes", the + * PCM named by the CapturePCM value is a dummy device, + * meaning that opening it enables an audio path in the + * hardware, but reading from the PCM device has no + * effect. * PlaybackRate - playback device sample rate * PlaybackChannels - playback device channel count * PlaybackCTL - playback control device name -- 1.9.3

3 2

[alsa-devel] [PATCH 01/16] ASoC: cs4270: Remove redundant OOM message
by Sachin Kamat 27 Jun '14

27 Jun '14

Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/codecs/cs4270.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 9947a9583679..e6d4ff9fd992 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -664,10 +664,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private), GFP_KERNEL); - if (!cs4270) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs4270) return -ENOMEM; - } /* get the power supply regulators */ for (i = 0; i < ARRAY_SIZE(supply_names); i++) -- 1.7.9.5

6 28

[alsa-devel] [PATCH 1/1] ASoC: samsung: Remove unused variable from idma.c
by Sachin Kamat 27 Jun '14

27 Jun '14

‘iiscon’ is not used in the function. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/samsung/idma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 8cc5770abb39..db6cefa18017 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -261,10 +261,9 @@ static int idma_mmap(struct snd_pcm_substream *substream, static irqreturn_t iis_irq(int irqno, void *dev_id) { struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; - u32 iiscon, iisahb, val, addr; + u32 iisahb, val, addr; iisahb = readl(idma.regs + I2SAHB); - iiscon = readl(idma.regs + I2SCON); val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; -- 1.7.9.5

2 1

Re: [alsa-devel] No sound on Thinkpad T440s in Docking Station
by Jim Minter 27 Jun '14

27 Jun '14

Hello, I'm getting the same issue reported by Joschi in May, but with a docked T540p. I can confirm that doing the following fixes the issue for me: ==== 8< ==== # cat <<EOF >/lib/firmware/t540p.fw [codec] 0x10ec0292 0x17aa2210 0 [pincfg] 0x16 0x21211010 0x19 0x21a11010 EOF # cat <<EOF >/etc/modprobe.d/hda-intel.conf options snd-hda-intel patch=t540p.fw,t540p.fw,t540p.fw EOF Then rmmod/modprobe the snd-hda-intel kernel module, or just reboot. ==== 8< ==== I can see http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=1c37… which I presume resolves this problem for T440s and T440p users. Please could an additional line, I'm guessing as follows, be added to sound/pci/hda/patch_realtek.c to include T540p users in this happy group? + SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), Thanks! Jim

2 1

[alsa-devel] [PATCH] ALSA: hda - Add another headset pin quirk for some Dell machines
by David Henningsson 27 Jun '14

27 Jun '14

Another quirk to make the headset mic work on some new Dell machines. Cc: Hui Wang <hui.wang(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1297581 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 60bf5ee..4ebbe03 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5199,6 +5199,18 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x411111f0}, {0x1d, 0x40700001}, {0x1e, 0x411111f0}), + SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x13, 0x90a60140}, + {0x14, 0x90170110}, + {0x15, 0x0221401f}, + {0x16, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x411111f0}, + {0x1d, 0x40700001}, + {0x1e, 0x411111f0}), {} }; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3 v3] arizona: Improvements to codec DVFS control
by Charles Keepax 27 Jun '14

27 Jun '14

Hi, Respin of Richard's DVFS series, from me since he is on holiday. I have based these on Mark's ASoC for-next branch as there a few changes in there to the same parts of code. But let me know if you would rather have a series based on Lee's tree instead. This set of changes cleans up the way the drivers handle the control of internal subsystem clocks for codecs with DVFS. This previously relied on the clock control happening as a side-effect of changing the regulator voltage, rather than explicitly, which assumed that you were using the built-in regulator. Also DVFS control was being done from the ADSP driver despite that driver supposedly being codec- agnostic. There are other cases on some codecs where DVFS control is needed so this sequence of patches cleans up the way DVFS is implemented and adds the extra case of DVFS boost for higher sample rates on WM8997 and WM5102 Changes since v2: - Correct some small issues (wrong printf specifier, clarity on PRE_PMD/POST_PMD event) - Added warning for calling arizona_dvfs_down with flags that are not enabled - Squashed patches 2 and 4, so the SUBSYS bit is not set from two seperate places Changes since v1: - Changed mask variable to flags to be more consistent with its usage - Fixed minor tabbing issue - Adding missing SND_SOC_DAPM_PRE_PMD event flag in WM_ADSP2_E Richard Fitzgerald (3): mfd: arizona: Export function to control subsystem DVFS ASoC: wm_adsp: Move DVFS control into codec driver ASoC: arizona: Add DVFS handling for sample rate control drivers/mfd/arizona-core.c | 87 ++++++++++++++++++++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 5 -- include/linux/mfd/arizona/core.h | 12 +++++ sound/soc/codecs/arizona.c | 22 +++++++++- sound/soc/codecs/wm5102.c | 47 +++++++++++++++++++- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm_adsp.c | 73 +------------------------------- sound/soc/codecs/wm_adsp.h | 13 +++--- 8 files changed, 174 insertions(+), 87 deletions(-) -- 1.7.2.5

2 4

[alsa-devel] [PATCH V2] ASoC: mcbsp: remove unnecessary null test before kfree
by Fabian Frederick 27 Jun '14

27 Jun '14

Fix checkpatch warning: WARNING: kfree(NULL) is safe this check is probably not required Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- V2: replace subject "sound/soc/omap/mcbsp.c" by "ASoC: mcbsp:" (suggested by Peter Ujfalusi) sound/soc/omap/mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 86c7538..68a1252 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -621,8 +621,7 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = NULL; spin_unlock(&mcbsp->lock); - if (reg_cache) - kfree(reg_cache); + kfree(reg_cache); } /* -- 1.8.4.5

2 1

[alsa-devel] infinity wait in snd_pcm_wait_nocheck
by 김은희 27 Jun '14

27 Jun '14

My system plays and captures short streams using by dmix device after boot. But that system sometimes go infinity wait state when stream plays. I use alsa-lib-1.0.24.1. but v1.0.27 and 1.0.28 also occur same err. So I have below questions. - What event that poll (in snd_pcm_wait_nocheck) wait? I know snd_pcm_wait function is for wait PCM ready. But I want to know details. - How can I avoid that problem? I changed timeout(in snd_pcm_wait) from -1 to 100. But system go infinity loop. And I attach status and params when this problem occurred. - status state: RUNNING owner_pid : 1213 trigger_time: 5.285075511 tstamp : 5.333190511 delay : -832 avail : 17216 avail_max : 17216 ----- hw_ptr : 832 appl_ptr : 0 - hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 44100 (44100/1) period_size: 1024 buffer_size: 16384 - sw_params tstamp_mode: ENABLE period_step: 1 avail_min: 1024 start_threshold: 1 stop_threshold: 1073741824 silence_threshold: 0 silence_size: 1073741824 boundary: 1073741824 Please reply about it. Thanks. ----- Eunhee Kim

1 0

[alsa-devel] [PATCH 12/12] OMAPDSS: Remove all references obsolete HDMI audio callbacks
by Jyri Sarha 26 Jun '14

26 Jun '14

In new model these callbacks are obsolete since the ASoC component drivers are integrated into the HDMI drivers and no callbacks are needed anymore. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- .../fbdev/omap2/displays-new/connector-hdmi.c | 99 -------------------- .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 ----------- include/video/omapdss.h | 31 ------ 3 files changed, 186 deletions(-) diff --git a/drivers/video/fbdev/omap2/displays-new/connector-hdmi.c b/drivers/video/fbdev/omap2/displays-new/connector-hdmi.c index 131c6e2..dd36d71 100644 --- a/drivers/video/fbdev/omap2/displays-new/connector-hdmi.c +++ b/drivers/video/fbdev/omap2/displays-new/connector-hdmi.c @@ -170,98 +170,6 @@ static bool hdmic_detect(struct omap_dss_device *dssdev) return in->ops.hdmi->detect(in); } -static int hdmic_audio_enable(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - int r; - - /* enable audio only if the display is active */ - if (!omapdss_device_is_enabled(dssdev)) - return -EPERM; - - r = in->ops.hdmi->audio_enable(in); - if (r) - return r; - - dssdev->audio_state = OMAP_DSS_AUDIO_ENABLED; - - return 0; -} - -static void hdmic_audio_disable(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - in->ops.hdmi->audio_disable(in); - - dssdev->audio_state = OMAP_DSS_AUDIO_DISABLED; -} - -static int hdmic_audio_start(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - int r; - - /* - * No need to check the panel state. It was checked when trasitioning - * to AUDIO_ENABLED. - */ - if (dssdev->audio_state != OMAP_DSS_AUDIO_ENABLED) - return -EPERM; - - r = in->ops.hdmi->audio_start(in); - if (r) - return r; - - dssdev->audio_state = OMAP_DSS_AUDIO_PLAYING; - - return 0; -} - -static void hdmic_audio_stop(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - in->ops.hdmi->audio_stop(in); - - dssdev->audio_state = OMAP_DSS_AUDIO_ENABLED; -} - -static bool hdmic_audio_supported(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - if (!omapdss_device_is_enabled(dssdev)) - return false; - - return in->ops.hdmi->audio_supported(in); -} - -static int hdmic_audio_config(struct omap_dss_device *dssdev, - struct omap_dss_audio *audio) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - int r; - - /* config audio only if the display is active */ - if (!omapdss_device_is_enabled(dssdev)) - return -EPERM; - - r = in->ops.hdmi->audio_config(in, audio); - if (r) - return r; - - dssdev->audio_state = OMAP_DSS_AUDIO_CONFIGURED; - - return 0; -} - static int hdmic_set_hdmi_mode(struct omap_dss_device *dssdev, bool hdmi_mode) { struct panel_drv_data *ddata = to_panel_data(dssdev); @@ -296,13 +204,6 @@ static struct omap_dss_driver hdmic_driver = { .detect = hdmic_detect, .set_hdmi_mode = hdmic_set_hdmi_mode, .set_hdmi_infoframe = hdmic_set_infoframe, - - .audio_enable = hdmic_audio_enable, - .audio_disable = hdmic_audio_disable, - .audio_start = hdmic_audio_start, - .audio_stop = hdmic_audio_stop, - .audio_supported = hdmic_audio_supported, - .audio_config = hdmic_audio_config, }; static int hdmic_probe_pdata(struct platform_device *pdev) diff --git a/drivers/video/fbdev/omap2/displays-new/encoder-tpd12s015.c b/drivers/video/fbdev/omap2/displays-new/encoder-tpd12s015.c index c891d8f..235e3d5 100644 --- a/drivers/video/fbdev/omap2/displays-new/encoder-tpd12s015.c +++ b/drivers/video/fbdev/omap2/displays-new/encoder-tpd12s015.c @@ -193,55 +193,6 @@ static bool tpd_detect(struct omap_dss_device *dssdev) return gpio_get_value_cansleep(ddata->hpd_gpio); } -static int tpd_audio_enable(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - return in->ops.hdmi->audio_enable(in); -} - -static void tpd_audio_disable(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - in->ops.hdmi->audio_disable(in); -} - -static int tpd_audio_start(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - return in->ops.hdmi->audio_start(in); -} - -static void tpd_audio_stop(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - in->ops.hdmi->audio_stop(in); -} - -static bool tpd_audio_supported(struct omap_dss_device *dssdev) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - return in->ops.hdmi->audio_supported(in); -} - -static int tpd_audio_config(struct omap_dss_device *dssdev, - struct omap_dss_audio *audio) -{ - struct panel_drv_data *ddata = to_panel_data(dssdev); - struct omap_dss_device *in = ddata->in; - - return in->ops.hdmi->audio_config(in, audio); -} - static int tpd_set_infoframe(struct omap_dss_device *dssdev, const struct hdmi_avi_infoframe *avi) { @@ -275,13 +226,6 @@ static const struct omapdss_hdmi_ops tpd_hdmi_ops = { .detect = tpd_detect, .set_infoframe = tpd_set_infoframe, .set_hdmi_mode = tpd_set_hdmi_mode, - - .audio_enable = tpd_audio_enable, - .audio_disable = tpd_audio_disable, - .audio_start = tpd_audio_start, - .audio_stop = tpd_audio_stop, - .audio_supported = tpd_audio_supported, - .audio_config = tpd_audio_config, }; static int tpd_probe_pdata(struct platform_device *pdev) diff --git a/include/video/omapdss.h b/include/video/omapdss.h index 069dfca..9035325 100644 --- a/include/video/omapdss.h +++ b/include/video/omapdss.h @@ -635,19 +635,6 @@ struct omapdss_hdmi_ops { int (*set_hdmi_mode)(struct omap_dss_device *dssdev, bool hdmi_mode); int (*set_infoframe)(struct omap_dss_device *dssdev, const struct hdmi_avi_infoframe *avi); - - /* - * Note: These functions might sleep. Do not call while - * holding a spinlock/readlock. - */ - int (*audio_enable)(struct omap_dss_device *dssdev); - void (*audio_disable)(struct omap_dss_device *dssdev); - bool (*audio_supported)(struct omap_dss_device *dssdev); - int (*audio_config)(struct omap_dss_device *dssdev, - struct omap_dss_audio *audio); - /* Note: These functions may not sleep */ - int (*audio_start)(struct omap_dss_device *dssdev); - void (*audio_stop)(struct omap_dss_device *dssdev); }; struct omapdss_dsi_ops { @@ -858,24 +845,6 @@ struct omap_dss_driver { int (*set_hdmi_mode)(struct omap_dss_device *dssdev, bool hdmi_mode); int (*set_hdmi_infoframe)(struct omap_dss_device *dssdev, const struct hdmi_avi_infoframe *avi); - - /* - * For display drivers that support audio. This encompasses - * HDMI and DisplayPort at the moment. - */ - /* - * Note: These functions might sleep. Do not call while - * holding a spinlock/readlock. - */ - int (*audio_enable)(struct omap_dss_device *dssdev); - void (*audio_disable)(struct omap_dss_device *dssdev); - bool (*audio_supported)(struct omap_dss_device *dssdev); - int (*audio_config)(struct omap_dss_device *dssdev, - struct omap_dss_audio *audio); - /* Note: These functions may not sleep */ - int (*audio_start)(struct omap_dss_device *dssdev); - void (*audio_stop)(struct omap_dss_device *dssdev); - }; enum omapdss_version omapdss_get_version(void); -- 1.7.9.5

1 1

[alsa-devel] Wandboard Quad - SPDIF issue, no sound
by Michael Howard 26 Jun '14

26 Jun '14

Hello All, I'm resending this email as orignally I wasn't subcribed and I've been told the message hasn't made it through. I was pointed here from the Wandboard Google Groups forum. If I'm in the wrong place please accept my appologies. With known good cabling and receiver I'm not getting sound from the Wandboard Quad's S/PDIF output. I have built various kernels from mainline, the last two sources I tried are 3.15.1 and 3.16-rc2. With sound built as modules I get nothing on any outputs, errors from dmesg are; dmesg | grep snd 5.206990] imx-spdif sound-spdif.13: snd_soc_register_card failed: -517 [ 5.241976] imx-spdif sound-spdif.13: snd_soc_register_card failed: -517 [ 5.258174] imx-spdif sound-spdif.13: snd-soc-dummy-dai <-> 2004000.spdif mapping ok [ 5.281704] imx-sgtl5000 sound.12: snd_soc_register_card failed (-517) With sound built in I get sound on the 'line out' output and from dmesg; dmesg | grep snd imx-spdif sound-spdif: snd-soc-dummy-dai <-> 2004000.spdif mapping ok In both cases I get (or similar); dmesg | grep firm imx-sdma 20ec000.sdma: Direct firmware load failed with error -2 imx-sdma 20ec000.sdma: loaded firmware 1.1 Not sure if the firmware message above is a problem or not. aplay output; aplay -l **** List of PLAYBACK Hardware Devices **** card 0: imx6wandboardsg [imx6-wandboard-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: imxspdif [imx-spdif], device 0: S/PDIF PCM snd-soc-dummy-dai-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server sysdefault:CARD=imx6wandboardsg imx6-wandboard-sgtl5000, Default Audio Device sysdefault:CARD=imxspdif imx-spdif, Default Audio Device aplay -vv ImABeliever.wav Playing WAVE 'ImABeliever.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Hardware PCM card 1 'imx-spdif' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 ################# + | 55% but no sound form SPDIF. If this is right place for this and if I can be of any help debuuging or you need more info I'm more than happy to help in any way I can. Since I sent the original message Ive realised that data is being sent to and received at the SPDIF output of the wandboard quad it can't seem to handle it. If I connect a DAC to the SPDIF output and then connect the DAC to my recievers aux input I hear the sound in all it's glory. The same receiver and cabling works with a cubietruck and the sunxi kernel. Cheers, Mike --

2 7

[alsa-devel] [PATCH] ALSA: usb-audio: Remove from list in snd_usb_endpoint_free()
by Takashi Iwai 26 Jun '14

26 Jun '14

snd_usb_endpoint_free() deallocates the object but it doesn't unlink from the list. This is actually no problem in the current usage because all elements are freed at once, but it's more consistent to do a proper unlinking there. Reported-by: Julia Lawall <julia.lawall(a)lip6.fr> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a 3.17 material. sound/usb/endpoint.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 114e3e7ff511..304f36bf29cb 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1011,6 +1011,7 @@ void snd_usb_endpoint_free(struct list_head *head) struct snd_usb_endpoint *ep; ep = list_entry(head, struct snd_usb_endpoint, list); + list_del(head); kfree(ep); } -- 2.0.0

1 0

[alsa-devel] [PATCH 1/1] sound/soc/omap/mcbsp.c: remove unnecessary null test before kfree
by Fabian Frederick 26 Jun '14

26 Jun '14

Fix checkpatch warning: WARNING: kfree(NULL) is safe this check is probably not required Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/soc/omap/mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 86c7538..68a1252 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -621,8 +621,7 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = NULL; spin_unlock(&mcbsp->lock); - if (reg_cache) - kfree(reg_cache); + kfree(reg_cache); } /* -- 1.8.4.5

1 0

[alsa-devel] [PATCH] ALSA: hda - restore BCLK M/N values when resuming HSW/BDW display controller
by mengdong.lin＠intel.com 26 Jun '14

26 Jun '14

From: Mengdong Lin <mengdong.lin(a)intel.com> For Intel Haswell/Broadwell display HD-A controller, the 24MHz HD-A link BCLK is converted from Core Display Clock (CDCLK): BCLK = CDCLK * M / N And there are two registers EM4 and EM5 to program M, N value respectively. The EM4/EM5 values will be lost and when the display power well is disabled. BIOS programs CDCLK selected by OEM and EM4/EM5, but BIOS has no idea about display power well on/off at runtime. So the M/N can be wrong if non-default CDCLK is used when the audio controller resumes, which results in an invalid BCLK and abnormal audio playback rate. So this patch saves and restores valid M/N values on controller suspend/resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bb65a124..90a2750 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -580,6 +580,18 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ +static void haswell_save_bclk(struct azx *chip) +{ + chip->bclk_m = azx_readw(chip, EM4); + chip->bclk_n = azx_readw(chip, EM5); +} + +static void haswell_restore_bclk(struct azx *chip) +{ + azx_writew(chip, EM4, chip->bclk_m); + azx_writew(chip, EM5, chip->bclk_n); +} + #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -606,6 +618,13 @@ static int azx_suspend(struct device *dev) free_irq(chip->irq, chip); chip->irq = -1; } + + /* Save BCLK M/N values before they become invalid in D3. + * Will test if display power well can be released now. + */ + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + haswell_save_bclk(chip); + if (chip->msi) pci_disable_msi(chip->pci); pci_disable_device(pci); @@ -625,8 +644,10 @@ static int azx_resume(struct device *dev) if (chip->disabled) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); + haswell_restore_bclk(chip); + } pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); if (pci_enable_device(pci) < 0) { @@ -670,8 +691,10 @@ static int azx_runtime_suspend(struct device *dev) azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + haswell_save_bclk(chip); hda_display_power(false); + } return 0; } @@ -689,8 +712,10 @@ static int azx_runtime_resume(struct device *dev) if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(true); + haswell_restore_bclk(chip); + } /* Read STATESTS before controller reset */ status = azx_readw(chip, STATESTS); diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index 4a7cb01..1469edb 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -96,6 +96,14 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ICH6_REG_SD_BDLPL 0x18 #define ICH6_REG_SD_BDLPU 0x1c +/* Intel HSW/BDW display HD-A controller Extended Mode registers. + * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display + * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled. + */ +#define ICH6_REG_EM4 0x100c +#define ICH6_REG_EM5 0x1010 + /* PCI space */ #define ICH6_PCIREG_TCSEL 0x44 @@ -417,6 +425,10 @@ struct azx { /* secondary power domain for hdmi audio under vga device */ struct dev_pm_domain hdmi_pm_domain; + + /* for HSW/BDW display HDA controller to restore BCLK from CDCLK */ + unsigned int bclk_m; + unsigned int bclk_n; }; #ifdef CONFIG_SND_VERBOSE_PRINTK -- 1.8.1.2

3 2

[alsa-devel] [PATCH v7 0/3] Add ES8328 audio codec
by Sean Cross 26 Jun '14

26 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks, in addition to the i.MX audmux routing. Changes since v6: - Minor clarification of AUDMUX in imx-audio-es8328.txt - Use u32 instead of int in imx-es8328.c for int-mux and ext-mux - Bounds checking on mux-int-port and mux-ext-port in imx-es8328.c - Remove NULL checks before calling of_node_put() Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 61 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 719 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 233 +++++++ 12 files changed, 1510 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

1 3

[alsa-devel] alsa-tools 1.0.28 regression with sblive/emu10k1: init_live shell script broken.
by Roc Vallès 25 Jun '14

25 Jun '14

I've recently upgraded to alsa-tools 1.0.28. There's apparently a regression: The script that is bundled for sound blaster live (emu10k1 in my case) soundcard initialization has been broken. Without this, the soundcard can only be used at a very basic level (just 2 speakers, no equalization or other effects, most inputs unavailable). Reporting here as the bugtracker is down. Correct run (1.0.17): # sh /usr/bin/init_live No Emux synth hwdep device is found Found hardware: "EMU10K1" "SigmaTel STAC9721,23" "AC97a:83847609" "0x1102" "0x8040" Hardware is initialized using a generic method Broken run (1.0.28): # /usr/bin/init_live /usr/bin/init_live: line 49: log: command not found /usr/bin/init_live: line 51: local: can only be used in a function /usr/bin/init_live: line 55: [: 0: unary operator expected /usr/bin/init_live: line 60: runlo10k1: command not found /usr/bin/init_live: line 62: log: command not found rename: not enough arguments Usage: rename [options] expression replacement file... Options: -v, --verbose explain what is being done -s, --symlink act on symlink target -h, --help display this help and exit -V, --version output version information and exit For more details see rename(1). /usr/bin/init_live: line 79: log: command not found /usr/bin/init_live: line 81: runlo10k1: command not found /usr/bin/init_live: line 84: connect: command not found /usr/bin/init_live: line 90: log: command not found /usr/bin/init_live: line 92: runlo10k1: command not found /usr/bin/init_live: line 95: copy: command not found /usr/bin/init_live: line 97: local: can only be used in a function /usr/bin/init_live: line 98: local: can only be used in a function /usr/bin/init_live: line 100: log: command not found /usr/bin/init_live: line 103: runlo10k1: command not found /usr/bin/init_live: line 105: connect: command not found /usr/bin/init_live: line 108: volume: command not found /usr/bin/init_live: line 110: local: can only be used in a function /usr/bin/init_live: line 111: local: can only be used in a function /usr/bin/init_live: line 112: local: can only be used in a function /usr/bin/init_live: line 113: local: can only be used in a function /usr/bin/init_live: line 115: log: command not found /usr/bin/init_live: line 118: runlo10k1: command not found /usr/bin/init_live: line 120: connect: command not found /usr/bin/init_live: line 123: master_volume: command not found /usr/bin/init_live: line 125: local: can only be used in a function /usr/bin/init_live: line 126: local: can only be used in a function /usr/bin/init_live: line 127: local: can only be used in a function /usr/bin/init_live: line 128: local: can only be used in a function /usr/bin/init_live: line 130: log: command not found /usr/bin/init_live: line 133: runlo10k1: command not found /usr/bin/init_live: line 135: connect: command not found /usr/bin/init_live: line 138: switch: command not found /usr/bin/init_live: line 140: local: can only be used in a function /usr/bin/init_live: line 141: local: can only be used in a function /usr/bin/init_live: line 142: local: can only be used in a function /usr/bin/init_live: line 144: log: command not found /usr/bin/init_live: line 148: runlo10k1: command not found /usr/bin/init_live: line 150: connect: command not found /usr/bin/init_live: line 153: switch_2x2: command not found /usr/bin/init_live: line 155: local: can only be used in a function /usr/bin/init_live: line 156: local: can only be used in a function /usr/bin/init_live: line 157: local: can only be used in a function /usr/bin/init_live: line 158: local: can only be used in a function /usr/bin/init_live: line 160: log: command not found /usr/bin/init_live: line 163: runlo10k1: command not found /usr/bin/init_live: line 165: connect: command not found /usr/bin/init_live: line 166: connect: command not found /usr/bin/init_live: line 169: switch_5x2: command not found /usr/bin/init_live: line 171: local: can only be used in a function /usr/bin/init_live: line 172: local: can only be used in a function /usr/bin/init_live: line 173: local: can only be used in a function /usr/bin/init_live: line 174: local: can only be used in a function /usr/bin/init_live: line 176: log: command not found /usr/bin/init_live: line 179: runlo10k1: command not found /usr/bin/init_live: line 181: connect: command not found /usr/bin/init_live: line 182: connect: command not found /usr/bin/init_live: line 185: master_switch: command not found /usr/bin/init_live: line 187: local: can only be used in a function /usr/bin/init_live: line 188: local: can only be used in a function /usr/bin/init_live: line 189: local: can only be used in a function /usr/bin/init_live: line 190: local: can only be used in a function /usr/bin/init_live: line 192: log: command not found /usr/bin/init_live: line 195: runlo10k1: command not found /usr/bin/init_live: line 197: connect: command not found /usr/bin/init_live: line 200: playback: command not found /usr/bin/init_live: line 205: log: command not found /usr/bin/init_live: line 208: switch: command not found /usr/bin/init_live: line 209: volume: command not found /usr/bin/init_live: line 211: add_output: command not found /usr/bin/init_live: line 214: playback_noswitch: command not found /usr/bin/init_live: line 219: log: command not found /usr/bin/init_live: line 222: volume: command not found /usr/bin/init_live: line 224: add_output: command not found /usr/bin/init_live: line 227: capture: command not found /usr/bin/init_live: line 232: log: command not found /usr/bin/init_live: line 235: switch: command not found /usr/bin/init_live: line 236: volume: command not found /usr/bin/init_live: line 238: connect: command not found /usr/bin/init_live: line 241: capture_noswitch: command not found /usr/bin/init_live: line 246: log: command not found /usr/bin/init_live: line 249: volume: command not found /usr/bin/init_live: line 251: connect: command not found /usr/bin/init_live: line 254: master: command not found /usr/bin/init_live: line 256: local: can only be used in a function /usr/bin/init_live: line 257: local: can only be used in a function /usr/bin/init_live: line 258: local: can only be used in a function /usr/bin/init_live: line 260: log: command not found /usr/bin/init_live: line 263: master_volume: command not found /usr/bin/init_live: line 264: master_switch: command not found /usr/bin/init_live: line 267: dump_patches: command not found

1 0

[alsa-devel] snd_usb_endpoint_free
by Julia Lawall 25 Jun '14

25 Jun '14

The function snd_usb_endpoint_free in sound/usb/endpoint.c is defined as follows: void snd_usb_endpoint_free(struct list_head *head) { struct snd_usb_endpoint *ep; ep = list_entry(head, struct snd_usb_endpoint, list); release_urbs(ep, 1); kfree(ep); } I wonder if the final kfree should be list_del? In practice, this function is only used from snd_usb_audio_disconnect in sound/usb/card.c where the entire list is destroyed, but it seems like quite a generic function that someone may someday want to use for just freeing one entry. julia

3 4

[alsa-devel] [PATCH] sound: oss: mpu401.c: Cleaning up variable is set more than once
by Rickard Strandqvist 25 Jun '14

25 Jun '14

A struct member variable is set to the same value more than once This was found using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/oss/mpu401.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 3bbc3ec..31d7f72 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -972,7 +972,6 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner) devc->m_busy = 0; devc->m_state = ST_INIT; devc->shared_irq = hw_config->always_detect; - devc->irq = hw_config->irq; spin_lock_init(&devc->lock); if (devc->irq < 0) -- 1.7.10.4

2 1

[alsa-devel] [PATCH v6 0/3] Add ES8328 audio codec
by Sean Cross 25 Jun '14

25 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks. Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 61 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 719 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 224 +++++++ 12 files changed, 1501 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

2 4

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: aic32x4_set_dai_fmt(): drop unneded register access
by Peter Korsgaard 25 Jun '14

25 Jun '14

There's no point in reading the IFACE2 register when the result isn't used, so drop it. Signed-off-by: Peter Korsgaard <peter(a)korsgaard.com> --- sound/soc/codecs/tlv320aic32x4.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 89e41d2..630367f 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -336,7 +336,6 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1); iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2); - iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2); iface_reg_2 = 0; iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3); iface_reg_3 = iface_reg_3 & ~(1 << 3); -- 2.0.0

1 0

[alsa-devel] [PATCH 1/6] ASoC: Intel: byt-max98090: Move MICBIAS widget to supply of Headset Mic
by Jarkko Nikula 24 Jun '14

24 Jun '14

Move "MICBIAS" as a supply widget to "Headset Mic" instead of keeping it between input pin "IN34" and "Headset Mic". Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-max98090.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 5fc98c64a3f4..a3c5f02a60fc 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { {"IN34", NULL, "Headset Mic"}, - {"IN34", NULL, "MICBIAS"}, - {"MICBIAS", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS"}, {"DMICL", NULL, "Int Mic"}, {"Headphone", NULL, "HPL"}, {"Headphone", NULL, "HPR"}, -- 2.0.0

2 6

[alsa-devel] [PATCH v2 1/2] ASoC: cs4265: bindings: sound: Add binding for CS4265 CODEC.
by Paul Handrigan 24 Jun '14

24 Jun '14

This patch adds binding documentation for the Cirrus Logic CS4265 I2C CODEC. Changes for v2: * Change DT reset pin name to "reset-gpios" Signed-off-by: Paul Handrigan <paul.handrigan(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4265.txt | 29 ++++++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4265.txt diff --git a/Documentation/devicetree/bindings/sound/cs4265.txt b/Documentation/devicetree/bindings/sound/cs4265.txt new file mode 100644 index 0000000..380fff8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4265.txt @@ -0,0 +1,29 @@ +CS4265 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "cirrus,cs4265" + + - reg : the I2C address of the device for I2C. The I2C address depends on + the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f. + If it is low, the i2c address is 0x4e. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Examples: + +codec_ad0_high: cs4265@4f { /* AD0 Pin is high */ + compatible = "cirrus,cs4265"; + reg = <0x4f>; +}; + + +codec_ad0_low: cs4265@4e { /* AD0 Pin is low */ + compatible = "cirrus,cs4265"; + reg = <0x4e>; +}; -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: hda - Adjust speaker HPF and add LED support for HP Spectre 13
by Takashi Iwai 24 Jun '14

24 Jun '14

HP Spectre 13 has the IDT 92HD95 codec, and BIOS seems to set the default high-pass filter in some "safer" range, which results in the very soft tone from the built-in speakers in contrast to Windows. Also, the mute LED control is missing, since 92HD95 codec still has no HP-specific fixups for GPIO setups. This patch adds these missing features: the HPF is adjusted by the vendor-specific verb, and the LED is set up from a DMI string (but with the default polarity = 0 assumption due to the incomplete BIOS on the given machine). Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=74841 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 5 +++ sound/pci/hda/patch_sigmatel.c | 58 +++++++++++++++++++++++++++- 2 files changed, 62 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 85c362d8ea34..d1ab5e17eb13 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -286,6 +286,11 @@ STAC92HD83* hp-inv-led HP with broken BIOS for inverted mute LED auto BIOS setup (default) +STAC92HD95 +========== + hp-led LED support for HP laptops + hp-bass Bass HPF setup for HP Spectre 13 + STAC9872 ======== vaio VAIO laptop without SPDIF diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7f40a150899c..3744ea4e843d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -122,6 +122,12 @@ enum { }; enum { + STAC_92HD95_HP_LED, + STAC_92HD95_HP_BASS, + STAC_92HD95_MODELS +}; + +enum { STAC_925x_REF, STAC_M1, STAC_M1_2, @@ -4128,6 +4134,48 @@ static const struct snd_pci_quirk stac9205_fixup_tbl[] = { {} /* terminator */ }; +static void stac92hd95_fixup_hp_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + if (find_mute_led_cfg(codec, spec->default_polarity)) + codec_dbg(codec, "mute LED gpio %d polarity %d\n", + spec->gpio_led, + spec->gpio_led_polarity); +} + +static const struct hda_fixup stac92hd95_fixups[] = { + [STAC_92HD95_HP_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd95_fixup_hp_led, + }, + [STAC_92HD95_HP_BASS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x1a, 0x795, 0x00}, /* HPF to 100Hz */ + {} + }, + .chained = true, + .chain_id = STAC_92HD95_HP_LED, + }, +}; + +static const struct snd_pci_quirk stac92hd95_fixup_tbl[] = { + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1911, "HP Spectre 13", STAC_92HD95_HP_BASS), + {} /* terminator */ +}; + +static const struct hda_model_fixup stac92hd95_models[] = { + { .id = STAC_92HD95_HP_LED, .name = "hp-led" }, + { .id = STAC_92HD95_HP_BASS, .name = "hp-bass" }, + {} +}; + + static int stac_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -4580,10 +4628,16 @@ static int patch_stac92hd95(struct hda_codec *codec) spec->gen.beep_nid = 0x19; /* digital beep */ spec->pwr_nids = stac92hd95_pwr_nids; spec->num_pwrs = ARRAY_SIZE(stac92hd95_pwr_nids); - spec->default_polarity = -1; /* no default cfg */ + spec->default_polarity = 0; codec->patch_ops = stac_patch_ops; + snd_hda_pick_fixup(codec, stac92hd95_models, stac92hd95_fixup_tbl, + stac92hd95_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + + stac_setup_gpio(codec); + err = stac_parse_auto_config(codec); if (err < 0) { stac_free(codec); @@ -4592,6 +4646,8 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->proc_widget_hook = stac92hd_proc_hook; + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; } -- 2.0.0

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Make a SND_HDA_PIN_QUIRK macro
by David Henningsson 24 Jun '14

24 Jun '14

This is cosmetical - it makes the new pin quirk table look better. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_local.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) Hi Takashi, If I were you, I'd take these patches for 3.16 even though it's not strictly a regression - as it would make maintenance easier not to have any release with the quirk table in non-macro style. Sorry for not making this macro earlier. diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ebd1fa6..4e2d486 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -417,6 +417,27 @@ struct snd_hda_pin_quirk { int value; /* quirk value */ }; +#ifdef CONFIG_SND_DEBUG_VERBOSE + +#define SND_HDA_PIN_QUIRK(_codec, _subvendor, _name, _value, _pins...) \ + { .codec = _codec,\ + .subvendor = _subvendor,\ + .name = _name,\ + .value = _value,\ + .pins = (const struct hda_pintbl[]) { _pins } \ + } +#else + +#define SND_HDA_PIN_QUIRK(_codec, _subvendor, _name, _value, _pins...) \ + { .codec = _codec,\ + .subvendor = _subvendor,\ + .value = _value,\ + .pins = (const struct hda_pintbl[]) { _pins } \ + } + +#endif + + /* fixup types */ enum { HDA_FIXUP_INVALID, -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: hda - Add pin quirk for Dell XPS 15
by David Henningsson 24 Jun '14

24 Jun '14

Two bug reporters with Dell XPS 15 report that they need to use the dell-headset-multi model to get the headset mic working. The two bug reporters have different PCI SSID (1028:05fd and 1028:05fe) but this pin quirk matches both. BugLink: https://bugs.launchpad.net/bugs/1331915 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f52f096..229a189 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6258,6 +6258,27 @@ static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { }, .value = ALC668_FIXUP_AUTO_MUTE, }, + { + .codec = 0x10ec0668, + .subvendor = 0x1028, +#ifdef CONFIG_SND_DEBUG_VERBOSE + .name = "Dell XPS 15", +#endif + .pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x15, 0x0321101f}, + {0x16, 0x40000000}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x411111f0}, + {0x1d, 0x40d6832d}, + {0x1e, 0x411111f0}, + {0x1f, 0x411111f0}, + }, + .value = ALC668_FIXUP_AUTO_MUTE, + }, {} }; -- 1.9.1

2 1

[alsa-devel] [PATCH] sound: oss: mpu401.c: Cleaning up missing break in a case
by Rickard Strandqvist 24 Jun '14

24 Jun '14

Added a missed break in a case statement This was found using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/oss/mpu401.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 25e4609..400896b 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -316,6 +316,7 @@ static int mpu_input_scanner(struct mpu_config *devc, unsigned char midic) case 0xf6: /* printk( "tune_request\n"); */ devc->m_state = ST_INIT; + break; /* * Real time messages -- 1.7.10.4

2 1

[alsa-devel] [PATCH v3 0/4] ASoc: samsung: update s3c24xx to dmaengine and common clk framework
by Vasily Khoruzhick 24 Jun '14

24 Jun '14

This series updates s3c24{xx,12}-i2s driver to use newer DMA and CLK APIs. v2: use hardcoded dma channel number v3: rebase onto v3.16-rc1, add common clk framework-related changes Vasily Khoruzhick (4): ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API ASoC: samsung: drop support for legacy S3C24XX DMA API ASoC: samsung: s3c2412-i2s: Move to clk_prepare_enable/clk_disable_unprepare ASoC: samsung: s3c24xx-i2s: Move to clk_prepare_enable/clk_disable_unprepare sound/soc/samsung/Kconfig | 12 +- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/dma.c | 454 ---------------------------------------- sound/soc/samsung/dmaengine.c | 3 + sound/soc/samsung/s3c-i2s-v2.c | 17 +- sound/soc/samsung/s3c2412-i2s.c | 51 +++-- sound/soc/samsung/s3c24xx-i2s.c | 62 +++--- 7 files changed, 61 insertions(+), 540 deletions(-) delete mode 100644 sound/soc/samsung/dma.c -- 2.0.0

2 5

[alsa-devel] [PATCH 0/3] More simple-card dai node clock parameters
by Jyri Sarha 24 Jun '14

24 Jun '14

These changes target to sii9022 HDMI audio support, but are independent fixes and additions. To cover more cases with simple-card it should be possible to select all parameters for set_sysclk() callback. The default behavior is not changed by these patches. Jyri Sarha (3): ASoC: simple-card: Make u32 DT parameter handling 64-bit proof ASoC: simple-card: Add system-clock-direction DT parameter to dai nodes ASoC: simple-card: Add system-clock-id DT parameter to dai nodes .../devicetree/bindings/sound/simple-card.txt | 3 ++ include/sound/simple_card.h | 2 ++ sound/soc/generic/simple-card.c | 31 +++++++++++++++----- 3 files changed, 29 insertions(+), 7 deletions(-) -- 1.7.9.5

3 8

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Correct S24_3LE support
by Peter Ujfalusi 24 Jun '14

24 Jun '14

Correct the hw_params callback to configure the codec correctly in case of S24_3LE format. S24_LE is not defined as supported format for the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e12fafbb1e09..5360772bc1ad 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: -- 2.0.0

2 5

Re: [alsa-devel] [PATCH] ASoC: cs42l56: Fix stereo channel register assignment for Headphone and LineOut volume mixers
by Brian Austin 23 Jun '14

23 Jun '14

On Mon, 23 Jun 2014, Ryan Harvey wrote: > Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- > sound/soc/codecs/cs42l56.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c > index fdc4bd2..8e68ef5 100644 > --- a/sound/soc/codecs/cs42l56.c > +++ b/sound/soc/codecs/cs42l56.c > @@ -445,9 +445,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { > SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), > > SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, > - CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv), > + CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv), > SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, > - CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv), > + CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv), > > SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, > 0, 0x00, 1, tone_tlv), > -- > 1.7.9.5 > > Acked-by: Brian Austin <brian.austin(a)cirrus.com> Thanks,

2 4

[alsa-devel] [PATCH 0/4 v2] arizona: Improvements to codec DVFS control
by Charles Keepax 23 Jun '14

23 Jun '14

Hi, Respin of Richard's DVFS series, from me since he is on holiday. I have based these on Mark's ASoC for-next branch as there a few changes in there to the same parts of code. But let me know if you would rather have a series based on Lee's tree instead. This set of changes cleans up the way the drivers handle the control of internal subsystem clocks for codecs with DVFS. This previously relied on the clock control happening as a side-effect of changing the regulator voltage, rather than explicitly, which assumed that you were using the built-in regulator. Also DVFS control was being done from the ADSP driver despite that driver supposedly being codec- agnostic. There are other cases on some codecs where DVFS control is needed so this sequence of patches cleans up the way DVFS is implemented and adds the extra case of DVFS boost for higher sample rates on WM8997 and WM5102 Changes since v1: - Changed mask variable to flags to be more consistent with its usage - Fixed minor tabbing issue - Adding missing SND_SOC_DAPM_PRE_PMD event flag in WM_ADSP2_E Richard Fitzgerald (4): mfd: arizona: Export function to control subsystem DVFS ASoC: wm_adsp: Move DVFS control into codec driver ASoC: arizona: Add DVFS handling for sample rate control regulator: arizona-ldo1: Do not control clocking from regulator drivers/mfd/arizona-core.c | 84 ++++++++++++++++++++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 5 -- include/linux/mfd/arizona/core.h | 12 +++++ sound/soc/codecs/arizona.c | 22 +++++++++- sound/soc/codecs/wm5102.c | 47 ++++++++++++++++++++- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm_adsp.c | 73 +-------------------------------- sound/soc/codecs/wm_adsp.h | 13 +++--- 8 files changed, 171 insertions(+), 87 deletions(-) -- 1.7.2.5

2 13

Re: [alsa-devel] [PATCH] ASoC: Fix SOC_DOUBLE_R_SX_TLV volume mixer arguments
by Brian Austin 23 Jun '14

23 Jun '14

On Mon, 23 Jun 2014, Ryan Harvey wrote: > Remove unnecessary bit shifts. Correct min value to match datasheet. #Steps equals number of steps between min and max. > > This patch depends on 8f0367b53d887726b1f4b0b39e0f09b7cca2f79f > ASoC: cs42l56: Fix stereo channel register assignment for Headphone and > LineOut volume mixers. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- > sound/soc/codecs/cs42l52.c | 14 +++++++------- > sound/soc/codecs/cs42l56.c | 12 ++++++------ > sound/soc/codecs/cs42l73.c | 2 +- > 3 files changed, 14 insertions(+), 14 deletions(-) > > diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c > index c0581c5..7aed07e 100644 > --- a/sound/soc/codecs/cs42l52.c > +++ b/sound/soc/codecs/cs42l52.c > @@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { > CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), > > SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, > - CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), > + CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv), > > SOC_ENUM("Headphone Analog Gain", hp_gain_enum), > > SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, > - CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), > + CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), > > SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, > - CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), > + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), > > SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), > > @@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { > SOC_ENUM("MIC Bias Level", mic_bias_level_enum), > > SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, > - CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), > + CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), > SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", > CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, > - 6, 0x7f, 0x19, ipd_tlv), > + 0, 0x19, 0x7F, ipd_tlv), > > SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), > > @@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { > CS42L52_ADCB_MIXER_VOL, 7, 1, 1), > > SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, > - CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), > + CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv), > > SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", > CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, > - 0, 0x7f, 0x19, mix_tlv), > + 0, 0x19, 0x7f, mix_tlv), > SOC_DOUBLE_R("PCM Mixer Switch", > CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), > > diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c > index 8e68ef5..24fbffe 100644 > --- a/sound/soc/codecs/cs42l56.c > +++ b/sound/soc/codecs/cs42l56.c > @@ -421,15 +421,15 @@ static const struct soc_enum ng_delay_enum = > static const struct snd_kcontrol_new cs42l56_snd_controls[] = { > > SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME, > - CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv), > + CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv), > SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1), > > SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME, > - CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), > + CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), > SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1), > > SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME, > - CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), > + CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), > SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1), > > SOC_SINGLE_TLV("Analog Advisory Volume", > @@ -438,16 +438,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { > CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv), > > SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME, > - CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv), > + CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv), > SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR, > CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv), > SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1), > SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), > > SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, > - CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv), > + CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), > SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, > - CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv), > + CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), > > SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, > 0, 0x00, 1, tone_tlv), > diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c > index 7f28c58..bdb8316 100644 > --- a/sound/soc/codecs/cs42l73.c > +++ b/sound/soc/codecs/cs42l73.c > @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { > CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), > > SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, > - CS42L73_MICBPREPGABVOL, 5, 0x34, > + CS42L73_MICBPREPGABVOL, 0, 0x34, > 0x24, micpga_tlv), > > SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, > -- > 1.7.9.5 > > Acked-by: Brian Austin <brian.austin(a)cirrus.com> Thanks,

1 0

Re: [alsa-devel] [PATCH] Sound: Core: fixed assignment in if condition style error in hwdep.c
by Takashi Iwai 23 Jun '14

23 Jun '14

At Fri, 20 Jun 2014 08:50:35 -0700, Vincent Heuken wrote: > > This is a patch to the hwdep.c file that fixes one instance of > the following error: > ERROR: do not use assignment in if condition > > It also, incidentally, fixes one instance of the following warning: > WARNING: line over 80 characters > > Signed-off-by: Vincent Heuken <me(a)vincentheuken.com> If this is a preliminary work for any real fix, I'd happily apply this patch. But, if it's only about a coding style fix, I won't buy it alone. Let me know if it's the former case. thanks, Takashi > --- > sound/core/hwdep.c | 4 +++- > 1 file changed, 3 insertions(+), 1 deletion(-) > > diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c > index 69459e5..42d8324 100644 > --- a/sound/core/hwdep.c > +++ b/sound/core/hwdep.c > @@ -527,7 +527,9 @@ static void __init snd_hwdep_proc_init(void) > { > struct snd_info_entry *entry; > > - if ((entry = snd_info_create_module_entry(THIS_MODULE, "hwdep", NULL)) != NULL) { > + entry = snd_info_create_module_entry(THIS_MODULE, "hwdep", NULL); > + > + if (entry != NULL) { > entry->c.text.read = snd_hwdep_proc_read; > if (snd_info_register(entry) < 0) { > snd_info_free_entry(entry); > -- > 2.0.0 >

1 0

[alsa-devel] [PATCH] trivial: sound/core/seq/seq_memory.c: Fix closing brace followed by if
by Rasmus Villemoes 23 Jun '14

23 Jun '14

Add a newline, and remove a space while at it. Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- sound/core/seq/seq_memory.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 1e206de..4854c21 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -101,7 +101,8 @@ int snd_seq_dump_var_event(const struct snd_seq_event *event, len -= size; } return 0; - } if (! (event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) { + } + if (!(event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) { return func(private_data, event->data.ext.ptr, len); } -- 1.9.2

2 3

[alsa-devel] [RFT v3 0/3] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 23 Jun '14

23 Jun '14

Hi Mark, Liam and Lars, Sorry for the delay. Here is the third version of the multicodec series. I changed the way to store the multiple codec_dai as suggested by Lars and added the missing iterations in all the places pointed by Lars in his comments. Please note that I have a very simple setup based on BBB to check if the series is working, so it will be good to test it on various platforms to ensure this is not generating any regression. Hence this version is still RFT. The series is based on asoc/for-next (v3.15-rc2). Test and comments are welcome. Thanks, Benoit v1: http://comments.gmane.org/gmane.linux.alsa.devel/120532 v2: http://comments.gmane.org/gmane.linux.alsa.devel/121024 - Split the first version in several patches. - Remove the TDM fixup v3: - Fixed all the places that were not iterating over all the codecs. - Change the structure and split it. Benoit Cousson (2): ASoC: core: Add one dai_get_widget helper instead of two rtd based ones ASoC: core: Add support for DAI multicodec Misael Lopez Cruz (1): ASoC: core: Add helpers for dai link and aux dev init include/sound/soc-dai.h | 5 + include/sound/soc.h | 13 ++ sound/soc/soc-compress.c | 83 +++++--- sound/soc/soc-core.c | 401 +++++++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 ++++--- sound/soc/soc-pcm.c | 529 +++++++++++++++++++++++++++++++---------------- 6 files changed, 763 insertions(+), 339 deletions(-) -- 1.8.3.2

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[alsa-devel] [PATCH] ALSA: compress: fix the struct alignment to 4 bytes
by Vinod Koul 23 Jun '14

23 Jun '14

In 64bit systems the compiler can default align to 8bytes causing mis-match with 32bit usermode. Avoid this is future by ensuring all the structures shared with usermode are packed and aligned to 4 bytes irrespective of arch used Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/compress_offload.h | 14 +++++++------- include/uapi/sound/compress_params.h | 14 +++++++------- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 21eed48..bd06dd0 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -39,7 +39,7 @@ struct snd_compressed_buffer { __u32 fragment_size; __u32 fragments; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_compr_params: compressed stream params @@ -51,7 +51,7 @@ struct snd_compr_params { struct snd_compressed_buffer buffer; struct snd_codec codec; __u8 no_wake_mode; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_compr_tstamp: timestamp descriptor @@ -70,7 +70,7 @@ struct snd_compr_tstamp { __u32 pcm_frames; __u32 pcm_io_frames; __u32 sampling_rate; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_compr_avail: avail descriptor @@ -80,7 +80,7 @@ struct snd_compr_tstamp { struct snd_compr_avail { __u64 avail; struct snd_compr_tstamp tstamp; -} __attribute__((packed)); +}__attribute__((packed, aligned(4))); enum snd_compr_direction { SND_COMPRESS_PLAYBACK = 0, @@ -107,7 +107,7 @@ struct snd_compr_caps { __u32 max_fragments; __u32 codecs[MAX_NUM_CODECS]; __u32 reserved[11]; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_compr_codec_caps: query capability of codec @@ -119,7 +119,7 @@ struct snd_compr_codec_caps { __u32 codec; __u32 num_descriptors; struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; -}; +}__attribute__((packed, aligned(4))); /** * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the @@ -140,7 +140,7 @@ enum { struct snd_compr_metadata { __u32 key; __u32 value[8]; -}; +}__attribute__((packed, aligned(4))); /** * compress path ioctl definitions diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index 165e705..54299ba 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -268,7 +268,7 @@ struct snd_enc_vorbis { __u32 max_bit_rate; __u32 min_bit_rate; __u32 downmix; -}; +}__attribute__((packed, aligned(4))); /** @@ -284,7 +284,7 @@ struct snd_enc_real { __u32 quant_bits; __u32 start_region; __u32 num_regions; -}; +}__attribute__((packed, aligned(4))); /** * struct snd_enc_flac @@ -308,12 +308,12 @@ struct snd_enc_real { struct snd_enc_flac { __u32 num; __u32 gain; -}; +}__attribute__((packed, aligned(4))); struct snd_enc_generic { __u32 bw; /* encoder bandwidth */ __s32 reserved[15]; -}; +}__attribute__((packed, aligned(4))); union snd_codec_options { struct snd_enc_wma wma; @@ -321,7 +321,7 @@ union snd_codec_options { struct snd_enc_real real; struct snd_enc_flac flac; struct snd_enc_generic generic; -}; +}__attribute__((packed, aligned(4))); /** struct snd_codec_desc - description of codec capabilities * @max_ch: Maximum number of audio channels @@ -358,7 +358,7 @@ struct snd_codec_desc { __u32 formats; __u32 min_buffer; __u32 reserved[15]; -}; +}__attribute__((packed, aligned(4))); /** struct snd_codec * @id: Identifies the supported audio encoder/decoder. @@ -399,6 +399,6 @@ struct snd_codec { __u32 align; union snd_codec_options options; __u32 reserved[3]; -}; +}__attribute__((packed, aligned(4))); #endif -- 1.7.0.4

2 1

[alsa-devel] [PATCH v2 00/11] ASoC: Add support for DAPM at the component level
by Lars-Peter Clausen 23 Jun '14

23 Jun '14

Changes since v1: * Rebased onto asoc/for-next and 3.16-rc1 * Added patch to remove platform field from widget and DAPM context to prevent new users from being added Original cover letter: Hi, This series is the next step in componentization. It moves the DAPM support to the component level. This will allow any component to have DAPM widgets and routes, which was previously only possible for CODECs, and will allow any component to have DAPM widgets with controls (i.e. Mixers and MUXs), which was previously only possible for CODECs. At the end of this patchset the DAPM code will have (almost) no dependency on snd_soc_codec or snd_soc_platform and the dependency on snd_soc_component is confined to 4 small helper functions. The only reason why there is still snd_soc_codec/snd_soc_platform dependent code is because there are still a lot of drivers which use widget->codec/platform and dapm->codec/platform. So those still need to be initialized. This will be cleaned up in a separate series and once that's done the codec and platform fields will be removed from the widget and DAPM context struct. One of the major problems of moving DAPM to the component level is that the DAPM context is directly embedded in the snd_soc_codec and snd_soc_platform structs and there are many drivers directly accessing that embedded field. This means it is not possible to move the field without causing a bit of code churn in the drivers. The patch series solves this by adding two fields to the component struct. One pointer to a DAPM context, which points to the DAPM context that is used by the component. And one DAPM context, which is used as the default if no other DAPM context was provided. Neither of the fields is accessed directly, instead a inline helper function is added that returns the DAPM context for a component. This will make it possible to eventually seamlessly switch over once all direct references to CODEC and platform DAPM contexts have been eliminated. - Lars Lars-Peter Clausen (11): ASoC: Move name_prefix from CODEC to component ASoC: Move name and id from CODEC/platform to component ASoC: Split component registration into two steps ASoC: Auto disconnect pins from all DAPM contexts ASoC: Add a set_bias_level() callback to the DAPM context struct ASoC: Add DAPM support at the component level ASoC: Use component DAPM context for platforms ASoC: Add component level stream_event() and seq_notifier() support ASoC: Add component level set_bias_level() support ASoC: dapm: Remove DAI DAPM context ASoC: dapm: Remove platform field from widget and dapm context struct include/sound/soc-dai.h | 1 - include/sound/soc-dapm.h | 8 +- include/sound/soc.h | 53 +++++-- include/trace/events/asoc.h | 6 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/soc-cache.c | 7 +- sound/soc/soc-compress.c | 8 +- sound/soc/soc-core.c | 330 +++++++++++++++++++++++------------------ sound/soc/soc-dapm.c | 183 +++++++++++++---------- sound/soc/soc-pcm.c | 4 +- 10 files changed, 344 insertions(+), 258 deletions(-) -- 1.8.0

3 27

[alsa-devel] [RESEND PATCH v2 1/2] ASoC: samsung: s3c24{xx, 12}-i2s: port to use generic dmaengine API
by Vasily Khoruzhick 23 Jun '14

23 Jun '14

Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412 Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v2: use hardcoded dma channel number sound/soc/samsung/Kconfig | 9 +++---- sound/soc/samsung/dmaengine.c | 3 +++ sound/soc/samsung/s3c-i2s-v2.c | 17 +------------ sound/soc/samsung/s3c2412-i2s.c | 47 +++++++++++++++++----------------- sound/soc/samsung/s3c24xx-i2s.c | 56 +++++++++++++++++++---------------------- 5 files changed, 56 insertions(+), 76 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3507574..022c7a4 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,11 +1,10 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX + select S3C24XX_DMAC if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA if !ARCH_S3C24XX - select SND_S3C_DMA_LEGACY if ARCH_S3C24XX - select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX + select SND_S3C_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to @@ -19,7 +18,6 @@ config SND_S3C_DMA_LEGACY config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +25,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index 750ce58..72f27d1 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/amba/pl08x.h> +#include <linux/platform_data/dma-s3c24xx.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,8 @@ #ifdef CONFIG_ARCH_S3C64XX #define filter_fn pl08x_filter_id +#elif defined(CONFIG_ARCH_S3C24XX) +#define filter_fn s3c24xx_dma_filter #else #define filter_fn NULL #endif diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 79e7efb..1a74051 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -392,8 +392,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -424,13 +422,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, local_irq_restore(irqs); - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; case SNDRV_PCM_TRIGGER_STOP: @@ -644,12 +635,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, /* record our i2s structure for later use in the callbacks */ snd_soc_dai_set_drvdata(dai, i2s); - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->iis_pclk = clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); @@ -729,7 +714,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id, struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) { - struct snd_soc_dai_ops *ops = dai_drv->ops; + struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops; ops->trigger = s3c2412_i2s_trigger; if (!ops->hw_params) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d079445..ba084e3 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,25 +33,19 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .ch_name = "tx", .dma_size = 4, }; static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .ch_name = "rx", .dma_size = 4, }; @@ -63,6 +57,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) pr_debug("Entered %s\n", __func__); + samsung_asoc_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, + &s3c2412_i2s_pcm_stereo_in); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); if (ret) return ret; @@ -70,10 +67,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); + s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); return PTR_ERR(s3c2412_i2s.iis_cclk); } @@ -94,8 +90,6 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { clk_disable(s3c2412_i2s.iis_cclk); - clk_put(s3c2412_i2s.iis_cclk); - iounmap(s3c2412_i2s.regs); return 0; } @@ -105,18 +99,10 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); - struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); @@ -169,6 +155,19 @@ static const struct snd_soc_component_driver s3c2412_i2s_component = { static int s3c2412_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c2412_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c2412_i2s.regs == NULL) + return -ENXIO; + + s3c2412_i2s_pcm_stereo_out.dma_addr = res->start + S3C2412_IISTXD; + s3c2412_i2s_pcm_stereo_in.dma_addr = res->start + S3C2412_IISRXD; ret = s3c_i2sv2_register_component(&pdev->dev, -1, &s3c2412_i2s_component, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916..97600df 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,25 +31,19 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "tx", .dma_size = 2, }; static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "rx", .dma_size = 2, }; @@ -231,18 +225,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -251,11 +239,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; + dma_data->addr_width = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; + dma_data->addr_width = 2; break; default: return -EINVAL; @@ -270,8 +258,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(dai, substream); pr_debug("Entered %s\n", __func__); @@ -290,7 +276,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -380,14 +365,12 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) { pr_debug("Entered %s\n", __func__); - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; + samsung_asoc_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out, + &s3c24xx_i2s_pcm_stereo_in); - s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); + s3c24xx_i2s.iis_clk = devm_clk_get(dai->dev, "iis"); if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); @@ -474,6 +457,19 @@ static const struct snd_soc_component_driver s3c24xx_i2s_component = { static int s3c24xx_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO; + s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO; ret = snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1); -- 1.9.3

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[alsa-devel] [PATCH 1/2] SOUND: TLV320AIC32x4: Fixed Coding style Issues
by Shahina Shaik 22 Jun '14

22 Jun '14

From: Shahina Shaik <sharab.shaik(a)gmail.com> Fixed a brace coding style issue in the tlv320aic32x4.c Signed-off-by: Shahina Shaik <sharab.shaik(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117..1e50f15 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -626,21 +626,18 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } - if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); - } tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? AIC32X4_LDOCTLEN : 0; snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) tmp_reg |= AIC32X4_LDOIN_18_36; - } - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) tmp_reg |= AIC32X4_LDOIN2HP; - } snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); /* Mic PGA routing */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: fsl: refine DMA/FIQ dependencies
by Arnd Bergmann 22 Jun '14

22 Jun '14

Commit 31ee2bfd724ab ("ASoC: fsl: select SND_SOC_IMX_PCM_DMA where needed") started selecting SND_SOC_IMX_PCM_DMA and SND_SOC_IMX_PCM_FIQ for two drivers when building for i.MX. This has turned out too aggressive, as FIQ is only available for i.mx2 through i.mx5, but not i.mx6 or vybrid. Further, two more drivers have become user-selectable in the meantime, and they both depend on DMA for the imx platform as well. This changes the selection of FIQ to depend on the TZIC or AVIC interrupt controllers that actually export the imx specific FIQ interfaces, and adds the missing select statements for SAI and ESAI. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3793362..52bbe9f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -5,6 +5,7 @@ comment "Common SoC Audio options for Freescale CPUs:" config SND_SOC_FSL_SAI tristate "Synchronous Audio Interface (SAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to add Synchronous Audio Interface (SAI) @@ -15,7 +16,7 @@ config SND_SOC_FSL_SAI config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO help Say Y if you want to add Synchronous Serial Interface (SSI) @@ -27,7 +28,7 @@ config SND_SOC_FSL_SPDIF tristate "Sony/Philips Digital Interface module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) help Say Y if you want to add Sony/Philips Digital Interface (SPDIF) support for the Freescale CPUs. @@ -37,6 +38,7 @@ config SND_SOC_FSL_SPDIF config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface

2 1

[alsa-devel] [PATCH v5 0/3] Add ES8328 audio codec
by Sean Cross 22 Jun '14

22 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks. This requires support for DT clock reparenting in order to get the codec its 22.5792 MHz supply frequency. Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 24 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 63 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 694 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 313 ++++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 263 ++++++++ 12 files changed, 1502 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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Re: [alsa-devel] [PATCH] ASoC: Fix SOC_DOUBLE_R_SX_TLV volume mixer arguments
by Brian Austin 21 Jun '14

21 Jun '14

On Fri, 20 Jun 2014, Ryan Harvey wrote: > Remove unnecessary bit shifts. Correct min value to match datasheet. #Steps equals number of steps between min and max. > > This patch depends on 8f0367b53d887726b1f4b0b39e0f09b7cca2f79f > ASoC: cs42l56: Fix stereo channel register assignment for Headphone and > LineOut volume mixers. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- Acked-by: Brian Austin <brian.austin(a)cirrus.com> Thanks,

3 2

[alsa-devel] [PATCH] ASoC: wm5110: Power both channels for differential mono output
by Charles Keepax 21 Jun '14

21 Jun '14

On the wm5110 CODEC both the left and right channel must be powered when an output is being used as a mono output, although no audio is routed to the right output channel. This patch adds additional DAPM routes to link the right channel to the left in the case where an output is marked as mono. Audio must always be brought in on the left channel for mono operation. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 1 + 3 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 41b56ee..d441d80 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +static const struct snd_soc_dapm_route arizona_mono_routes[] = { + { "OUT1R", NULL, "OUT1L" }, + { "OUT2R", NULL, "OUT2L" }, + { "OUT3R", NULL, "OUT3L" }, + { "OUT4R", NULL, "OUT4L" }, + { "OUT5R", NULL, "OUT5L" }, + { "OUT6R", NULL, "OUT6L" }, +}; + +int arizona_init_mono(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) { + if (arizona->pdata.out_mono[i]) + snd_soc_dapm_add_routes(&codec->dapm, + &arizona_mono_routes[i], 1); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_mono); + int arizona_init_gpio(struct snd_soc_codec *codec) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 05ae17f..942cfb1 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); +extern int arizona_init_mono(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e5fcb5..62ef544 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1596,6 +1596,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + arizona_init_mono(codec); ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5677: Replace the string "micbias1" to "MICBIAS1"
by Oder Chiou 21 Jun '14

21 Jun '14

The patch replaces the string "micbias1" to "MICBIAS1". Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 833231e..75268a7 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1685,7 +1685,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* Input Side */ /* micbias */ - SND_SOC_DAPM_SUPPLY("micbias1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, 0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU), /* Input Lines */ -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH 1/3] ASoC: rt5640: Remove unneeded goto in rt5640_i2c_probe
by Axel Lin 21 Jun '14

21 Jun '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5640.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index de80e89..6bc6efd 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2215,14 +2215,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, rt5640->hp_mute = 1; - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, - rt5640_dai, ARRAY_SIZE(rt5640_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); } static int rt5640_i2c_remove(struct i2c_client *i2c) -- 1.8.3.2

2 5

[alsa-devel] [PATCH] ASoC: pcm: fix dpcm_path_put in dpcm runtime update
by Qiao Zhou 21 Jun '14

21 Jun '14

we need to release dapm widget list after dpcm_path_get in soc_dpcm_runtime_update. otherwise, there will be potential memory leak. add dpcm_path_put to fix it. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a391de0..3bb8b80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2072,6 +2072,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); } + dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ if (!fe->cpu_dai->driver->capture.channels_min) -- 1.7.9.5

2 2

[alsa-devel] [PATCH 0/2] Couple of fixes to davinci-mcasp driver
by Jyri Sarha 21 Jun '14

21 Jun '14

These changes target to sii9022 HDMI audio support, but are independent fixes. Jyri Sarha (2): ASoC: davinci-mcasp: Add dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC ASoC: davinci-mcasp: Allow best effort in selecting BCLK divider sound/soc/davinci/Kconfig | 1 + sound/soc/davinci/davinci-mcasp.c | 22 +++++++++++++++++----- 2 files changed, 18 insertions(+), 5 deletions(-) -- 1.7.9.5

2 4

Re: [alsa-devel] [PATCH] ASoC: cs42l56: Fix stereo channel register assignment for Headphone and LineOut volume mixers
by Brian Austin 20 Jun '14

20 Jun '14

On Fri, 20 Jun 2014, Ryan Harvey wrote: > Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- Acked-by: Brian Austin <brian.austin(a)cirrus.com>

1 0

[alsa-devel] [PATCH Resend 1/1] ASoC: mc13783: Add missing of_node_put
by Sachin Kamat 20 Jun '14

20 Jun '14

of_get_child_by_name expects of_node_put be called when done. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/codecs/mc13783.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } -- 1.7.9.5

1 0

[alsa-devel] [PATCH 1/2] ARM: dts: imx6: remove wrong spdif rxtx2 clock
by Anssi Hannula 20 Jun '14

20 Jun '14

The rxtx2 clock of i.MX6 S/PDIF is currently set to "asrc" clock. However, according to SoC documentation, rxtx2 is connected to ASRC_EXT_CLK, a different external clock. Testing on Cubox-i system seems to confirm that: when fsl_spdif is forced to select rxtx2 as input clock, audio playback fails with an I/O error. Set rxtx2 to the dummy clock by default to prevent fsl_spdif from selecting it. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Cc: Mark Brown <broonie(a)kernel.org> --- This one seems clear enough. arch/arm/boot/dts/imx6qdl.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/boot/dts/imx6qdl.dtsi b/arch/arm/boot/dts/imx6qdl.dtsi index ce0599134a69..3aef9a2f5fda 100644 --- a/arch/arm/boot/dts/imx6qdl.dtsi +++ b/arch/arm/boot/dts/imx6qdl.dtsi @@ -181,7 +181,7 @@ <&sdma 15 18 0>; dma-names = "rx", "tx"; clocks = <&clks 197>, <&clks 3>, - <&clks 197>, <&clks 107>, + <&clks 197>, <&clks 0>, <&clks 0>, <&clks 118>, <&clks 0>, <&clks 139>, <&clks 0>; -- 1.8.4.5

3 4

[alsa-devel] [PATCH] *** ASoC: core: refine of node id parse of cpu/codec dai ***
by Qiao Zhou 20 Jun '14

20 Jun '14

Hi Mark, Liam This patch is to add another check besides cpu_dai/codec_dai name during dai_link bind. currently if the cpu_dai/codec_dai name match corresponding dai_link cpu/codec name, then a match is found. in this patch, it also checks whether cpu/codec dai id match dailink cpu_dai_id/codec_dai_id. Still check name first. 1. if it doesn't match, it will keep checking whether cpu_id/codec_id match corresponding dai_link cpu_dai_id/codec_dai_id. if the ids are equal, then a match is found. 2. if it does match, then a match is already found. no need to further check. By this way, it's convenient to maintain dai_link in DT files, and machine driver can parse related node point to get dai id info. Then there is no need to parse cpu_dai/codec_dai name for each dai. Like below: In DT files: sound { compatible = "sound card name xx"; dai_link_1:dai_link_1 { compatible = "xx-dailink-1"; cpu-dai = <&cpu-dai-node CPU_DAI_ID_1>; codec-dai = <&codec-dai-node CODEC_DAI_ID_1>; ...... } dai_link_2:dai_link_2 { compatible = "xx-dailink-2"; cpu-dai = <&cpu-dai-node CPU_DAI_ID_2>; codec-dai = <&codec-dai-node CODEC_DAI_ID_2>; ...... } ...... } In machine driver: static xx_sound_card_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; i = 0; for_each_child_of_node(np, dailink_np) { of_parse_phandle_with_args(dailink_np, "cpu-dai", "#dailink-cells", 0, &out_args); dailink[i].cpu_of_node = out_args.np; dailink[i].cpu_dai_id = out_args.args[0]; dailink[i].platform_of_node = out_args.np; of_parse_phandle_with_args(dailink_np, "codec-dai", "#dailink-cells", 0, &out_args); dailink[i].codec_of_node = out_args.np; dailink[i].codec_dai_id = out_args.args[0]; ...... i++ } ...... ret = snd_soc_register_card(card); ...... } If some drivers don't have DT support, then we have no choice but to add correct cpu/codec(dai) name in DT and parse name info in machine driver. Could you help share your opinions? Thanks in advance. Qiao Zhou (1): ASoC: core: refine of node id parse of cpu/codec dai include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 32 +++++++++++++++++++++++--------- 2 files changed, 25 insertions(+), 9 deletions(-) -- 1.7.9.5

2 5

[alsa-devel] [PATCH] ALSA: hda - Fix usage of "model" module parameter
by David Henningsson 19 Jun '14

19 Jun '14

A recent refactoring broke the possibility to manually specify model name as a module parameter. This patch restores the desired functionality. Reported-by: Kent Baxley <kent.baxley(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index b684c6e..dabe419 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -898,6 +898,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (!strcmp(codec->modelname, models->name)) { codec->fixup_id = models->id; codec->fixup_name = models->name; + codec->fixup_list = fixlist; codec->fixup_forced = 1; return; } -- 1.9.1

1 0

[alsa-devel] [PATCH v4 0/3] Add ES8328 audio codec
by Sean Cross 19 Jun '14

19 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the codec itself and speaker amps, have headphone support, and must supply an unusual frequency to the codec chip. Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 18 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 68 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 62 ++ sound/soc/codecs/es8328-spi.c | 51 ++ sound/soc/codecs/es8328.c | 625 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 313 +++++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 359 ++++++++++++ 12 files changed, 1532 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

3 11

[alsa-devel] [PATCH] ASoC: max98090: Remove needless defines and line feeds
by Jarkko Nikula 19 Jun '14

19 Jun '14

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/codecs/max98090.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index d97f1ce7ff7d..6ea0fdd42eb5 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -26,10 +26,6 @@ #include <sound/max98090.h> #include "max98090.h" -#define DEBUG -#define EXTMIC_METHOD -#define EXTMIC_METHOD_TEST - /* Allows for sparsely populated register maps */ static struct reg_default max98090_reg[] = { { 0x00, 0x00 }, /* 00 Software Reset */ @@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, else val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; - if (val >= 1) { if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { max98090->pa1en = val - 1; /* Update for volatile */ @@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("DMICL"), @@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("DMIC3"), SND_SOC_DAPM_INPUT("DMIC4"), @@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { - {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, @@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"SPKR", NULL, "SPK Right Out"}, {"RCVL", NULL, "RCV Left Out"}, {"RCVR", NULL, "RCV Right Out"}, - }; static const struct snd_soc_dapm_route max98091_dapm_routes[] = { - /* DMIC inputs */ {"DMIC3", NULL, "DMIC3_ENA"}, {"DMIC4", NULL, "DMIC4_ENA"}, {"DMIC3", NULL, "AHPF"}, {"DMIC4", NULL, "AHPF"}, - }; static int max98090_add_widgets(struct snd_soc_codec *codec) @@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, ARRAY_SIZE(max98091_dapm_routes)); - } return 0; @@ -2221,7 +2209,6 @@ static void max98090_handle_pdata(struct snd_soc_codec *codec) dev_err(codec->dev, "No platform data\n"); return; } - } static int max98090_probe(struct snd_soc_codec *codec) -- 2.0.0

2 1

[alsa-devel] [PATCH] ASoC: max98090: Fix missing free_irq
by Jarkko Nikula 19 Jun '14

19 Jun '14

max98090.c doesn't free the threaded interrupt it requests. This causes an oops when doing "cat /proc/interrupts" after snd-soc-max98090.ko is unloaded. Fix this by requesting the interrupt by using devm_request_threaded_irq(). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Stable <stable(a)vger.kernel.org> # 3.10+ --- This issue has been in max98090.c since v3.8 commit 685e42154dcf ("ASoC: Replace max98090 Device Driver"). However this patch applies only after v3.10 which has 3d15aacbb802 ("ASoC: max98090: request IRQF_ONESHOT interrupt"). I'm not sure is there need to go before that since first upstream max98090 user was introduced in a v3.14 commit 7637af2e17f1 ("ASoC: tegra: add tegra+MAX98090 machine driver"). However I'm aware there is an out-of-tree 3.10 user. --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7a8e89..d97f1ce7ff7d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2284,7 +2284,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Register for interrupts */ dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - ret = request_threaded_irq(max98090->irq, NULL, + ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { -- 2.0.0

2 1

[alsa-devel] [PATCH] ASoC: wm5100/wm8903/wm8996: Replace open-coded snd_soc_dapm_to_codec()
by Lars-Peter Clausen 19 Jun '14

19 Jun '14

We now have a generic helper function to cast from a DAPM context to a CODEC. Make use of it in the places which previously open-coded it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm5100.c | 3 +-- sound/soc/codecs/wm8903.c | 3 +-- sound/soc/codecs/wm8996.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 91a9ea2..7bb0d36 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -735,8 +735,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); u16 val, expect, i; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b84940c..ec3250d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -281,8 +281,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; int i, val; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 6926633..9304a91 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -690,8 +690,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); u16 val, mask; -- 1.8.0

3 2

[alsa-devel] [PATCH 1/2] tinycompress: update compress headers for sample rate field
by Vinod Koul 19 Jun '14

19 Jun '14

commit "929559b: ALSA: compress: add num_sample_rates in snd_codec_desc" adds array for passing sample rate instead of bit map. Update the local header as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/compress_params.h | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h index deda425..378417a 100644 --- a/include/sound/compress_params.h +++ b/include/sound/compress_params.h @@ -23,6 +23,7 @@ #define MAX_NUM_CODECS 32 #define MAX_NUM_CODEC_DESCRIPTORS 32 #define MAX_NUM_BITRATES 32 +#define MAX_NUM_SAMPLE_RATES 32 #define SND_AUDIOCODEC_PCM ((__u32) 0x00000001) #define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002) @@ -209,7 +210,8 @@ union snd_codec_options { struct snd_codec_desc { __u32 max_ch; - __u32 sample_rates; + __u32 sample_rates[MAX_NUM_SAMPLE_RATES]; + __u32 num_sample_rates; __u32 bit_rate[MAX_NUM_BITRATES]; __u32 num_bitrates; __u32 rate_control; -- 1.7.0.4

1 1

[alsa-devel] Enable HP machine to enable capture mute led and output mute led
by Kailang 19 Jun '14

19 Jun '14

Hi Takashi, Attach was HP mute led support patch. It use Line1 vref and Mic1 vref to control mute led for ALC282. Line1 vref to control output mute led. Mic1 vref to control input mute led. It use GPIO3 and Mic1 vref to control mute led for ALC280. GPIO3 to control output mute led. Mic1 vref to control input mute led. For the DMI string for output mute led application. I already request from HP to support it in next release. HP_Mute_LED_0_A => use Mic1 vref to control mute led. Is the string right? I have tested upper string for Mic1 mute led. BR, Kailang

2 2

Re: [alsa-devel] [PATCH v11 2/8] ASoC: core: Generic ac97 link reset functions
by Markus Pargmann 19 Jun '14

19 Jun '14

Hi Silvio, On Wed, Jun 18, 2014 at 08:42:34PM +0200, S. Fricke wrote: > > Hi Markus, > > I'm fighting with your ac97 pinctrl configuration per dts. > > > [...] > > +ssi { > > + ... > > + > > + pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset"; > > + pinctrl-0 = <&ac97link_running>; > > + pinctrl-1 = <&ac97link_running>; > > + pinctrl-2 = <&ac97link_reset>; > > + pinctrl-3 = <&ac97link_warm_reset>; > > + ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>; > > I see here the gpios for sync, sdata and reset. On my board I have two sdata > lines (sdata-in and sdata-out) which line should I specify here for this setting? Try to use sdata-out. This is the only data output line. sdata-in is the input to the SoC so we don't have to use this for an AC97 reset. Regards, Markus -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

1 0

Re: [alsa-devel] [RFC PATCH] ASoC: tas2552: Support TI TAS2552 Amplifier
by Mark Brown 18 Jun '14

18 Jun '14

On Wed, Jun 18, 2014 at 12:45:46PM -0500, Dan Murphy wrote: > +static int tas2552_i2c_read(struct tas2552_data *tas_data, int reg) > +{ > + int val; > + > + if (WARN_ON(!tas_data->tas2552_client)) > + return -EINVAL; > + > + /* If powered off, return the cached value */ > + mutex_lock(&tas_data->mutex); > + if (tas_data->power_state) { > + val = i2c_smbus_read_byte_data(tas_data->tas2552_client, reg); Don't open code your I/O, use regmap like any other CODEC driver. This will remove a very large proportion of the code in the driver which replicates features of the regmap framework. > +static int tas2552_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); > + > + tas2552_sw_shutdown(tas2552, 1); > + tas2552_power(tas2552, 1); > + /* Turn on Class D amplifier */ > + tas2552_i2c_write(tas2552, TAS2552_CFG_2, 0x80); > + tas2552_sw_shutdown(tas2552, 0); I'd expect this to be done using DAPM. > +static const struct regmap_config tas2552_regmap_config = { > + .reg_bits = 8, > + .val_bits = 8, > + > + .max_register = TAS2552_MAX_REG, > + .reg_defaults = NULL, > + .num_reg_defaults = 0, No need to set things to zero or NULL in static variables, that's the default. > + dev = &client->dev; > + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); > + if (data == NULL) { > + dev_err(dev, "Can not allocate memory\n"); No need to print an error, OOM messages are already quite noisy.

1 0

[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: set up channel status bits for S/PDIF mode
by Daniel Mack 18 Jun '14

18 Jun '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- v3 just changes variable names. sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4500cf0..a21a7d4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -576,8 +577,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 cs_value = 0; + u8 *cs_bytes = (u8*) &cs_value; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -599,6 +604,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; + cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; + + switch (rate) { + case 22050: + cs_bytes[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + cs_bytes[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + cs_bytes[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + cs_bytes[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + cs_bytes[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + cs_bytes[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + cs_bytes[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + cs_bytes[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + cs_bytes[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + return 0; } @@ -634,7 +679,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

4 4

[alsa-devel] [PATCH 1/2] ASoC: ak5386: Add regulators to documentation and fix sparse warning
by Daniel Mack 18 Jun '14

18 Jun '14

Document the newly added regulators to the DT binding document. Also, "static const char const *x" is not identical to "static const char * const x", which sparse now complains about. Fix it. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/ak5386.txt | 4 ++++ sound/soc/codecs/ak5386.c | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt index dc3914f..ec3df3a 100644 --- a/Documentation/devicetree/bindings/sound/ak5386.txt +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -10,10 +10,14 @@ Optional properties: - reset-gpio : a GPIO spec for the reset/power down pin. If specified, it will be deasserted at probe time. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V Example: spdif: ak5386@0 { compatible = "asahi-kasei,ak5386"; reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; }; diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index a30be5c..8107a1c 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -19,7 +19,7 @@ #include <sound/pcm.h> #include <sound/initval.h> -static const char const *supply_names[] = { +static const char * const supply_names[] = { "va", "vd" }; -- 1.8.5.3

3 5

Re: [alsa-devel] [v3.13.y][SRU][PATCH 0/1] ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while DEBUG not defined
by Greg KH 18 Jun '14

18 Jun '14

On Wed, Jun 18, 2014 at 01:14:05PM -0400, Joseph Salisbury wrote: > BugLink: http://bugs.launchpad.net/bugs/1319457 > > Hello, > > Please consider including upstream commit b7a77235 in the next v3.13.y release. It was included upstream as of v3.15-rc5. It has been tested and confirmed to resolve http://bugs.launchpad.net/bugs/1319457 . Please get a good email client and wrap your lines :( > commit b7a7723513dc89f83d6df13206df55d4dc26e825 > Author: Sander Eikelenboom <linux(a)eikelenboom.it> > Date: Fri May 2 15:09:27 2014 +0200 > > ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while DEBUG not defined > > > This commit does not apply cleanly to v3.13, so I performed a backport, which is in email 1/1. Any reason this shouldn't also be in 3.14-stable? thanks, greg k-h

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Remove the obsoleted static quirk codes from patch_conexant.c
by Takashi Iwai 18 Jun '14

18 Jun '14

The static quirk code has been disabled for a while and it seems working fine, so it's time to actually get rid of it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 2631 +--------------------------------------- 1 file changed, 9 insertions(+), 2622 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1dc7e974f3b1..7627a69ca6d7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -34,27 +34,6 @@ #include "hda_jack.h" #include "hda_generic.h" -#undef ENABLE_CXT_STATIC_QUIRKS - -#define CXT_PIN_DIR_IN 0x00 -#define CXT_PIN_DIR_OUT 0x01 -#define CXT_PIN_DIR_INOUT 0x02 -#define CXT_PIN_DIR_IN_NOMICBIAS 0x03 -#define CXT_PIN_DIR_INOUT_NOMICBIAS 0x04 - -#define CONEXANT_HP_EVENT 0x37 -#define CONEXANT_MIC_EVENT 0x38 -#define CONEXANT_LINE_EVENT 0x39 - -/* Conexant 5051 specific */ - -#define CXT5051_SPDIF_OUT 0x12 -#define CXT5051_PORTB_EVENT 0x38 -#define CXT5051_PORTC_EVENT 0x39 - -#define AUTO_MIC_PORTB (1 << 1) -#define AUTO_MIC_PORTC (1 << 2) - struct conexant_spec { struct hda_gen_spec gen; @@ -72,64 +51,6 @@ struct conexant_spec { bool dc_enable; unsigned int dc_input_bias; /* offset into olpc_xo_dc_bias */ struct nid_path *dc_mode_path; - -#ifdef ENABLE_CXT_STATIC_QUIRKS - const struct snd_kcontrol_new *mixers[5]; - int num_mixers; - hda_nid_t vmaster_nid; - - const struct hda_verb *init_verbs[5]; /* initialization verbs - * don't forget NULL - * termination! - */ - unsigned int num_init_verbs; - - /* playback */ - struct hda_multi_out multiout; /* playback set-up - * max_channels, dacs must be set - * dig_out_nid and hp_nid are optional - */ - unsigned int cur_eapd; - unsigned int hp_present; - unsigned int line_present; - unsigned int auto_mic; - - /* capture */ - unsigned int num_adc_nids; - const hda_nid_t *adc_nids; - hda_nid_t dig_in_nid; /* digital-in NID; optional */ - - unsigned int cur_adc_idx; - hda_nid_t cur_adc; - unsigned int cur_adc_stream_tag; - unsigned int cur_adc_format; - - const struct hda_pcm_stream *capture_stream; - - /* capture source */ - const struct hda_input_mux *input_mux; - const hda_nid_t *capsrc_nids; - unsigned int cur_mux[3]; - - /* channel model */ - const struct hda_channel_mode *channel_mode; - int num_channel_mode; - - /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in build_pcms() */ - - unsigned int spdif_route; - - unsigned int port_d_mode; - unsigned int dell_automute:1; - unsigned int dell_vostro:1; - unsigned int ideapad:1; - unsigned int thinkpad:1; - unsigned int hp_laptop:1; - unsigned int asus:1; - - unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ -#endif /* ENABLE_CXT_STATIC_QUIRKS */ }; @@ -173,2533 +94,6 @@ static int add_beep_ctls(struct hda_codec *codec) #define add_beep_ctls(codec) 0 #endif - -#ifdef ENABLE_CXT_STATIC_QUIRKS -static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int conexant_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, - format, substream); -} - -static int conexant_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - -/* - * Digital out - */ -static int conexant_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int conexant_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int conexant_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, - stream_tag, - format, substream); -} - -/* - * Analog capture - */ -static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); - return 0; -} - -static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); - return 0; -} - - - -static const struct hda_pcm_stream conexant_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .open = conexant_playback_pcm_open, - .prepare = conexant_playback_pcm_prepare, - .cleanup = conexant_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream conexant_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .prepare = conexant_capture_pcm_prepare, - .cleanup = conexant_capture_pcm_cleanup - }, -}; - - -static const struct hda_pcm_stream conexant_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .open = conexant_dig_playback_pcm_open, - .close = conexant_dig_playback_pcm_close, - .prepare = conexant_dig_playback_pcm_prepare - }, -}; - -static const struct hda_pcm_stream conexant_pcm_digital_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in alc_build_pcms */ -}; - -static int cx5051_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - spec->cur_adc = spec->adc_nids[spec->cur_adc_idx]; - spec->cur_adc_stream_tag = stream_tag; - spec->cur_adc_format = format; - snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); - return 0; -} - -static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->cur_adc); - spec->cur_adc = 0; - return 0; -} - -static const struct hda_pcm_stream cx5051_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .prepare = cx5051_capture_pcm_prepare, - .cleanup = cx5051_capture_pcm_cleanup - }, -}; - -static int conexant_build_pcms(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "CONEXANT Analog"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = conexant_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = - spec->multiout.max_channels; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dac_nids[0]; - if (spec->capture_stream) - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->capture_stream; - else { - if (codec->vendor_id == 0x14f15051) - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - cx5051_pcm_analog_capture; - else { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - conexant_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = - spec->num_adc_nids; - } - } - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - - if (spec->multiout.dig_out_nid) { - info++; - codec->num_pcms++; - info->name = "Conexant Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = - conexant_pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dig_out_nid; - if (spec->dig_in_nid) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - conexant_pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = - spec->dig_in_nid; - } - } - - return 0; -} - -static int conexant_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - - return snd_hda_input_mux_info(spec->input_mux, uinfo); -} - -static int conexant_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; - return 0; -} - -static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->capsrc_nids[adc_idx], - &spec->cur_mux[adc_idx]); -} - -static void conexant_set_power(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) -{ - if (power_state == AC_PWRST_D3) - msleep(100); - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - power_state); - /* partial workaround for "azx_get_response timeout" */ - if (power_state == AC_PWRST_D0) - msleep(10); - snd_hda_codec_set_power_to_all(codec, fg, power_state); -} - -static int conexant_init(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_init_verbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); - return 0; -} - -static void conexant_free(struct hda_codec *codec) -{ - kfree(codec->spec); -} - -static const struct snd_kcontrol_new cxt_capture_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, - {} -}; - -static const char * const slave_pfxs[] = { - "Headphone", "Speaker", "Bass Speaker", "Front", "Surround", "CLFE", - NULL -}; - -static int conexant_build_controls(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int i; - int err; - - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - if (spec->multiout.dig_out_nid) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; - } - if (spec->dig_in_nid) { - err = snd_hda_create_spdif_in_ctls(codec,spec->dig_in_nid); - if (err < 0) - return err; - } - - /* if we have no master control, let's create it */ - if (spec->vmaster_nid && - !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - unsigned int vmaster_tlv[4]; - snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_pfxs, - "Playback Volume"); - if (err < 0) - return err; - } - if (spec->vmaster_nid && - !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch"); - if (err < 0) - return err; - } - - if (spec->input_mux) { - err = snd_hda_add_new_ctls(codec, cxt_capture_mixers); - if (err < 0) - return err; - } - - err = add_beep_ctls(codec); - if (err < 0) - return err; - - return 0; -} - -static const struct hda_codec_ops conexant_patch_ops = { - .build_controls = conexant_build_controls, - .build_pcms = conexant_build_pcms, - .init = conexant_init, - .free = conexant_free, - .set_power_state = conexant_set_power, -}; - -static int patch_conexant_auto(struct hda_codec *codec); -/* - * EAPD control - * the private value = nid | (invert << 8) - */ - -#define cxt_eapd_info snd_ctl_boolean_mono_info - -static int cxt_eapd_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int invert = (kcontrol->private_value >> 8) & 1; - if (invert) - ucontrol->value.integer.value[0] = !spec->cur_eapd; - else - ucontrol->value.integer.value[0] = spec->cur_eapd; - return 0; - -} - -static int cxt_eapd_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int invert = (kcontrol->private_value >> 8) & 1; - hda_nid_t nid = kcontrol->private_value & 0xff; - unsigned int eapd; - - eapd = !!ucontrol->value.integer.value[0]; - if (invert) - eapd = !eapd; - if (eapd == spec->cur_eapd) - return 0; - - spec->cur_eapd = eapd; - snd_hda_codec_write_cache(codec, nid, - 0, AC_VERB_SET_EAPD_BTLENABLE, - eapd ? 0x02 : 0x00); - return 1; -} - -/* controls for test mode */ -#ifdef CONFIG_SND_DEBUG - -#define CXT_EAPD_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = cxt_eapd_info, \ - .get = cxt_eapd_get, \ - .put = cxt_eapd_put, \ - .private_value = nid | (mask<<16) } - - - -static int conexant_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - -static int conexant_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - spec->multiout.max_channels); -} - -static int conexant_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); - return err; -} - -#define CXT_PIN_MODE(xname, nid, dir) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = conexant_ch_mode_info, \ - .get = conexant_ch_mode_get, \ - .put = conexant_ch_mode_put, \ - .private_value = nid | (dir<<16) } - -#endif /* CONFIG_SND_DEBUG */ - -/* Conexant 5045 specific */ - -static const hda_nid_t cxt5045_dac_nids[1] = { 0x19 }; -static const hda_nid_t cxt5045_adc_nids[1] = { 0x1a }; -static const hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a }; -#define CXT5045_SPDIF_OUT 0x18 - -static const struct hda_channel_mode cxt5045_modes[1] = { - { 2, NULL }, -}; - -static const struct hda_input_mux cxt5045_capture_source = { - .num_items = 2, - .items = { - { "Internal Mic", 0x1 }, - { "Mic", 0x2 }, - } -}; - -static const struct hda_input_mux cxt5045_capture_source_benq = { - .num_items = 4, - .items = { - { "Internal Mic", 0x1 }, - { "Mic", 0x2 }, - { "Line", 0x3 }, - { "Mixer", 0x0 }, - } -}; - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - /* toggle internal speakers mute depending of presence of - * the headphone jack - */ - bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - - bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x11, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - return 1; -} - -/* bind volumes of both NID 0x10 and 0x11 */ -static const struct hda_bind_ctls cxt5045_hp_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5045_hp_automic(struct hda_codec *codec) -{ - static const struct hda_verb mic_jack_on[] = { - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - static const struct hda_verb mic_jack_off[] = { - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x12); - if (present) - snd_hda_sequence_write(codec, mic_jack_on); - else - snd_hda_sequence_write(codec, mic_jack_off); -} - - -/* mute internal speaker if HP is plugged */ -static void cxt5045_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_jack_detect(codec, 0x11); - - bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5045_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case CONEXANT_HP_EVENT: - cxt5045_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5045_hp_automic(codec); - break; - - } -} - -static const struct snd_kcontrol_new cxt5045_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x17, 0x2, HDA_INPUT), - HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5045_hp_master_sw_put, - .private_value = 0x10, - }, - - {} -}; - -static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { - HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT), - - {} -}; - -static const struct hda_verb cxt5045_init_verbs[] = { - /* Line in, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - /* HP, Amp */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Record selector: Internal mic */ - {0x1a, AC_VERB_SET_CONNECT_SEL,0x1}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* SPDIF route: PCM */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x13, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* EAPD */ - {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */ - { } /* end */ -}; - -static const struct hda_verb cxt5045_benq_init_verbs[] = { - /* Internal Mic, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - /* Line In,HP, Amp */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Record selector: Internal mic */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* SPDIF route: PCM */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - { } /* end */ -}; - -static const struct hda_verb cxt5045_hp_sense_init_verbs[] = { - /* pin sensing on HP jack */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5045_mic_sense_init_verbs[] = { - /* pin sensing on HP jack */ - {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -#ifdef CONFIG_SND_DEBUG -/* Test configuration for debugging, modelled after the ALC260 test - * configuration. - */ -static const struct hda_input_mux cxt5045_test_capture_source = { - .num_items = 5, - .items = { - { "MIXER", 0x0 }, - { "MIC1 pin", 0x1 }, - { "LINE1 pin", 0x2 }, - { "HP-OUT pin", 0x3 }, - { "CD pin", 0x4 }, - }, -}; - -static const struct snd_kcontrol_new cxt5045_test_mixer[] = { - - /* Output controls */ - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT), - - /* Modes for retasking pin widgets */ - CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT), - CXT_PIN_MODE("LINE1 pin mode", 0x12, CXT_PIN_DIR_INOUT), - - /* EAPD Switch Control */ - CXT_EAPD_SWITCH("External Amplifier", 0x10, 0x0), - - /* Loopback mixer controls */ - - HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Input Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put, - }, - /* Audio input controls */ - HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb cxt5045_test_init_verbs[] = { - /* Set connections */ - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, - { 0x11, AC_VERB_SET_CONNECT_SEL, 0x0 }, - { 0x12, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* Enable retasking pins as output, initially without power amp */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* Disable digital (SPDIF) pins initially, but users can enable - * them via a mixer switch. In the case of SPDIF-out, this initverb - * payload also sets the generation to 0, output to be in "consumer" - * PCM format, copyright asserted, no pre-emphasis and no validity - * control. - */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Unmute retasking pin widget output buffers since the default - * state appears to be output. As the pin mode is changed by the - * user the pin mode control will take care of enabling the pin's - * input/output buffers as needed. - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mute capture amp left and right */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - /* Set ADC connection select to match default mixer setting (mic1 - * pin) - */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x01}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - - { } -}; -#endif - - -/* initialize jack-sensing, too */ -static int cxt5045_init(struct hda_codec *codec) -{ - conexant_init(codec); - cxt5045_hp_automute(codec); - return 0; -} - - -enum { - CXT5045_LAPTOP_HPSENSE, - CXT5045_LAPTOP_MICSENSE, - CXT5045_LAPTOP_HPMICSENSE, - CXT5045_BENQ, -#ifdef CONFIG_SND_DEBUG - CXT5045_TEST, -#endif - CXT5045_AUTO, - CXT5045_MODELS -}; - -static const char * const cxt5045_models[CXT5045_MODELS] = { - [CXT5045_LAPTOP_HPSENSE] = "laptop-hpsense", - [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", - [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", - [CXT5045_BENQ] = "benq", -#ifdef CONFIG_SND_DEBUG - [CXT5045_TEST] = "test", -#endif - [CXT5045_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), - SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), - SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", - CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell", - CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE), - {} -}; - -static int patch_cxt5045(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5045_MODELS, - cxt5045_models, - cxt5045_cfg_tbl); - if (board_config < 0) - board_config = CXT5045_AUTO; /* model=auto as default */ - if (board_config == CXT5045_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->single_adc_amp = 1; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); - spec->multiout.dac_nids = cxt5045_dac_nids; - spec->multiout.dig_out_nid = CXT5045_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = cxt5045_adc_nids; - spec->capsrc_nids = cxt5045_capsrc_nids; - spec->input_mux = &cxt5045_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = cxt5045_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5045_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes); - spec->channel_mode = cxt5045_modes; - - set_beep_amp(spec, 0x16, 0, 1); - - codec->patch_ops = conexant_patch_ops; - - switch (board_config) { - case CXT5045_LAPTOP_HPSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - case CXT5045_LAPTOP_MICSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - default: - case CXT5045_LAPTOP_HPMICSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 3; - spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; - spec->init_verbs[2] = cxt5045_mic_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - case CXT5045_BENQ: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source_benq; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5045_benq_init_verbs; - spec->mixers[0] = cxt5045_mixers; - spec->mixers[1] = cxt5045_benq_mixers; - spec->num_mixers = 2; - codec->patch_ops.init = cxt5045_init; - break; -#ifdef CONFIG_SND_DEBUG - case CXT5045_TEST: - spec->input_mux = &cxt5045_test_capture_source; - spec->mixers[0] = cxt5045_test_mixer; - spec->init_verbs[0] = cxt5045_test_init_verbs; - break; - -#endif - } - - switch (codec->subsystem_id >> 16) { - case 0x103c: - case 0x1631: - case 0x1734: - case 0x17aa: - /* HP, Packard Bell, Fujitsu-Siemens & Lenovo laptops have - * really bad sound over 0dB on NID 0x17. Fix max PCM level to - * 0 dB (originally it has 0x2b steps with 0dB offset 0x14) - */ - snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, - (0x14 << AC_AMPCAP_OFFSET_SHIFT) | - (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - - -/* Conexant 5047 specific */ -#define CXT5047_SPDIF_OUT 0x11 - -static const hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ -static const hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; -static const hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; - -static const struct hda_channel_mode cxt5047_modes[1] = { - { 2, NULL }, -}; - -static const struct hda_input_mux cxt5047_toshiba_capture_source = { - .num_items = 2, - .items = { - { "ExtMic", 0x2 }, - { "Line-In", 0x1 }, - } -}; - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - /* toggle internal speakers mute depending of presence of - * the headphone jack - */ - bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - /* NOTE: Conexat codec needs the index for *OUTPUT* amp of - * pin widgets unlike other codecs. In this case, we need to - * set index 0x01 for the volume from the mixer amp 0x19. - */ - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, - HDA_AMP_MUTE, bits); - bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - return 1; -} - -/* mute internal speaker if HP is plugged */ -static void cxt5047_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_jack_detect(codec, 0x13); - - bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - /* See the note in cxt5047_hp_master_sw_put */ - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, - HDA_AMP_MUTE, bits); -} - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5047_hp_automic(struct hda_codec *codec) -{ - static const struct hda_verb mic_jack_on[] = { - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {} - }; - static const struct hda_verb mic_jack_off[] = { - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {} - }; - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x15); - if (present) - snd_hda_sequence_write(codec, mic_jack_on); - else - snd_hda_sequence_write(codec, mic_jack_off); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5047_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5047_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5047_hp_automic(codec); - break; - } -} - -static const struct snd_kcontrol_new cxt5047_base_mixers[] = { - HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5047_hp_master_sw_put, - .private_value = 0x13, - }, - - {} -}; - -static const struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { - /* See the note in cxt5047_hp_master_sw_put */ - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT), - { } /* end */ -}; - -static const struct hda_verb cxt5047_init_verbs[] = { - /* Line in, Mic, Built-in Mic */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, - /* HP, Speaker */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */ - /* Record selector: Mic */ - {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - {0x1A, AC_VERB_SET_CONNECT_SEL,0x02}, - {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x00}, - {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x03}, - /* SPDIF route: PCM */ - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* Enable unsolicited events */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -/* configuration for Toshiba Laptops */ -static const struct hda_verb cxt5047_toshiba_init_verbs[] = { - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ - {} -}; - -/* Test configuration for debugging, modelled after the ALC260 test - * configuration. - */ -#ifdef CONFIG_SND_DEBUG -static const struct hda_input_mux cxt5047_test_capture_source = { - .num_items = 4, - .items = { - { "LINE1 pin", 0x0 }, - { "MIC1 pin", 0x1 }, - { "MIC2 pin", 0x2 }, - { "CD pin", 0x3 }, - }, -}; - -static const struct snd_kcontrol_new cxt5047_test_mixer[] = { - - /* Output only controls */ - HDA_CODEC_VOLUME("OutAmp-1 Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("OutAmp-1 Switch", 0x10,0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("OutAmp-2 Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("OutAmp-2 Switch", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("HeadPhone Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("HeadPhone Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line1-Out Playback Volume", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line1-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line2-Out Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line2-Out Playback Switch", 0x15, 0x0, HDA_OUTPUT), - - /* Modes for retasking pin widgets */ - CXT_PIN_MODE("LINE1 pin mode", 0x14, CXT_PIN_DIR_INOUT), - CXT_PIN_MODE("MIC1 pin mode", 0x15, CXT_PIN_DIR_INOUT), - - /* EAPD Switch Control */ - CXT_EAPD_SWITCH("External Amplifier", 0x13, 0x0), - - /* Loopback mixer controls */ - HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x12, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("MIC1 Playback Switch", 0x12, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x12, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("MIC2 Playback Switch", 0x12, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LINE Playback Volume", 0x12, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("LINE Playback Switch", 0x12, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x12, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x12, 0x04, HDA_INPUT), - - HDA_CODEC_VOLUME("Capture-1 Volume", 0x19, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture-1 Switch", 0x19, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-2 Volume", 0x19, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Capture-2 Switch", 0x19, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-3 Volume", 0x19, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Capture-3 Switch", 0x19, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-4 Volume", 0x19, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Capture-4 Switch", 0x19, 0x3, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Input Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put, - }, - HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), - - { } /* end */ -}; - -static const struct hda_verb cxt5047_test_init_verbs[] = { - /* Enable retasking pins as output, initially without power amp */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* Disable digital (SPDIF) pins initially, but users can enable - * them via a mixer switch. In the case of SPDIF-out, this initverb - * payload also sets the generation to 0, output to be in "consumer" - * PCM format, copyright asserted, no pre-emphasis and no validity - * control. - */ - {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure mic1, mic2, line1 pin widgets take input from the - * OUT1 sum bus when acting as an output. - */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* Unmute retasking pin widget output buffers since the default - * state appears to be output. As the pin mode is changed by the - * user the pin mode control will take care of enabling the pin's - * input/output buffers as needed. - */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mute capture amp left and right */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - /* Set ADC connection select to match default mixer setting (mic1 - * pin) - */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; -#endif - - -/* initialize jack-sensing, too */ -static int cxt5047_hp_init(struct hda_codec *codec) -{ - conexant_init(codec); - cxt5047_hp_automute(codec); - return 0; -} - - -enum { - CXT5047_LAPTOP, /* Laptops w/o EAPD support */ - CXT5047_LAPTOP_HP, /* Some HP laptops */ - CXT5047_LAPTOP_EAPD, /* Laptops with EAPD support */ -#ifdef CONFIG_SND_DEBUG - CXT5047_TEST, -#endif - CXT5047_AUTO, - CXT5047_MODELS -}; - -static const char * const cxt5047_models[CXT5047_MODELS] = { - [CXT5047_LAPTOP] = "laptop", - [CXT5047_LAPTOP_HP] = "laptop-hp", - [CXT5047_LAPTOP_EAPD] = "laptop-eapd", -#ifdef CONFIG_SND_DEBUG - [CXT5047_TEST] = "test", -#endif - [CXT5047_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5047_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", - CXT5047_LAPTOP), - SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), - {} -}; - -static int patch_cxt5047(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5047_MODELS, - cxt5047_models, - cxt5047_cfg_tbl); - if (board_config < 0) - board_config = CXT5047_AUTO; /* model=auto as default */ - if (board_config == CXT5047_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->pin_amp_workaround = 1; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids); - spec->multiout.dac_nids = cxt5047_dac_nids; - spec->multiout.dig_out_nid = CXT5047_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = cxt5047_adc_nids; - spec->capsrc_nids = cxt5047_capsrc_nids; - spec->num_mixers = 1; - spec->mixers[0] = cxt5047_base_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5047_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5047_modes), - spec->channel_mode = cxt5047_modes, - - codec->patch_ops = conexant_patch_ops; - - switch (board_config) { - case CXT5047_LAPTOP: - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_spk_mixers; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - break; - case CXT5047_LAPTOP_HP: - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_only_mixers; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - codec->patch_ops.init = cxt5047_hp_init; - break; - case CXT5047_LAPTOP_EAPD: - spec->input_mux = &cxt5047_toshiba_capture_source; - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_spk_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5047_toshiba_init_verbs; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - break; -#ifdef CONFIG_SND_DEBUG - case CXT5047_TEST: - spec->input_mux = &cxt5047_test_capture_source; - spec->mixers[0] = cxt5047_test_mixer; - spec->init_verbs[0] = cxt5047_test_init_verbs; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; -#endif - } - spec->vmaster_nid = 0x13; - - switch (codec->subsystem_id >> 16) { - case 0x103c: - /* HP laptops have really bad sound over 0 dB on NID 0x10. - * Fix max PCM level to 0 dB (originally it has 0x1e steps - * with 0 dB offset 0x17) - */ - snd_hda_override_amp_caps(codec, 0x10, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - } - - return 0; -} - -/* Conexant 5051 specific */ -static const hda_nid_t cxt5051_dac_nids[1] = { 0x10 }; -static const hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 }; - -static const struct hda_channel_mode cxt5051_modes[1] = { - { 2, NULL }, -}; - -static void cxt5051_update_speaker(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int pinctl; - /* headphone pin */ - pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_set_pin_ctl(codec, 0x16, pinctl); - /* speaker pin */ - pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_set_pin_ctl(codec, 0x1a, pinctl); - /* on ideapad there is an additional speaker (subwoofer) to mute */ - if (spec->ideapad) - snd_hda_set_pin_ctl(codec, 0x1b, pinctl); -} - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - cxt5051_update_speaker(codec); - return 1; -} - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5051_portb_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int present; - - if (!(spec->auto_mic & AUTO_MIC_PORTB)) - return; - present = snd_hda_jack_detect(codec, 0x17); - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_CONNECT_SEL, - present ? 0x01 : 0x00); -} - -/* switch the current ADC according to the jack state */ -static void cxt5051_portc_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int present; - hda_nid_t new_adc; - - if (!(spec->auto_mic & AUTO_MIC_PORTC)) - return; - present = snd_hda_jack_detect(codec, 0x18); - if (present) - spec->cur_adc_idx = 1; - else - spec->cur_adc_idx = 0; - new_adc = spec->adc_nids[spec->cur_adc_idx]; - if (spec->cur_adc && spec->cur_adc != new_adc) { - /* stream is running, let's swap the current ADC */ - __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); - spec->cur_adc = new_adc; - snd_hda_codec_setup_stream(codec, new_adc, - spec->cur_adc_stream_tag, 0, - spec->cur_adc_format); - } -} - -/* mute internal speaker if HP is plugged */ -static void cxt5051_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - spec->hp_present = snd_hda_jack_detect(codec, 0x16); - cxt5051_update_speaker(codec); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5051_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5051_hp_automute(codec); - break; - case CXT5051_PORTB_EVENT: - cxt5051_portb_automic(codec); - break; - case CXT5051_PORTC_EVENT: - cxt5051_portc_automic(codec); - break; - } -} - -static const struct snd_kcontrol_new cxt5051_playback_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} -}; - -static const struct snd_kcontrol_new cxt5051_capture_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Dock Mic Switch", 0x15, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_hp_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x15, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_f700_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - {} -}; - -static const struct hda_verb cxt5051_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5051_f700_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, - unsigned int event) -{ - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | event); -} - -static const struct hda_verb cxt5051_ideapad_init_verbs[] = { - /* Subwoofer */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } /* end */ -}; - -/* initialize jack-sensing, too */ -static int cxt5051_init(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - conexant_init(codec); - - if (spec->auto_mic & AUTO_MIC_PORTB) - cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT); - if (spec->auto_mic & AUTO_MIC_PORTC) - cxt5051_init_mic_port(codec, 0x18, CXT5051_PORTC_EVENT); - - if (codec->patch_ops.unsol_event) { - cxt5051_hp_automute(codec); - cxt5051_portb_automic(codec); - cxt5051_portc_automic(codec); - } - return 0; -} - - -enum { - CXT5051_LAPTOP, /* Laptops w/ EAPD support */ - CXT5051_HP, /* no docking */ - CXT5051_HP_DV6736, /* HP without mic switch */ - CXT5051_F700, /* HP Compaq Presario F700 */ - CXT5051_TOSHIBA, /* Toshiba M300 & co */ - CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ - CXT5051_AUTO, /* auto-parser */ - CXT5051_MODELS -}; - -static const char *const cxt5051_models[CXT5051_MODELS] = { - [CXT5051_LAPTOP] = "laptop", - [CXT5051_HP] = "hp", - [CXT5051_HP_DV6736] = "hp-dv6736", - [CXT5051_F700] = "hp-700", - [CXT5051_TOSHIBA] = "toshiba", - [CXT5051_IDEAPAD] = "ideapad", - [CXT5051_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5051_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), - SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), - SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), - SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba M30x", CXT5051_TOSHIBA), - SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", - CXT5051_LAPTOP), - SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), - {} -}; - -static int patch_cxt5051(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, - cxt5051_models, - cxt5051_cfg_tbl); - if (board_config < 0) - board_config = CXT5051_AUTO; /* model=auto as default */ - if (board_config == CXT5051_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->pin_amp_workaround = 1; - - codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = cxt5051_init; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5051_dac_nids); - spec->multiout.dac_nids = cxt5051_dac_nids; - spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT; - spec->num_adc_nids = 1; /* not 2; via auto-mic switch */ - spec->adc_nids = cxt5051_adc_nids; - spec->num_mixers = 2; - spec->mixers[0] = cxt5051_capture_mixers; - spec->mixers[1] = cxt5051_playback_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5051_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5051_modes); - spec->channel_mode = cxt5051_modes; - spec->cur_adc = 0; - spec->cur_adc_idx = 0; - - set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); - - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; - - spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC; - switch (board_config) { - case CXT5051_HP: - spec->mixers[0] = cxt5051_hp_mixers; - break; - case CXT5051_HP_DV6736: - spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; - spec->mixers[0] = cxt5051_hp_dv6736_mixers; - spec->auto_mic = 0; - break; - case CXT5051_F700: - spec->init_verbs[0] = cxt5051_f700_init_verbs; - spec->mixers[0] = cxt5051_f700_mixers; - spec->auto_mic = 0; - break; - case CXT5051_TOSHIBA: - spec->mixers[0] = cxt5051_toshiba_mixers; - spec->auto_mic = AUTO_MIC_PORTB; - break; - case CXT5051_IDEAPAD: - spec->init_verbs[spec->num_init_verbs++] = - cxt5051_ideapad_init_verbs; - spec->ideapad = 1; - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - -/* Conexant 5066 specific */ - -static const hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; -static const hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; -static const hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; -static const hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 }; - -static const struct hda_channel_mode cxt5066_modes[1] = { - { 2, NULL }, -}; - -#define HP_PRESENT_PORT_A (1 << 0) -#define HP_PRESENT_PORT_D (1 << 1) -#define hp_port_a_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_A) -#define hp_port_d_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_D) - -static void cxt5066_update_speaker(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int pinctl; - - codec_dbg(codec, - "CXT5066: update speaker, hp_present=%d, cur_eapd=%d\n", - spec->hp_present, spec->cur_eapd); - - /* Port A (HP) */ - pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_set_pin_ctl(codec, 0x19, pinctl); - - /* Port D (HP/LO) */ - pinctl = spec->cur_eapd ? spec->port_d_mode : 0; - if (spec->dell_automute || spec->thinkpad) { - /* Mute if Port A is connected */ - if (hp_port_a_present(spec)) - pinctl = 0; - } else { - /* Thinkpad/Dell doesn't give pin-D status */ - if (!hp_port_d_present(spec)) - pinctl = 0; - } - snd_hda_set_pin_ctl(codec, 0x1c, pinctl); - - /* CLASS_D AMP */ - pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_set_pin_ctl(codec, 0x1f, pinctl); -} - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - cxt5066_update_speaker(codec); - return 1; -} - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_vostro_automic(struct hda_codec *codec) -{ - unsigned int present; - - struct hda_verb ext_mic_present[] = { - /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - - /* switch to external mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - - /* disable internal digital mic */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - /* enable internal mic, port C */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* switch to internal mic input */ - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - - /* disable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - present = snd_hda_jack_detect(codec, 0x1a); - if (present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_ideapad_automic(struct hda_codec *codec) -{ - unsigned int present; - - struct hda_verb ext_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - present = snd_hda_jack_detect(codec, 0x1b); - if (present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_asus_automic(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x1b); - codec_dbg(codec, "CXT5066: external microphone present=%d\n", present); - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, - present ? 1 : 0); -} - - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_hp_laptop_automic(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x1b); - codec_dbg(codec, "CXT5066: external microphone present=%d\n", present); - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, - present ? 1 : 3); -} - - -/* toggle input of built-in digital mic and mic jack appropriately - order is: external mic -> dock mic -> interal mic */ -static void cxt5066_thinkpad_automic(struct hda_codec *codec) -{ - unsigned int ext_present, dock_present; - - static const struct hda_verb ext_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb dock_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - ext_present = snd_hda_jack_detect(codec, 0x1b); - dock_present = snd_hda_jack_detect(codec, 0x1a); - if (ext_present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else if (dock_present) { - codec_dbg(codec, "CXT5066: dock microphone detected\n"); - snd_hda_sequence_write(codec, dock_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - -/* mute internal speaker if HP is plugged */ -static void cxt5066_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int portA, portD; - - /* Port A */ - portA = snd_hda_jack_detect(codec, 0x19); - - /* Port D */ - portD = snd_hda_jack_detect(codec, 0x1c); - - spec->hp_present = portA ? HP_PRESENT_PORT_A : 0; - spec->hp_present |= portD ? HP_PRESENT_PORT_D : 0; - codec_dbg(codec, "CXT5066: hp automute portA=%x portD=%x present=%d\n", - portA, portD, spec->hp_present); - cxt5066_update_speaker(codec); -} - -/* Dispatch the right mic autoswitch function */ -static void cxt5066_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - if (spec->dell_vostro) - cxt5066_vostro_automic(codec); - else if (spec->ideapad) - cxt5066_ideapad_automic(codec); - else if (spec->thinkpad) - cxt5066_thinkpad_automic(codec); - else if (spec->hp_laptop) - cxt5066_hp_laptop_automic(codec); - else if (spec->asus) - cxt5066_asus_automic(codec); -} - -/* unsolicited event for jack sensing */ -static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) -{ - codec_dbg(codec, "CXT5066: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5066_automic(codec); - break; - } -} - - -static const struct hda_input_mux cxt5066_analog_mic_boost = { - .num_items = 5, - .items = { - { "0dB", 0 }, - { "10dB", 1 }, - { "20dB", 2 }, - { "30dB", 3 }, - { "40dB", 4 }, - }, -}; - -static void cxt5066_set_mic_boost(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_write_cache(codec, 0x17, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | - cxt5066_analog_mic_boost.items[spec->mic_boost].index); - if (spec->ideapad || spec->thinkpad) { - /* adjust the internal mic as well...it is not through 0x17 */ - snd_hda_codec_write_cache(codec, 0x23, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_INPUT | - cxt5066_analog_mic_boost. - items[spec->mic_boost].index); - } -} - -static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo); -} - -static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = spec->mic_boost; - return 0; -} - -static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; - unsigned int idx; - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - - spec->mic_boost = idx; - cxt5066_set_mic_boost(codec); - return 1; -} - -static void conexant_check_dig_outs(struct hda_codec *codec, - const hda_nid_t *dig_pins, - int num_pins) -{ - struct conexant_spec *spec = codec->spec; - hda_nid_t *nid_loc = &spec->multiout.dig_out_nid; - int i; - - for (i = 0; i < num_pins; i++, dig_pins++) { - unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins); - if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE) - continue; - if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1) - continue; - } -} - -static const struct hda_input_mux cxt5066_capture_source = { - .num_items = 4, - .items = { - { "Mic B", 0 }, - { "Mic C", 1 }, - { "Mic E", 2 }, - { "Mic F", 3 }, - }, -}; - -static const struct hda_bind_ctls cxt5066_bind_capture_vol_others = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), - 0 - }, -}; - -static const struct hda_bind_ctls cxt5066_bind_capture_sw_others = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new cxt5066_mixer_master[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5066_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5066_hp_master_sw_put, - .private_value = 0x1d, - }, - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Mic Boost Capture Enum", - .info = cxt5066_mic_boost_mux_enum_info, - .get = cxt5066_mic_boost_mux_enum_get, - .put = cxt5066_mic_boost_mux_enum_put, - }, - - HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), - HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others), - {} -}; - -static const struct snd_kcontrol_new cxt5066_vostro_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Internal Mic Boost Capture Enum", - .info = cxt5066_mic_boost_mux_enum_info, - .get = cxt5066_mic_boost_mux_enum_get, - .put = cxt5066_mic_boost_mux_enum_put, - .private_value = 0x23 | 0x100, - }, - {} -}; - -static const struct hda_verb cxt5066_init_verbs[] = { - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* no digital microphone support yet */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* not handling these yet */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_vostro[] = { - /* Port A: headphones */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port C: unused */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port D: unused */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port E: unused, but has primary EAPD */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* Port F: unused */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port G: internal speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* DAC2: unused */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - - /* Digital microphone port */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* Audio input selectors */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - - /* Disable SPDIF */ - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* enable unsolicited events for Port A and B */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_ideapad[] = { - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* internal microphone */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */ - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_thinkpad[] = { - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Port G: internal speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port A: HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port B: Mic Dock */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port C: Mic */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port D: HP Dock, Amp */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* internal microphone */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */ - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* enable unsolicited events for Port A, B, C and D */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_portd_lo[] = { - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { } /* end */ -}; - - -static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -/* initialize jack-sensing, too */ -static int cxt5066_init(struct hda_codec *codec) -{ - codec_dbg(codec, "CXT5066: init\n"); - conexant_init(codec); - if (codec->patch_ops.unsol_event) { - cxt5066_hp_automute(codec); - cxt5066_automic(codec); - } - cxt5066_set_mic_boost(codec); - return 0; -} - -enum { - CXT5066_LAPTOP, /* Laptops w/ EAPD support */ - CXT5066_DELL_LAPTOP, /* Dell Laptop */ - CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ - CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ - CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ - CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ - CXT5066_HP_LAPTOP, /* HP Laptop */ - CXT5066_AUTO, /* BIOS auto-parser */ - CXT5066_MODELS -}; - -static const char * const cxt5066_models[CXT5066_MODELS] = { - [CXT5066_LAPTOP] = "laptop", - [CXT5066_DELL_LAPTOP] = "dell-laptop", - [CXT5066_DELL_VOSTRO] = "dell-vostro", - [CXT5066_IDEAPAD] = "ideapad", - [CXT5066_THINKPAD] = "thinkpad", - [CXT5066_ASUS] = "asus", - [CXT5066_HP_LAPTOP] = "hp-laptop", - [CXT5066_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), - SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), - SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS), - SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", - CXT5066_LAPTOP), - SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), - {} -}; - -static int patch_cxt5066(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, - cxt5066_models, cxt5066_cfg_tbl); - if (board_config < 0) - board_config = CXT5066_AUTO; /* model=auto as default */ - if (board_config == CXT5066_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - - codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = conexant_init; - - spec->dell_automute = 0; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); - spec->multiout.dac_nids = cxt5066_dac_nids; - conexant_check_dig_outs(codec, cxt5066_digout_pin_nids, - ARRAY_SIZE(cxt5066_digout_pin_nids)); - spec->num_adc_nids = 1; - spec->adc_nids = cxt5066_adc_nids; - spec->capsrc_nids = cxt5066_capsrc_nids; - spec->input_mux = &cxt5066_capture_source; - - spec->port_d_mode = PIN_HP; - - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5066_init_verbs; - spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes); - spec->channel_mode = cxt5066_modes; - spec->cur_adc = 0; - spec->cur_adc_idx = 0; - - set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); - - switch (board_config) { - default: - case CXT5066_LAPTOP: - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - break; - case CXT5066_DELL_LAPTOP: - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - - spec->port_d_mode = PIN_OUT; - spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo; - spec->num_init_verbs++; - spec->dell_automute = 1; - break; - case CXT5066_ASUS: - case CXT5066_HP_LAPTOP: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->init_verbs[spec->num_init_verbs] = - cxt5066_init_verbs_hp_laptop; - spec->num_init_verbs++; - spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; - spec->asus = board_config == CXT5066_ASUS; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - /* no S/PDIF out */ - if (board_config == CXT5066_HP_LAPTOP) - spec->multiout.dig_out_nid = 0; - /* input source automatically selected */ - spec->input_mux = NULL; - spec->port_d_mode = 0; - spec->mic_boost = 3; /* default 30dB gain */ - break; - - case CXT5066_DELL_VOSTRO: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->init_verbs[0] = cxt5066_init_verbs_vostro; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; - spec->port_d_mode = 0; - spec->dell_vostro = 1; - spec->mic_boost = 3; /* default 30dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - case CXT5066_IDEAPAD: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->init_verbs[0] = cxt5066_init_verbs_ideapad; - spec->port_d_mode = 0; - spec->ideapad = 1; - spec->mic_boost = 2; /* default 20dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - case CXT5066_THINKPAD: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->init_verbs[0] = cxt5066_init_verbs_thinkpad; - spec->thinkpad = 1; - spec->port_d_mode = PIN_OUT; - spec->mic_boost = 2; /* default 20dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - -#endif /* ENABLE_CXT_STATIC_QUIRKS */ - - /* * Automatic parser for CX20641 & co */ @@ -3487,35 +881,28 @@ static int patch_conexant_auto(struct hda_codec *codec) return err; } -#ifndef ENABLE_CXT_STATIC_QUIRKS -#define patch_cxt5045 patch_conexant_auto -#define patch_cxt5047 patch_conexant_auto -#define patch_cxt5051 patch_conexant_auto -#define patch_cxt5066 patch_conexant_auto -#endif - /* */ static const struct hda_codec_preset snd_hda_preset_conexant[] = { { .id = 0x14f15045, .name = "CX20549 (Venice)", - .patch = patch_cxt5045 }, + .patch = patch_conexant_auto }, { .id = 0x14f15047, .name = "CX20551 (Waikiki)", - .patch = patch_cxt5047 }, + .patch = patch_conexant_auto }, { .id = 0x14f15051, .name = "CX20561 (Hermosa)", - .patch = patch_cxt5051 }, + .patch = patch_conexant_auto }, { .id = 0x14f15066, .name = "CX20582 (Pebble)", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15068, .name = "CX20584", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15069, .name = "CX20585", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f1506c, .name = "CX20588", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f1506e, .name = "CX20590", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15097, .name = "CX20631", .patch = patch_conexant_auto }, { .id = 0x14f15098, .name = "CX20632", -- 1.9.3

1 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Kill snd_printd*() in HDMI debug / info prints
by Takashi Iwai 18 Jun '14

18 Jun '14

Pass codec instance to each function that still prints info and debug outputs via snd_printd*(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_eld.c | 46 ++++++++++++++++++++++------------------------ sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 11 ++++++----- 3 files changed, 30 insertions(+), 31 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 46690a7f48f6..e1cd34d9011d 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -167,7 +167,8 @@ static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid, (buf[byte] >> (lowbit)) & ((1 << (bits)) - 1); \ }) -static void hdmi_update_short_audio_desc(struct cea_sad *a, +static void hdmi_update_short_audio_desc(struct hda_codec *codec, + struct cea_sad *a, const unsigned char *buf) { int i; @@ -188,8 +189,7 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: - snd_printd(KERN_INFO - "HDMI: audio coding type 0 not expected\n"); + codec_info(codec, "HDMI: audio coding type 0 not expected\n"); break; case AUDIO_CODING_TYPE_LPCM: @@ -233,9 +233,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->format = GRAB_BITS(buf, 2, 3, 5); if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT || a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) { - snd_printd(KERN_INFO - "HDMI: audio coding xtype %d not expected\n", - a->format); + codec_info(codec, + "HDMI: audio coding xtype %d not expected\n", + a->format); a->format = 0; } else a->format += AUDIO_CODING_TYPE_HE_AAC - @@ -247,7 +247,7 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, /* * Be careful, ELD buf could be totally rubbish! */ -int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, +int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e, const unsigned char *buf, int size) { int mnl; @@ -256,8 +256,7 @@ int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, e->eld_ver = GRAB_BITS(buf, 0, 3, 5); if (e->eld_ver != ELD_VER_CEA_861D && e->eld_ver != ELD_VER_PARTIAL) { - snd_printd(KERN_INFO "HDMI: Unknown ELD version %d\n", - e->eld_ver); + codec_info(codec, "HDMI: Unknown ELD version %d\n", e->eld_ver); goto out_fail; } @@ -280,20 +279,20 @@ int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, e->product_id = get_unaligned_le16(buf + 18); if (mnl > ELD_MAX_MNL) { - snd_printd(KERN_INFO "HDMI: MNL is reserved value %d\n", mnl); + codec_info(codec, "HDMI: MNL is reserved value %d\n", mnl); goto out_fail; } else if (ELD_FIXED_BYTES + mnl > size) { - snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl); + codec_info(codec, "HDMI: out of range MNL %d\n", mnl); goto out_fail; } else strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1); for (i = 0; i < e->sad_count; i++) { if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { - snd_printd(KERN_INFO "HDMI: out of range SAD %d\n", i); + codec_info(codec, "HDMI: out of range SAD %d\n", i); goto out_fail; } - hdmi_update_short_audio_desc(e->sad + i, + hdmi_update_short_audio_desc(codec, e->sad + i, buf + ELD_FIXED_BYTES + mnl + 3 * i); } @@ -394,7 +393,8 @@ static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) #define SND_PRINT_RATES_ADVISED_BUFSIZE 80 -static void hdmi_show_short_audio_desc(struct cea_sad *a) +static void hdmi_show_short_audio_desc(struct hda_codec *codec, + struct cea_sad *a) { char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; char buf2[8 + SND_PRINT_BITS_ADVISED_BUFSIZE] = ", bits ="; @@ -412,12 +412,10 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) else buf2[0] = '\0'; - _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:" - " channels = %d, rates =%s%s\n", - cea_audio_coding_type_names[a->format], - a->channels, - buf, - buf2); + codec_dbg(codec, + "HDMI: supports coding type %s: channels = %d, rates =%s%s\n", + cea_audio_coding_type_names[a->format], + a->channels, buf, buf2); } void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) @@ -432,22 +430,22 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) buf[j] = '\0'; /* necessary when j == 0 */ } -void snd_hdmi_show_eld(struct parsed_hdmi_eld *e) +void snd_hdmi_show_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e) { int i; - _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n", + codec_dbg(codec, "HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf); + codec_dbg(codec, "HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) - hdmi_show_short_audio_desc(e->sad + i); + hdmi_show_short_audio_desc(codec, e->sad + i); } #ifdef CONFIG_PROC_FS diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e51d15529215..7820fd2d9ba8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -738,9 +738,9 @@ struct hdmi_eld { int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid); int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size); -int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, +int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e, const unsigned char *buf, int size); -void snd_hdmi_show_eld(struct parsed_hdmi_eld *e); +void snd_hdmi_show_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e); void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, struct hda_pcm_stream *hinfo); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b4218a19df22..39f86fad0330 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -648,7 +648,8 @@ static int get_channel_allocation_order(int ca) * * TODO: it could select the wrong CA from multiple candidates. */ -static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) +static int hdmi_channel_allocation(struct hda_codec *codec, + struct hdmi_eld *eld, int channels) { int i; int ca = 0; @@ -694,7 +695,7 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) } snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf)); - snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", + codec_dbg(codec, "HDMI: select CA 0x%x for %d-channel allocation: %s\n", ca, channels, buf); return ca; @@ -1135,7 +1136,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!non_pcm && per_pin->chmap_set) ca = hdmi_manual_channel_allocation(channels, per_pin->chmap); else - ca = hdmi_channel_allocation(eld, channels); + ca = hdmi_channel_allocation(codec, eld, channels); if (ca < 0) ca = 0; @@ -1561,13 +1562,13 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld->eld_valid = false; else { memset(&eld->info, 0, sizeof(struct parsed_hdmi_eld)); - if (snd_hdmi_parse_eld(&eld->info, eld->eld_buffer, + if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, eld->eld_size) < 0) eld->eld_valid = false; } if (eld->eld_valid) { - snd_hdmi_show_eld(&eld->info); + snd_hdmi_show_eld(codec, &eld->info); update_eld = true; } else if (repoll) { -- 1.9.3

1 1

[alsa-devel] ALSA 1.0.28 release
by Jaroslav Kysela 18 Jun '14

18 Jun '14

Hello all, the ALSA 1.0.28 packages are available on the main ALSA project web site: http://www.alsa-project.org . Thank you to all authors of code updates. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

1 0

Re: [alsa-devel] Plug In Creation - aplay can't find existing .so
by Takashi Iwai 18 Jun '14

18 Jun '14

At Tue, 17 Jun 2014 15:08:31 -0400, Brandon Yates wrote: > > Interestingly strace shows the failure occurs on the open call (not > dlopen) > > open("/usr/lib/i386-linux-gnu/alsa-lib/libasound_module_pcm_adt_dsp.so", > O_RDONLY|O_CLOEXEC) = 3 This succeeded and returned the proper fd number 3. > running ldd on my shared library shows undefined symbols that are part > of libasound, for example > undefined symbol: snd_pcm_ioplug_create > > I wonder if I have to build the .so a different way so these can be > resolved? Any advice would be greatly appreciated. Maybe you forgot to link your shared object with -lasound -lm? Takashi > > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Friday, June 13, 2014 8:42 AM > To: Brandon Yates > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Plug In Creation - aplay can't find existing > .so > > At Tue, 10 Jun 2014 13:55:39 -0400, > Brandon Yates wrote: > > > > Hello, > > > > I am working on a new pcm plugin. At this point it is basically just a > > > Hello World plugin, to make sure I have the tools working and > > understand the workflow. I created a simple file which is just > basically: > > > > #include <alsa/asoundlib.h> > > #include <alsa/pcm_external.h> > > SND_PCM_PLUGIN_DEFINE_FUNC(my_mod) > > { > > printf("hello world\n"); > > return 0; > > } > > SND_PCM_PLUGIN_SYMBOL(my_mod); > > > > > > I compiled this as a shared lib named > 'libasound_module_pcm_my_mod.so' > > and copied it to '/usr/lib/i386-linux-gnu/alsa-lib/' > > > > I edited my ~/.asoundrc to create a new device that uses this plugin. > > Then I try to use it with `aplay --device=my_mod_test out.pcm` > > > > I get the error message > > > > ALSA lib dlmisc.c:254:(snd1_dlobj_cache_get) Cannot open shared > > library > > /usr/lib/i386-linux-gnu/alsa-lib/libasound_module_pcm_my_mod.so > > aplay: main:682: audio open error: No such device or address > > > > > > But this file actually does exist, I just copied it there. > > > > > > Any idea why this happens? > > Try strace to see what happens. If it fails in dlopen(), it's likely > because your shared object is invalid. > > > Takashi >

1 0

[alsa-devel] Plug In Creation - aplay can't find existing .so
by Brandon Yates 18 Jun '14

18 Jun '14

Hello, I am working on a new pcm plugin. At this point it is basically just a Hello World plugin, to make sure I have the tools working and understand the workflow. I created a simple file which is just basically: #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> SND_PCM_PLUGIN_DEFINE_FUNC(my_mod) { printf("hello world\n"); return 0; } SND_PCM_PLUGIN_SYMBOL(my_mod); I compiled this as a shared lib named 'libasound_module_pcm_my_mod.so' and copied it to '/usr/lib/i386-linux-gnu/alsa-lib/' I edited my ~/.asoundrc to create a new device that uses this plugin. Then I try to use it with `aplay --device=my_mod_test out.pcm` I get the error message ALSA lib dlmisc.c:254:(snd1_dlobj_cache_get) Cannot open shared library /usr/lib/i386-linux-gnu/alsa-lib/libasound_module_pcm_my_mod.so aplay: main:682: audio open error: No such device or address But this file actually does exist, I just copied it there. Any idea why this happens? Thanks much, Brandon

2 3

[alsa-devel] Enable more HP machine to enable mute led
by Kailang 18 Jun '14

18 Jun '14

Hi Takashi, Add more HP machine to support mute led functions. Thanks. BR, Kailang

2 4

[alsa-devel] Possible regression caused by "ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr"
by Simon Horman 18 Jun '14

18 Jun '14

Hi Morimoto-san, Hi All, ad32d0c7b0e99343 ("ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr"), which is included in v3.16-rc1, appears to cause the following panic on the bockw board when booted using the bockw_defconfig. Booting Linux on physical CPU 0x0 Linux version 3.15.0-rc1-00017-gad32d0c (horms(a)ayumi.isobedori.kobe.vergenet.net) (gcc version 4.6.3 (GCC) ) #123 Tue Jun 17 20:56:53 JST 2014 CPU: ARMv7 Processor [412fc094] revision 4 (ARMv7), cr=10c53c7d CPU: PIPT / VIPT nonaliasing data cache, VIPT aliasing instruction cache Machine model: bockw debug: ignoring loglevel setting. Memory policy: Data cache writeback On node 0 totalpages: 65536 free_area_init_node: node 0, pgdat c03f7f88, node_mem_map cfdfb000 Normal zone: 512 pages used for memmap Normal zone: 0 pages reserved Normal zone: 65536 pages, LIFO batch:15 DT missing boot CPU MPIDR[23:0], fall back to default cpu_logical_map CPU: All CPU(s) started in SVC mode. pcpu-alloc: s0 r0 d32768 u32768 alloc=1*32768 pcpu-alloc: [0] 0 Built 1 zonelists in Zone order, mobility grouping on. Total pages: 65024 Kernel command line: console=ttySC0,115200 ignore_loglevel ip=dhcp root=/dev/nfs rw PID hash table entries: 1024 (order: 0, 4096 bytes) Dentry cache hash table entries: 32768 (order: 5, 131072 bytes) Inode-cache hash table entries: 16384 (order: 4, 65536 bytes) Memory: 255600K/262144K available (3123K kernel code, 146K rwdata, 628K rodata, 130K init, 165K bss, 6544K reserved, 0K highmem) Virtual kernel memory layout: vector : 0xffff0000 - 0xffff1000 ( 4 kB) fixmap : 0xfff00000 - 0xfffe0000 ( 896 kB) vmalloc : 0xd0800000 - 0xff000000 ( 744 MB) lowmem : 0xc0000000 - 0xd0000000 ( 256 MB) pkmap : 0xbfe00000 - 0xc0000000 ( 2 MB) .text : 0xc0008000 - 0xc03b1e14 (3752 kB) .init : 0xc03b2000 - 0xc03d2a38 ( 131 kB) .data : 0xc03d4000 - 0xc03f8a20 ( 147 kB) .bss : 0xc03f8a20 - 0xc0422100 ( 166 kB) NR_IRQS:16 nr_irqs:16 16 GIC CPU mask not found - kernel will fail to boot. GIC CPU mask not found - kernel will fail to boot. sched_clock: 32 bits at 1kHz, resolution 976562ns, wraps every 2097152000000000ns Calibrating delay loop (skipped) preset value.. 532.00 BogoMIPS (lpj=260000) pid_max: default: 32768 minimum: 301 Mount-cache hash table entries: 1024 (order: 0, 4096 bytes) Mountpoint-cache hash table entries: 1024 (order: 0, 4096 bytes) CPU: Testing write buffer coherency: ok Setting up static identity map for 0x60312b28 - 0x60312b5c devtmpfs: initialized VFP support v0.3: implementor 41 architecture 3 part 30 variant 9 rev 4 pinctrl core: initialized pinctrl subsystem regulator-dummy: no parameters NET: Registered protocol family 16 DMA: preallocated 256 KiB pool for atomic coherent allocations renesas_intc_irqpin renesas_intc_irqpin: driving 4 irqs L310 cache controller enabled l2x0: 16 ways, CACHE_ID 0x410000c6, AUX_CTRL 0x42460000, Cache size: 1024 kB sh-pfc pfc-r8a7778: r8a7778_pfc support registered fixed-dummy: no parameters fixed-3.3V: at 3300 mV SCSI subsystem initialized usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb Linux video capture interface: v2.00 sh_tmu sh_tmu.0: used for clock events sh_tmu sh_tmu.0: used for periodic clock events sh_tmu sh_tmu.1: used as clock source Advanced Linux Sound Architecture Driver Initialized. Switched to clocksource sh_tmu.1 NET: Registered protocol family 2 TCP established hash table entries: 2048 (order: 1, 8192 bytes) TCP bind hash table entries: 2048 (order: 1, 8192 bytes) TCP: Hash tables configured (established 2048 bind 2048) TCP: reno registered UDP hash table entries: 256 (order: 0, 4096 bytes) UDP-Lite hash table entries: 256 (order: 0, 4096 bytes) NET: Registered protocol family 1 RPC: Registered named UNIX socket transport module. RPC: Registered udp transport module. RPC: Registered tcp transport module. RPC: Registered tcp NFSv4.1 backchannel transport module. futex hash table entries: 256 (order: -1, 3072 bytes) NFS: Registering the id_resolver key type Key type id_resolver registered Key type id_legacy registered nfs4filelayout_init: NFSv4 File Layout Driver Registering... Block layer SCSI generic (bsg) driver version 0.4 loaded (major 253) io scheduler noop registered io scheduler deadline registered (default) SuperH (H)SCI(F) driver initialized sh-sci.0: ttySC0 at MMIO 0xffe40000 (irq = 102, base_baud = 0) is a scif console [ttySC0] enabled sh-sci.1: ttySC1 at MMIO 0xffe41000 (irq = 103, base_baud = 0) is a scif sh-sci.2: ttySC2 at MMIO 0xffe42000 (irq = 104, base_baud = 0) is a scif sh-sci.3: ttySC3 at MMIO 0xffe43000 (irq = 105, base_baud = 0) is a scif sh-sci.4: ttySC4 at MMIO 0xffe44000 (irq = 106, base_baud = 0) is a scif sh-sci.5: ttySC5 at MMIO 0xffe45000 (irq = 107, base_baud = 0) is a scif m25p80 spi0.0: found at26df081a, expected s25fl008k m25p80 spi0.0: at26df081a (1024 Kbytes) Creating 1 MTD partitions on "m25p80": 0x000000000000-0x000000100000 : "data(spi)" libphy: smsc911x-mdio: probed smsc911x smsc911x eth0: attached PHY driver [Generic PHY] (mii_bus:phy_addr=smsc911x-fffffff:01, irq=-1) smsc911x smsc911x eth0: MAC Address: 2e:09:0a:00:2f:04 ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) Driver ehci-platform: EHCI generic platform driver ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver ohci-platform: OHCI generic platform driver usbcore: registered new interface driver usb-storage rtc-rx8581 0-0051: chip found, driver version 0.1 rtc-rx8581 0-0051: low voltage detected, date/time is not reliable. rtc-rx8581 0-0051: rtc core: registered rtc-rx8581 as rtc0 i2c-rcar i2c-rcar.0: probed i2c-rcar i2c-rcar.1: probed i2c-rcar i2c-rcar.2: probed i2c-rcar i2c-rcar.3: probed soc-camera-pdrv soc-camera-pdrv.0: Probing soc-camera-pdrv.0 ml86v7667 0-0041: chip found @ 0x41 (i2c-rcar) soc-camera-pdrv soc-camera-pdrv.1: Probing soc-camera-pdrv.1 ml86v7667 0-0043: chip found @ 0x43 (i2c-rcar) sh_mobile_sdhi.0 supply vmmc not found, using dummy regulator sh_mobile_sdhi sh_mobile_sdhi.0: Failed getting OCR mask: -22 sh_mobile_sdhi sh_mobile_sdhi.0: could not set regulator OCR (-22) sh_mobile_sdhi sh_mobile_sdhi.0: mmc0 base at 0xffe4c000 clock rate 66 MHz sh_mmcif sh_mmcif: driver version 2010-04-28 asoc-simple-card asoc-simple-card.0: ASoC: CPU DAI rsnd-dai.0 not registered platform asoc-simple-card.0: Driver asoc-simple-card requests probe deferral asoc-simple-card asoc-simple-card.1: ASoC: CPU DAI rsnd-dai.1 not registered platform asoc-simple-card.1: Driver asoc-simple-card requests probe deferral asoc-simple-card asoc-simple-card.2: ASoC: CPU DAI rsnd-dai.2 not registered platform asoc-simple-card.2: Driver asoc-simple-card requests probe deferral asoc-simple-card asoc-simple-card.3: ASoC: CPU DAI rsnd-dai.3 not registered platform asoc-simple-card.3: Driver asoc-simple-card requests probe deferral asoc-simple-card asoc-simple-card.4: ASoC: CPU DAI rsnd-dai.4 not registered platform asoc-simple-card.4: Driver asoc-simple-card requests probe deferral Unable to handle kernel NULL pointer dereference at virtual address 00000000 pgd = c0004000 [00000000] *pgd=00000000 Internal error: Oops: 5 [#1] ARM CPU: 0 PID: 1 Comm: swapper Not tainted 3.15.0-rc1-00017-gad32d0c #123 task: cf827a40 ti: cf828000 task.ti: cf828000 PC is at rsnd_gen_dma_addr+0x40/0x1dc LR is at rsnd_gen_dma_addr+0x38/0x1dc pc : [<c026402c>] lr : [<c0264024>] psr: 60000113 sp : cf829c38 ip : 00000003 fp : cf829cdc r10: cf8e5600 r9 : 00000000 r8 : cf8e5610 r7 : 00000001 r6 : cf9a3e0c r5 : cf9a7c24 r4 : cf829ce8 r3 : 00000003 r2 : 00000000 r1 : 00000200 r0 : 00000000 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c53c7d Table: 60004059 DAC: 00000015 Process swapper (pid: 1, stack limit = 0xcf828238) Stack: (0xcf829c38 to 0xcf82a000) 9c20: c0088644 c00886f8 9c40: cf9b01e0 c032a990 00000000 cf9b5990 cf829c8c cf829c60 c016f7c8 c016e798 9c60: c016f70c cf9b01e0 c03ea34c c03ea35c 0000001f c016fab8 00200200 cf829d2c 9c80: cf829ca4 cf829c90 c016d628 c016f718 00000003 cf9b04d0 cf829cdc cf829ca8 9ca0: c016dc04 c0311854 00000000 cf9b01e0 cf829cd4 cf9a3e0c 00000001 cf9b5990 9cc0: cf829d1c cf8e5610 00000000 0000001f cf829d5c cf829ce0 c0263348 c0263ff8 9ce0: 0000001f ffffffff cf829d14 cf8e1f80 cf9a3630 cf9a7d90 00000060 00000004 9d00: 00000000 c03e0fd8 c03e1098 cf9a7dc8 cf829d34 cf829d20 c0262c14 c0007874 9d20: cf8e1f80 cf9a3630 cf829d7c 00000200 c0262e98 cf8e5600 c03e0fb4 00000005 9d40: cf9a7c1c 00000000 00000000 cf9b5990 cf829d7c cf829d60 c0265d08 c0263208 9d60: c03f40e0 cf9a7c18 cf9a7c10 cf9a7c10 cf829dcc cf829d80 c0262850 c0265c90 9d80: cf829db4 cf8e5610 c02641c8 c0265f6c c0264c7c c026663c c0265630 c0262cd8 9da0: 00000000 cf8e5610 c03f3d14 c0416bb8 00000000 00000000 c03d29e0 c03f3d14 9dc0: cf829de4 cf829dd0 c018fd88 c026274c cf8e5610 c0416bb0 cf829e0c cf829de8 9de0: c018ecb8 c018fd74 cf8e5610 c03f3d14 cf8e5644 00000000 c03b24c0 00000000 9e00: cf829e2c cf829e10 c018ee5c c018ebac cf8e8e3c c03f3d14 00000000 c018edc8 9e20: cf829e54 cf829e30 c018d32c c018edd4 cf818ecc cf8e8e30 cf9b5a34 c03f3d14 9e40: cf9b5a00 c03eb348 cf829e64 cf829e58 c018e990 c018d2dc cf829e8c cf829e68 9e60: c018db54 c018e978 c0380c40 cf829e78 c03f3d14 00000006 c03cd314 c03c69dc 9e80: cf829ea4 cf829e90 c018f27c c018da80 c03cd334 00000006 cf829eb4 cf829ea8 9ea0: c0190590 c018f208 cf829ec4 cf829eb8 c03c69f4 c019054c cf829f54 cf829ec8 9ec0: c03b2af4 c03c69e8 c040c9b4 c03f8a40 00000000 c03b0410 cf829f04 cf829ee8 9ee0: cf829f0c cf829ef0 cf829f00 cf829ef8 c03b24c0 c03b0410 cfdf7a3c 0000005c 9f00: cf829f54 cf829f10 c0030398 c03b24cc 00000000 00000006 00000006 c03afe50 9f20: 00000000 c039d11c cf829f54 c03cd334 00000006 c03cd314 c03f8a40 c03b24c0 9f40: c03d29e0 0000005c cf829f94 cf829f58 c03b2c98 c03b2a60 00000006 00000006 9f60: c03b24c0 3e9b7ef7 dfdf4fed 00000000 c030ce98 00000000 00000000 00000000 9f80: 00000000 00000000 cf829fac cf829f98 c030cea8 c03b2bb4 00000000 00000000 9fa0: 00000000 cf829fb0 c000e298 c030cea4 00000000 00000000 00000000 00000000 9fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9fe0: 00000000 00000000 00000000 00000000 00000013 00000000 5e3decfa e74e5bde Backtrace: [<c0263fec>] (rsnd_gen_dma_addr) from [<c0263348>] (rsnd_dma_init+0x14c/0x24c) r10:0000001f r9:00000000 r8:cf8e5610 r7:cf829d1c r6:cf9b5990 r5:00000001 r4:cf9a3e0c [<c02631fc>] (rsnd_dma_init) from [<c0265d08>] (rsnd_ssi_dma_probe+0x84/0xa4) r10:cf9b5990 r9:00000000 r8:00000000 r7:cf9a7c1c r6:00000005 r5:c03e0fb4 r4:cf8e5600 [<c0265c84>] (rsnd_ssi_dma_probe) from [<c0262850>] (rsnd_probe+0x110/0x258) r6:cf9a7c10 r5:cf9a7c10 r4:cf9a7c18 r3:c03f40e0 [<c0262740>] (rsnd_probe) from [<c018fd88>] (platform_drv_probe+0x20/0x50) r10:c03f3d14 r9:c03d29e0 r8:00000000 r7:00000000 r6:c0416bb8 r5:c03f3d14 r4:cf8e5610 [<c018fd68>] (platform_drv_probe) from [<c018ecb8>] (driver_probe_device+0x118/0x228) r5:c0416bb0 r4:cf8e5610 [<c018eba0>] (driver_probe_device) from [<c018ee5c>] (__driver_attach+0x94/0x98) r10:00000000 r8:c03b24c0 r7:00000000 r6:cf8e5644 r5:c03f3d14 r4:cf8e5610 [<c018edc8>] (__driver_attach) from [<c018d32c>] (bus_for_each_dev+0x5c/0x90) r6:c018edc8 r5:00000000 r4:c03f3d14 r3:cf8e8e3c [<c018d2d0>] (bus_for_each_dev) from [<c018e990>] (driver_attach+0x24/0x28) r6:c03eb348 r5:cf9b5a00 r4:c03f3d14 [<c018e96c>] (driver_attach) from [<c018db54>] (bus_add_driver+0xe0/0x1c8) [<c018da74>] (bus_add_driver) from [<c018f27c>] (driver_register+0x80/0xfc) r7:c03c69dc r6:c03cd314 r5:00000006 r4:c03f3d14 [<c018f1fc>] (driver_register) from [<c0190590>] (__platform_driver_register+0x50/0x64) r5:00000006 r4:c03cd334 [<c0190540>] (__platform_driver_register) from [<c03c69f4>] (rsnd_driver_init+0x18/0x20) [<c03c69dc>] (rsnd_driver_init) from [<c03b2af4>] (do_one_initcall+0xa0/0x154) [<c03b2a54>] (do_one_initcall) from [<c03b2c98>] (kernel_init_freeable+0xf0/0x1b8) r10:0000005c r9:c03d29e0 r8:c03b24c0 r7:c03f8a40 r6:c03cd314 r5:00000006 r4:c03cd334 [<c03b2ba8>] (kernel_init_freeable) from [<c030cea8>] (kernel_init+0x10/0xec) r10:00000000 r9:00000000 r8:00000000 r7:00000000 r6:00000000 r5:c030ce98 r4:00000000 [<c030ce98>] (kernel_init) from [<c000e298>] (ret_from_fork+0x14/0x3c) r4:00000000 r3:00000000 Code: e1a07003 ebfcaf68 e3a01c02 e3a02000 (e5908000) ---[ end trace 421d78826f73f05e ]--- Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b ---[ end Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b

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[alsa-devel] [PATCH 0/4] ASoC: fsi/rsnd: cyclic transfer support
by Kuninori Morimoto 18 Jun '14

18 Jun '14

Hi Mark Current Renesas sound driver (= FSI/RSND) is not using ALSA common DMAEngine method for some reasons. The 1st reason was that Renesas DMA didn't support cyclic transfer mode. But now, it is supported ! These patches support cyclic transfer for FSI/RSND It needs this branch git://git.infradead.org/users/vkoul/slave-dma.git :: next especially dfbb85cab5f0819d0424a3637b03e7892704fa42 (DMA: shdma: add cyclic transfer support) These are based on mark/topic/rcar branch But, #4 patch needs mark/fix/rcar merge Kuninori Morimoto (4): ASoC: fsi: use SNDRV_DMA_TYPE_DEV for sound buffer ASoC: fsi: add fsi_pointer_update() for common pointer method ASoC: fsi: use dmaengine_prep_dma_cyclic() for DMA transfer ASoC: rsnd: use dmaengine_prep_dma_cyclic() instead of original method arch/arm/mach-shmobile/board-armadillo800eva.c | 4 + arch/arm/mach-shmobile/board-kzm9g.c | 2 + arch/arm/mach-shmobile/board-mackerel.c | 4 + arch/sh/boards/mach-ecovec24/setup.c | 2 + sound/soc/sh/fsi.c | 189 +++++++----------------- sound/soc/sh/rcar/core.c | 76 +++------- sound/soc/sh/rcar/rsnd.h | 4 - 7 files changed, 82 insertions(+), 199 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/2] ASoC: rsnd: add BUSIF support for DMA
by Kuninori Morimoto 18 Jun '14

18 Jun '14

Hi Mark These patches are for R-Car sound BUSIF support for DMA transfer. If we use BUSIF via DMA, it will work as FIFO. It can reduce underrun error. Kuninori Morimoto (2): ASoC: rsnd: fixup loop exit timing of dma name search ASoC: rsnd: SSI + DMA can select BUSIF .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/core.c | 26 +++++--- sound/soc/sh/rcar/gen.c | 64 +++++++++++++------- sound/soc/sh/rcar/rsnd.h | 10 ++- sound/soc/sh/rcar/src.c | 37 ++++++++--- sound/soc/sh/rcar/ssi.c | 33 +++++++++- 7 files changed, 126 insertions(+), 46 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/3] Auvitek au0828 fixups
by Mauro Carvalho Chehab 18 Jun '14

18 Jun '14

Testing auvitek au0828-based devices with xawtv caused lots of overruns, as the period_bytes is set to a too low value, with causes troubles when syncing the audio capture from au0828 card with the audio playback, at the main audio card. It turns that we need a period_bytes size that it is big enough to receive data from two consecutive URB intervals. While here, also add the missing USB IDs from other au0828-based devices. Mauro Carvalho Chehab (3): sound: Add a quirk to enforce period_bytes sound: simplify au0828 quirk table sound: Update au0828 quirks table sound/usb/card.h | 1 + sound/usb/pcm.c | 34 ++++++++++ sound/usb/quirks-table.h | 167 ++++++++++++----------------------------------- sound/usb/quirks.c | 14 ++-- sound/usb/stream.c | 1 + sound/usb/usbaudio.h | 3 +- 6 files changed, 88 insertions(+), 132 deletions(-) -- 1.9.3

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[alsa-devel] Trying to enable digital audio output - Intel HDA - AD1983
by Scott D'Vileskis 18 Jun '14

18 Jun '14

Hi-- I have a Lenovo M55 SFF desktop I use for a media center computer (XBMCbuntu). It has the AD1983 chip on it. From the factory, it had 2 ports in the back (mic + speaker) , and 2 ports in the front (mic + speaker). It also has an on-board speaker. It never shipped with a SPDIF output, but I soldered a wire to the AD1983 and installed an RCA jack on the back of the case. Earlier versions of XBMC shipped with a different version of alsa, and I was able to use this custom SPDIF output without issue for the last year or two, but the current version only enables the 'proper' inputs/outputs I have tried editing the module options and specifying several different models, but the documentation doesn't list any 'model' options for the AD1983. I suspect I need to overwrite the pin configs. cat init_pin_configs: 0x05 0x01014010 0x06 0x0221401f 0x07 0x991301f0 0x08 0x02a19020 0x09 0x0181302e 0x0a 0x414511f0 [ 2421.907510] hda-codec: reconfiguring [ 2421.980324] autoconfig: line_outs=1 (0x5/0x0/0x0/0x0/0x0) type:line [ 2421.980328] speaker_outs=1 (0x7/0x0/0x0/0x0/0x0) [ 2421.980330] hp_outs=1 (0x6/0x0/0x0/0x0/0x0) [ 2421.980332] mono: mono_out=0x0 [ 2421.980333] inputs: [ 2421.980335] Mic=0x8 [ 2421.980336] Line=0x9 [ 2421.981858] input: HDA Intel Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input34 [ 2421.984829] input: HDA Intel Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input33 [ 2421.985848] input: HDA Intel Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input32 Does anyone have any guidance for how I may re-enable the SPDIF on the AD1983 when my BIOS disables it by default? Where else should I look for information on pin configs? Thanks!

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[alsa-devel] [PATCH 1/3] ASoC: Add SND_SOC_BYTES_EXT control type
by Charles Keepax 17 Jun '14

17 Jun '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..92c1b99 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,6 +256,13 @@ ((unsigned long)&(struct soc_bytes) \ {.base = xbase, .num_regs = xregs }) } +#define SND_SOC_BYTES_EXT(xname, xbase, xregs, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = xhandler_get, \ + .put = xhandler_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) \ + {.base = xbase, .num_regs = xregs }) } + #define SND_SOC_BYTES_MASK(xname, xbase, xregs, xmask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ -- 1.7.2.5

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[alsa-devel] alsa-lib: pcm_plug: fix float conversion for user specified ttable
by Sergey 17 Jun '14

17 Jun '14

Move custom ttable with equal channels case from a separate ttable_last exception into a common plugins insertion loop. Fixes plug with ttable for float pcms (jack, ladspa). Example: aplay -fFLOAT_LE /dev/zero pcm.!default { type plug slave.pcm { type null } ttable.0.0 1 } --- src/pcm/pcm_plug.c | 36 +++++++++++------------------------- 1 file changed, 11 insertions(+), 25 deletions(-) diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index ede9c15..6d0454f 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -53,7 +53,7 @@ typedef struct { const snd_config_t *rate_converter; enum snd_pcm_plug_route_policy route_policy; snd_pcm_route_ttable_entry_t *ttable; - int ttable_ok, ttable_last; + int ttable_ok; unsigned int tt_ssize, tt_cused, tt_sused; } snd_pcm_plug_t; @@ -380,7 +380,7 @@ static int snd_pcm_plug_change_channels(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm snd_pcm_route_ttable_entry_t *ttable; int err; if (clt->channels == slv->channels && - (!plug->ttable || !plug->ttable_last)) + (!plug->ttable || plug->ttable_ok)) return 0; if (clt->rate != slv->rate && clt->channels > slv->channels) @@ -485,13 +485,15 @@ static int snd_pcm_plug_change_format(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm_p /* No conversion is needed */ if (clt->format == slv->format && clt->rate == slv->rate && - clt->channels == slv->channels) + clt->channels == slv->channels && + (!plug->ttable || plug->ttable_ok)) return 0; if (snd_pcm_format_linear(slv->format)) { /* Conversion is done in another plugin */ if (clt->rate != slv->rate || - clt->channels != slv->channels) + clt->channels != slv->channels || + (plug->ttable && !plug->ttable_ok)) return 0; cfmt = clt->format; switch (clt->format) { @@ -525,7 +527,8 @@ static int snd_pcm_plug_change_format(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm_p if (snd_pcm_format_linear(clt->format)) { cfmt = clt->format; f = snd_pcm_lfloat_open; - } else if (clt->rate != slv->rate || clt->channels != slv->channels) { + } else if (clt->rate != slv->rate || clt->channels != slv->channels || + (plug->ttable && !plug->ttable_ok)) { cfmt = SND_PCM_FORMAT_S16; f = snd_pcm_lfloat_open; } else @@ -641,11 +644,12 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, }; snd_pcm_plug_params_t p = *slave; unsigned int k = 0; - plug->ttable_ok = plug->ttable_last = 0; + plug->ttable_ok = 0; while (client->format != p.format || client->channels != p.channels || client->rate != p.rate || - client->access != p.access) { + client->access != p.access || + (plug->ttable && !plug->ttable_ok)) { snd_pcm_t *new; int err; if (k >= sizeof(funcs)/sizeof(*funcs)) @@ -660,24 +664,6 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, } k++; } -#ifdef BUILD_PCM_PLUGIN_ROUTE - /* it's exception, user specified ttable, but no reduction/expand */ - if (plug->ttable && !plug->ttable_ok) { - snd_pcm_t *new; - int err; - plug->ttable_last = 1; - err = snd_pcm_plug_change_channels(pcm, &new, client, &p); - if (err < 0) { - snd_pcm_plug_clear(pcm); - return err; - } - assert(err); - assert(plug->ttable_ok); - plug->gen.slave = new; - pcm->fast_ops = new->fast_ops; - pcm->fast_op_arg = new->fast_op_arg; - } -#endif return 0; } -- 1.7.10.4

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