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[alsa-devel] [PATCH V2 0/2] ASoC: max98090/max98095: Add master clock handing
by Tushar Behera 26 May '14

26 May '14

These CODECs are used on Snow/Peach-pit boards and the master clock needs to be set at 24MHz. Adding support in respective codec drivers so that we can update the master clock frequency from sound card driver through set_sysclk. Changes for V2: 1. Added PROBE_DEFER if clock driver has not yet been probed 2. Moved clk_prepare_enable from .probe to .set_bias_level 3. Moved clk_disable_unprepare from .remove to .set_bias_level Tushar Behera (2): ASoC: max98090: Add master clock handling ASoC: max98095: Add master clock handling .../devicetree/bindings/sound/max98090.txt | 6 +++++ .../devicetree/bindings/sound/max98095.txt | 6 +++++ sound/soc/codecs/max98090.c | 23 +++++++++++++++++++ sound/soc/codecs/max98090.h | 1 + sound/soc/codecs/max98095.c | 24 ++++++++++++++++++++ 5 files changed, 60 insertions(+) -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ALSA: hda - Pop noises fix for XPS13 9333
by Gabriele Mazzotta 26 May '14

26 May '14

When headphones are plugged in, force AFG and node 0x02 ("Headphone Playback Volume") to D0 to avoid pop noises. Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 32 +++++++++++++++++++++++++++++++- 1 file changed, 31 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce28dc4..b7a435c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3726,6 +3726,29 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec, } } +static unsigned int alc_power_filter_xps13(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) +{ + struct alc_spec *spec = codec->spec; + + /* Avoid pop noises when headphones are plugged in */ + if (spec->gen.hp_jack_present) + if (nid == codec->afg || nid == 0x02) + return AC_PWRST_D0; + return power_state; +} + +static void alc_fixup_dell_xps13(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PROBE) { + struct alc_spec *spec = codec->spec; + spec->shutup = alc_no_shutup; + codec->power_filter = alc_power_filter_xps13; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5068,6 +5091,7 @@ enum { ALC662_FIXUP_BASS_1A_CHMAP, ALC668_FIXUP_AUTO_MUTE, ALC668_FIXUP_DELL_DISABLE_AAMIX, + ALC668_FIXUP_DELL_XPS13, }; static const struct hda_fixup alc662_fixups[] = { @@ -5228,6 +5252,12 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_DELL_XPS13] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_dell_xps13, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_DISABLE_AAMIX + }, [ALC668_FIXUP_DELL_DISABLE_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -5285,7 +5315,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_XPS13), SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), -- 2.0.0.rc4

1 0

[alsa-devel] [PATCH] ASoC: alc5623: Fix Kconfig dependency
by Takashi Iwai 26 May '14

26 May '14

Add "depends on I2C" to shut up the build errors from randconfig. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c59943a19f94..3f7bbe61326d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -274,6 +274,7 @@ config SND_SOC_AK5386 config SND_SOC_ALC5623 tristate "Realtek ALC5623 CODEC" + depends on I2C config SND_SOC_ALC5632 tristate -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: omap-pcm: Move omap-pcm under include/sound
by Jyri Sarha 26 May '14

26 May '14

Make including the omap-pcm.h outside sound/soc/omap more convenient. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- It would be nice to get this applied at ASAP for OMAP HDMI audio rework patches depend on this and it would help a lot if this would already be there when applying to OMAPDSS side patches. include/sound/omap-pcm.h | 30 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/omap/omap-dmic.c | 2 +- sound/soc/omap/omap-hdmi.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/omap-pcm.h | 30 ------------------------------ 7 files changed, 35 insertions(+), 35 deletions(-) create mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.h diff --git a/include/sound/omap-pcm.h b/include/sound/omap-pcm.h new file mode 100644 index 0000000..c1d2f31 --- /dev/null +++ b/include/sound/omap-pcm.h @@ -0,0 +1,30 @@ +/* + * omap-pcm.h - OMAP PCM driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Peter Ujfalusi <peter.ujfalusi(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 __OMAP_PCM_H__ +#define __OMAP_PCM_H__ + +#if IS_ENABLED(CONFIG_SND_OMAP_SOC) +int omap_pcm_platform_register(struct device *dev); +#else +static inline int omap_pcm_platform_register(struct device *dev) +{ + return 0; +} +#endif /* CONFIG_SND_OMAP_SOC */ + +#endif /* __OMAP_PCM_H__ */ diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 14058dc..9afb146 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -33,10 +33,10 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <sound/omap-pcm.h> #include "davinci-pcm.h" #include "davinci-mcasp.h" -#include "../omap/omap-pcm.h" #define MCASP_MAX_AFIFO_DEPTH 64 diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 53da041..6925d71 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -40,9 +40,9 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <sound/omap-pcm.h> #include "omap-dmic.h" -#include "omap-pcm.h" struct omap_dmic { struct device *dev; diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 537a1ec..eb9c392 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -34,9 +34,9 @@ #include <sound/asoundef.h> #include <sound/dmaengine_pcm.h> #include <video/omapdss.h> +#include <sound/omap-pcm.h> #include "omap-hdmi.h" -#include "omap-pcm.h" #define DRV_NAME "omap-hdmi-audio-dai" diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 71d2266..efe2cd6 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -34,11 +34,11 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <sound/omap-pcm.h> #include <linux/platform_data/asoc-ti-mcbsp.h> #include "mcbsp.h" #include "omap-mcbsp.h" -#include "omap-pcm.h" #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index d8ebb52..f0e2ebe 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -40,9 +40,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <sound/omap-pcm.h> #include "omap-mcpdm.h" -#include "omap-pcm.h" struct mcpdm_link_config { u32 link_mask; /* channel mask for the direction */ diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h deleted file mode 100644 index c1d2f31..0000000 --- a/sound/soc/omap/omap-pcm.h +++ /dev/null @@ -1,30 +0,0 @@ -/* - * omap-pcm.h - OMAP PCM driver - * - * Copyright (C) 2014 Texas Instruments, Inc. - * - * Author: Peter Ujfalusi <peter.ujfalusi(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 __OMAP_PCM_H__ -#define __OMAP_PCM_H__ - -#if IS_ENABLED(CONFIG_SND_OMAP_SOC) -int omap_pcm_platform_register(struct device *dev); -#else -static inline int omap_pcm_platform_register(struct device *dev) -{ - return 0; -} -#endif /* CONFIG_SND_OMAP_SOC */ - -#endif /* __OMAP_PCM_H__ */ -- 1.7.9.5

3 2

[alsa-devel] [PATCH 1/2] ASoC: jack: Basic GPIO descriptor conversion
by Jarkko Nikula 26 May '14

26 May '14

This patch does basic GPIO descriptor conversion to soc-jack. Even the GPIOs are still passed and requested using legacy GPIO numbers the driver internals are converted to use GPIO descriptor API. Motivation for this is to prepare soc-jack so that it will allow registering jack GPIO pins using both GPIO descriptors and legacy GPIO numbers. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 19 +++++++++++-------- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index dcdcc95efd1d..e9ec645911d9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -607,6 +607,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack *jack; struct delayed_work work; + struct gpio_desc *desc; void *data; int (*jack_status_check)(void *data); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b903f822d1b2..7203842fea66 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,7 @@ #include <sound/jack.h> #include <sound/soc.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/interrupt.h> #include <linux/workqueue.h> #include <linux/delay.h> @@ -240,7 +241,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) int enable; int report; - enable = gpio_get_value_cansleep(gpio->gpio); + enable = gpiod_get_value_cansleep(gpio->desc); if (gpio->invert) enable = !enable; @@ -314,14 +315,16 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (ret) goto undo; - ret = gpio_direction_input(gpios[i].gpio); + gpios[i].desc = gpio_to_desc(gpios[i].gpio); + + ret = gpiod_direction_input(gpios[i].desc); if (ret) goto err; INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; - ret = request_any_context_irq(gpio_to_irq(gpios[i].gpio), + ret = request_any_context_irq(gpiod_to_irq(gpios[i].desc), gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, @@ -331,7 +334,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, goto err; if (gpios[i].wake) { - ret = irq_set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); + ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1); if (ret != 0) dev_err(jack->codec->dev, "ASoC: " "Failed to mark GPIO %d as wake source: %d\n", @@ -339,7 +342,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, } /* Expose GPIO value over sysfs for diagnostic purposes */ - gpio_export(gpios[i].gpio, false); + gpiod_export(gpios[i].desc, false); /* Update initial jack status */ schedule_delayed_work(&gpios[i].work, @@ -372,10 +375,10 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, int i; for (i = 0; i < count; i++) { - gpio_unexport(gpios[i].gpio); - free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); + gpiod_unexport(gpios[i].desc); + free_irq(gpiod_to_irq(gpios[i].desc), &gpios[i]); cancel_delayed_work_sync(&gpios[i].work); - gpio_free(gpios[i].gpio); + gpiod_put(gpios[i].desc); gpios[i].jack = NULL; } } -- 2.0.0.rc2

2 3

[alsa-devel] [PATCH] ASoC: tegra: free jack GPIOs before the sound card is freed
by Stephen Warren 26 May '14

26 May '14

From: Stephen Warren <swarren(a)nvidia.com> snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, gGuard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. This change fixes all Tegra machine drivers. By code inspection, I believe some non-Tegra machine drivers have the same issue. I'll send a patch for that separately, once this is reviewed. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Should this be Cc: stable? --- sound/soc/tegra/tegra_alc5632.c | 16 +++++++++++++--- sound/soc/tegra/tegra_max98090.c | 16 +++++++++++++--- sound/soc/tegra/tegra_rt5640.c | 16 +++++++++++++--- sound/soc/tegra/tegra_wm8903.c | 11 ++++++----- 4 files changed, 45 insertions(+), 14 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c61ea3a1030f..02734bd4f09b 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -125,6 +125,18 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_alc5632_card_remove(struct snd_soc_card *card) +{ + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1, + &tegra_alc5632_hp_jack_gpio); + } + + return 0; +} + static struct snd_soc_dai_link tegra_alc5632_dai = { .name = "ALC5632", .stream_name = "ALC5632 PCM", @@ -139,6 +151,7 @@ static struct snd_soc_dai_link tegra_alc5632_dai = { static struct snd_soc_card snd_soc_tegra_alc5632 = { .name = "tegra-alc5632", .owner = THIS_MODULE, + .remove = tegra_alc5632_card_remove, .dai_link = &tegra_alc5632_dai, .num_links = 1, .controls = tegra_alc5632_controls, @@ -223,9 +236,6 @@ static int tegra_alc5632_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); - snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1, - &tegra_alc5632_hp_jack_gpio); - snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 0283cfb7c031..ce73e1f62c4b 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -145,6 +145,18 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_max98090_card_remove(struct snd_soc_card *card) +{ + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_max98090_hp_jack, 1, + &tegra_max98090_hp_jack_gpio); + } + + return 0; +} + static struct snd_soc_dai_link tegra_max98090_dai = { .name = "max98090", .stream_name = "max98090 PCM", @@ -158,6 +170,7 @@ static struct snd_soc_dai_link tegra_max98090_dai = { static struct snd_soc_card snd_soc_tegra_max98090 = { .name = "tegra-max98090", .owner = THIS_MODULE, + .remove = tegra_max98090_card_remove, .dai_link = &tegra_max98090_dai, .num_links = 1, .controls = tegra_max98090_controls, @@ -241,9 +254,6 @@ static int tegra_max98090_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); - snd_soc_jack_free_gpios(&tegra_max98090_hp_jack, 1, - &tegra_max98090_hp_jack_gpio); - snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 4511c5a875ec..4feb16a99e02 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -128,6 +128,18 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_rt5640_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5640_hp_jack, 1, + &tegra_rt5640_hp_jack_gpio); + } + + return 0; +} + static struct snd_soc_dai_link tegra_rt5640_dai = { .name = "RT5640", .stream_name = "RT5640 PCM", @@ -141,6 +153,7 @@ static struct snd_soc_dai_link tegra_rt5640_dai = { static struct snd_soc_card snd_soc_tegra_rt5640 = { .name = "tegra-rt5640", .owner = THIS_MODULE, + .remove = tegra_rt5640_card_remove, .dai_link = &tegra_rt5640_dai, .num_links = 1, .controls = tegra_rt5640_controls, @@ -224,9 +237,6 @@ static int tegra_rt5640_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); - snd_soc_jack_free_gpios(&tegra_rt5640_hp_jack, 1, - &tegra_rt5640_hp_jack_gpio); - snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 4ac73730d79a..0939661df60b 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -206,6 +206,12 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1, + &tegra_wm8903_hp_jack_gpio); + } wm8903_mic_detect(codec, NULL, 0, 0); @@ -228,9 +234,7 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .owner = THIS_MODULE, .dai_link = &tegra_wm8903_dai, .num_links = 1, - .remove = tegra_wm8903_remove, - .controls = tegra_wm8903_controls, .num_controls = ARRAY_SIZE(tegra_wm8903_controls), .dapm_widgets = tegra_wm8903_dapm_widgets, @@ -368,9 +372,6 @@ static int tegra_wm8903_driver_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1, - &tegra_wm8903_hp_jack_gpio); - snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: Intel: avoid format string leak to thread name
by Kees Cook 26 May '14

26 May '14

This makes sure a format string can never get processed into the worker thread name from the device name. Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/soc/intel/sst-baytrail-ipc.c | 2 +- sound/soc/intel/sst-haswell-ipc.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 0d31dbbf4806..1b25bf168beb 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -809,7 +809,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) /* start the IPC message thread */ init_kthread_worker(&byt->kworker); byt->tx_thread = kthread_run(kthread_worker_fn, - &byt->kworker, + &byt->kworker, "%s", dev_name(byt->dev)); if (IS_ERR(byt->tx_thread)) { err = PTR_ERR(byt->tx_thread); diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index e7996b39a484..a8fd60c67341 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1735,7 +1735,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* start the IPC message thread */ init_kthread_worker(&hsw->kworker); hsw->tx_thread = kthread_run(kthread_worker_fn, - &hsw->kworker, + &hsw->kworker, "%s", dev_name(hsw->dev)); if (IS_ERR(hsw->tx_thread)) { ret = PTR_ERR(hsw->tx_thread); -- 1.7.9.5 -- Kees Cook Chrome OS Security

3 2

[alsa-devel] [PATCH v2 0/9] DT instantiation of sound on T5325
by Andrew Lunn 26 May '14

26 May '14

This patchset uses DT to instantiate sound on the T5325. It adds a DT binding to the codec, then removes the platform device for the codec and adds a DT node. simple-card is extended to support mlck_fs. The platform device for the audio device is then replaced with a DT node. Lastly, two _defconfig's are updated to build the needed modules. Andrew Lunn (9): ASoC: alc5623: Add device tree binding ARM: Kirkwood: Remove platform driver for codec ARM: Kirkwood: Add node for audio codec ARM: Kirkwood: DT: Add missing #sound-dai-cells property ASoC: simple-card: Support setting mclk via a fixed factor ARM: Kirkwood: t5325: Remove platform device to instantiate audio ARM: Kirkwood: t5325: Use simple card to instantiate audio ARM: mvebu_v5_defconfig: Enable sound modules needed for t5325 ARM: multi_v5_defconfig: Enable sound modules needed for t5325 .../devicetree/bindings/sound/alc5623.txt | 25 +++++++++++++ .../devicetree/bindings/sound/simple-card.txt | 2 ++ arch/arm/boot/dts/kirkwood-t5325.dts | 33 +++++++++++++++++ arch/arm/boot/dts/kirkwood.dtsi | 1 + arch/arm/configs/multi_v5_defconfig | 3 +- arch/arm/configs/mvebu_v5_defconfig | 3 +- arch/arm/mach-mvebu/Kconfig | 7 ---- arch/arm/mach-mvebu/Makefile | 1 - arch/arm/mach-mvebu/board-t5325.c | 41 ---------------------- arch/arm/mach-mvebu/board.h | 6 ---- arch/arm/mach-mvebu/kirkwood.c | 3 -- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/alc5623.c | 20 +++++++++++ sound/soc/generic/simple-card.c | 28 +++++++++++++++ 14 files changed, 114 insertions(+), 61 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5623.txt delete mode 100644 arch/arm/mach-mvebu/board-t5325.c -- 2.0.0.rc0

3 26

[alsa-devel] [PATCH] ASoC: max98090: Add NI/MI values for user pclk 19.2 MHz
by Sylwester Nawrocki 26 May '14

26 May '14

From: Chen Zhen <zhen1.chen(a)samsung.com> This patch adds the clock divisor and multiplier NI, MI values for audio sampling frequencies 44100 and 48000 Hz and PCLK 19.2 MHz. This is useful for the Odroid X2/U2 boards when the codec works in master mode and its MCLK clock is fed from the I2S CDCLK output. Signed-off-by: Chen Zhen <zhen1.chen(a)samsung.com> [s.nawrocki(a)samsung.com: edited the commit description] Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/max98090.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f7b0b37..ea44471 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1544,19 +1544,19 @@ static const int lrclk_rates[] = { }; static const int user_pclk_rates[] = { - 13000000, 13000000 + 13000000, 13000000, 19200000, 19200000, }; static const int user_lrclk_rates[] = { - 44100, 48000 + 44100, 48000, 44100, 48000, }; static const unsigned long long ni_value[] = { - 3528, 768 + 3528, 768, 441, 8 }; static const unsigned long long mi_value[] = { - 8125, 1625 + 8125, 1625, 1500, 25 }; static void max98090_configure_bclk(struct snd_soc_codec *codec) -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 26 May '14

26 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. According to Mark Brown: "Looking at the code what I'd expect to happen here is that set_bias_level() manages the cache enable, turning on cache only mode just before it turns the regulators off and restoring normal mode just after enabling them, then calling _restore_regs() after that. The resume call should just be a call to set_bias_level() then.." Do as suggested and also complete the cache array with the missing registers so that sgtl5000_restore_regs() can properly cache the all the registers it needs. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v3: - Call sgtl5000_restore_regs from set_bias_level and let set_bias_level handle the cache handling. Changes since v2: - Use regmap_update_bits sound/soc/codecs/sgtl5000.c | 134 ++++++++++++++++++++++++-------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 76 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..57bda9b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -897,6 +912,64 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec) #endif /* + * restore all sgtl5000 registers, + * since a big hole between dap and regular registers, + * we will restore them respectively. + */ +static int sgtl5000_restore_regs(struct snd_soc_codec *codec) +{ + u16 reg; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + /* restore regular registers */ + for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { + + /* These regs should restore in particular order */ + if (reg == SGTL5000_CHIP_ANA_POWER || + reg == SGTL5000_CHIP_CLK_CTRL || + reg == SGTL5000_CHIP_LINREG_CTRL || + reg == SGTL5000_CHIP_LINE_OUT_CTRL || + reg == SGTL5000_CHIP_REF_CTRL) + continue; + + regmap_update_bits(sgtl5000->regmap, reg, reg, 0); + } + + /* restore dap registers */ + for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) + regmap_update_bits(sgtl5000->regmap, reg, reg, 0); + + /* + * restore these regs according to the power setting sequence in + * sgtl5000_set_power_regs() and clock setting sequence in + * sgtl5000_set_clock(). + * + * The order of restore is: + * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after + * SGTL5000_CHIP_ANA_POWER PLL bits set + * 2. SGTL5000_CHIP_LINREG_CTRL should be set before + * SGTL5000_CHIP_ANA_POWER LINREG_D restored + * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, + * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored + */ + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_LINREG_CTRL, + SGTL5000_CHIP_LINREG_CTRL, 0); + + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP_BIT, 0); + + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_CLK_CTRL, + SGTL5000_MCLK_FREQ_PLL, 0); + + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_REF_CTRL, + SGTL5000_CHIP_REF_CTRL, 0); + + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_LINE_OUT_CTRL, + SGTL5000_CHIP_LINE_OUT_CTRL, 0); + return 0; +} + +/* * set dac bias * common state changes: * startup: @@ -926,6 +999,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, udelay(10); regcache_cache_only(sgtl5000->regmap, false); + sgtl5000_restore_regs(codec); ret = regcache_sync(sgtl5000->regmap); if (ret != 0) { @@ -1068,71 +1142,11 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) return 0; } -/* - * restore all sgtl5000 registers, - * since a big hole between dap and regular registers, - * we will restore them respectively. - */ -static int sgtl5000_restore_regs(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - u16 reg; - - /* restore regular registers */ - for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { - - /* These regs should restore in particular order */ - if (reg == SGTL5000_CHIP_ANA_POWER || - reg == SGTL5000_CHIP_CLK_CTRL || - reg == SGTL5000_CHIP_LINREG_CTRL || - reg == SGTL5000_CHIP_LINE_OUT_CTRL || - reg == SGTL5000_CHIP_REF_CTRL) - continue; - - snd_soc_write(codec, reg, cache[reg]); - } - - /* restore dap registers */ - for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); - - /* - * restore these regs according to the power setting sequence in - * sgtl5000_set_power_regs() and clock setting sequence in - * sgtl5000_set_clock(). - * - * The order of restore is: - * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after - * SGTL5000_CHIP_ANA_POWER PLL bits set - * 2. SGTL5000_CHIP_LINREG_CTRL should be set before - * SGTL5000_CHIP_ANA_POWER LINREG_D restored - * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, - * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored - */ - snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); - - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); - return 0; -} - static int sgtl5000_resume(struct snd_soc_codec *codec) { /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Restore registers by cached in memory */ - sgtl5000_restore_regs(codec); return 0; } #else diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 2f8c889..72b68c0 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -341,6 +341,8 @@ #define SGTL5000_ADC_POWERUP 0x0002 #define SGTL5000_LINE_OUT_POWERUP 0x0001 +#define SGTL5000_PLL_POWERUP_BIT 10 + /* * SGTL5000_CHIP_PLL_CTRL */ -- 1.8.3.2

4 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Add suspend/resume support
by Fabio Estevam 26 May '14

26 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Doing a suspend/resume sequence while playing an audio track in the backgroung causes broken audio right after resume: root@freescale /$ aplay clarinet.wav & root@freescale /home$ Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono root@freescale /home$ echo mem > /sys/power/state PM: Syncing filesystems ... done. Freezing user space processes ... (elapsed 0.002 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.002 seconds) done. Suspending console(s) (use no_console_suspend to debug) PM: suspend of devices complete after 37.082 msecs PM: suspend devices took 0.040 seconds PM: late suspend of devices complete after 4.234 msecs PM: noirq suspend of devices complete after 4.618 msecs Disabling non-boot CPUs ... PM: noirq resume of devices complete after 4.013 msecs PM: early resume of devices complete after 4.000 msecs PM: resume of devices complete after 68.907 msecs PM: resume devices took 0.070 seconds Restarting tasks ... Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. .... Add SNDRV_PCM_TRIGGER_RESUME/SUSPEND cases so that we can gracefully handle system suspend/resume. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f233d91..090f449 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -874,6 +874,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int 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_PLAYBACK) fsl_ssi_tx_config(ssi_private, true); @@ -882,6 +883,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, break; case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fsl_ssi_tx_config(ssi_private, false); -- 1.8.3.2

3 2

[alsa-devel] Recovering from DAI xrun condition
by Josef Ahmad 26 May '14

26 May '14

Hi, I see the ALSA framework is capable of detecting and recovering from ring buffer xrun conditions. What I'm not sure about is whether it is possible to recover from a DAI xrun error as well. Let's take the example of an I2S interface operating in tx mode. If the tx fifo overruns, one or more words will be discarded by the interface. If an odd number of words gets discarded, this may result in the left and right word being swapped. This is actually a quite common weakness in I2S controllers. Is there a mechanism to notify the ALSA subsystem that a DAI xrun condition has occurred? Regards Josef

1 0

[alsa-devel] [PATCH] alsa-plugins: jack: fix polling and recovering
by Sergey 26 May '14

26 May '14

This patch fixes polling in alsa-to-jack plugin. It makes poll()+_poll_revents() return correct values when there's something to read/write. It also makes jack pcm survive snd_pcm_recover(-EPIPE). --- jack/pcm_jack.c | 62 ++++++++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 55 insertions(+), 7 deletions(-) diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 7a8b24d..36fbdc9 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -42,6 +42,7 @@ typedef struct { unsigned int num_ports; unsigned int hw_ptr; unsigned int sample_bits; + snd_pcm_uframes_t min_avail; unsigned int channels; snd_pcm_channel_area_t *areas; @@ -50,6 +51,41 @@ typedef struct { jack_client_t *client; } snd_pcm_jack_t; +static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); + +static int pcm_poll_block_check(snd_pcm_ioplug_t *io) +{ + static char buf[32]; + snd_pcm_sframes_t avail; + snd_pcm_jack_t *jack = io->private_data; + + if (io->state == SND_PCM_STATE_RUNNING || + (io->state == SND_PCM_STATE_PREPARED && io->stream == SND_PCM_STREAM_CAPTURE)) { + avail = snd_pcm_avail_update(io->pcm); + if (avail >= 0 && avail < jack->min_avail) { + while (read(io->poll_fd, &buf, sizeof(buf)) == sizeof(buf)); + return 1; + } + } + + return 0; +} + +static int pcm_poll_unblock_check(snd_pcm_ioplug_t *io) +{ + static char buf[1]; + snd_pcm_sframes_t avail; + snd_pcm_jack_t *jack = io->private_data; + + avail = snd_pcm_avail_update(io->pcm); + if (avail < 0 || avail >= jack->min_avail) { + write(jack->fd, &buf, 1); + return 1; + } + + return 0; +} + static void snd_pcm_jack_free(snd_pcm_jack_t *jack) { if (jack) { @@ -81,14 +117,10 @@ static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { - static char buf[1]; - assert(pfds && nfds == 1 && revents); - read(pfds[0].fd, buf, 1); - *revents = pfds[0].revents & ~(POLLIN | POLLOUT); - if (pfds[0].revents & POLLIN) + if (pfds[0].revents & POLLIN && !pcm_poll_block_check(io)) *revents |= (io->stream == SND_PCM_STREAM_PLAYBACK) ? POLLOUT : POLLIN; return 0; } @@ -105,7 +137,6 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) snd_pcm_jack_t *jack = io->private_data; const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t xfer = 0; - static char buf[1]; unsigned int channel; for (channel = 0; channel < io->channels; channel++) { @@ -145,7 +176,7 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) xfer += frames; } - write(jack->fd, buf, 1); /* for polling */ + pcm_poll_unblock_check(io); /* unblock socket for polling if needed */ return 0; } @@ -154,9 +185,26 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; unsigned int i; + snd_pcm_sw_params_t *swparams; + int err; jack->hw_ptr = 0; + jack->min_avail = io->period_size; + snd_pcm_sw_params_alloca(&swparams); + err = snd_pcm_sw_params_current(io->pcm, swparams); + if (err == 0) { + snd_pcm_sw_params_get_avail_min(swparams, &jack->min_avail); + } + + /* deactivate jack connections if this is XRUN recovery */ + snd_pcm_jack_stop(io); + + if (io->stream == SND_PCM_STREAM_PLAYBACK) + pcm_poll_unblock_check(io); /* playback pcm initially accepts writes */ + else + pcm_poll_block_check(io); /* block capture pcm if that's XRUN recovery */ + if (jack->ports) return 0; -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: snd-usb: mixer: coding style fixups
by Daniel Mack 25 May '14

25 May '14

Shorten some over-long lines, multi-line comments, spurious whitespaces, curly brakets etc. No functional change. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/mixer.c | 400 +++++++++++++++++++++++++++++++++--------------------- 1 file changed, 246 insertions(+), 154 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6fe83e4..9765e1c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -162,7 +162,7 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid, { const struct usbmix_selector_map *p; - if (! state->selector_map) + if (!state->selector_map) return 0; for (p = state->selector_map; p->id; p++) { if (p->id == unitid && index < p->count) @@ -174,7 +174,8 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid, /* * find an audio control unit with the given unit id */ -static void *find_audio_control_unit(struct mixer_build *state, unsigned char unit) +static void *find_audio_control_unit(struct mixer_build *state, + unsigned char unit) { /* we just parse the header */ struct uac_feature_unit_descriptor *hdr = NULL; @@ -194,7 +195,8 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un /* * copy a string with the given id */ -static int snd_usb_copy_string_desc(struct mixer_build *state, int index, char *buf, int maxlen) +static int snd_usb_copy_string_desc(struct mixer_build *state, + int index, char *buf, int maxlen) { int len = usb_string(state->chip->dev, index, buf, maxlen - 1); buf[len] = 0; @@ -253,7 +255,7 @@ static int convert_bytes_value(struct usb_mixer_elem_info *cval, int val) static int get_relative_value(struct usb_mixer_elem_info *cval, int val) { - if (! cval->res) + if (!cval->res) cval->res = 1; if (val < cval->min) return 0; @@ -267,7 +269,7 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) { if (val < 0) return cval->min; - if (! cval->res) + if (!cval->res) cval->res = 1; val *= cval->res; val += cval->min; @@ -281,7 +283,8 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) * retrieve a mixer value */ -static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, + int validx, int *value_ret) { struct snd_usb_audio *chip = cval->mixer->chip; unsigned char buf[2]; @@ -292,6 +295,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v err = snd_usb_autoresume(cval->mixer->chip); if (err < 0) return -EIO; + down_read(&chip->shutdown_rwsem); while (timeout-- > 0) { if (chip->shutdown) @@ -316,10 +320,11 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v return err; } -static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, + int validx, int *value_ret) { struct snd_usb_audio *chip = cval->mixer->chip; - unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; @@ -339,9 +344,9 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v goto error; down_read(&chip->shutdown_rwsem); - if (chip->shutdown) + if (chip->shutdown) { ret = -ENODEV; - else { + } else { idx = snd_usb_ctrl_intf(chip) | (cval->id << 8); ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, @@ -382,7 +387,8 @@ error: return 0; } -static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, + int validx, int *value_ret) { validx += cval->idx_off; @@ -391,7 +397,8 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali get_ctl_value_v2(cval, request, validx, value_ret); } -static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value) +static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, + int validx, int *value) { return get_ctl_value(cval, UAC_GET_CUR, validx, value); } @@ -400,7 +407,9 @@ static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int * static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, int channel, int *value) { - return get_ctl_value(cval, UAC_GET_CUR, (cval->control << 8) | channel, value); + return get_ctl_value(cval, UAC_GET_CUR, + (cval->control << 8) | channel, + value); } static int get_cur_mix_value(struct usb_mixer_elem_info *cval, @@ -417,7 +426,7 @@ static int get_cur_mix_value(struct usb_mixer_elem_info *cval, if (!cval->mixer->ignore_ctl_error) usb_audio_dbg(cval->mixer->chip, "cannot get current value for control %d ch %d: err = %d\n", - cval->control, channel, err); + cval->control, channel, err); return err; } cval->cached |= 1 << channel; @@ -425,7 +434,6 @@ static int get_cur_mix_value(struct usb_mixer_elem_info *cval, return 0; } - /* * set a mixer value */ @@ -474,7 +482,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, } } usb_audio_dbg(chip, "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, data = %#x/%#x\n", - request, validx, idx, cval->val_type, buf[0], buf[1]); + request, validx, idx, cval->val_type, buf[0], buf[1]); err = -EINVAL; out: @@ -483,7 +491,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, return err; } -static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value) +static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, + int validx, int value) { return snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, validx, value); } @@ -503,8 +512,9 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, return 0; } - err = snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, - value); + err = snd_usb_mixer_set_ctl_value(cval, + UAC_SET_CUR, (cval->control << 8) | channel, + value); if (err < 0) return err; cval->cached |= 1 << channel; @@ -541,13 +551,13 @@ static int parse_audio_unit(struct mixer_build *state, int unitid); * check if the input/output channel routing is enabled on the given bitmap. * used for mixer unit parser */ -static int check_matrix_bitmap(unsigned char *bmap, int ich, int och, int num_outs) +static int check_matrix_bitmap(unsigned char *bmap, + int ich, int och, int num_outs) { int idx = ich * num_outs + och; return bmap[idx >> 3] & (0x80 >> (idx & 7)); } - /* * add an alsa control element * search and increment the index until an empty slot is found. @@ -564,7 +574,8 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, while (snd_ctl_find_id(mixer->chip->card, &kctl->id)) kctl->id.index++; if ((err = snd_ctl_add(mixer->chip->card, kctl)) < 0) { - usb_audio_dbg(mixer->chip, "cannot add control (err = %d)\n", err); + usb_audio_dbg(mixer->chip, "cannot add control (err = %d)\n", + err); return err; } cval->elem_id = &kctl->id; @@ -573,7 +584,6 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, return 0; } - /* * get a terminal name string */ @@ -627,7 +637,8 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm struct iterm_name_combo *names; if (iterm->name) - return snd_usb_copy_string_desc(state, iterm->name, name, maxlen); + return snd_usb_copy_string_desc(state, iterm->name, + name, maxlen); /* virtual type - not a real terminal */ if (iterm->type >> 16) { @@ -635,13 +646,17 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm return 0; switch (iterm->type >> 16) { case UAC_SELECTOR_UNIT: - strcpy(name, "Selector"); return 8; + strcpy(name, "Selector"); + return 8; case UAC1_PROCESSING_UNIT: - strcpy(name, "Process Unit"); return 12; + strcpy(name, "Process Unit"); + return 12; case UAC1_EXTENSION_UNIT: - strcpy(name, "Ext Unit"); return 8; + strcpy(name, "Ext Unit"); + return 8; case UAC_MIXER_UNIT: - strcpy(name, "Mixer"); return 5; + strcpy(name, "Mixer"); + return 5; default: return sprintf(name, "Unit %d", iterm->id); } @@ -649,29 +664,35 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm switch (iterm->type & 0xff00) { case 0x0100: - strcpy(name, "PCM"); return 3; + strcpy(name, "PCM"); + return 3; case 0x0200: - strcpy(name, "Mic"); return 3; + strcpy(name, "Mic"); + return 3; case 0x0400: - strcpy(name, "Headset"); return 7; + strcpy(name, "Headset"); + return 7; case 0x0500: - strcpy(name, "Phone"); return 5; + strcpy(name, "Phone"); + return 5; } - for (names = iterm_names; names->type; names++) + for (names = iterm_names; names->type; names++) { if (names->type == iterm->type) { strcpy(name, names->name); return strlen(names->name); } + } + return 0; } - /* * parse the source unit recursively until it reaches to a terminal * or a branched unit. */ -static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) +static int check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) { int err; void *p1; @@ -766,7 +787,6 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ return -ENODEV; } - /* * Feature Unit */ @@ -794,7 +814,6 @@ static struct usb_feature_control_info audio_feature_info[] = { { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; - /* private_free callback */ static void usb_mixer_elem_free(struct snd_kcontrol *kctl) { @@ -802,7 +821,6 @@ static void usb_mixer_elem_free(struct snd_kcontrol *kctl) kctl->private_data = NULL; } - /* * interface to ALSA control for feature/mixer units */ @@ -906,7 +924,6 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 384; } break; - } } @@ -939,21 +956,26 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { usb_audio_err(cval->mixer->chip, "%d:%d: cannot get min/max values for control %d (id %d)\n", - cval->id, snd_usb_ctrl_intf(cval->mixer->chip), cval->control, cval->id); + cval->id, snd_usb_ctrl_intf(cval->mixer->chip), + cval->control, cval->id); return -EINVAL; } - if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { + if (get_ctl_value(cval, UAC_GET_RES, + (cval->control << 8) | minchn, + &cval->res) < 0) { cval->res = 1; } else { int last_valid_res = cval->res; while (cval->res > 1) { if (snd_usb_mixer_set_ctl_value(cval, UAC_SET_RES, - (cval->control << 8) | minchn, cval->res / 2) < 0) + (cval->control << 8) | minchn, + cval->res / 2) < 0) break; cval->res /= 2; } - if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) + if (get_ctl_value(cval, UAC_GET_RES, + (cval->control << 8) | minchn, &cval->res) < 0) cval->res = last_valid_res; } if (cval->res == 0) @@ -1017,7 +1039,8 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, #define get_min_max(cval, def) get_min_max_with_quirks(cval, def, NULL) /* get a feature/mixer unit info */ -static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *cval = kcontrol->private_data; @@ -1051,7 +1074,8 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ } /* get the current value from feature/mixer unit */ -static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int c, cnt, val, err; @@ -1082,7 +1106,8 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e } /* put the current value to feature/mixer unit */ -static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int c, cnt, val, oval, err; @@ -1136,22 +1161,25 @@ static struct snd_kcontrol_new usb_feature_unit_ctl_ro = { .put = NULL, }; -/* This symbol is exported in order to allow the mixer quirks to - * hook up to the standard feature unit control mechanism */ +/* + * This symbol is exported in order to allow the mixer quirks to + * hook up to the standard feature unit control mechanism + */ struct snd_kcontrol_new *snd_usb_feature_unit_ctl = &usb_feature_unit_ctl; /* * build a feature control */ - static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) { return strlcat(kctl->id.name, str, sizeof(kctl->id.name)); } -/* A lot of headsets/headphones have a "Speaker" mixer. Make sure we - rename it to "Headphone". We determine if something is a headphone - similar to how udev determines form factor. */ +/* + * A lot of headsets/headphones have a "Speaker" mixer. Make sure we + * rename it to "Headphone". We determine if something is a headphone + * similar to how udev determines form factor. + */ static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, struct snd_card *card) { @@ -1201,7 +1229,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (! cval) { + if (!cval) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); return; } @@ -1222,15 +1250,17 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval->ch_readonly = readonly_mask; } - /* if all channels in the mask are marked read-only, make the control + /* + * If all channels in the mask are marked read-only, make the control * read-only. set_cur_mix_value() will check the mask again and won't - * issue write commands to read-only channels. */ + * issue write commands to read-only channels. + */ if (cval->channels == readonly_mask) kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); else kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); - if (! kctl) { + if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(cval); return; @@ -1239,48 +1269,53 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); mapped_name = len != 0; - if (! len && nameid) + if (!len && nameid) len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); switch (control) { case UAC_FU_MUTE: case UAC_FU_VOLUME: - /* determine the control name. the rule is: + /* + * determine the control name. the rule is: * - if a name id is given in descriptor, use it. * - if the connected input can be determined, then use the name * of terminal type. * - if the connected output can be determined, use it. * - otherwise, anonymous name. */ - if (! len) { - len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 1); - if (! len) - len = get_term_name(state, &state->oterm, kctl->id.name, sizeof(kctl->id.name), 1); - if (! len) - len = snprintf(kctl->id.name, sizeof(kctl->id.name), + if (!len) { + len = get_term_name(state, iterm, kctl->id.name, + sizeof(kctl->id.name), 1); + if (!len) + len = get_term_name(state, &state->oterm, + kctl->id.name, + sizeof(kctl->id.name), 1); + if (!len) + len = snprintf(kctl->id.name, + sizeof(kctl->id.name), "Feature %d", unitid); } if (!mapped_name) check_no_speaker_on_headset(kctl, state->mixer->chip->card); - /* determine the stream direction: + /* + * determine the stream direction: * if the connected output is USB stream, then it's likely a * capture stream. otherwise it should be playback (hopefully :) */ - if (! mapped_name && ! (state->oterm.type >> 16)) { - if ((state->oterm.type & 0xff00) == 0x0100) { + if (!mapped_name && !(state->oterm.type >> 16)) { + if ((state->oterm.type & 0xff00) == 0x0100) len = append_ctl_name(kctl, " Capture"); - } else { + else len = append_ctl_name(kctl, " Playback"); - } } append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); break; default: - if (! len) + if (!len) strlcpy(kctl->id.name, audio_feature_info[control-1].name, sizeof(kctl->id.name)); break; @@ -1300,33 +1335,35 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } range = (cval->max - cval->min) / cval->res; - /* Are there devices with volume range more than 255? I use a bit more + /* + * Are there devices with volume range more than 255? I use a bit more * to be sure. 384 is a resolution magic number found on Logitech * devices. It will definitively catch all buggy Logitech devices. */ if (range > 384) { - usb_audio_warn(state->chip, "Warning! Unlikely big " - "volume range (=%u), cval->res is probably wrong.", - range); + usb_audio_warn(state->chip, + "Warning! Unlikely big volume range (=%u), " + "cval->res is probably wrong.", + range); usb_audio_warn(state->chip, "[%d] FU [%s] ch = %d, " - "val = %d/%d/%d", cval->id, - kctl->id.name, cval->channels, - cval->min, cval->max, cval->res); + "val = %d/%d/%d", cval->id, + kctl->id.name, cval->channels, + cval->min, cval->max, cval->res); } usb_audio_dbg(state->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", - cval->id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res); + cval->id, kctl->id.name, cval->channels, + cval->min, cval->max, cval->res); snd_usb_mixer_add_control(state->mixer, kctl); } - - /* * parse a feature unit * * most of controls are defined here. */ -static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr) +static int parse_audio_feature_unit(struct mixer_build *state, int unitid, + void *_ftr) { int channels, i, j; struct usb_audio_term iterm; @@ -1400,15 +1437,25 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void for (i = 0; i < 10; i++) { unsigned int ch_bits = 0; for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); + unsigned int mask; + + mask = snd_usb_combine_bytes(bmaControls + + csize * (j+1), csize); if (mask & (1 << i)) ch_bits |= (1 << j); } /* audio class v1 controls are never read-only */ - if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, 0); + + /* + * The first channel must be set + * (for ease of programming). + */ + if (ch_bits & 1) + build_feature_ctl(state, _ftr, ch_bits, i, + &iterm, unitid, 0); if (master_bits & (1 << i)) - build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0); + build_feature_ctl(state, _ftr, 0, i, &iterm, + unitid, 0); } } else { /* UAC_VERSION_2 */ for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) { @@ -1416,7 +1463,10 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void unsigned int ch_read_only = 0; for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); + unsigned int mask; + + mask = snd_usb_combine_bytes(bmaControls + + csize * (j+1), csize); if (uac2_control_is_readable(mask, i)) { ch_bits |= (1 << j); if (!uac2_control_is_writeable(mask, i)) @@ -1424,12 +1474,22 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } } - /* NOTE: build_feature_ctl() will mark the control read-only if all channels - * are marked read-only in the descriptors. Otherwise, the control will be - * reported as writeable, but the driver will not actually issue a write - * command for read-only channels */ - if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only); + /* + * NOTE: build_feature_ctl() will mark the control + * read-only if all channels are marked read-only in + * the descriptors. Otherwise, the control will be + * reported as writeable, but the driver will not + * actually issue a write command for read-only + * channels. + */ + + /* + * The first channel must be set + * (for ease of programming). + */ + if (ch_bits & 1) + build_feature_ctl(state, _ftr, ch_bits, i, + &iterm, unitid, ch_read_only); if (uac2_control_is_readable(master_bits, i)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, !uac2_control_is_writeable(master_bits, i)); @@ -1439,7 +1499,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void return 0; } - /* * Mixer Unit */ @@ -1450,7 +1509,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void * the callbacks are identical with feature unit. * input channel number (zero based) is given in control field instead. */ - static void build_mixer_unit_ctl(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc, int in_pin, int in_ch, int unitid, @@ -1467,7 +1525,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (! cval) + if (!cval) return; cval->mixer = state->mixer; @@ -1475,7 +1533,9 @@ static void build_mixer_unit_ctl(struct mixer_build *state, cval->control = in_ch + 1; /* based on 1 */ cval->val_type = USB_MIXER_S16; for (i = 0; i < num_outs; i++) { - if (check_matrix_bitmap(uac_mixer_unit_bmControls(desc, state->mixer->protocol), in_ch, i, num_outs)) { + __u8 *c = uac_mixer_unit_bmControls(desc, state->mixer->protocol); + + if (check_matrix_bitmap(c, in_ch, i, num_outs)) { cval->cmask |= (1 << i); cval->channels++; } @@ -1485,7 +1545,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, get_min_max(cval, 0); kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); - if (! kctl) { + if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(cval); return; @@ -1493,9 +1553,10 @@ static void build_mixer_unit_ctl(struct mixer_build *state, kctl->private_free = usb_mixer_elem_free; len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); - if (! len) - len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0); - if (! len) + if (!len) + len = get_term_name(state, iterm, kctl->id.name, + sizeof(kctl->id.name), 0); + if (!len) len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1); append_ctl_name(kctl, " Volume"); @@ -1504,24 +1565,28 @@ static void build_mixer_unit_ctl(struct mixer_build *state, snd_usb_mixer_add_control(state->mixer, kctl); } - /* * parse a mixer unit */ -static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *raw_desc) +static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, + void *raw_desc) { struct uac_mixer_unit_descriptor *desc = raw_desc; struct usb_audio_term iterm; int input_pins, num_ins, num_outs; int pin, ich, err; - if (desc->bLength < 11 || ! (input_pins = desc->bNrInPins) || ! (num_outs = uac_mixer_unit_bNrChannels(desc))) { - usb_audio_err(state->chip, "invalid MIXER UNIT descriptor %d\n", unitid); + if (desc->bLength < 11 || !(input_pins = desc->bNrInPins) || + !(num_outs = uac_mixer_unit_bNrChannels(desc))) { + usb_audio_err(state->chip, + "invalid MIXER UNIT descriptor %d\n", + unitid); return -EINVAL; } /* no bmControls field (e.g. Maya44) -> ignore */ if (desc->bLength <= 10 + input_pins) { - usb_audio_dbg(state->chip, "MU %d has no bmControls field\n", unitid); + usb_audio_dbg(state->chip, "MU %d has no bmControls field\n", + unitid); return 0; } @@ -1535,12 +1600,14 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r if (err < 0) return err; num_ins += iterm.channels; - for (; ich < num_ins; ++ich) { + for (; ich < num_ins; ich++) { int och, ich_has_controls = 0; - for (och = 0; och < num_outs; ++och) { - if (check_matrix_bitmap(uac_mixer_unit_bmControls(desc, state->mixer->protocol), - ich, och, num_outs)) { + for (och = 0; och < num_outs; och++) { + __u8 *c = uac_mixer_unit_bmControls(desc, + state->mixer->protocol); + + if (check_matrix_bitmap(c, ich, och, num_outs)) { ich_has_controls = 1; break; } @@ -1553,13 +1620,13 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r return 0; } - /* * Processing Unit / Extension Unit */ /* get callback for processing/extension unit */ -static int mixer_ctl_procunit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_procunit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int err, val; @@ -1577,7 +1644,8 @@ static int mixer_ctl_procunit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ } /* put callback for processing/extension unit */ -static int mixer_ctl_procunit_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_procunit_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, oval, err; @@ -1606,7 +1674,6 @@ static struct snd_kcontrol_new mixer_procunit_ctl = { .put = mixer_ctl_procunit_put, }; - /* * predefined data for processing units */ @@ -1697,10 +1764,13 @@ static struct procunit_info extunits[] = { { USB_XU_DEVICE_OPTIONS, "AnalogueIn Soft Limit", soft_limit_xu_info }, { 0 } }; + /* * build a processing/extension unit */ -static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw_desc, struct procunit_info *list, char *name) +static int build_audio_procunit(struct mixer_build *state, int unitid, + void *raw_desc, struct procunit_info *list, + char *name) { struct uac_processing_unit_descriptor *desc = raw_desc; int num_ins = desc->bNrInPins; @@ -1733,19 +1803,19 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw for (info = list; info && info->type; info++) if (info->type == type) break; - if (! info || ! info->type) + if (!info || !info->type) info = &default_info; for (valinfo = info->values; valinfo->control; valinfo++) { __u8 *controls = uac_processing_unit_bmControls(desc, state->mixer->protocol); - if (! (controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1)))) + if (!(controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1)))) continue; map = find_map(state, unitid, valinfo->control); if (check_ignored_ctl(map)) continue; cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (! cval) { + if (!cval) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); return -ENOMEM; } @@ -1765,7 +1835,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre/0204 + /* + * E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; @@ -1777,60 +1848,70 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw } kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); - if (! kctl) { + if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(cval); return -ENOMEM; } kctl->private_free = usb_mixer_elem_free; - if (check_mapped_name(map, kctl->id.name, - sizeof(kctl->id.name))) + if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) { /* nothing */ ; - else if (info->name) + } else if (info->name) { strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name)); - else { + } else { nameid = uac_processing_unit_iProcessing(desc, state->mixer->protocol); len = 0; if (nameid) - len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); - if (! len) + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, + sizeof(kctl->id.name)); + if (!len) strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); } append_ctl_name(kctl, " "); append_ctl_name(kctl, valinfo->suffix); usb_audio_dbg(state->chip, - "[%d] PU [%s] ch = %d, val = %d/%d\n", - cval->id, kctl->id.name, cval->channels, cval->min, cval->max); - if ((err = snd_usb_mixer_add_control(state->mixer, kctl)) < 0) + "[%d] PU [%s] ch = %d, val = %d/%d\n", + cval->id, kctl->id.name, cval->channels, + cval->min, cval->max); + + err = snd_usb_mixer_add_control(state->mixer, kctl); + if (err < 0) return err; } return 0; } - -static int parse_audio_processing_unit(struct mixer_build *state, int unitid, void *raw_desc) +static int parse_audio_processing_unit(struct mixer_build *state, int unitid, + void *raw_desc) { - return build_audio_procunit(state, unitid, raw_desc, procunits, "Processing Unit"); + return build_audio_procunit(state, unitid, raw_desc, + procunits, "Processing Unit"); } -static int parse_audio_extension_unit(struct mixer_build *state, int unitid, void *raw_desc) +static int parse_audio_extension_unit(struct mixer_build *state, int unitid, + void *raw_desc) { - /* Note that we parse extension units with processing unit descriptors. - * That's ok as the layout is the same */ - return build_audio_procunit(state, unitid, raw_desc, extunits, "Extension Unit"); + /* + * Note that we parse extension units with processing unit descriptors. + * That's ok as the layout is the same. + */ + return build_audio_procunit(state, unitid, raw_desc, + extunits, "Extension Unit"); } - /* * Selector Unit */ -/* info callback for selector unit +/* + * info callback for selector unit * use an enumerator type for routing */ -static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *cval = kcontrol->private_data; const char **itemlist = (const char **)kcontrol->private_value; @@ -1841,7 +1922,8 @@ static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl } /* get callback for selector unit */ -static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, err; @@ -1860,7 +1942,8 @@ static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ } /* put callback for selector unit */ -static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, oval, err; @@ -1889,8 +1972,8 @@ static struct snd_kcontrol_new mixer_selectunit_ctl = { .put = mixer_ctl_selector_put, }; - -/* private free callback. +/* + * private free callback. * free both private_data and private_value */ static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl) @@ -1915,7 +1998,8 @@ static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl) /* * parse a selector unit */ -static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void *raw_desc) +static int parse_audio_selector_unit(struct mixer_build *state, int unitid, + void *raw_desc) { struct uac_selector_unit_descriptor *desc = raw_desc; unsigned int i, nameid, len; @@ -1944,7 +2028,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void return 0; cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (! cval) { + if (!cval) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); return -ENOMEM; } @@ -1963,7 +2047,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void cval->control = 0; namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); - if (! namelist) { + if (!namelist) { usb_audio_err(state->chip, "cannot malloc\n"); kfree(cval); return -ENOMEM; @@ -1973,7 +2057,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void struct usb_audio_term iterm; len = 0; namelist[i] = kmalloc(MAX_ITEM_NAME_LEN, GFP_KERNEL); - if (! namelist[i]) { + if (!namelist[i]) { usb_audio_err(state->chip, "cannot malloc\n"); while (i--) kfree(namelist[i]); @@ -2004,11 +2088,12 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void if (len) ; else if (nameid) - snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); + snd_usb_copy_string_desc(state, nameid, kctl->id.name, + sizeof(kctl->id.name)); else { len = get_term_name(state, &state->oterm, kctl->id.name, sizeof(kctl->id.name), 0); - if (! len) + if (!len) strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) @@ -2027,7 +2112,6 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void return 0; } - /* * parse an audio unit recursively */ @@ -2125,14 +2209,16 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } p = NULL; - while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen, + while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, + mixer->hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) { if (mixer->protocol == UAC_VERSION_1) { struct uac1_output_terminal_descriptor *desc = p; if (desc->bLength < sizeof(*desc)) continue; /* invalid descriptor? */ - set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + /* mark terminal ID as visited */ + set_bit(desc->bTerminalID, state.unitbitmap); state.oterm.id = desc->bTerminalID; state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; @@ -2144,7 +2230,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (desc->bLength < sizeof(*desc)) continue; /* invalid descriptor? */ - set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + /* mark terminal ID as visited */ + set_bit(desc->bTerminalID, state.unitbitmap); state.oterm.id = desc->bTerminalID; state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; @@ -2152,7 +2239,10 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (err < 0 && err != -EINVAL) return err; - /* for UAC2, use the same approach to also add the clock selectors */ + /* + * For UAC2, use the same approach to also add the + * clock selectors + */ err = parse_audio_unit(&state, desc->bCSourceID); if (err < 0 && err != -EINVAL) return err; @@ -2306,7 +2396,9 @@ static void snd_usb_mixer_interrupt(struct urb *urb) } requeue: - if (ustatus != -ENOENT && ustatus != -ECONNRESET && ustatus != -ESHUTDOWN) { + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { urb->dev = mixer->chip->dev; usb_submit_urb(urb, GFP_ATOMIC); } -- 1.9.0

2 2

[alsa-devel] Sound popping on playback start stop Audigy 2 ZS Notebook
by Rim Botede 24 May '14

24 May '14

To whom it may be of concern.https://bugzilla.kernel.org/show_bug.cgi?id=76841

1 0

[alsa-devel] [PATCH v3] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 24 May '14

24 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. According to Mark Brown: "Yes, reg_cache isn't there if we're not using ASoC level caching. The fix should just be to replace the direct cache references with snd_soc_read()s which will end up in a cache lookup if the register is cached." Do as suggested and also complete the cache array with the missing registers so that sgtl5000_restore_regs() can properly cache the all the registers it needs. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v2: - Use regmap_update_bits Changes since v1: - Also fix sgtl5000_set_bias_level so that suspend/resume/play sequence does not fail after several interactions sound/soc/codecs/sgtl5000.c | 61 +++++++++++++++++++++++++-------------------- sound/soc/codecs/sgtl5000.h | 2 ++ 2 files changed, 36 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..6b4bf9a 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -924,20 +939,6 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; udelay(10); - - regcache_cache_only(sgtl5000->regmap, false); - - ret = regcache_sync(sgtl5000->regmap); - if (ret != 0) { - dev_err(codec->dev, - "Failed to restore cache: %d\n", ret); - - regcache_cache_only(sgtl5000->regmap, true); - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - return ret; - } } break; @@ -1063,7 +1064,10 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) #ifdef CONFIG_SUSPEND static int sgtl5000_suspend(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(sgtl5000->regmap, true); return 0; } @@ -1075,8 +1079,8 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) */ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; u16 reg; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* restore regular registers */ for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { @@ -1089,12 +1093,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) reg == SGTL5000_CHIP_REF_CTRL) continue; - snd_soc_write(codec, reg, cache[reg]); + regmap_update_bits(sgtl5000->regmap, reg, reg, 0); } /* restore dap registers */ for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); + regmap_update_bits(sgtl5000->regmap, reg, reg, 0); /* * restore these regs according to the power setting sequence in @@ -1109,30 +1113,33 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored */ - snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_LINREG_CTRL, + SGTL5000_CHIP_LINREG_CTRL, 0); - snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP_BIT, 0); - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_CLK_CTRL, + SGTL5000_MCLK_FREQ_PLL, 0); - snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_REF_CTRL, + SGTL5000_CHIP_REF_CTRL, 0); - snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_LINE_OUT_CTRL, + SGTL5000_CHIP_LINE_OUT_CTRL, 0); return 0; } static int sgtl5000_resume(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Restore registers by cached in memory */ sgtl5000_restore_regs(codec); + + regcache_cache_only(sgtl5000->regmap, false); return 0; } #else diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 2f8c889..72b68c0 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -341,6 +341,8 @@ #define SGTL5000_ADC_POWERUP 0x0002 #define SGTL5000_LINE_OUT_POWERUP 0x0001 +#define SGTL5000_PLL_POWERUP_BIT 10 + /* * SGTL5000_CHIP_PLL_CTRL */ -- 1.8.3.2

2 2

[alsa-devel] MiniDSP USBStreamer / UAC2 / Alsamixer bogus
by Jan Baumgart 24 May '14

24 May '14

Hi there, Today I received a MiniDSP USBStreamer Box. (http://www.minidsp.com/products/usb-audio-interface/usbstreamer-box) It's an USB to ADAT optical interface and claims to be UAC2 compliant. It's based on a XMOS L2 core. Audio worked out of the box! 8-Channels In & Out :) But the alsamixer controls seem to be bogus. Here's a screenshot of qasmixer: icem.folkwang-uni.de/~jb/geheim/usbstreamer_qasmixer.png All controls are named "USBStreamer Clock Selector"... The first control always snaps back to zero. The following 8 Multi-Controls don't have any effect. Only the last two Input/Ouput Master controls work as expected. The most important problem is that I cannot set the clock source: > amixer -D hw:USBStreamer cset numid=7 1 numid=7,iface=MIXER,name='USBStreamer Clock Selector' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'USBStreamer Internal Clock' ; Item #1 'USBStreamer ADAT Clock' : values=0 The value always stays 0 (internal). On OS X 10.8.5 without installing any driver I see 8 Input/Output Volume Sliders + Master Sliders and a working Clock Selector. I'm running debian testing with a 3.13-1-amd64 stock kernel. Any ideas? I've attached lsusb and amixer infos below... Cheers & Thanks, Jan _________________________________________________ > amixer -D hw:USBStreamer controls numid=7,iface=MIXER,name='USBStreamer Clock Selector' numid=8,iface=MIXER,name='USBStreamer Clock Selector Capture Switch' numid=9,iface=MIXER,name='USBStreamer Clock Selector Capture Switch',index=1 numid=10,iface=MIXER,name='USBStreamer Clock Selector Capture Volume' numid=11,iface=MIXER,name='USBStreamer Clock Selector Capture Volume',index=1 numid=3,iface=MIXER,name='USBStreamer Clock Selector Playback Switch' numid=4,iface=MIXER,name='USBStreamer Clock Selector Playback Switch',index=1 numid=5,iface=MIXER,name='USBStreamer Clock Selector Playback Volume' numid=6,iface=MIXER,name='USBStreamer Clock Selector Playback Volume',index=1 numid=2,iface=PCM,name='Capture Channel Map' numid=1,iface=PCM,name='Playback Channel Map' ______________________________________________________________- > amixer -D hw:USBStreamer contents numid=7,iface=MIXER,name='USBStreamer Clock Selector' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'USBStreamer Internal Clock' ; Item #1 'USBStreamer ADAT Clock' : values=0 numid=8,iface=MIXER,name='USBStreamer Clock Selector Capture Switch' ; type=BOOLEAN,access=rw------,values=8 : values=on,on,on,on,on,on,on,on numid=9,iface=MIXER,name='USBStreamer Clock Selector Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=10,iface=MIXER,name='USBStreamer Clock Selector Capture Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=255,step=0 : values=255,255,255,255,255,255,255,255 | dBminmax-min=-127.50dB,max=0.00dB numid=11,iface=MIXER,name='USBStreamer Clock Selector Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=1,min=0,max=255,step=0 : values=255 | dBminmax-min=-127.50dB,max=0.00dB numid=3,iface=MIXER,name='USBStreamer Clock Selector Playback Switch' ; type=BOOLEAN,access=rw------,values=8 : values=on,on,on,on,on,on,on,on numid=4,iface=MIXER,name='USBStreamer Clock Selector Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=5,iface=MIXER,name='USBStreamer Clock Selector Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=255,step=0 : values=255,255,255,255,255,255,255,255 | dBminmax-min=-127.50dB,max=0.00dB numid=6,iface=MIXER,name='USBStreamer Clock Selector Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=1,min=0,max=255,step=0 : values=255 | dBminmax-min=-127.50dB,max=0.00dB numid=2,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r----R--,values=8,min=0,max=36,step=0 : values=0,0,0,0,0,0,0,0 | | chmap-fixed=FL,FR,FC,LFE,RL,RR,FLC,FRC numid=1,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=8,min=0,max=36,step=0 : values=0,0,0,0,0,0,0,0 | | chmap-fixed=FL,FR,FC,LFE,RL,RR,FLC,FRC ______________________________________________________________ > lsusb -s 3:10 -v Bus 003 Device 010: ID 2752:0016 Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x2752 idProduct 0x0016 bcdDevice 5.30 iManufacturer 1 iProduct 2 iSerial 3 bNumConfigurations 2 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 524 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 2 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 257 bmControl 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 41 bmAttributes 0x03 Internal programmable Clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 9 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 43 bmAttributes 0x00 External Clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 11 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bUnitID 40 bNrInPins 2 baCSourceID( 0) 41 baCSourceID( 1) 43 bmControls 0x03 Clock Selector Control (read/write) iClockSelector 8 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 40 bNrChannels 8 bmChannelConfig 0x00000000 bmControls 0x0000 iChannelNames 15 iTerminal 6 AudioControl Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 9 (EXTENSION_UNIT) Warning: Descriptor too short bUnitID 51 wExtensionCode 0 bNrPins 1 baSourceID( 0) 2 bNrChannels 8 wChannelConfig 0 iChannelNames 3 bmControls 0x00 iExtension 42 AudioControl Interface Descriptor: bLength 42 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 51 bmaControls( 0) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 1) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 2) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 3) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 4) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 5) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 6) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 7) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 8) 0x0000000f Mute Control (read/write) Volume Control (read/write) iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 20 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 10 bCSourceID 40 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 40 bNrChannels 8 bmChannelConfig 0x00000000 bmControls 0x0000 iChannelNames 33 iTerminal 0 AudioControl Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 9 (EXTENSION_UNIT) Warning: Descriptor too short bUnitID 52 wExtensionCode 0 bNrPins 1 baSourceID( 0) 1 bNrChannels 8 wChannelConfig 0 iChannelNames 3 bmControls 0x00 iExtension 42 AudioControl Interface Descriptor: bLength 42 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 52 bmaControls( 0) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 1) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 2) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 3) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 4) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 5) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 6) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 7) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 8) 0x0000000f Mute Control (read/write) Volume Control (read/write) iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 22 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 11 bCSourceID 40 bmControls 0x0000 iTerminal 7 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 9 (EXTENSION_UNIT) Warning: Descriptor too short bUnitID 50 wExtensionCode 0 bNrPins 2 baSourceID( 0) 2 baSourceID( 1) 1 bNrChannels 18 wChannelConfig 0 iChannelNames 3 bmControls 0x00 iExtension 32 AudioControl Interface Descriptor: bLength 32 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 60 bNrPins 1 baSourceID( 0) 50 bNrChannels 8 bmChannelConfig 0x00000000 iChannelNames 49 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff bmControls 00 iMixer 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0006 1x 6 bytes bInterval 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 8 bmChannelConfig 0x00000000 iChannelNames 15 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 8 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 8 bmChannelConfig 0x00000000 iChannelNames 15 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 8 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 8 bmChannelConfig 0x00000000 iChannelNames 33 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 4 bmChannelConfig 0x00000000 iChannelNames 33 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 254 Application Specific Interface bInterfaceSubClass 1 Device Firmware Update bInterfaceProtocol 1 iInterface 12 Device Firmware Upgrade Interface Descriptor: bLength 9 bDescriptorType 33 bmAttributes 7 Will Not Detach Manifestation Tolerant Upload Supported Download Supported wDetachTimeout 250 milliseconds wTransferSize 64 bytes bcdDFUVersion 1.10 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 524 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 2 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 257 bmControl 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 41 bmAttributes 0x03 Internal programmable Clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 9 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 43 bmAttributes 0x00 External Clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 11 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bUnitID 40 bNrInPins 2 baCSourceID( 0) 41 baCSourceID( 1) 43 bmControls 0x03 Clock Selector Control (read/write) iClockSelector 8 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 40 bNrChannels 8 bmChannelConfig 0x00000000 bmControls 0x0000 iChannelNames 15 iTerminal 6 AudioControl Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 9 (EXTENSION_UNIT) Warning: Descriptor too short bUnitID 51 wExtensionCode 0 bNrPins 1 baSourceID( 0) 2 bNrChannels 8 wChannelConfig 0 iChannelNames 3 bmControls 0x00 iExtension 42 AudioControl Interface Descriptor: bLength 42 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 51 bmaControls( 0) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 1) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 2) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 3) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 4) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 5) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 6) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 7) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 8) 0x0000000f Mute Control (read/write) Volume Control (read/write) iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 20 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 10 bCSourceID 40 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 40 bNrChannels 8 bmChannelConfig 0x00000000 bmControls 0x0000 iChannelNames 33 iTerminal 0 AudioControl Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 9 (EXTENSION_UNIT) Warning: Descriptor too short bUnitID 52 wExtensionCode 0 bNrPins 1 baSourceID( 0) 1 bNrChannels 8 wChannelConfig 0 iChannelNames 3 bmControls 0x00 iExtension 42 AudioControl Interface Descriptor: bLength 42 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 52 bmaControls( 0) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 1) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 2) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 3) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 4) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 5) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 6) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 7) 0x0000000f Mute Control (read/write) Volume Control (read/write) bmaControls( 8) 0x0000000f Mute Control (read/write) Volume Control (read/write) iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 22 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 11 bCSourceID 40 bmControls 0x0000 iTerminal 7 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 9 (EXTENSION_UNIT) Warning: Descriptor too short bUnitID 50 wExtensionCode 0 bNrPins 2 baSourceID( 0) 2 baSourceID( 1) 1 bNrChannels 18 wChannelConfig 0 iChannelNames 3 bmControls 0x00 iExtension 32 AudioControl Interface Descriptor: bLength 32 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 60 bNrPins 1 baSourceID( 0) 50 bNrChannels 8 bmChannelConfig 0x00000000 iChannelNames 49 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff bmControls 00 iMixer 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0006 1x 6 bytes bInterval 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 8 bmChannelConfig 0x00000000 iChannelNames 15 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 8 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 8 bmChannelConfig 0x00000000 iChannelNames 15 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 8 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 8 bmChannelConfig 0x00000000 iChannelNames 33 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 4 bmChannelConfig 0x00000000 iChannelNames 33 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 254 Application Specific Interface bInterfaceSubClass 1 Device Firmware Update bInterfaceProtocol 1 iInterface 12 Device Firmware Upgrade Interface Descriptor: bLength 9 bDescriptorType 33 bmAttributes 7 Will Not Detach Manifestation Tolerant Upload Supported Download Supported wDetachTimeout 250 milliseconds wTransferSize 64 bytes bcdDFUVersion 1.10

2 3

[alsa-devel] [PATCH 1/2] ALSA: snd-usb: mixer: coding style fixups
by Daniel Mack 24 May '14

24 May '14

Shorten some over-long lines, multi-line comments, spurious whitespaces, curly brakets etc. No functional change. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/mixer.c | 406 +++++++++++++++++++++++++++++++++--------------------- 1 file changed, 249 insertions(+), 157 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d40a285..e13e0c1 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -162,7 +162,7 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid, { const struct usbmix_selector_map *p; - if (! state->selector_map) + if (!state->selector_map) return 0; for (p = state->selector_map; p->id; p++) { if (p->id == unitid && index < p->count) @@ -174,7 +174,8 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid, /* * find an audio control unit with the given unit id */ -static void *find_audio_control_unit(struct mixer_build *state, unsigned char unit) +static void *find_audio_control_unit(struct mixer_build *state, + unsigned char unit) { /* we just parse the header */ struct uac_feature_unit_descriptor *hdr = NULL; @@ -194,7 +195,8 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un /* * copy a string with the given id */ -static int snd_usb_copy_string_desc(struct mixer_build *state, int index, char *buf, int maxlen) +static int snd_usb_copy_string_desc(struct mixer_build *state, + int index, char *buf, int maxlen) { int len = usb_string(state->chip->dev, index, buf, maxlen - 1); buf[len] = 0; @@ -253,7 +255,7 @@ static int convert_bytes_value(struct usb_mixer_elem_info *cval, int val) static int get_relative_value(struct usb_mixer_elem_info *cval, int val) { - if (! cval->res) + if (!cval->res) cval->res = 1; if (val < cval->min) return 0; @@ -267,7 +269,7 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) { if (val < 0) return cval->min; - if (! cval->res) + if (!cval->res) cval->res = 1; val *= cval->res; val += cval->min; @@ -281,7 +283,8 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) * retrieve a mixer value */ -static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, + int validx, int *value_ret) { struct snd_usb_audio *chip = cval->mixer->chip; unsigned char buf[2]; @@ -292,6 +295,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v err = snd_usb_autoresume(cval->mixer->chip); if (err < 0) return -EIO; + down_read(&chip->shutdown_rwsem); while (timeout-- > 0) { if (chip->shutdown) @@ -316,10 +320,11 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v return err; } -static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, + int validx, int *value_ret) { struct snd_usb_audio *chip = cval->mixer->chip; - unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; @@ -339,9 +344,9 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v goto error; down_read(&chip->shutdown_rwsem); - if (chip->shutdown) + if (chip->shutdown) { ret = -ENODEV; - else { + } else { idx = snd_usb_ctrl_intf(chip) | (cval->id << 8); ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, @@ -382,7 +387,8 @@ error: return 0; } -static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, + int validx, int *value_ret) { validx += cval->idx_off; @@ -391,7 +397,8 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali get_ctl_value_v2(cval, request, validx, value_ret); } -static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value) +static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, + int validx, int *value) { return get_ctl_value(cval, UAC_GET_CUR, validx, value); } @@ -400,7 +407,9 @@ static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int * static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, int channel, int *value) { - return get_ctl_value(cval, UAC_GET_CUR, (cval->control << 8) | channel, value); + return get_ctl_value(cval, UAC_GET_CUR, + (cval->control << 8) | channel, + value); } static int get_cur_mix_value(struct usb_mixer_elem_info *cval, @@ -417,7 +426,7 @@ static int get_cur_mix_value(struct usb_mixer_elem_info *cval, if (!cval->mixer->ignore_ctl_error) usb_audio_dbg(cval->mixer->chip, "cannot get current value for control %d ch %d: err = %d\n", - cval->control, channel, err); + cval->control, channel, err); return err; } cval->cached |= 1 << channel; @@ -425,7 +434,6 @@ static int get_cur_mix_value(struct usb_mixer_elem_info *cval, return 0; } - /* * set a mixer value */ @@ -474,7 +482,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, } } usb_audio_dbg(chip, "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, data = %#x/%#x\n", - request, validx, idx, cval->val_type, buf[0], buf[1]); + request, validx, idx, cval->val_type, buf[0], buf[1]); err = -EINVAL; out: @@ -483,7 +491,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, return err; } -static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value) +static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, + int validx, int value) { return snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, validx, value); } @@ -503,8 +512,9 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, return 0; } - err = snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, - value); + err = snd_usb_mixer_set_ctl_value(cval, + UAC_SET_CUR, (cval->control << 8) | channel, + value); if (err < 0) return err; cval->cached |= 1 << channel; @@ -541,13 +551,13 @@ static int parse_audio_unit(struct mixer_build *state, int unitid); * check if the input/output channel routing is enabled on the given bitmap. * used for mixer unit parser */ -static int check_matrix_bitmap(unsigned char *bmap, int ich, int och, int num_outs) +static int check_matrix_bitmap(unsigned char *bmap, + int ich, int och, int num_outs) { int idx = ich * num_outs + och; return bmap[idx >> 3] & (0x80 >> (idx & 7)); } - /* * add an alsa control element * search and increment the index until an empty slot is found. @@ -564,7 +574,8 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, while (snd_ctl_find_id(mixer->chip->card, &kctl->id)) kctl->id.index++; if ((err = snd_ctl_add(mixer->chip->card, kctl)) < 0) { - usb_audio_dbg(mixer->chip, "cannot add control (err = %d)\n", err); + usb_audio_dbg(mixer->chip, "cannot add control (err = %d)\n", + err); return err; } cval->elem_id = &kctl->id; @@ -573,7 +584,6 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, return 0; } - /* * get a terminal name string */ @@ -627,7 +637,8 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm struct iterm_name_combo *names; if (iterm->name) - return snd_usb_copy_string_desc(state, iterm->name, name, maxlen); + return snd_usb_copy_string_desc(state, iterm->name, + name, maxlen); /* virtual type - not a real terminal */ if (iterm->type >> 16) { @@ -635,13 +646,17 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm return 0; switch (iterm->type >> 16) { case UAC_SELECTOR_UNIT: - strcpy(name, "Selector"); return 8; + strcpy(name, "Selector"); + return 8; case UAC1_PROCESSING_UNIT: - strcpy(name, "Process Unit"); return 12; + strcpy(name, "Process Unit"); + return 12; case UAC1_EXTENSION_UNIT: - strcpy(name, "Ext Unit"); return 8; + strcpy(name, "Ext Unit"); + return 8; case UAC_MIXER_UNIT: - strcpy(name, "Mixer"); return 5; + strcpy(name, "Mixer"); + return 5; default: return sprintf(name, "Unit %d", iterm->id); } @@ -649,29 +664,35 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm switch (iterm->type & 0xff00) { case 0x0100: - strcpy(name, "PCM"); return 3; + strcpy(name, "PCM"); + return 3; case 0x0200: - strcpy(name, "Mic"); return 3; + strcpy(name, "Mic"); + return 3; case 0x0400: - strcpy(name, "Headset"); return 7; + strcpy(name, "Headset"); + return 7; case 0x0500: - strcpy(name, "Phone"); return 5; + strcpy(name, "Phone"); + return 5; } - for (names = iterm_names; names->type; names++) + for (names = iterm_names; names->type; names++) { if (names->type == iterm->type) { strcpy(name, names->name); return strlen(names->name); } + } + return 0; } - /* * parse the source unit recursively until it reaches to a terminal * or a branched unit. */ -static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) +static int check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) { int err; void *p1; @@ -766,7 +787,6 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ return -ENODEV; } - /* * Feature Unit */ @@ -794,7 +814,6 @@ static struct usb_feature_control_info audio_feature_info[] = { { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; - /* private_free callback */ static void usb_mixer_elem_free(struct snd_kcontrol *kctl) { @@ -802,7 +821,6 @@ static void usb_mixer_elem_free(struct snd_kcontrol *kctl) kctl->private_data = NULL; } - /* * interface to ALSA control for feature/mixer units */ @@ -906,7 +924,6 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 384; } break; - } } @@ -939,21 +956,26 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { usb_audio_err(cval->mixer->chip, "%d:%d: cannot get min/max values for control %d (id %d)\n", - cval->id, snd_usb_ctrl_intf(cval->mixer->chip), cval->control, cval->id); + cval->id, snd_usb_ctrl_intf(cval->mixer->chip), + cval->control, cval->id); return -EINVAL; } - if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { + if (get_ctl_value(cval, UAC_GET_RES, + (cval->control << 8) | minchn, + &cval->res) < 0) { cval->res = 1; } else { int last_valid_res = cval->res; while (cval->res > 1) { if (snd_usb_mixer_set_ctl_value(cval, UAC_SET_RES, - (cval->control << 8) | minchn, cval->res / 2) < 0) + (cval->control << 8) | minchn, + cval->res / 2) < 0) break; cval->res /= 2; } - if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) + if (get_ctl_value(cval, UAC_GET_RES, + (cval->control << 8) | minchn, &cval->res) < 0) cval->res = last_valid_res; } if (cval->res == 0) @@ -1017,7 +1039,8 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, #define get_min_max(cval, def) get_min_max_with_quirks(cval, def, NULL) /* get a feature/mixer unit info */ -static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *cval = kcontrol->private_data; @@ -1051,7 +1074,8 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ } /* get the current value from feature/mixer unit */ -static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int c, cnt, val, err; @@ -1082,7 +1106,8 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e } /* put the current value to feature/mixer unit */ -static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int c, cnt, val, oval, err; @@ -1136,22 +1161,25 @@ static struct snd_kcontrol_new usb_feature_unit_ctl_ro = { .put = NULL, }; -/* This symbol is exported in order to allow the mixer quirks to - * hook up to the standard feature unit control mechanism */ +/* + * This symbol is exported in order to allow the mixer quirks to + * hook up to the standard feature unit control mechanism + */ struct snd_kcontrol_new *snd_usb_feature_unit_ctl = &usb_feature_unit_ctl; /* * build a feature control */ - static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) { return strlcat(kctl->id.name, str, sizeof(kctl->id.name)); } -/* A lot of headsets/headphones have a "Speaker" mixer. Make sure we - rename it to "Headphone". We determine if something is a headphone - similar to how udev determines form factor. */ +/* + * A lot of headsets/headphones have a "Speaker" mixer. Make sure we + * rename it to "Headphone". We determine if something is a headphone + * similar to how udev determines form factor. + */ static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, struct snd_card *card) { @@ -1201,7 +1229,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (! cval) { + if (!cval) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); return; } @@ -1222,15 +1250,17 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval->ch_readonly = readonly_mask; } - /* if all channels in the mask are marked read-only, make the control + /* + * If all channels in the mask are marked read-only, make the control * read-only. set_cur_mix_value() will check the mask again and won't - * issue write commands to read-only channels. */ + * issue write commands to read-only channels. + */ if (cval->channels == readonly_mask) kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); else kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); - if (! kctl) { + if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(cval); return; @@ -1239,48 +1269,53 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); mapped_name = len != 0; - if (! len && nameid) + if (!len && nameid) len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); switch (control) { case UAC_FU_MUTE: case UAC_FU_VOLUME: - /* determine the control name. the rule is: + /* + * determine the control name. the rule is: * - if a name id is given in descriptor, use it. * - if the connected input can be determined, then use the name * of terminal type. * - if the connected output can be determined, use it. * - otherwise, anonymous name. */ - if (! len) { - len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 1); - if (! len) - len = get_term_name(state, &state->oterm, kctl->id.name, sizeof(kctl->id.name), 1); - if (! len) - len = snprintf(kctl->id.name, sizeof(kctl->id.name), + if (!len) { + len = get_term_name(state, iterm, kctl->id.name, + sizeof(kctl->id.name), 1); + if (!len) + len = get_term_name(state, &state->oterm, + kctl->id.name, + sizeof(kctl->id.name), 1); + if (!len) + len = snprintf(kctl->id.name, + sizeof(kctl->id.name), "Feature %d", unitid); } if (!mapped_name) check_no_speaker_on_headset(kctl, state->mixer->chip->card); - /* determine the stream direction: + /* + * determine the stream direction: * if the connected output is USB stream, then it's likely a * capture stream. otherwise it should be playback (hopefully :) */ - if (! mapped_name && ! (state->oterm.type >> 16)) { - if ((state->oterm.type & 0xff00) == 0x0100) { + if (!mapped_name && !(state->oterm.type >> 16)) { + if ((state->oterm.type & 0xff00) == 0x0100) len = append_ctl_name(kctl, " Capture"); - } else { + else len = append_ctl_name(kctl, " Playback"); - } } append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); break; default: - if (! len) + if (!len) strlcpy(kctl->id.name, audio_feature_info[control-1].name, sizeof(kctl->id.name)); break; @@ -1300,33 +1335,35 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } range = (cval->max - cval->min) / cval->res; - /* Are there devices with volume range more than 255? I use a bit more + /* + * Are there devices with volume range more than 255? I use a bit more * to be sure. 384 is a resolution magic number found on Logitech * devices. It will definitively catch all buggy Logitech devices. */ if (range > 384) { - usb_audio_warn(state->chip, "Warning! Unlikely big " - "volume range (=%u), cval->res is probably wrong.", - range); + usb_audio_warn(state->chip, + "Warning! Unlikely big volume range (=%u), " + "cval->res is probably wrong.", + range); usb_audio_warn(state->chip, "[%d] FU [%s] ch = %d, " - "val = %d/%d/%d", cval->id, - kctl->id.name, cval->channels, - cval->min, cval->max, cval->res); + "val = %d/%d/%d", cval->id, + kctl->id.name, cval->channels, + cval->min, cval->max, cval->res); } usb_audio_dbg(state->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", - cval->id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res); + cval->id, kctl->id.name, cval->channels, + cval->min, cval->max, cval->res); snd_usb_mixer_add_control(state->mixer, kctl); } - - /* * parse a feature unit * * most of controls are defined here. */ -static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr) +static int parse_audio_feature_unit(struct mixer_build *state, int unitid, + void *_ftr) { int channels, i, j; struct usb_audio_term iterm; @@ -1400,15 +1437,25 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void for (i = 0; i < 10; i++) { unsigned int ch_bits = 0; for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); + unsigned int mask; + + mask = snd_usb_combine_bytes(bmaControls + + csize * (j+1), csize); if (mask & (1 << i)) ch_bits |= (1 << j); } /* audio class v1 controls are never read-only */ - if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, 0); + + /* + * The first channel must be set + * (for ease of programming). + */ + if (ch_bits & 1) + build_feature_ctl(state, _ftr, ch_bits, i, + &iterm, unitid, 0); if (master_bits & (1 << i)) - build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0); + build_feature_ctl(state, _ftr, 0, i, &iterm, + unitid, 0); } } else { /* UAC_VERSION_2 */ for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) { @@ -1416,7 +1463,10 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void unsigned int ch_read_only = 0; for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); + unsigned int mask; + + mask = snd_usb_combine_bytes(bmaControls + + csize * (j+1), csize); if (uac2_control_is_readable(mask, i)) { ch_bits |= (1 << j); if (!uac2_control_is_writeable(mask, i)) @@ -1424,12 +1474,22 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } } - /* NOTE: build_feature_ctl() will mark the control read-only if all channels - * are marked read-only in the descriptors. Otherwise, the control will be - * reported as writeable, but the driver will not actually issue a write - * command for read-only channels */ - if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only); + /* + * NOTE: build_feature_ctl() will mark the control + * read-only if all channels are marked read-only in + * the descriptors. Otherwise, the control will be + * reported as writeable, but the driver will not + * actually issue a write command for read-only + * channels. + */ + + /* + * The first channel must be set + * (for ease of programming). + */ + if (ch_bits & 1) + build_feature_ctl(state, _ftr, ch_bits, i, + &iterm, unitid, ch_read_only); if (uac2_control_is_readable(master_bits, i)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, !uac2_control_is_writeable(master_bits, i)); @@ -1439,7 +1499,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void return 0; } - /* * Mixer Unit */ @@ -1450,7 +1509,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void * the callbacks are identical with feature unit. * input channel number (zero based) is given in control field instead. */ - static void build_mixer_unit_ctl(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc, int in_pin, int in_ch, int unitid, @@ -1467,7 +1525,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (! cval) + if (!cval) return; cval->mixer = state->mixer; @@ -1475,7 +1533,9 @@ static void build_mixer_unit_ctl(struct mixer_build *state, cval->control = in_ch + 1; /* based on 1 */ cval->val_type = USB_MIXER_S16; for (i = 0; i < num_outs; i++) { - if (check_matrix_bitmap(uac_mixer_unit_bmControls(desc, state->mixer->protocol), in_ch, i, num_outs)) { + __u8 *c = uac_mixer_unit_bmControls(desc, state->mixer->protocol); + + if (check_matrix_bitmap(c, in_ch, i, num_outs)) { cval->cmask |= (1 << i); cval->channels++; } @@ -1485,7 +1545,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, get_min_max(cval, 0); kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); - if (! kctl) { + if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(cval); return; @@ -1493,9 +1553,10 @@ static void build_mixer_unit_ctl(struct mixer_build *state, kctl->private_free = usb_mixer_elem_free; len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); - if (! len) - len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0); - if (! len) + if (!len) + len = get_term_name(state, iterm, kctl->id.name, + sizeof(kctl->id.name), 0); + if (!len) len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1); append_ctl_name(kctl, " Volume"); @@ -1504,24 +1565,28 @@ static void build_mixer_unit_ctl(struct mixer_build *state, snd_usb_mixer_add_control(state->mixer, kctl); } - /* * parse a mixer unit */ -static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *raw_desc) +static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, + void *raw_desc) { struct uac_mixer_unit_descriptor *desc = raw_desc; struct usb_audio_term iterm; int input_pins, num_ins, num_outs; int pin, ich, err; - if (desc->bLength < 11 || ! (input_pins = desc->bNrInPins) || ! (num_outs = uac_mixer_unit_bNrChannels(desc))) { - usb_audio_err(state->chip, "invalid MIXER UNIT descriptor %d\n", unitid); + if (desc->bLength < 11 || !(input_pins = desc->bNrInPins) || + !(num_outs = uac_mixer_unit_bNrChannels(desc))) { + usb_audio_err(state->chip, + "invalid MIXER UNIT descriptor %d\n", + unitid); return -EINVAL; } /* no bmControls field (e.g. Maya44) -> ignore */ if (desc->bLength <= 10 + input_pins) { - usb_audio_dbg(state->chip, "MU %d has no bmControls field\n", unitid); + usb_audio_dbg(state->chip, "MU %d has no bmControls field\n", + unitid); return 0; } @@ -1535,12 +1600,14 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r if (err < 0) return err; num_ins += iterm.channels; - for (; ich < num_ins; ++ich) { + for (; ich < num_ins; ich++) { int och, ich_has_controls = 0; - for (och = 0; och < num_outs; ++och) { - if (check_matrix_bitmap(uac_mixer_unit_bmControls(desc, state->mixer->protocol), - ich, och, num_outs)) { + for (och = 0; och < num_outs; och++) { + __u8 *c = uac_mixer_unit_bmControls(desc, + state->mixer->protocol); + + if (check_matrix_bitmap(c, ich, och, num_outs)) { ich_has_controls = 1; break; } @@ -1553,13 +1620,13 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r return 0; } - /* * Processing Unit / Extension Unit */ /* get callback for processing/extension unit */ -static int mixer_ctl_procunit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_procunit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int err, val; @@ -1577,7 +1644,8 @@ static int mixer_ctl_procunit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ } /* put callback for processing/extension unit */ -static int mixer_ctl_procunit_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_procunit_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, oval, err; @@ -1606,7 +1674,6 @@ static struct snd_kcontrol_new mixer_procunit_ctl = { .put = mixer_ctl_procunit_put, }; - /* * predefined data for processing units */ @@ -1697,10 +1764,13 @@ static struct procunit_info extunits[] = { { USB_XU_DEVICE_OPTIONS, "AnalogueIn Soft Limit", soft_limit_xu_info }, { 0 } }; + /* * build a processing/extension unit */ -static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw_desc, struct procunit_info *list, char *name) +static int build_audio_procunit(struct mixer_build *state, int unitid, + void *raw_desc, struct procunit_info *list, + char *name) { struct uac_processing_unit_descriptor *desc = raw_desc; int num_ins = desc->bNrInPins; @@ -1733,19 +1803,19 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw for (info = list; info && info->type; info++) if (info->type == type) break; - if (! info || ! info->type) + if (!info || !info->type) info = &default_info; for (valinfo = info->values; valinfo->control; valinfo++) { __u8 *controls = uac_processing_unit_bmControls(desc, state->mixer->protocol); - if (! (controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1)))) + if (!(controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1)))) continue; map = find_map(state, unitid, valinfo->control); if (check_ignored_ctl(map)) continue; cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (! cval) { + if (!cval) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); return -ENOMEM; } @@ -1765,7 +1835,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre/0204 + /* + * E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; @@ -1777,60 +1848,70 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw } kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); - if (! kctl) { + if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(cval); return -ENOMEM; } kctl->private_free = usb_mixer_elem_free; - if (check_mapped_name(map, kctl->id.name, - sizeof(kctl->id.name))) + if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) { /* nothing */ ; - else if (info->name) + } else if (info->name) { strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name)); - else { + } else { nameid = uac_processing_unit_iProcessing(desc, state->mixer->protocol); len = 0; if (nameid) - len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); - if (! len) + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, + sizeof(kctl->id.name)); + if (!len) strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); } append_ctl_name(kctl, " "); append_ctl_name(kctl, valinfo->suffix); usb_audio_dbg(state->chip, - "[%d] PU [%s] ch = %d, val = %d/%d\n", - cval->id, kctl->id.name, cval->channels, cval->min, cval->max); - if ((err = snd_usb_mixer_add_control(state->mixer, kctl)) < 0) + "[%d] PU [%s] ch = %d, val = %d/%d\n", + cval->id, kctl->id.name, cval->channels, + cval->min, cval->max); + + err = snd_usb_mixer_add_control(state->mixer, kctl); + if (err < 0) return err; } return 0; } - -static int parse_audio_processing_unit(struct mixer_build *state, int unitid, void *raw_desc) +static int parse_audio_processing_unit(struct mixer_build *state, int unitid, + void *raw_desc) { - return build_audio_procunit(state, unitid, raw_desc, procunits, "Processing Unit"); + return build_audio_procunit(state, unitid, raw_desc, + procunits, "Processing Unit"); } -static int parse_audio_extension_unit(struct mixer_build *state, int unitid, void *raw_desc) +static int parse_audio_extension_unit(struct mixer_build *state, int unitid, + void *raw_desc) { - /* Note that we parse extension units with processing unit descriptors. - * That's ok as the layout is the same */ - return build_audio_procunit(state, unitid, raw_desc, extunits, "Extension Unit"); + /* + * Note that we parse extension units with processing unit descriptors. + * That's ok as the layout is the same. + */ + return build_audio_procunit(state, unitid, raw_desc, + extunits, "Extension Unit"); } - /* * Selector Unit */ -/* info callback for selector unit +/* + * info callback for selector unit * use an enumerator type for routing */ -static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *cval = kcontrol->private_data; const char **itemlist = (const char **)kcontrol->private_value; @@ -1841,7 +1922,8 @@ static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl } /* get callback for selector unit */ -static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, err; @@ -1860,7 +1942,8 @@ static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ } /* put callback for selector unit */ -static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, oval, err; @@ -1889,8 +1972,8 @@ static struct snd_kcontrol_new mixer_selectunit_ctl = { .put = mixer_ctl_selector_put, }; - -/* private free callback. +/* + * private free callback. * free both private_data and private_value */ static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl) @@ -1915,7 +1998,8 @@ static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl) /* * parse a selector unit */ -static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void *raw_desc) +static int parse_audio_selector_unit(struct mixer_build *state, int unitid, + void *raw_desc) { struct uac_selector_unit_descriptor *desc = raw_desc; unsigned int i, nameid, len; @@ -1944,7 +2028,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void return 0; cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (! cval) { + if (!cval) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); return -ENOMEM; } @@ -1963,7 +2047,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void cval->control = 0; namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); - if (! namelist) { + if (!namelist) { usb_audio_err(state->chip, "cannot malloc\n"); kfree(cval); return -ENOMEM; @@ -1973,7 +2057,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void struct usb_audio_term iterm; len = 0; namelist[i] = kmalloc(MAX_ITEM_NAME_LEN, GFP_KERNEL); - if (! namelist[i]) { + if (!namelist[i]) { usb_audio_err(state->chip, "cannot malloc\n"); while (i--) kfree(namelist[i]); @@ -1983,14 +2067,14 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void } len = check_mapped_selector_name(state, unitid, i, namelist[i], MAX_ITEM_NAME_LEN); - if (! len && check_input_term(state, desc->baSourceID[i], &iterm) >= 0) + if (!len && check_input_term(state, desc->baSourceID[i], &iterm) >= 0) len = get_term_name(state, &iterm, namelist[i], MAX_ITEM_NAME_LEN, 0); - if (! len) + if (!len) sprintf(namelist[i], "Input %d", i); } kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval); - if (! kctl) { + if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(namelist); kfree(cval); @@ -2004,11 +2088,12 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void if (len) ; else if (nameid) - snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); + snd_usb_copy_string_desc(state, nameid, kctl->id.name, + sizeof(kctl->id.name)); else { len = get_term_name(state, &state->oterm, kctl->id.name, sizeof(kctl->id.name), 0); - if (! len) + if (!len) strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) @@ -2027,7 +2112,6 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void return 0; } - /* * parse an audio unit recursively */ @@ -2125,14 +2209,16 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } p = NULL; - while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen, + while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, + mixer->hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) { if (mixer->protocol == UAC_VERSION_1) { struct uac1_output_terminal_descriptor *desc = p; if (desc->bLength < sizeof(*desc)) continue; /* invalid descriptor? */ - set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + /* mark terminal ID as visited */ + set_bit(desc->bTerminalID, state.unitbitmap); state.oterm.id = desc->bTerminalID; state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; @@ -2144,7 +2230,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (desc->bLength < sizeof(*desc)) continue; /* invalid descriptor? */ - set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + /* mark terminal ID as visited */ + set_bit(desc->bTerminalID, state.unitbitmap); state.oterm.id = desc->bTerminalID; state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; @@ -2152,7 +2239,10 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (err < 0 && err != -EINVAL) return err; - /* for UAC2, use the same approach to also add the clock selectors */ + /* + * For UAC2, use the same approach to also add the + * clock selectors + */ err = parse_audio_unit(&state, desc->bCSourceID); if (err < 0 && err != -EINVAL) return err; @@ -2306,7 +2396,9 @@ static void snd_usb_mixer_interrupt(struct urb *urb) } requeue: - if (ustatus != -ENOENT && ustatus != -ECONNRESET && ustatus != -ESHUTDOWN) { + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { urb->dev = mixer->chip->dev; usb_submit_urb(urb, GFP_ATOMIC); } -- 1.9.0

2 3

[alsa-devel] alsa platform driver testing without codec driver
by Babu Balu 24 May '14

24 May '14

Hi, Need an input from you guys please. Is there a possibility to play out an audio using aplay or any alsa tools without the presence of the codec driver In our target platform, we have the SoC's I2S platform driver working, and a dummy codec driver is registered The actual codec chip sits on a external board and connected to the main SoC via I2S and I2C lines. And that codec chip is configured via a middleware package(not alsa) So we do not have a codec driver at the kernel space, we have just provided a plain I2C interface driver for the middleware package. so that it can send I2C commands directly to the audio codec chip. Based on the requirement have to give the control interface of the codec chip to the middleware package In this case is there a possibility to play out an audio file using alsa tools(eg. aplay) Just a raw PCM format should be enough. Thanks, Babu ::DISCLAIMER:: ----------------------------------------------------------------------- ----------------------------------------------------------------------- ------ The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------- ----------------------------------------------------------------------- ------

1 0

[alsa-devel] [PATCH 00/13] Fix OMAP4+ HDMI audio
by Jyri Sarha 23 May '14

23 May '14

This patch set fixes the resource sharing problems between omap-hdmi-dai driver and OMAPDSS HDMI driver. It does it by registering the OMAP HDMI audio related ASoC drivers from OMAPDSS HDMI driver. Platform data structs have been added for omap-hdmi-dai and omap-hdmi-card drivers to pass the information and resources from OMAPDSS HDMI driver. The idea of this patch set is to fix HDMI audio for the next release after it got broken by OMAPDSS DT changes. It does not mean that I have abandoned the patch set that integrates the omap-hdmi-dai driver into OMAPDSS HDMI driver. The OMAPDSS side of those patches just had some dependencies to the recent ASoC side patches that would have caused problems in the next merge. I'll mail a revised and rebased version of those patches soon. Best regards, Jyri Jyri Sarha (13): ARM: omap4.dtsi: Add audio related parametes to hdmi node ARM: omap5.dtsi: Add audio related parameters to hdmi node ARM: OMAP2+: Remove non working OMAP HDMI audio initialization OMAPDSS: hdmi_wp: Add function for getting hdmi_wp physical base address OMAPDSS: hdmi_audio: Add hdmi_audio.c for registering HDMI audio support ASoC: omap-hdmi-dai: Add platform data struct for omap-hdmi-dai driver ASoC: omap-hdmi-card: Add platform data stuct for omap-hdmi-card driver ASoC: omap-hdmi: Changes for registeing the driver from OMAPDSS ASoC: omap-hdmi-card: Changes for registeing the driver from OMAPDSS OMAPDSS: hdmi4: Register HDMI audio ASoC drivers from HDMI driver OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi5: Register HDMI audio ASoC drivers from HDMI driver ASoC: omap: Add Kconfig option for OMAP5 HDMI audio arch/arm/boot/dts/omap4.dtsi | 2 + arch/arm/boot/dts/omap5.dtsi | 2 + arch/arm/mach-omap2/devices.c | 28 --------- drivers/video/fbdev/omap2/dss/Makefile | 2 +- drivers/video/fbdev/omap2/dss/hdmi.h | 17 ++++- drivers/video/fbdev/omap2/dss/hdmi4.c | 15 +++++ drivers/video/fbdev/omap2/dss/hdmi5.c | 15 +++++ drivers/video/fbdev/omap2/dss/hdmi_audio.c | 92 ++++++++++++++++++++++++++++ drivers/video/fbdev/omap2/dss/hdmi_wp.c | 6 ++ include/sound/omap-hdmi-card-pdata.h | 28 +++++++++ include/sound/omap-hdmi-dai-pdata.h | 31 ++++++++++ sound/soc/omap/Kconfig | 13 +++- sound/soc/omap/omap-hdmi-card.c | 20 ++++-- sound/soc/omap/omap-hdmi.c | 65 +++++--------------- 14 files changed, 253 insertions(+), 83 deletions(-) create mode 100644 drivers/video/fbdev/omap2/dss/hdmi_audio.c create mode 100644 include/sound/omap-hdmi-card-pdata.h create mode 100644 include/sound/omap-hdmi-dai-pdata.h -- 1.7.9.5

1 13

Re: [alsa-devel] [PATCH] ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
by Mark Brown 23 May '14

23 May '14

On Fri, May 23, 2014 at 07:19:27PM +0400, Alexander Shiyan wrote: > Yes, now I see. > So on my opinion the true method is transform this board (for DT case) is > to use generic sound card. In this patch , I just try to remove non-existent > symbol dependencies, so if we have is_machine_xx checks I do not know > the true way, since cannot test this. Yes, transitioning to simple-card would be ideal if you can do that. I guess we can apply this one for now since it doesn't make things any worse at least.

1 0

[alsa-devel] [PATCH 00/19] Rework OMAP4+ HDMI audio support
by Jyri Sarha 23 May '14

23 May '14

Since RFC version of the patch set: - Split callbacks removal patch away from "Integrated ASoC DAI component driver implementation" patches for easier reading This set of patches fixes OMAP4+ HDMI audio. The structure of the implementatin looks a bit different than before. Instead of creating a driver specific API for a separate ASoC component driver to connect to, this implementation integrates an the component driver into the HDMI driver. The idea is to use an existing ASoC component driver API instead of creating a new custom API for each HDMI IP and to avoid splitting the driver to half for separate video and audio parts connected with the API. The new implementation also uses simple-audio-card for a machine driver instead of having its own HW specific machine driver. The patches are based on 3.15-rc2 merged with git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next and git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git work/dss-dt-omap5 Everything is pushed here here: git://git.ti.com/~jyrisarha/ti-linux-kernel/jyrisarhas-audio-video-linux-fe… omap-hdmi-audio Best regards, Jyri Jyri Sarha (19): ASoC: omap-pcm: Move omap-pcm under include/sound OMAPDSS: hdmi-common: Add hdmi_dss_audio_from_hw_params() OMAPDSS: hdmi4: Remove callbacks for an external ASoC DAI driver OMAPDSS: hdmi4: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Add depencies and help section to OMAP4_DSS_HDMI_AUDIO OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi5_core: Fix compilation with OMAP5_DSS_HDMI_AUDIO OMAPDSS: hdmi5: Remove callbacks for an external ASoC DAI driver OMAPDSS: hdmi5: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Add depencies and help section to OMAP5_DSS_HDMI_AUDIO ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove obsolete audio code ARM: omap4.dtsi: Add audio related parametes to hdmi node ARM: omap4-panda-common.dtsi: Add HDMI audio nodes ARM: omap5.dtsi: Add audio related parameters to hdmi node ARM: omap5-uevm.dts: Add hdmi audio related nodes ARM: omap2plus_defconfig: Build DSS HDMI support for OMAP4 in kernel ARM: omap2plus_defconfig: Enable OMAP5 HDMI support ARM: omap2plus_defconfig: Enable OMAP4+ HDMI audio support .../devicetree/bindings/video/ti,omap4-dss.txt | 4 + .../devicetree/bindings/video/ti,omap5-dss.txt | 4 + arch/arm/boot/dts/omap4-panda-common.dtsi | 21 +- arch/arm/boot/dts/omap4.dtsi | 2 + arch/arm/boot/dts/omap5-uevm.dts | 19 + arch/arm/boot/dts/omap5.dtsi | 2 + arch/arm/configs/omap2plus_defconfig | 20 +- .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ------ .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 19 +- drivers/video/fbdev/omap2/dss/hdmi.h | 8 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 216 ++++++++---- drivers/video/fbdev/omap2/dss/hdmi5.c | 216 ++++++++---- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 2 +- drivers/video/fbdev/omap2/dss/hdmi_common.c | 117 ++++++- include/sound/omap-pcm.h | 30 ++ include/video/omapdss.h | 31 -- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/omap/Kconfig | 13 - sound/soc/omap/Makefile | 4 - sound/soc/omap/omap-dmic.c | 2 +- sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 -------------------- sound/soc/omap/omap-hdmi.h | 38 -- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/omap-pcm.h | 30 -- 27 files changed, 540 insertions(+), 870 deletions(-) create mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h delete mode 100644 sound/soc/omap/omap-pcm.h -- 1.7.9.5

5 34

[alsa-devel] [PATCH] hda-emu: Add new HDMI commands
by David Henningsson 23 May '14

23 May '14

Some new HDMI commands were added recently to the HDA driver. This patch makes hda-emu support these new commands (or at least, not output errors). Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-int.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ include/hda-types.h | 2 ++ 2 files changed, 55 insertions(+) diff --git a/hda-int.c b/hda-int.c index ef90fe8..6dd862d 100644 --- a/hda-int.c +++ b/hda-int.c @@ -823,6 +823,52 @@ static int get_cvt_channel_count(struct xhda_codec *codec, return node->cvt_channel_count; } +static int get_asp_channel_slot(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + return (node->asp_channel_slot[cmd & 0xf] << 4) + (cmd & 0xf); +} + +static int set_asp_channel_slot(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + node->asp_channel_slot[cmd & 0xf] = (cmd & 0xf0) >> 4; + return 0; +} + +static int set_dip_index(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + return 0; /* FIXME */ +} + +static int set_dip_data(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + return 0; /* FIXME */ +} + +static int get_dip_size(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + if (cmd & 0x8) + return 128; /* ELD buffer size */ + return 32; /* DIP buffer size */ +} + +static int set_dip_xmitctrl(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + node->dip_xmitctrl = cmd & 0xc0; + return 0; +} + +static int get_dip_xmitctrl(struct xhda_codec *codec, + struct xhda_node *node, unsigned int cmd) +{ + return node->dip_xmitctrl; +} + /* * parameters @@ -1063,6 +1109,10 @@ static struct xhda_verb_table verb_tbl[] = { { 0x71e, set_config_def_2, "set_config_def_2" }, { 0x71f, set_config_def_3, "set_config_def_3" }, { 0x72d, set_cvt_channel_count, "set_cvt_channel_count" }, + { 0x730, set_dip_index, "set_dip_index" }, + { 0x731, set_dip_data, "set_dip_data" }, + { 0x732, set_dip_xmitctrl, "set_dip_xmitctrl" }, + { 0x734, set_asp_channel_slot, "set_asp_channel_slot" }, { 0x7ff, set_codec_reset, "set_codec_reset" }, { 0xf00, get_parameters, "get_parameters" }, { 0xf01, get_connect_sel, "get_connect_sel" }, @@ -1087,6 +1137,9 @@ static struct xhda_verb_table verb_tbl[] = { { 0xf1c, get_config_default, "get_config_default" }, { 0xf20, get_ssid, "get_ssid" }, { 0xf2d, get_cvt_channel_count, "get_cvt_channel_count" }, + { 0xf2e, get_dip_size, "get_dip_size" }, + { 0xf32, get_dip_xmitctrl, "get_dip_xmitctrl" }, + { 0xf34, get_asp_channel_slot, "get_asp_channel_slot" }, {} }; diff --git a/include/hda-types.h b/include/hda-types.h index fe6b31f..46bfc67 100644 --- a/include/hda-types.h +++ b/include/hda-types.h @@ -58,6 +58,8 @@ struct xhda_node { unsigned char gpio_wake, gpio_unsol, gpio_sticky; unsigned int coef_idx; unsigned char cvt_channel_count; + unsigned char asp_channel_slot[16]; + unsigned char dip_xmitctrl; struct xhda_coef_table *coef_tbl; struct xhda_node *next; }; -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 23 May '14

23 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. According to Mark Brown: "Yes, reg_cache isn't there if we're not using ASoC level caching. The fix should just be to replace the direct cache references with snd_soc_read()s which will end up in a cache lookup if the register is cached." Do as suggested and also complete the cache array with the missing registers so that sgtl5000_restore_regs() can properly cache the all the registers it needs. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Also fix sgtl5000_set_bias_level so that suspend/resume/play sequence does not fail after several interactions sound/soc/codecs/sgtl5000.c | 50 +++++++++++++++++++++++++-------------------- 1 file changed, 28 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..53511c6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -924,20 +939,6 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; udelay(10); - - regcache_cache_only(sgtl5000->regmap, false); - - ret = regcache_sync(sgtl5000->regmap); - if (ret != 0) { - dev_err(codec->dev, - "Failed to restore cache: %d\n", ret); - - regcache_cache_only(sgtl5000->regmap, true); - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - return ret; - } } break; @@ -1063,7 +1064,10 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) #ifdef CONFIG_SUSPEND static int sgtl5000_suspend(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(sgtl5000->regmap, true); return 0; } @@ -1075,7 +1079,6 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) */ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; u16 reg; /* restore regular registers */ @@ -1089,12 +1092,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) reg == SGTL5000_CHIP_REF_CTRL) continue; - snd_soc_write(codec, reg, cache[reg]); + snd_soc_write(codec, reg, snd_soc_read(codec, reg)); } /* restore dap registers */ for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); + snd_soc_write(codec, reg, snd_soc_read(codec, reg)); /* * restore these regs according to the power setting sequence in @@ -1110,29 +1113,32 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); + snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER)); snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_CLK_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_REF_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_LINE_OUT_CTRL)); return 0; } static int sgtl5000_resume(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Restore registers by cached in memory */ sgtl5000_restore_regs(codec); + + regcache_cache_only(sgtl5000->regmap, false); return 0; } #else -- 1.8.3.2

2 2

[alsa-devel] [PATCH 0/2] ASoC: max98090/max98095: Add master clock handing
by Tushar Behera 23 May '14

23 May '14

These CODECs are used on Snow/Peach-pit boards and the master clock needs to be set at 24MHz. Adding support in respective codec drivers so that we can update the master clock frequency from sound card driver through set_sysclk. These patches are prepared as per review comments on following patch. [PATCH] ASoC: samsung: Add 'mclk' handling for Snow sound-card driver https://lkml.org/lkml/2014/5/20/4 Tushar Behera (2): ASoC: max98090: Add master clock handling ASoC: max98095: Add master clock handling .../devicetree/bindings/sound/max98090.txt | 6 ++++++ .../devicetree/bindings/sound/max98095.txt | 6 ++++++ sound/soc/codecs/max98090.c | 13 +++++++++++++ sound/soc/codecs/max98090.h | 1 + sound/soc/codecs/max98095.c | 14 ++++++++++++++ 5 files changed, 40 insertions(+) -- 1.7.9.5

3 15

Re: [alsa-devel] [PATCH 2/2] ASoC: samsung: Add machine driver for odroidx2
by Mark Brown 23 May '14

23 May '14

On Fri, May 23, 2014 at 02:18:39AM +0000, Zhen Chen wrote: > <HTML xmlns:o = "urn:schemas-microsoft-com:office:office"><HEAD><TITLE>Samsung Enterprise Portal mySingle</TITLE> > <META content=IE=5 http-equiv=X-UA-Compatible> > <META content="text/html; charset=windows-1252" http-equiv=Content-Type> Please fix your mail configuration to send in plain text, I can't read this - it's not even got a multipart/alternative plain text bit.

1 0

[alsa-devel] [PATCH] ALSA: Remove deprecated snd_card_create()
by Takashi Iwai 23 May '14

23 May '14

Now all calls have been fixed, let's get rid of the old definition. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 7 ------- 1 file changed, 7 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index d3f5f818e0b9..eedda2cdfe57 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -282,13 +282,6 @@ int snd_card_new(struct device *parent, int idx, const char *xid, struct module *module, int extra_size, struct snd_card **card_ret); -static inline int __deprecated -snd_card_create(int idx, const char *id, struct module *module, int extra_size, - struct snd_card **ret) -{ - return snd_card_new(NULL, idx, id, module, extra_size, ret); -} - int snd_card_disconnect(struct snd_card *card); int snd_card_free(struct snd_card *card); int snd_card_free_when_closed(struct snd_card *card); -- 1.9.3

1 0

Re: [alsa-devel] [ALC1150] combined with Intel H97/Z97
by Takashi Iwai 23 May '14

23 May '14

At Thu, 22 May 2014 21:34:27 +0200, Arthur Borsboom wrote: > > Hi Takashi, > > I have tested the patch and it fixes the issue. > Tested on kernel 3.15.0-rc6 > > Do you think you can get this patch in 3.15? > Or will it be 3.16? I'll queue the patch below for 3.15. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix onboard audio on Intel H97/Z97 chipsets The recent Intel H97/Z97 chipsets need the similar setups like other Intel chipsets for snooping, etc. Especially without snooping, the audio playback stutters or gets corrupted. This fix patch just adds the corresponding PCI ID entry with the proper flags. Reported-and-tested-by: Arthur Borsboom <arthurborsboom(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2c54629d62d1..6cc3cf285558 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1743,6 +1743,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + /* 9 Series */ + { PCI_DEVICE(0x8086, 0x8ca0), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Wellsburg */ { PCI_DEVICE(0x8086, 0x8d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, -- 1.9.3

1 0

[alsa-devel] [RFC PATCH] ALSA: hda - Add a new quirk match based on default pin configuration
by David Henningsson 23 May '14

23 May '14

Normally, we match on pci ssid only. This works but needs new code for every machine. To catch more machines in the same quirk, let's add a new type of quirk, where we match on 1) PCI Subvendor ID (i e, not device, just vendor) 2) Codec ID 3) Pin configuration default If all these three match, we could be reasonably certain that the quirk should apply to the machine even though it might not be the exact same device. In case a quirk matches both the pci ssid quirk and this type of quirk, then the pci ssid quirk is chosen. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Hi Takashi, Do you think this could be working? If so I'll ask Hui to move some machines over and come up with a patch for that. Two questions: 1) I'm unsure if the "name" entry makes sense here - I mean, since the same quirk hopefully matches many machines, we can't use the machine name here. Should I just skip it? 2) Should I make a macro (like SND_PCI_QUIRK) to make the actual pinquirk tables look more compact? sound/pci/hda/hda_auto_parser.c | 39 ++++++++++++++++++++++++++++++++++----- sound/pci/hda/hda_local.h | 27 +++++++++++++++++++++++---- 2 files changed, 57 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 90d2fda..6904779 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -839,10 +839,22 @@ void snd_hda_apply_fixup(struct hda_codec *codec, int action) } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); -void snd_hda_pick_fixup(struct hda_codec *codec, - const struct hda_model_fixup *models, - const struct snd_pci_quirk *quirk, - const struct hda_fixup *fixlist) +static bool pin_config_match(struct hda_codec *codec, + const struct hda_pintbl *pins) +{ + for (; pins->nid; pins++) { + u32 def_conf = snd_hda_codec_get_pincfg(codec, pins->nid); + if (pins->val != def_conf) + return false; + } + return true; +} + +void snd_hda_pick_pin_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct snd_hda_pin_quirk *pin_quirk, + const struct hda_fixup *fixlist) { const struct snd_pci_quirk *q; int id = -1; @@ -889,10 +901,27 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } + if (id < 0 && pin_quirk) { + const struct snd_hda_pin_quirk *pq; + for (pq = pin_quirk; pq->subvendor; pq++) { + if (codec->bus->pci->subsystem_vendor != pq->subvendor) + continue; + if (codec->vendor_id != pq->codec) + continue; + if (pin_config_match(codec, pq->pins)) { + id = pq->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + name = pq->name; +#endif + break; + } + } + } + codec->fixup_id = id; if (id >= 0) { codec->fixup_list = fixlist; codec->fixup_name = name; } } -EXPORT_SYMBOL_GPL(snd_hda_pick_fixup); +EXPORT_SYMBOL_GPL(snd_hda_pick_pin_fixup); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e51d155..41a6817 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -425,15 +425,34 @@ enum { HDA_FIXUP_ACT_FREE, }; +struct snd_hda_pin_quirk { + unsigned int codec; /* Codec vendor/device ID */ + unsigned short subvendor; /* PCI subvendor ID */ + const struct hda_pintbl *pins; /* list of matching pins */ +#ifdef CONFIG_SND_DEBUG_VERBOSE + const char *name; +#endif + int value; /* quirk value */ +}; + int snd_hda_add_verbs(struct hda_codec *codec, const struct hda_verb *list); void snd_hda_apply_verbs(struct hda_codec *codec); void snd_hda_apply_pincfgs(struct hda_codec *codec, const struct hda_pintbl *cfg); void snd_hda_apply_fixup(struct hda_codec *codec, int action); -void snd_hda_pick_fixup(struct hda_codec *codec, - const struct hda_model_fixup *models, - const struct snd_pci_quirk *quirk, - const struct hda_fixup *fixlist); +void snd_hda_pick_pin_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct snd_hda_pin_quirk *pin_quirk, + const struct hda_fixup *fixlist); + +static inline void snd_hda_pick_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct hda_fixup *fixlist) +{ + snd_hda_pick_pin_fixup(codec, models, quirk, NULL, fixlist); +} /* * unsolicited event handler -- 1.9.1

2 8

[alsa-devel] [PATCH RFT] ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls
by Axel Lin 23 May '14

23 May '14

The new value argument needs proper shift to match the mask bit fields. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Brian, This patch replaces [PATCH RFT] ASoC: cs42l56: Fix update mute register bits in cs42l56_digital_mute. I just found more places of snd_soc_update_bits calls needs fix. Regards, Axel sound/soc/codecs/cs42l56.c | 71 ++++++++++++++++++---------------------------- sound/soc/codecs/cs42l56.h | 14 ++++----- 2 files changed, 35 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 5bb134b..c0be526 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -763,14 +763,14 @@ static int cs42l56_set_sysclk(struct snd_soc_dai *codec_dai, case CS42L56_MCLK_11P2896MHZ: case CS42L56_MCLK_12MHZ: case CS42L56_MCLK_12P288MHZ: - cs42l56->mclk_div2 = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; cs42l56->mclk_prediv = 0; break; case CS42L56_MCLK_22P5792MHZ: case CS42L56_MCLK_24MHZ: case CS42L56_MCLK_24P576MHZ: - cs42l56->mclk_div2 = 1; - cs42l56->mclk_prediv = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; + cs42l56->mclk_prediv = CS42L56_MCLK_PREDIV; break; default: return -EINVAL; @@ -839,62 +839,47 @@ static int cs42l56_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + unsigned int mask; if (mute) { /* Hit the DSP Mixer first */ - snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, - CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_MUTE); + mask = CS42L56_ADCAMIX_MUTE_MASK | CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | CS42L56_MSTA_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, mask, mask); + /* Mute ADC's */ - snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_MUTE); + mask = CS42L56_ADCA_MUTE_MASK | CS42L56_ADCB_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, mask, mask); + /* HP And LO */ snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); } else { - snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, - CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_UNMUTE); - snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_UNMUTE); + mask = CS42L56_ADCAMIX_MUTE_MASK | CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | CS42L56_MSTA_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, mask , 0); + + mask = CS42L56_ADCA_MUTE_MASK | CS42L56_ADCB_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, mask, 0); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, 0); } return 0; } diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h index ad2b50a..d2f68c6 100644 --- a/sound/soc/codecs/cs42l56.h +++ b/sound/soc/codecs/cs42l56.h @@ -80,19 +80,21 @@ #define CS42L56_PDN_HPB_MASK 0xc0 /* serial port and clk masks */ -#define CS42L56_MASTER_MODE 1 -#define CS42L56_SLAVE_MODE 0 +#define CS42L56_MASTER_MODE (1 << 6) +#define CS42L56_SLAVE_MODE (0 << 6) #define CS42L56_MS_MODE_MASK 0x40 -#define CS42L56_SCLK_INV 1 +#define CS42L56_SCLK_INV (1 << 5) #define CS42L56_SCLK_INV_MASK 0x20 #define CS42L56_SCLK_MCLK_MASK 0x18 +#define CS42L56_MCLK_PREDIV (1 << 6) #define CS42L56_MCLK_PREDIV_MASK 0x04 +#define CS42L56_MCLK_DIV2 (1 << 1) #define CS42L56_MCLK_DIV2_MASK 0x02 #define CS42L56_MCLK_DIS_MASK 0x01 #define CS42L56_CLK_AUTO_MASK 0x20 #define CS42L56_CLK_RATIO_MASK 0x1f -#define CS42L56_DIG_FMT_I2S 0 -#define CS42L56_DIG_FMT_LEFT_J 1 +#define CS42L56_DIG_FMT_I2S (0 << 3) +#define CS42L56_DIG_FMT_LEFT_J (1 << 3) #define CS42L56_DIG_FMT_MASK 0x08 /* Class H and misc ctl masks */ @@ -116,8 +118,6 @@ #define CS42L56_DEEMPH_MASK 0x40 #define CS42L56_PLYBCK_GANG_MASK 0x10 #define CS42L56_PCM_INV_MASK 0x0c -#define CS42L56_MUTE 1 -#define CS42L56_UNMUTE 0 #define CS42L56_ADCAMIX_MUTE_MASK 0x40 #define CS42L56_ADCBMIX_MUTE_MASK 0x80 #define CS42L56_PCMAMIX_MUTE_MASK 0x10 -- 1.8.3.2

2 2

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 23 May '14

23 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. According to Mark Brown: "Yes, reg_cache isn't there if we're not using ASoC level caching. The fix should just be to replace the direct cache references with snd_soc_read()s which will end up in a cache lookup if the register is cached." Do as suggested and also complete the cache array with the missing registers so that sgtl5000_restore_regs() can properly cache the all the registers it needs. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 39 ++++++++++++++++++++++++++++----------- 1 file changed, 28 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..aac4fbc 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -925,14 +940,11 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, return ret; udelay(10); - regcache_cache_only(sgtl5000->regmap, false); - ret = regcache_sync(sgtl5000->regmap); if (ret != 0) { dev_err(codec->dev, "Failed to restore cache: %d\n", ret); - regcache_cache_only(sgtl5000->regmap, true); regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1063,7 +1075,10 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) #ifdef CONFIG_SUSPEND static int sgtl5000_suspend(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(sgtl5000->regmap, true); return 0; } @@ -1075,7 +1090,6 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) */ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; u16 reg; /* restore regular registers */ @@ -1089,12 +1103,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) reg == SGTL5000_CHIP_REF_CTRL) continue; - snd_soc_write(codec, reg, cache[reg]); + snd_soc_write(codec, reg, snd_soc_read(codec, reg)); } /* restore dap registers */ for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); + snd_soc_write(codec, reg, snd_soc_read(codec, reg)); /* * restore these regs according to the power setting sequence in @@ -1110,29 +1124,32 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); + snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER)); snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_CLK_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_REF_CTRL)); snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); + snd_soc_read(codec, SGTL5000_CHIP_LINE_OUT_CTRL)); return 0; } static int sgtl5000_resume(struct snd_soc_codec *codec) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Restore registers by cached in memory */ sgtl5000_restore_regs(codec); + + regcache_cache_only(sgtl5000->regmap, false); return 0; } #else -- 1.8.3.2

1 0

[alsa-devel] [PATCH] sound: soc: samsung: dma.c: Fix to remove null pointer checks that could never happen
by Rickard Strandqvist 23 May '14

23 May '14

Removal of null pointer checks that could never happen Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/samsung/dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index dc09b71..0f17ae0 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -124,8 +124,7 @@ static void audio_buffdone(void *data) if (prtd->dma_pos >= prtd->dma_end) prtd->dma_pos = prtd->dma_start; - if (substream) - snd_pcm_period_elapsed(substream); + snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); if (!samsung_dma_has_circular()) { -- 1.7.10.4

2 1

[alsa-devel] The resume of sgtl5000 causes kernel Oops
by Shawn Guo 23 May '14

23 May '14

The suspend/resume test on imx51-babbage board discovers a kernel Oops caused by sgtl5000_resume(). I believe it happens due to commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly). After the commit, it seems that the codec->reg_cache in function sgtl5000_restore_regs() becomes invalid. It looks that the register restoring in the function requires some strict sequence, I'm not sure what the correct fix is. Shawn $ echo mem > /sys/power/state PM: Syncing filesystems ... done. PM: Preparing system for mem sleep Freezing user space processes ... (elapsed 0.006 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.004 seconds) done. PM: Entering mem sleep PM: suspend of devices complete after 41.598 msecs PM: suspend devices took 0.050 seconds PM: late suspend of devices complete after 6.434 msecs PM: noirq suspend of devices complete after 6.735 msecs Disabling non-boot CPUs ... PM: noirq resume of devices complete after 10.871 msecs PM: early resume of devices complete after 5.241 msecs Unable to handle kernel NULL pointer dereference at virtual address 00000010 pgd = 80004000 [00000010] *pgd=00000000 Internal error: Oops: 17 [#1] SMP ARM Modules linked in: CPU: 0 PID: 21 Comm: kworker/0:1 Not tainted 3.15.0-rc3-00071-g49a96400a142-dirty #118 Workqueue: events soc_resume_deferred task: 9e11c440 ti: 9e14e000 task.ti: 9e14e000 PC is at sgtl5000_resume+0x7c/0x124 LR is at mark_held_locks+0x70/0x94 pc : [<804e8fb8>] lr : [<800661b8>] psr: 80000013 sp : 9e14fe40 ip : 80a26e60 fp : 9e14fe5c r10: 9fbd6b00 r9 : 00000001 r8 : 00000000 r7 : 00000010 r6 : 9e369a00 r5 : 00000002 r4 : 00000000 r3 : 00000000 r2 : 00000000 r1 : 00000000 r0 : 9e369a00 Flags: Nzcv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: ac828019 DAC: 00000015 Process kworker/0:1 (pid: 21, stack limit = 0x9e14e240) Stack: (0x9e14fe40 to 0x9e150000) fe40: 9e387d84 9e369a00 9e387c64 9e387da4 9e14fe84 9e14fe60 804d90d0 804e8f48 fe60: 9e135300 9e387d84 9fbd36c0 9e14e000 00000000 9fbd6b00 9e14fee4 9e14fe88 fe80: 8003f060 804d8fd0 00000001 00000000 8003efec 9fbd36c0 8003f620 00000000 fea0: 00000000 9e14fea8 80e981a0 80a26e60 00000000 80843dfc 80661fac 9e135300 fec0: 9fbd36f0 9e14e000 9e135318 9e14e000 9fbd36c0 8094e442 9e14ff24 9e14fee8 fee0: 8003f700 8003eec4 80662510 00000000 9e134ac0 9e14e000 8003f5dc 00000000 ff00: 9e134ac0 9e135300 8003f5dc 00000000 00000000 00000000 9e14ffac 9e14ff28 ff20: 800463bc 8003f5e8 800663bc 00000000 9e14ff54 9e135300 00000000 00000000 ff40: dead4ead ffffffff ffffffff 80952424 00000000 00000000 807cc28c 9e14ff5c ff60: 9e14ff5c 00000000 00000000 dead4ead ffffffff ffffffff 80952424 00000000 ff80: 00000000 807cc28c 9e14ff88 9e14ff88 9e134ac0 800462ec 00000000 00000000 ffa0: 00000000 9e14ffb0 8000eb68 800462f8 00000000 00000000 00000000 00000000 ffc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 ffe0: 00000000 00000000 00000000 00000000 00000013 00000000 00c37f70 76f142d0 Backtrace: [<804e8f3c>] (sgtl5000_resume) from [<804d90d0>] (soc_resume_deferred+0x10c/0x30c) r7:9e387da4 r6:9e387c64 r5:9e369a00 r4:9e387d84 [<804d8fc4>] (soc_resume_deferred) from [<8003f060>] (process_one_work+0x1a8/0x444) r10:9fbd6b00 r8:00000000 r7:9e14e000 r6:9fbd36c0 r5:9e387d84 r4:9e135300 [<8003eeb8>] (process_one_work) from [<8003f700>] (worker_thread+0x124/0x398) r10:8094e442 r9:9fbd36c0 r8:9e14e000 r7:9e135318 r6:9e14e000 r5:9fbd36f0 r4:9e135300 [<8003f5dc>] (worker_thread) from [<800463bc>] (kthread+0xd0/0xec) r10:00000000 r9:00000000 r8:00000000 r7:8003f5dc r6:9e135300 r5:9e134ac0 r4:00000000 [<800462ec>] (kthread) from [<8000eb68>] (ret_from_fork+0x14/0x2c) r7:00000000 r6:00000000 r5:800462ec r4:9e134ac0 Code: e1a03084 e1a01004 e1a00006 0afffff0 (e19720b3) ---[ end trace eeb2712470f0de34 ]--- Unable to handle kernel paging request at virtual address ffffffd0 pgd = 80004000 [ffffffd0] *pgd=afff6821, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#2] SMP ARM Modules linked in: CPU: 0 PID: 21 Comm: kworker/0:1 Tainted: G D 3.15.0-rc3-00071-g49a96400a142-dirty #118 task: 9e11c440 ti: 9e14e000 task.ti: 9e14e000 PC is at kthread_data+0x10/0x18 LR is at wq_worker_sleeping+0x14/0xc8 pc : [<8004673c>] lr : [<8004047c>] psr: 00000193 sp : 9e14fb00 ip : 9e14fb10 fp : 9e14fb0c r10: 808f2d2c r9 : 9e11c698 r8 : 808e8c40 r7 : 00000000 r6 : 9e14e000 r5 : 9e11c440 r4 : 00000000 r3 : 00000000 r2 : 00000000 r1 : 00000000 r0 : 9e11c440 Flags: nzcv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c5387d Table: ac828019 DAC: 00000015 Process kworker/0:1 (pid: 21, stack limit = 0x9e14e240) Stack: (0x9e14fb00 to 0x9e150000) fb00: 9e14fb24 9e14fb10 8004047c 80046738 00000020 9fbd3c40 9e14fbc4 9e14fb28 fb20: 8065dd8c 80040474 9e14e000 60000113 9e14fb4c 9e14fb40 800663bc 800661e8 fb40: 9e14fb74 9e14fb50 808e8c40 808e8c40 9e11c440 9e1078c0 8094fe04 9e116100 fb60: 9e116604 9e11c440 9e14fb84 9e14fb78 800786f0 80078498 9e14fbd4 9e14fb88 fb80: 80029c60 800786e4 00000000 00000000 800299ec 9e11c438 808e7390 00000001 fba0: 9e11c438 9e11c440 9e11c438 9e14e000 9e11c614 9e068000 9e14fbd4 9e14fbc8 fbc0: 8065e114 8065da60 9e14fc0c 9e14fbd8 8002b088 8065e0e8 9e14e000 00000001 fbe0: 9e11c614 9e14fbf0 8094e944 9e11c65c 9e14fbf0 9e14fbf0 80028d90 808f760c fc00: 9e14fc8c 9e14fc10 80011ff4 8002aa90 9e14e240 0000000b 00000008 00000000 fc20: 00000000 60000113 65000001 33306131 20343830 30613165 34303031 61316520 fc40: 30303030 61302036 66666666 28203066 37393165 33623032 80002029 80659664 fc60: 80823f48 00000010 00000017 00000000 9e14fdf8 00000010 00000001 9e11c440 fc80: 9e14fca4 9e14fc90 80658498 80011dfc 9e14fdf8 00000017 9e14fd44 9e14fca8 fca0: 8001bcd8 80658448 00000037 00000000 9e14fccc 9e14fcc0 800613d0 800612f8 fcc0: 9e14fcf4 9e14fcd0 8007911c 800613c8 9fbd3c40 9e14e000 808e7ffc 9e14e000 fce0: 00000004 00000001 00000004 00000006 9e11c440 60000013 60000013 00000000 fd00: 9e14fd2c 9e14fd10 800661b8 80063254 9e14e000 80012928 9e11c440 00000017 fd20: 8001bb20 808f8314 00000010 9e14fdf8 00000001 9fbd6b00 9e14fdf4 9e14fd48 fd40: 8000840c 8001bb2c 800663bc 800661e8 9e14fe0c 9e14fd60 80012928 800663b4 fd60: 00000001 9e11c8a0 00000000 9e11c440 00000001 00000007 80903a24 80e3ea20 fd80: 60000013 60000013 00000000 9e14fe0c 80a26e60 9e14fda8 800661b8 8006d650 fda0: 20000013 ffffffff 00000000 00000000 00000000 00000000 80e3f308 00000007 fdc0: 60000013 80660be4 9e14fde4 00000000 00000000 804e8fb8 80000013 ffffffff fde0: 9e14fe2c 00000000 9e14fe5c 9e14fdf8 800128a4 800083dc 9e369a00 00000000 fe00: 00000000 00000000 00000000 00000002 9e369a00 00000010 00000000 00000001 fe20: 9fbd6b00 9e14fe5c 80a26e60 9e14fe40 800661b8 804e8fb8 80000013 ffffffff fe40: 9e387d84 9e369a00 9e387c64 9e387da4 9e14fe84 9e14fe60 804d90d0 804e8f48 fe60: 9e135300 9e387d84 9fbd36c0 9e14e000 00000000 9fbd6b00 9e14fee4 9e14fe88 fe80: 8003f060 804d8fd0 00000001 00000000 8003efec 9fbd36c0 8003f620 00000000 fea0: 00000000 9e14fea8 80e981a0 80a26e60 00000000 80843dfc 80661fac 9e135300 fec0: 9fbd36f0 9e14e000 9e135318 9e14e000 9fbd36c0 8094e442 9e14ff24 9e14fee8 fee0: 8003f700 8003eec4 80662510 00000000 9e134ac0 9e14e000 8003f5dc 00000000 ff00: 9e134ac0 9e135300 8003f5dc 00000000 00000000 00000000 9e14ffac 9e14ff28 ff20: 800463bc 8003f5e8 800663bc 00000000 9e14ff54 9e135300 00000000 00000000 ff40: dead4ead ffffffff ffffffff 80952424 00000000 00000000 807cc28c 9e14ff5c ff60: 9e14ff5c 00000001 00010001 dead4ead ffffffff ffffffff 80952424 00000000 ff80: 00000000 807cc28c 9e14ff88 9e14ff88 9e134ac0 800462ec 00000000 00000000 ffa0: 00000000 9e14ffb0 8000eb68 800462f8 00000000 00000000 00000000 00000000 ffc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 ffe0: 00000000 00000000 00000000 00000000 00000013 00000000 00c37f70 76f142d0 Backtrace: [<8004672c>] (kthread_data) from [<8004047c>] (wq_worker_sleeping+0x14/0xc8) [<80040468>] (wq_worker_sleeping) from [<8065dd8c>] (__schedule+0x338/0x5cc) r4:9fbd3c40 r3:00000020 [<8065da54>] (__schedule) from [<8065e114>] (schedule+0x38/0x88) r10:9e068000 r9:9e11c614 r8:9e14e000 r7:9e11c438 r6:9e11c440 r5:9e11c438 r4:00000001 [<8065e0dc>] (schedule) from [<8002b088>] (do_exit+0x604/0x8d8) [<8002aa84>] (do_exit) from [<80011ff4>] (die+0x204/0x3c0) r7:808f760c [<80011df0>] (die) from [<80658498>] (__do_kernel_fault.part.9+0x5c/0x7c) r10:9e11c440 r9:00000001 r8:00000010 r7:9e14fdf8 r6:00000000 r5:00000017 r4:00000010 [<8065843c>] (__do_kernel_fault.part.9) from [<8001bcd8>] (do_page_fault+0x1b8/0x3e4) r7:00000017 r3:9e14fdf8 [<8001bb20>] (do_page_fault) from [<8000840c>] (do_DataAbort+0x3c/0xa0) r10:9fbd6b00 r9:00000001 r8:9e14fdf8 r7:00000010 r6:808f8314 r5:8001bb20 r4:00000017 [<800083d0>] (do_DataAbort) from [<800128a4>] (__dabt_svc+0x44/0x80) Exception stack(0x9e14fdf8 to 0x9e14fe40) fde0: 9e369a00 00000000 fe00: 00000000 00000000 00000000 00000002 9e369a00 00000010 00000000 00000001 fe20: 9fbd6b00 9e14fe5c 80a26e60 9e14fe40 800661b8 804e8fb8 80000013 ffffffff r8:00000000 r7:9e14fe2c r6:ffffffff r5:80000013 r4:804e8fb8 [<804e8f3c>] (sgtl5000_resume) from [<804d90d0>] (soc_resume_deferred+0x10c/0x30c) r7:9e387da4 r6:9e387c64 r5:9e369a00 r4:9e387d84 [<804d8fc4>] (soc_resume_deferred) from [<8003f060>] (process_one_work+0x1a8/0x444) r10:9fbd6b00 r8:00000000 r7:9e14e000 r6:9fbd36c0 r5:9e387d84 r4:9e135300 [<8003eeb8>] (process_one_work) from [<8003f700>] (worker_thread+0x124/0x398) r10:8094e442 r9:9fbd36c0 r8:9e14e000 r7:9e135318 r6:9e14e000 r5:9fbd36f0 r4:9e135300 [<8003f5dc>] (worker_thread) from [<800463bc>] (kthread+0xd0/0xec) r10:00000000 r9:00000000 r8:00000000 r7:8003f5dc r6:9e135300 r5:9e134ac0 r4:00000000 [<800462ec>] (kthread) from [<8000eb68>] (ret_from_fork+0x14/0x2c) r7:00000000 r6:00000000 r5:800462ec r4:9e134ac0 Code: e1a0c00d e92dd800 e24cb004 e590322c (e5130030) ---[ end trace eeb2712470f0de35 ]--- Fixing recursive fault but reboot is needed! BUG: spinlock lockup suspected on CPU#0, kworker/0:1/21 lock: 0x9fbd3c40, .magic: dead4ead, .owner: kworker/0:1/21, .owner_cpu: 0 CPU: 0 PID: 21 Comm: kworker/0:1 Tainted: G D 3.15.0-rc3-00071-g49a96400a142-dirty #118 Backtrace: [<80011c50>] (dump_backtrace) from [<80011dec>] (show_stack+0x18/0x1c) r6:016e3600 r5:00000000 r4:00000000 r3:00000000 [<80011dd4>] (show_stack) from [<8065b720>] (dump_stack+0x88/0xa4) [<8065b698>] (dump_stack) from [<806595ac>] (spin_dump+0x7c/0x90) r5:9fbd3c40 r4:9e11c440 [<80659530>] (spin_dump) from [<80068a74>] (do_raw_spin_lock+0x110/0x198) r5:00000000 r4:016e3600 [<80068964>] (do_raw_spin_lock) from [<80661fac>] (_raw_spin_lock_irq+0x44/0x4c) r10:808f2d2c r9:8094e944 r8:808e8c40 r7:00000000 r6:9e14e000 r5:8065dab4 r4:9fbd3c40 r3:9e11c440 [<80661f68>] (_raw_spin_lock_irq) from [<8065dab4>] (__schedule+0x60/0x5cc) r5:9e11c440 r4:9fbd3c40 [<8065da54>] (__schedule) from [<8065e114>] (schedule+0x38/0x88) r10:00000001 r9:8094e944 r8:9e14e000 r7:808f760c r6:9e11c440 r5:0000000b r4:0000000b [<8065e0dc>] (schedule) from [<8002b2dc>] (do_exit+0x858/0x8d8) [<8002aa84>] (do_exit) from [<80011ff4>] (die+0x204/0x3c0) r7:808f760c [<80011df0>] (die) from [<80658498>] (__do_kernel_fault.part.9+0x5c/0x7c) r10:9e11c440 r9:9e11c698 r8:ffffffd0 r7:9e14fab8 r6:00000000 r5:00000017 r4:ffffffd0 [<8065843c>] (__do_kernel_fault.part.9) from [<8001bcd8>] (do_page_fault+0x1b8/0x3e4) r7:00000017 r3:9e14fab8 [<8001bb20>] (do_page_fault) from [<8000840c>] (do_DataAbort+0x3c/0xa0) r10:808f2d2c r9:9e11c698 r8:9e14fab8 r7:ffffffd0 r6:808f8314 r5:8001bb20 r4:00000017 [<800083d0>] (do_DataAbort) from [<800128a4>] (__dabt_svc+0x44/0x80) Exception stack(0x9e14fab8 to 0x9e14fb00) faa0: 9e11c440 00000000 fac0: 00000000 00000000 00000000 9e11c440 9e14e000 00000000 808e8c40 9e11c698 fae0: 808f2d2c 9e14fb0c 9e14fb10 9e14fb00 8004047c 8004673c 00000193 ffffffff r8:808e8c40 r7:9e14faec r6:ffffffff r5:00000193 r4:8004673c [<8004672c>] (kthread_data) from [<8004047c>] (wq_worker_sleeping+0x14/0xc8) [<80040468>] (wq_worker_sleeping) from [<8065dd8c>] (__schedule+0x338/0x5cc) r4:9fbd3c40 r3:00000020 [<8065da54>] (__schedule) from [<8065e114>] (schedule+0x38/0x88) r10:9e068000 r9:9e11c614 r8:9e14e000 r7:9e11c438 r6:9e11c440 r5:9e11c438 r4:00000001 [<8065e0dc>] (schedule) from [<8002b088>] (do_exit+0x604/0x8d8) [<8002aa84>] (do_exit) from [<80011ff4>] (die+0x204/0x3c0) r7:808f760c [<80011df0>] (die) from [<80658498>] (__do_kernel_fault.part.9+0x5c/0x7c) r10:9e11c440 r9:00000001 r8:00000010 r7:9e14fdf8 r6:00000000 r5:00000017 r4:00000010 [<8065843c>] (__do_kernel_fault.part.9) from [<8001bcd8>] (do_page_fault+0x1b8/0x3e4) r7:00000017 r3:9e14fdf8 [<8001bb20>] (do_page_fault) from [<8000840c>] (do_DataAbort+0x3c/0xa0) r10:9fbd6b00 r9:00000001 r8:9e14fdf8 r7:00000010 r6:808f8314 r5:8001bb20 r4:00000017 [<800083d0>] (do_DataAbort) from [<800128a4>] (__dabt_svc+0x44/0x80) Exception stack(0x9e14fdf8 to 0x9e14fe40) fde0: 9e369a00 00000000 fe00: 00000000 00000000 00000000 00000002 9e369a00 00000010 00000000 00000001 fe20: 9fbd6b00 9e14fe5c 80a26e60 9e14fe40 800661b8 804e8fb8 80000013 ffffffff r8:00000000 r7:9e14fe2c r6:ffffffff r5:80000013 r4:804e8fb8 [<804e8f3c>] (sgtl5000_resume) from [<804d90d0>] (soc_resume_deferred+0x10c/0x30c) r7:9e387da4 r6:9e387c64 r5:9e369a00 r4:9e387d84 [<804d8fc4>] (soc_resume_deferred) from [<8003f060>] (process_one_work+0x1a8/0x444) r10:9fbd6b00 r8:00000000 r7:9e14e000 r6:9fbd36c0 r5:9e387d84 r4:9e135300 [<8003eeb8>] (process_one_work) from [<8003f700>] (worker_thread+0x124/0x398) r10:8094e442 r9:9fbd36c0 r8:9e14e000 r7:9e135318 r6:9e14e000 r5:9fbd36f0 r4:9e135300 [<8003f5dc>] (worker_thread) from [<800463bc>] (kthread+0xd0/0xec) r10:00000000 r9:00000000 r8:00000000 r7:8003f5dc r6:9e135300 r5:9e134ac0 r4:00000000 [<800462ec>] (kthread) from [<8000eb68>] (ret_from_fork+0x14/0x2c) r7:00000000 r6:00000000 r5:800462ec r4:9e134ac0

3 4

[alsa-devel] [PATCH v2] ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls
by Axel Lin 23 May '14

23 May '14

The new value argument needs proper shift to match the mask bit fields. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l56.c | 68 +++++++++++++++++++++------------------------- sound/soc/codecs/cs42l56.h | 13 +++++---- 2 files changed, 39 insertions(+), 42 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 5bb134b..5d7a800 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -763,14 +763,14 @@ static int cs42l56_set_sysclk(struct snd_soc_dai *codec_dai, case CS42L56_MCLK_11P2896MHZ: case CS42L56_MCLK_12MHZ: case CS42L56_MCLK_12P288MHZ: - cs42l56->mclk_div2 = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; cs42l56->mclk_prediv = 0; break; case CS42L56_MCLK_22P5792MHZ: case CS42L56_MCLK_24MHZ: case CS42L56_MCLK_24P576MHZ: - cs42l56->mclk_div2 = 1; - cs42l56->mclk_prediv = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; + cs42l56->mclk_prediv = CS42L56_MCLK_PREDIV; break; default: return -EINVAL; @@ -844,57 +844,51 @@ static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) /* Hit the DSP Mixer first */ snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_MUTE); + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_MUTE_ALL_DSP); /* Mute ADC's */ snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_MUTE); + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_MUTE_ALL_ADC); /* HP And LO */ snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); } else { snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); } return 0; } diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h index ad2b50a..3fb73dc 100644 --- a/sound/soc/codecs/cs42l56.h +++ b/sound/soc/codecs/cs42l56.h @@ -80,19 +80,21 @@ #define CS42L56_PDN_HPB_MASK 0xc0 /* serial port and clk masks */ -#define CS42L56_MASTER_MODE 1 +#define CS42L56_MASTER_MODE 0x40 #define CS42L56_SLAVE_MODE 0 #define CS42L56_MS_MODE_MASK 0x40 -#define CS42L56_SCLK_INV 1 +#define CS42L56_SCLK_INV 0x20 #define CS42L56_SCLK_INV_MASK 0x20 #define CS42L56_SCLK_MCLK_MASK 0x18 +#define CS42L56_MCLK_PREDIV 0x04 #define CS42L56_MCLK_PREDIV_MASK 0x04 +#define CS42L56_MCLK_DIV2 0x02 #define CS42L56_MCLK_DIV2_MASK 0x02 #define CS42L56_MCLK_DIS_MASK 0x01 #define CS42L56_CLK_AUTO_MASK 0x20 #define CS42L56_CLK_RATIO_MASK 0x1f #define CS42L56_DIG_FMT_I2S 0 -#define CS42L56_DIG_FMT_LEFT_J 1 +#define CS42L56_DIG_FMT_LEFT_J 0x08 #define CS42L56_DIG_FMT_MASK 0x08 /* Class H and misc ctl masks */ @@ -116,8 +118,6 @@ #define CS42L56_DEEMPH_MASK 0x40 #define CS42L56_PLYBCK_GANG_MASK 0x10 #define CS42L56_PCM_INV_MASK 0x0c -#define CS42L56_MUTE 1 -#define CS42L56_UNMUTE 0 #define CS42L56_ADCAMIX_MUTE_MASK 0x40 #define CS42L56_ADCBMIX_MUTE_MASK 0x80 #define CS42L56_PCMAMIX_MUTE_MASK 0x10 @@ -128,6 +128,9 @@ #define CS42L56_ADCB_MUTE_MASK 0x02 #define CS42L56_HP_MUTE_MASK 0x80 #define CS42L56_LO_MUTE_MASK 0x80 +#define CS42L56_MUTE_ALL_DSP 0xf3 +#define CS42L56_MUTE_ALL_ADC 0x03 +#define CS42L56_UNMUTE 0 /* Beep masks */ #define CS42L56_BEEP_FREQ_MASK 0xf0 -- 1.8.3.2

2 1

[alsa-devel] Fwd: enabling a20 allwinner audio codec to 24bit
by Ricardo Crudo 22 May '14

22 May '14

Hello, I'm working with sunxi-codec of allwinner A20 device. Currently the codec driver works only for 16bit mode, the source code can be found here: https://github.com/linux-sunxi/linux-sunxi/tree/sunxi-3.4/sound/soc/sunxi, and the A20 datasheet here: https://github.com/OLIMEX/OLINUXINO/blob/master/HARDWARE/A20-PDFs/A20%20Use…. Reading the allwinner A20 datasheet and doing some modifications on sunxi code I managed the codec to work in 24bit mode. However, it isn't working properly. When I play a stereo wav file using the aplay command the audio data of both channels are played mixed in only one channel and the another channel didn't plays any sound. For example, picture a wav stereo file with this samples: left right 0.5 -0.5 0.4 -0.4 0.3 -0.3 0.2 -0.2 The played sound is: left right 0.5 0.0 -0.5 0.0 0.4 0.0 -0.4 0.0 0.3 0.0 -0.3 0.0 0.2 0.0 -0.2 0.0 The A20 datasheet says: "DAC TX DATA REGISTER (TXDATA): Transmitting left, right channel sample data should be written this register one by one. The left channel sample data is first and then the right channel sample." I guess that this is related with the problem, despite the datasheet didn't notify, I think that in 16bit mode the TXDATA transmits both channels data together as the register is 32bit. I'm not familiar writing alsa driver and other alsa working concepts. I hope someone can help me with this. The diff file of sunxi-codec.c is here: http://pastebin.com/raw.php?i=SvypkVU6 I appreciate any help. Regards, Ricardo

1 0

[alsa-devel] [PATCH] ASoC: samsung: Handle errors when getting the op_clk clock
by Sylwester Nawrocki 22 May '14

22 May '14

Ensure i2s->op_clk is not used when clk_get() for this clock fails. This prevents working with an incorrectly configured clock in some conditions. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ae02811..a634a66 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -488,7 +488,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, clk_id = 1; if (!any_active(i2s)) { - if (i2s->op_clk) { + if (i2s->op_clk && !IS_ERR(i2s->op_clk)) { if ((clk_id && !(mod & MOD_IMS_SYSMUX)) || (!clk_id && (mod & MOD_IMS_SYSMUX))) { clk_disable_unprepare(i2s->op_clk); @@ -506,6 +506,10 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, else i2s->op_clk = clk_get(&i2s->pdev->dev, "i2s_opclk0"); + + if (WARN_ON(IS_ERR(i2s->op_clk))) + return PTR_ERR(i2s->op_clk); + clk_prepare_enable(i2s->op_clk); i2s->rclk_srcrate = clk_get_rate(i2s->op_clk); -- 1.7.9.5

2 1

Re: [alsa-devel] [GIT PULL] ASoC updates for v3.16
by Takashi Iwai 22 May '14

22 May '14

At Thu, 22 May 2014 16:47:01 +0100, Mark Brown wrote: > > The following changes since commit d6d211db37e75de2ddc3a4f979038c40df7cc79c: > > Linux 3.15-rc5 (2014-05-09 13:10:52 -0700) > > are available in the git repository at: > > git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.16 > > for you to fetch changes up to cee429e5c53f7ffc1dd500001d9dff567833c23d: > > Merge remote-tracking branches 'asoc/topic/ux500', 'asoc/topic/wm8731', 'asoc/topic/wm8804', 'asoc/topic/wm8955' and 'asoc/topic/wm8985' into asoc-next (2014-05-22 00:24:04 +0100) > > ---------------------------------------------------------------- > > ASoC: Updates for v3.16 > > Lots of cleanup work going on in the core this release but very little > visible to external users except for the new drivers that have been > added. > > - Support for specifying aux CODECs in DT. > - Removal of the deprecated mux and enum macros. > - More moves towards full componentisation. > - Removal of some unused I/O code. > - Lots of cleanups, fixes and enhancements to the davinci, Freescale, > Haswell and Realtek drivers. > - Several drivers exposed directly in Kconfig for use with simple-card. > - New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and > ST STA350. Thanks, pulled now. Takashi > > ---------------------------------------------------------------- > Alexander Shiyan (3): > ASoC: fsl: Separation of the main audio options and boards > ASoC: fsl: Allow to select individual common options > ASoC: mc13783: Add devicetree support > > Andrew Lunn (1): > ASoC: alc5623: Add device tree binding > > Arnd Bergmann (15): > ASoC: cq93vc: fix cq93vc_get_regmap build error > ASoC: CS42L51 and WM8962 codecs depend on INPUT > ASoC: davinci: add dependencies for SND_SOC_TLV320AIC3X > ASoC: samsung: SMDK_WM8580_PCM needs REGMAP_I2C > ASoC: samsung-idma: avoid 64-bit division > ASoC: nuc900: export nuc900_ac97_data > ASoC: omap: RX-51 audio needs I2C > ASoC: samsung: UDA1380 needs I2C > ASoC: pxa: TTC DKB audio needs I2C > ASoC: atmel: Atmel WM8904 codec support needs I2C > ASoC: samsung: TLV320AIC23 and Simtec Hermes audio need I2C > ASoC: sh: Migo-R sound needs I2C > ASoC: omap: Amstrad E3 needs TTY support for codec > ASoC: fsl: select SND_SOC_IMX_PCM_DMA where needed > ASoC: pxa: remove mach header dependency > > Axel Lin (2): > ASoC: cs42l51: Fix mask for REVID > ASoC: rt5645: Fix updating wrong register for T5645_AIF2 case > > Bard Liao (6): > ASoC: core: Allow snd_soc_update_bits use 32 bits register > ASoC: dapm: Allow update_bits use 32 bits reg > ASoC: add RT5651 CODEC driver > ASoC: rt5640: Add minimal support for RT5642 > ASoC: rt5640: correct 5640's device ID > ASoC: rt5640: add default case for unexpected ID > > Benoit Cousson (1): > ASoC: core: Add one dai_get_widget helper instead of two rtd based ones > > Brian Austin (4): > ASoC: cs42l51: split i2c from codec driver > ASoC: Remove IS_ENABLED for INPUT in CS42L52 and WM8962 > ASoC: Add support for CS42L56 CODEC > ASoC: cs42l56: bindings: sound: Add bindings for CS42L56 CODEC > > Charles Keepax (3): > ASoC: wm5110: Add in OSR controls for OUT5/6 > ASoC: wm_adsp: Remove uneeded semicolon > ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile > > Christian Engelmayer (2): > ASoC: Intel: Fix incorrect sizeof() in sst_hsw_stream_get_volume() > ASoC: Intel: Fix a self assignment in sst_mem_block_alloc_scratch() > > Christoph Jaeger (1): > ASoC: core: use PTR_ERR instead of PTR_RET > > Dan Carpenter (1): > ASoC: compress: indent an if statement > > Daniel Mack (4): > ASoC: ak4104: add regulator consumer support > ASoC: ak4104: Add regulator to documentation > ASoC: sta350: fix DT bindings document > ASoC: sta350: add support for bits in miscellaneous registers > > Fabio Estevam (6): > ASoC: sgtl5000: Use devm_regulator_bulk_get() > ASoC: tlv320dac33: Use devm_regulator_bulk_get() > ASoC: wm8804: Use devm_regulator_bulk_get() > ASoC: wm8731: Use devm_regulator_bulk_get() > ASoC: wm8955: Use devm_regulator_bulk_get() > ASoC: wm8985: Use devm_regulator_bulk_get() > > Geert Uytterhoeven (4): > ARM: shmobile: r8a7791: Use rcar_gen2_read_mode_pins() helper > ARM: shmobile: koelsch-reference: Work around core clock issues > ARM: shmobile: koelsch legacy: Enable Quad SPI transfers for the SPI FLASH > ARM: shmobile: lager legacy: Enable Quad SPI transfers for the SPI FLASH > > Jarkko Nikula (15): > ASoC: Intel: Cancel hsw_notification_work before freeing the stream > ASoC: core: Fix component_list corruption when unloading modules > ASoC: core: Add support for machine specific trigger callback > ASoC: Intel: Sample Baytrail DSP DMA pointer only after each period > ASoC: Intel: Simplify Baytrail stream control IPC construction > ASoC: Intel: Pass stream start position to sst_byt_stream_start() > ASoC: Intel: Move Baytrail extended fw address saving to sst_byt_boot() > ASoC: Intel: Allow byt-5640 machine driver and SST core go to suspend > ASoC: Intel: Use ACPI device for Baytrail PCM buffer allocation > ASoC: Intel: Fix Baytrail SST DSP firmware loading > ASoC: Intel: Fix simultaneous Baytrail SST capture and playback > ASoC: max98090: Add ACPI probing support > ASoC: max98090: Move microphone bias voltage setting to probe function > ASoC: Remove needless snd_soc_dapm_sync() from machine driver inits > ASoC: Remove needless snd_soc_dapm_enable_pin() from machine driver inits > > Joe Perches (1): > ASoC: tlv320aic31xx: Convert /n to \n > > Jyri Sarha (2): > ASoC: core: Update snd_soc_of_parse_daifmt() interface > ASoC: simple-card: Move dai-link level properties away from dai subnodes > > Kevin Strasser (1): > ASoC: Intel: Fix pcm stream context restore crash > > Kuninori Morimoto (13): > ARM: shmobile: r8a7778: remove old style audio clock > ARM: shmobile: r8a7790: remove old style audio clock > ARM: shmobile: bockw: remove old style audio clock > ARM: shmobile: bockw: switch to use dai info for R-Car sound > ARM: shmobile: lager: switch to use dai info for R-Car sound > ASoC: rsnd: call rsnd_dai_pointer_update() from outside of lock > ASoC: rsnd: remove old clock style support > ASoC: rsnd: remove compatibility code > ASoC: rsnd: add rsnd_path_parse() macro > ASoC: rsnd: add rsnd_get_adinr() > ASoC: rsnd: remove duplicate parameter from rsnd_mod_ops > ASoC: rsnd: enable to use multi parameter on rsnd_dai_call/rsnd_mod_call > ASoC: rsnd: add DVC support > > Lars-Peter Clausen (65): > ASoC: Add snd_soc_kcontrol_codec() helper function > ASoC: Add snd_soc_kcontrol_platform() helper function > ASoC: Prepare SOC_SINGLE_XR_SX controls for regmap > ASoC: Move IO functions to soc-io.c > ASoC: Remove IO register modifier callbacks > ASoC: Track which components have been registered with snd_soc_register_component() > ASoC: Let snd_soc_platform subclass snd_soc_component > ASoC: ams-delta: Convert to table based DAPM and control setup > ASoC: omap: rx51: Convert to table based control and DAPM setup > ASoC: omap3pandora: Convert to table based DAPM setup > ASoC: adau1373: Replace usage deprecated MUX/ENUM macros > ASoC: adav80x: Replace usage deprecated MUX/ENUM macros > ASoC: arizona: Replace usage deprecated MUX/ENUM macros > ASoC: max98090: Replace usage deprecated MUX/ENUM macros > ASoC: mc13783: Replace usage deprecated MUX/ENUM macros > ASoC: pcm512x: Replace usage deprecated SOC_VALUE_ENUM macro > ASoC: rt5640: Replace usage deprecated MUX/ENUM macros > ASoC: wm2200: Replace usage deprecated MUX/ENUM macros > ASoC: wm5100: Replace usage deprecated MUX/ENUM macros > ASoC: wm5102: Replace usage deprecated MUX/ENUM macros > ASoC: wm5110: Replace usage deprecated MUX/ENUM macros > ASoC: wm8988: Replace usage deprecated MUX/ENUM macros > ASoC: wm5102: Replace usage deprecated SOC_VALUE_ENUM macro > ASoC: wm8994: Replace usage deprecated MUX/ENUM macros > ASoC: wm8995: Replace usage deprecated MUX/ENUM macros > ASoC: wm8997: Replace usage deprecated MUX/ENUM macros > ASoC: Remove deprecated ENUM/MUX macros > ASoC: wm8997: Fix compile error > ASoC: Fix snd_soc_kcontrol_platform() return type > ASoC: sta350: Use snd_soc_kcontrol_codec() > ASoC: Change return type of snd_soc_write() to int > ASoC: Move IO abstraction to the component level > ASoC: Remove snd_soc_update_bits_locked() > ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() > ASoC: Remove ASoC level IO tracing > ASoC: Move standard kcontrol helpers to the component level > ASoC: qi_lb60: Set fully_routed flag > ASoC: qi_lb60: Set .dai_fmt instead of calling snd_soc_set_dai_fmt() > ASoC: qi_lb60: Use devm_snd_soc_register_card() > ASoC: Export devm_snd_soc_register_platform() > ASoC: qi_lb60: Use GPIO descriptor API > ASoC: jz4740: Improve build test coverage > ASoC: imx-audmux: Fix section mismatch > ASoC: Remove name_prefix unset during DAI link init hack again > ASoC: dapm: Factor out duplicated code in soc_dapm_stream_event() > ASoC: Make soc_find_matching_codec() static > ASoC: dapm: Make sure to always update the DAPM graph in _put_volsw() > ASoC: Remove card's DAI list > ASoC: Remove unused 'list' field form card > ASoC: Remove unused num_dai field from CODEC > ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets() > ASoC: dapm: Revert "ASoC: dapm: Fix double prefix addition" > ASoC: dapm: Use snd_soc_dapm_add_path() in connect_dai_link_widgets() > ASoC: dapm: Use snd_soc_dapm_add_path() in snd_soc_dapm_new_pcm() > ASoC: dapm: Simplify snd_soc_dapm_link_dai_widgets() > ASoC: dapm: Handle SND_SOC_DAPM_REG() generically > ASoC: dapm: Fix SUSPEND -> OFF bias sequence > ASoC: atmel-pcm-pdc: Remove broken suspend/resume code > ASoC: Remove runtime field from DAI > ASoC: cs42xx8: Do not use rtd->codec > ASoC: rt5651: Do not use rtd->codec > ASoC: ad1980: Replace goto loop with do-while loop > ASoC: omap: Replace instances of rtd->codec->card with rtd->card > ASoC: mop500_ab8500: Replace instances of rtd->codec->card with rtd->card > ASoC: neo1973_wm8753: Automatically disconnected non-connected pins > > Laurent Pinchart (5): > ARM: shmobile: r8a7790: Fix the I2C clocks parents in legacy code > ARM: shmobile: r8a7791: Fix the I2C clocks parents in legacy code > ARM: shmobile: r8a7790: Rename VSP1_(SY|RT) clocks to VSP1_(S|R) > ARM: shmobile: r8a7791: Rename VSP1_SY clocks to VSP1_S > ASoC: rsnd: Fix warnings due to improper printk formats > > Liam Girdwood (18): > ASoC: Intel: Fix block allocation so we only allocate blocks once. > ASoC: Intel: Fix allocated block list usage when adding blocks. > ASoC: Intel: Fix Haswell/Broadwell DSP page table creation. > ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled. > ASoC: Intel: Fix allow hw_params to be called more than once. > ASoC: Intel: Fix stream position pointer. > ASoC: Intel: Fix check for pdata usage before dereference. > ASoC: Intel: Fix block offset calculations. > ASoC: Intel: Add support to unload/reload firmware modules. > ASoC: Intel: Allow Rx/Tx message list can be cleared prior to suspend > ASoC: Intel: Add Baytrail suspend/resume support > ASoC: Intel: Make Baytrail PCM data per stream rather than per DAI device > ASoC: Intel: Only export one Baytrail DAI > ASoC: max98090: Fix digital sidetone gain TLV > ASoC: max98090: Fix reset at resume time > ASoC: max98090: Reset codec on resume > ASoC: max98090: Mark cache as dirty prior to restoring > ASoC: max98090: Make sure we configure BCLK in one place > > Magnus Damm (5): > ARM: shmobile: Introduce shmobile_clk_workaround() > ARM: shmobile: Use shmobile_clk_workaround() on Lager > ARM: shmobile: Use shmobile_clk_workaround() on Koelsch > ARM: shmobile: Add Lager clock workarounds for SDHI and MMCIF > ARM: shmobile: Add Koelsch clock workarounds for SDHI > > Mark Brown (37): > Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-core > ASoC: fsl: Add explicit include of of.h > Merge branch 'topic/devm' of git://git.kernel.org/.../broonie/sound into asoc-omap > Merge commit 'topic/omap' into asoc-davinci > ASoC: rt5651: Staticise non-exported symbols > Merge branches 'topic/sta350', 'topic/core', 'topic/dapm' and 'topic/cache' of git://git.kernel.org/.../broonie/sound into asoc-component > Merge branch 'topic/multicodec' of git://git.kernel.org/.../broonie/sound into asoc-component > Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-dapm > Merge branch 'asoc-dt' into asoc-omap > Merge branch 'topic/input' of git://git.kernel.org/.../broonie/sound into asoc-cs42l51 > Merge tag 'v3.15-rc4' into asoc-fsl-spdif > Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-core > Merge tags 'renesas-clock-for-v3.16' and 'renesas-boards-for-v3.16' of git://git.kernel.org/.../horms/renesas into asoc-rcar > ASoC: Intel: Build Medfield compressed ops > Merge branch 'fix/intel' of git://git.kernel.org/.../broonie/sound into asoc-intel > Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', 'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus > Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus > Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus > Merge tag 'asoc-v3.15-rc5-core' into asoc-linus > Merge tag 'asoc-v3.15-rc5-drivers' into asoc-linus > Merge tag 'asoc-v3.15-rc5-intel' into asoc-linus > Merge remote-tracking branch 'asoc/fix/max98090' into asoc-linus > Merge remote-tracking branch 'asoc/topic/component' into asoc-next > Merge remote-tracking branch 'asoc/topic/core' into asoc-next > Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next > Merge remote-tracking branch 'asoc/topic/dapm-init' into asoc-next > Merge remote-tracking branch 'asoc/topic/dt' into asoc-next > Merge remote-tracking branch 'asoc/topic/enum' into asoc-next > Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next > Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next > Merge remote-tracking branches 'asoc/topic/ad1980', 'asoc/topic/adsp', 'asoc/topic/ak4104', 'asoc/topic/ak4642', 'asoc/topic/alc5623', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/cache' into asoc-next > Merge remote-tracking branches 'asoc/topic/cs42l56', 'asoc/topic/cs42xx8' and 'asoc/topic/davinci' into asoc-next > Merge remote-tracking branches 'asoc/topic/devm', 'asoc/topic/fsl', 'asoc/topic/fsl-esai', 'asoc/topic/fsl-sai', 'asoc/topic/fsl-spdif' and 'asoc/topic/fsl-ssi' into asoc-next > Merge remote-tracking branches 'asoc/topic/headers', 'asoc/topic/intel', 'asoc/topic/jz4740', 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/mc13783' and 'asoc/topic/multicodec' into asoc-next > Merge remote-tracking branches 'asoc/topic/nuc900', 'asoc/topic/omap', 'asoc/topic/pxa', 'asoc/topic/rcar', 'asoc/topic/rt5640' and 'asoc/topic/rt5645' into asoc-next > Merge remote-tracking branches 'asoc/topic/rt5651', 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/sta350' and 'asoc/topic/tlv320dac33' into asoc-next > Merge remote-tracking branches 'asoc/topic/ux500', 'asoc/topic/wm8731', 'asoc/topic/wm8804', 'asoc/topic/wm8955' and 'asoc/topic/wm8985' into asoc-next > > Markus Pargmann (10): > ASoC: fsl-ssi: Remove fsl_ssi_setup > ASoC: fsl-ssi: Fix i2s_mode variable setup > ASoC: fsl-ssi: Fix register values when disabling > ASoC: fsl-ssi: Move debugging to seperate file > ASoC: fsl-ssi: Use dev_name for DAI driver struct > ASoC: fsl-ssi: Move imx-specific probe to seperate function > ASoC: fsl-ssi: Remove useless DMA code > ASoC: fsl-ssi: Cleanup probe function > ASoC: fsl-ssi: Remove unnecessary variables from ssi_private > ASoC: fsl-ssi: Transmit enable synchronization > > Misael Lopez Cruz (5): > ASoC: core: Add helpers for codec and codec_dai search > ASoC: core: Add helpers for codec DAI probe & remove > ASoC: core: Add helper for DAI widgets linking > ASoC: core: Add function for ac97 codec registration > ASoC: core: Add helpers for dai link and aux dev init > > Nariman Poushin (2): > ASoC: core: Fail probe if we fail to add dai widgets > ASoC: core: Fail codec probe if we fail to add controls > > Nicolin Chen (24): > ASoC: fsl_sai: Improve fsl_sai_isr() > ASoC: fsl_sai: Fix buggy configurations in trigger() > ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams > ASoC: fsl_sai: Add imx6sx platform support > ASoC: fsl_sai: Add clock controls for SAI > ASoC: fsl_sai: Fix incorrect condition check in trigger() > ASoC: fsl_sai: Use FSL_SAI_xXR() and regmap_update_bits() to simplify code > ASoC: core: Don't break component searching if both id and num_dai are 0 > ASoC: fsl_spdif: Add core clock control for DMA access > ASoC: fsl: Drop formats limitation for imx-pcm-dma.c > ASoC: simple-card: Drop node->name checking > ASoC: simple-card: Simplify error msg in simple_card_dai_link_of() > ASoC: simple-card: Improve coding style > ASoC: fsl_spdif: Fix clock source for rxclk rate measurement > ASoC: fsl_spdif: Use clk_set_rate() for spdif root clock only > ASoC: fsl_spdif: Rename all _div to _df > ASoC: fsl_spdif: Add sysclk df support to derive txclk from sysclk > ASoC: fsl_spdif: Print actual sample rate for debug > ASoc: fsl_spdif: Add descriptions for fsl_spdif_priv > ASoC: pcm: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPEND > ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP > ASoC: fsl_esai: Only bypass sck_div for EXTAL source > ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params() > ASoC: fsl_esai: Bypass divider settings if clock requirement is not changed > > Oder Chiou (19): > ASoC: rt5640: Move cache sync() to resume() > ASoC: rt5640: Correct the judgement of data length > ASoC: rt5640: Use the platform data for DMIC settings > ASoC: rt5640: Remove the pre-allocated size of reg_default > ASoC: rt5640: Rename the function of clock checking > ASoC: rt5640: Remove the unused or incorrect setting of clock source > ASoC: rt5640: Remove the unused field in private data > ASoC: rt5640: Remove the unnecessary parentheses > ASoC: rt5640: Change the setting method of idle_bias_off > ASoC: rt5640: add rt5639 support > ASoC: rt5640: Add the string "rt5639" to the list of I2C device IDs > ASoC: rt5640: Add the info of rt5639 to the binding document > ASoC: rt5640: Add the rt5639 support to the OF match table > ASoC: rt5645: Add codec driver > ASoC: rt5645: Remove the unused variable > ASoC: rt5645: Staticise non-exported symbols > ASoC: rt5645: Move settings from probe() to reg_default struct > ASoC: rt5645: Correct the cache sync function > ASoC: rt5645: fix coccinelle warnings > > Paul Handrigan (1): > ASoC: Update Cirrus Logic CODEC maintainers. > > Peter Ujfalusi (37): > ASoC: davinci-pcm: Add empty functions for !CONFIG_SND_DAVINCI_SOC builds > ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback > ASoC: davinci-mcasp: Fix debug typo in davinci_mcasp_hw_params() > ASoC: davinci-mcasp: Simplify and clean up the AFIFO configuration code > ASoC: davinci-mcasp: Configure the AFIFO and DMA burst size at the same place > ASoC: davinic-mcasp: Adopt the AFIFO/DMA configuration to the stream (dynamic depth) > ASoC: davinci-mcasp: Fine tune and correct the DMA burst configuration > ASoC: davinci-mcasp: Format data delay configuration enhancement > ASoC: davinci-mcasp: Support for DSP_A format > ASoC: davinci-mcasp: Move the FS polarity change out from the switch case > ASoC: davinci-mcasp: Add support for I2S format > ASoC: davinci-mcasp: Support for LEFT_J format > ASoC: davinci-mcasp: Remove excess empty lines from davinci_mcasp_set_dai_fmt() > ASoC: davinci-mcasp: Do not touch 0x04 register above McASP_VERSION_2 > ASoC: Add resource managed snd_soc_register_platform() > ASoC: omap-mcpdm: Assign the dai DMA data at earlier time > ASoC: omap-mcpdm: Bind the platform driver to the dai driver when loading > ASoC: omap-mcbsp: Assign the dai DMA data at earlier time > ASoC: omap-mcbsp: Bind the platform driver to the dai driver when loading > ASoC: omap-pcm: Support for binding the platform driver to dai devices > ASoC: omap-dmic: Assign the dai DMA data at earlier time > ASoC: omap-dmic: Bind the platform driver to the dai driver when loading > ASoC: omap-hdmi: Bind the platform driver to the dai driver when loading > ASoC: omap-abe-twl6040: Use the cpu_dai node to specify the platform driver > ASoC: omap-twl4030: Use the same name/node for platform as the cpu_dai > ASoC: am3517evm: Use the same name for platform as the cpu_dai > ASoC: ams-delta: Use the same name for platform as the cpu_dai > ASoC: n810: Use the same name for platform as the cpu_dai > ASoC: omap3pandora: Use the same name for platform as the cpu_dai > ASoC: osk5912: Use the same name for platform as the cpu_dai > ASoC: rx51: Use the same name for platform as the cpu_dai > ASoC: omap-hdmi-card: Use the same name for platform as the cpu_dai > ASoC: omap-pcm: Drop the platform driver init code > ASoC: davinci-mcasp: Update MCASP_VERSION_4 platform driver registration > ASoC: omap-hdmi: Remove excess curly bracket > ASoC: davinci-pcm: Convert to use devm_snd_soc_register_platform() > ASoC: davinci-mcasp: Convert to use devm_snd_soc_register_component() > > Rongjun Ying (1): > ASoC: sirf: Move the tx rx enable from port to codec, that will not need register sharing > > Sachin Kamat (6): > ASoC: pcm512x: Use CONFIG_PM_RUNTIME macro > ASoC: hdmi: Include of.h > ASoC: max98090: Include of.h > ASoC: rt5640: Include of.h > ASoC: tlv320aic23: Include of.h > ASoC: tpa6130a2: Include of.h > > Sascha Hauer (5): > ASoC: ak4642: Fix typo zoro -> zero > ASoC: ak4642: Add ALC controls > ASoC: ak4642: Add driver data and driver private struct > ASoC: ak4642: Add support for extended sysclk frequencies of the ak4648 > ASoC: fsl-ssi: Move fsl_ssi_set_dai_sysclk above fsl_ssi_hw_params > > Sebastian Reichel (8): > 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: 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: dt: Allow Aux Codecs to be specified using DT > ASoC: omap: rx51: Add DT support > > Simon Horman (4): > ARM: shmobile: koelsch: Annotate clk_names with __initconst > ARM: shmobile: lager: Annotate clk_names with __initconst > ARM: shmobile: koelsch-reference: Annotate clk_enables as __initconst > ARM: shmobile: lager-reference: Work around core clock issues > > Sven Brandau (1): > ASoC: sta350: Add codec driver > > Sylwester Nawrocki (1): > ASoC: samsung: Allow setting OP_CLK of the IIS Multi Audio Interface > > Tushar Behera (8): > ASoC: max98095: Add an explicit of_match_table > ASoC: samsung: Don't clear clock setting during i2s_startup > ASoC: samsung: Add sound card driver for Snow board > ASoC: samsung: Add missing pm ops for Snow sound card driver > ASoC: samsung: Use devm_snd_soc_register_card > ASoC: samsung: Use devm_snd_soc_register_platform > ASoC: samsung: Use devm_snd_soc_register_component > ASoC: samsung: Add devm_clk_get to pcm.c > > Vinod Koul (11): > ASoC: add SND_SOC_BYTES_EXT > ASoC: Intel: move component registration blob > ASoC: Intel: remove FSF snail mail address > ASoc: Intel: rename sst-mfld-platform.c > ASoC: Intel: mark sst_set_stream_status as non static > ASoC: Intel: split the pcm and compress to different files > ASoC: Intel: remove unused sst-mfld platform dais > ASoC: Intel: rename pcm dias to media dai > ASoC: Intel: Revert "rename pcm dias to media dai" > ASoC: Intel: add drain_notify support > ASoC: Intel: remove codec memeber from codec structs > > Wenkai Du (1): > ASoC: Intel: Fix audio crash due to race condition in stream deletion > > Xiubo Li (18): > ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() > ASoC: 88pm860x: Remove the set_cache_io() entirely from ASoC probe. > ASoC: cq93vc: Remove the set_cache_io() entirely from ASoC probe. > ASoC: mc13783: Remove the set_cache_io() entirely from ASoC probe. > ASoC: si476x: Remove the set_cache_io() entirely from ASoC probe. > ASoC: wm5102: Remove the set_cache_io() entirely from ASoC probe. > ASoC: wm5110: Remove the set_cache_io() entirely from ASoC probe. > ASoC: wm8350: Remove the set_cache_io() entirely from ASoC probe. > ASoC: wm8400: Remove the set_cache_io() entirely from ASoC probe. > ASoC: wm8994: Remove the set_cache_io() entirely from ASoC probe. > ASoC: wm8997: Remove the set_cache_io() entirely from ASoC probe. > ASoC: codec: fix the sparse check warnings. > ASoC: esai: use the precise definition of 'ret'. > ASoC: spdif: Sort the header files alphabetically. > ASoC: esai: Add VF610+ compatibles support. > ASoC: spdif: Add VF610+ compatibles support. > ASoC: fsl: Allow to select SAI device individually > ASoC: fsl: Allow to select ESAI device individually > > Documentation/devicetree/bindings/mfd/mc13xxx.txt | 3 + > Documentation/devicetree/bindings/sound/ak4104.txt | 3 + > .../devicetree/bindings/sound/alc5623.txt | 25 + > .../devicetree/bindings/sound/cs42l56.txt | 63 + > .../devicetree/bindings/sound/fsl-sai.txt | 11 +- > .../devicetree/bindings/sound/max98095.txt | 16 + > .../devicetree/bindings/sound/nokia,rx51.txt | 27 + > Documentation/devicetree/bindings/sound/rt5640.txt | 13 +- > .../devicetree/bindings/sound/simple-card.txt | 88 +- > Documentation/devicetree/bindings/sound/snow.txt | 17 + > .../devicetree/bindings/sound/st,sta350.txt | 131 ++ > MAINTAINERS | 6 - > arch/arm/boot/dts/r8a7790.dtsi | 2 +- > arch/arm/boot/dts/r8a7791.dtsi | 2 +- > arch/arm/mach-shmobile/Makefile | 2 +- > arch/arm/mach-shmobile/board-bockw.c | 63 +- > arch/arm/mach-shmobile/board-koelsch-reference.c | 86 +- > arch/arm/mach-shmobile/board-koelsch.c | 2 +- > arch/arm/mach-shmobile/board-lager-reference.c | 79 +- > arch/arm/mach-shmobile/board-lager.c | 29 +- > arch/arm/mach-shmobile/clock-r8a7778.c | 22 +- > arch/arm/mach-shmobile/clock-r8a7790.c | 32 +- > arch/arm/mach-shmobile/clock-r8a7791.c | 23 +- > arch/arm/mach-shmobile/clock.c | 28 + > arch/arm/mach-shmobile/include/mach/clock.h | 16 + > arch/mips/include/asm/mach-jz4740/dma.h | 2 - > arch/mips/jz4740/board-qi_lb60.c | 11 + > drivers/mfd/mc13xxx-core.c | 10 +- > include/dt-bindings/clock/r8a7790-clock.h | 4 +- > include/dt-bindings/clock/r8a7791-clock.h | 2 +- > include/sound/cs42l56.h | 48 + > include/sound/rcar_snd.h | 32 +- > include/sound/rt5640.h | 4 + > include/sound/rt5645.h | 25 + > include/sound/rt5651.h | 21 + > include/sound/soc-dai.h | 2 - > include/sound/soc-dapm.h | 20 +- > include/sound/soc.h | 155 +- > include/sound/sta350.h | 57 + > include/trace/events/asoc.h | 92 - > sound/arm/pxa2xx-pcm.c | 2 + > sound/arm/pxa2xx-pcm.h | 3 +- > sound/soc/atmel/Kconfig | 2 +- > sound/soc/atmel/atmel-pcm-pdc.c | 63 - > sound/soc/atmel/snd-soc-afeb9260.c | 12 - > sound/soc/codecs/88pm860x-codec.c | 20 +- > sound/soc/codecs/Kconfig | 34 +- > sound/soc/codecs/Makefile | 10 + > sound/soc/codecs/ab8500-codec.c | 12 +- > sound/soc/codecs/ad1980.c | 39 +- > sound/soc/codecs/adau1373.c | 7 +- > sound/soc/codecs/adav80x.c | 12 +- > sound/soc/codecs/ak4104.c | 62 +- > sound/soc/codecs/ak4641.c | 4 +- > sound/soc/codecs/ak4642.c | 75 +- > sound/soc/codecs/alc5623.c | 20 + > sound/soc/codecs/arizona.h | 4 +- > sound/soc/codecs/cq93vc.c | 10 +- > sound/soc/codecs/cs4270.c | 2 +- > sound/soc/codecs/cs4271.c | 4 +- > sound/soc/codecs/cs42l51-i2c.c | 59 + > sound/soc/codecs/cs42l51.c | 80 +- > sound/soc/codecs/cs42l51.h | 6 + > sound/soc/codecs/cs42l52.c | 12 - > sound/soc/codecs/cs42l56.c | 1427 +++++++++++ > sound/soc/codecs/cs42l56.h | 175 ++ > sound/soc/codecs/cs42xx8.c | 3 +- > sound/soc/codecs/da7210.c | 4 +- > sound/soc/codecs/da7213.c | 4 +- > sound/soc/codecs/da732x.c | 4 +- > sound/soc/codecs/da9055.c | 2 +- > sound/soc/codecs/hdmi.c | 1 + > sound/soc/codecs/lm4857.c | 4 +- > sound/soc/codecs/max9768.c | 4 +- > sound/soc/codecs/max98088.c | 12 +- > sound/soc/codecs/max98090.c | 87 +- > sound/soc/codecs/max98090.h | 1 + > sound/soc/codecs/max98095.c | 23 +- > sound/soc/codecs/mc13783.c | 36 +- > sound/soc/codecs/pcm1681.c | 4 +- > sound/soc/codecs/pcm512x.c | 4 +- > sound/soc/codecs/rt5631.c | 4 +- > sound/soc/codecs/rt5640.c | 424 ++-- > sound/soc/codecs/rt5640.h | 10 +- > sound/soc/codecs/rt5645.c | 2475 ++++++++++++++++++++ > sound/soc/codecs/rt5645.h | 2188 +++++++++++++++++ > sound/soc/codecs/rt5651.c | 1897 +++++++++++++++ > sound/soc/codecs/rt5651.h | 2081 ++++++++++++++++ > sound/soc/codecs/sgtl5000.c | 15 +- > sound/soc/codecs/si476x.c | 14 +- > sound/soc/codecs/sirf-audio-codec.c | 74 +- > sound/soc/codecs/sirf-audio-codec.h | 50 + > sound/soc/codecs/sta32x.c | 4 +- > sound/soc/codecs/sta350.c | 1311 +++++++++++ > sound/soc/codecs/sta350.h | 238 ++ > sound/soc/codecs/tas5086.c | 4 +- > sound/soc/codecs/tlv320aic23-i2c.c | 1 + > sound/soc/codecs/tlv320aic23.c | 4 +- > sound/soc/codecs/tlv320aic31xx.c | 3 +- > sound/soc/codecs/tlv320dac33.c | 12 +- > sound/soc/codecs/tpa6130a2.c | 1 + > sound/soc/codecs/twl4030.c | 10 +- > sound/soc/codecs/twl6040.c | 8 +- > sound/soc/codecs/wl1273.c | 12 +- > sound/soc/codecs/wm2000.c | 8 +- > sound/soc/codecs/wm2200.c | 4 +- > sound/soc/codecs/wm5100.c | 4 +- > sound/soc/codecs/wm5102.c | 26 +- > sound/soc/codecs/wm5110.c | 35 +- > sound/soc/codecs/wm8350.c | 14 +- > sound/soc/codecs/wm8400.c | 12 +- > sound/soc/codecs/wm8580.c | 2 +- > sound/soc/codecs/wm8731.c | 11 +- > sound/soc/codecs/wm8753.c | 4 +- > sound/soc/codecs/wm8804.c | 11 +- > sound/soc/codecs/wm8903.c | 4 +- > sound/soc/codecs/wm8904.c | 14 +- > sound/soc/codecs/wm8955.c | 13 +- > sound/soc/codecs/wm8958-dsp2.c | 32 +- > sound/soc/codecs/wm8960.c | 4 +- > sound/soc/codecs/wm8962.c | 35 +- > sound/soc/codecs/wm8962.h | 4 + > sound/soc/codecs/wm8983.c | 4 +- > sound/soc/codecs/wm8985.c | 11 +- > sound/soc/codecs/wm8988.c | 8 +- > sound/soc/codecs/wm8990.c | 2 +- > sound/soc/codecs/wm8991.c | 2 +- > sound/soc/codecs/wm8994.c | 32 +- > sound/soc/codecs/wm8995.c | 10 +- > sound/soc/codecs/wm8996.c | 4 +- > sound/soc/codecs/wm8997.c | 25 +- > sound/soc/codecs/wm9081.c | 4 +- > sound/soc/codecs/wm_adsp.c | 6 +- > sound/soc/codecs/wm_hubs.c | 2 +- > sound/soc/davinci/Kconfig | 10 +- > sound/soc/davinci/davinci-i2s.c | 1 - > sound/soc/davinci/davinci-mcasp.c | 260 +- > sound/soc/davinci/davinci-mcasp.h | 1 + > sound/soc/davinci/davinci-pcm.c | 8 +- > sound/soc/davinci/davinci-pcm.h | 8 +- > sound/soc/davinci/davinci-vcif.c | 1 - > sound/soc/fsl/Kconfig | 77 +- > sound/soc/fsl/Makefile | 3 +- > sound/soc/fsl/fsl_esai.c | 54 +- > sound/soc/fsl/fsl_sai.c | 255 +- > sound/soc/fsl/fsl_sai.h | 16 + > sound/soc/fsl/fsl_spdif.c | 186 +- > sound/soc/fsl/fsl_spdif.h | 14 +- > sound/soc/fsl/fsl_ssi.c | 996 +++----- > sound/soc/fsl/fsl_ssi.h | 62 +- > sound/soc/fsl/fsl_ssi_dbg.c | 163 ++ > sound/soc/fsl/imx-audmux.c | 2 +- > sound/soc/fsl/imx-pcm-dma.c | 1 - > sound/soc/generic/simple-card.c | 250 +- > sound/soc/intel/Makefile | 2 +- > sound/soc/intel/byt-rt5640.c | 34 +- > sound/soc/intel/haswell.c | 6 - > sound/soc/intel/sst-acpi.c | 1 + > sound/soc/intel/sst-baytrail-dsp.c | 16 +- > sound/soc/intel/sst-baytrail-ipc.c | 132 +- > sound/soc/intel/sst-baytrail-ipc.h | 7 +- > sound/soc/intel/sst-baytrail-pcm.c | 162 +- > sound/soc/intel/sst-dsp-priv.h | 3 + > sound/soc/intel/sst-dsp.c | 1 + > sound/soc/intel/sst-dsp.h | 1 + > sound/soc/intel/sst-firmware.c | 67 +- > sound/soc/intel/sst-haswell-dsp.c | 4 +- > sound/soc/intel/sst-haswell-ipc.c | 31 +- > sound/soc/intel/sst-haswell-ipc.h | 4 +- > sound/soc/intel/sst-haswell-pcm.c | 115 +- > sound/soc/intel/sst-mfld-dsp.h | 8 - > sound/soc/intel/sst-mfld-platform-compress.c | 237 ++ > ...sst-mfld-platform.c => sst-mfld-platform-pcm.c} | 250 +- > sound/soc/intel/sst-mfld-platform.h | 11 +- > sound/soc/jz4740/Kconfig | 11 +- > sound/soc/jz4740/jz4740-i2s.c | 5 +- > sound/soc/jz4740/qi_lb60.c | 84 +- > sound/soc/kirkwood/kirkwood-t5325.c | 13 - > sound/soc/nuc900/Kconfig | 1 + > sound/soc/nuc900/nuc900-ac97.c | 1 + > sound/soc/omap/Kconfig | 4 +- > sound/soc/omap/am3517evm.c | 2 +- > sound/soc/omap/ams-delta.c | 68 +- > sound/soc/omap/n810.c | 2 +- > sound/soc/omap/omap-abe-twl6040.c | 13 +- > sound/soc/omap/omap-dmic.c | 9 +- > sound/soc/omap/omap-hdmi-card.c | 2 +- > sound/soc/omap/omap-hdmi.c | 6 +- > sound/soc/omap/omap-mcbsp.c | 25 +- > sound/soc/omap/omap-mcbsp.h | 2 +- > sound/soc/omap/omap-mcpdm.c | 16 +- > sound/soc/omap/omap-pcm.c | 25 +- > sound/soc/omap/omap-pcm.h | 30 + > sound/soc/omap/omap-twl4030.c | 15 +- > sound/soc/omap/omap3pandora.c | 35 +- > sound/soc/omap/osk5912.c | 2 +- > sound/soc/omap/rx51.c | 275 ++- > sound/soc/pxa/Kconfig | 2 +- > sound/soc/pxa/brownstone.c | 7 - > sound/soc/pxa/palm27x.c | 8 - > sound/soc/pxa/poodle.c | 1 - > sound/soc/pxa/pxa-ssp.c | 2 - > sound/soc/pxa/pxa2xx-pcm.c | 2 + > sound/soc/pxa/ttc-dkb.c | 4 - > sound/soc/samsung/Kconfig | 22 +- > sound/soc/samsung/Makefile | 2 + > sound/soc/samsung/ac97.c | 9 +- > sound/soc/samsung/bells.c | 16 +- > sound/soc/samsung/dma.c | 8 +- > sound/soc/samsung/dma.h | 1 - > sound/soc/samsung/dmaengine.c | 13 +- > sound/soc/samsung/goni_wm8994.c | 9 +- > sound/soc/samsung/h1940_uda1380.c | 5 - > sound/soc/samsung/i2s.c | 10 +- > sound/soc/samsung/i2s.h | 1 + > sound/soc/samsung/idma.c | 11 +- > sound/soc/samsung/littlemill.c | 18 +- > sound/soc/samsung/lowland.c | 18 +- > sound/soc/samsung/neo1973_wm8753.c | 8 +- > sound/soc/samsung/pcm.c | 17 +- > sound/soc/samsung/rx1950_uda1380.c | 5 - > sound/soc/samsung/s3c-i2s-v2.c | 2 +- > sound/soc/samsung/s3c2412-i2s.c | 15 +- > sound/soc/samsung/s3c24xx-i2s.c | 19 +- > sound/soc/samsung/s3c24xx_simtec_hermes.c | 8 - > sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 8 - > sound/soc/samsung/smartq_wm8987.c | 2 - > sound/soc/samsung/smdk_wm8580pcm.c | 15 +- > sound/soc/samsung/smdk_wm8994.c | 12 - > sound/soc/samsung/smdk_wm8994pcm.c | 15 +- > sound/soc/samsung/snow.c | 123 + > sound/soc/samsung/spdif.c | 11 +- > sound/soc/samsung/speyside.c | 18 +- > sound/soc/samsung/tobermory.c | 18 +- > sound/soc/sh/Kconfig | 2 +- > sound/soc/sh/rcar/Makefile | 2 +- > sound/soc/sh/rcar/adg.c | 57 +- > sound/soc/sh/rcar/core.c | 168 +- > sound/soc/sh/rcar/dvc.c | 273 +++ > sound/soc/sh/rcar/gen.c | 25 +- > sound/soc/sh/rcar/rsnd.h | 68 +- > sound/soc/sh/rcar/src.c | 190 +- > sound/soc/sh/rcar/ssi.c | 85 +- > sound/soc/sirf/sirf-audio-port.c | 107 - > sound/soc/sirf/sirf-audio-port.h | 62 - > sound/soc/soc-cache.c | 2 - > sound/soc/soc-compress.c | 6 +- > sound/soc/soc-core.c | 991 ++++---- > sound/soc/soc-dapm.c | 273 +-- > sound/soc/soc-devres.c | 35 + > sound/soc/soc-io.c | 296 ++- > sound/soc/soc-pcm.c | 29 +- > sound/soc/tegra/tegra_wm9712.c | 4 +- > sound/soc/ux500/mop500_ab8500.c | 2 +- > 254 files changed, 17727 insertions(+), 4331 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/alc5623.txt > create mode 100644 Documentation/devicetree/bindings/sound/cs42l56.txt > create mode 100644 Documentation/devicetree/bindings/sound/max98095.txt > create mode 100644 Documentation/devicetree/bindings/sound/nokia,rx51.txt > create mode 100644 Documentation/devicetree/bindings/sound/snow.txt > create mode 100644 Documentation/devicetree/bindings/sound/st,sta350.txt > create mode 100644 include/sound/cs42l56.h > create mode 100644 include/sound/rt5645.h > create mode 100644 include/sound/rt5651.h > create mode 100644 include/sound/sta350.h > create mode 100644 sound/soc/codecs/cs42l51-i2c.c > create mode 100644 sound/soc/codecs/cs42l56.c > create mode 100644 sound/soc/codecs/cs42l56.h > create mode 100644 sound/soc/codecs/rt5645.c > create mode 100644 sound/soc/codecs/rt5645.h > create mode 100644 sound/soc/codecs/rt5651.c > create mode 100644 sound/soc/codecs/rt5651.h > create mode 100644 sound/soc/codecs/sta350.c > create mode 100644 sound/soc/codecs/sta350.h > create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c > create mode 100644 sound/soc/intel/sst-mfld-platform-compress.c > rename sound/soc/intel/{sst-mfld-platform.c => sst-mfld-platform-pcm.c} (67%) > create mode 100644 sound/soc/omap/omap-pcm.h > create mode 100644 sound/soc/samsung/snow.c > create mode 100644 sound/soc/sh/rcar/dvc.c > delete mode 100644 sound/soc/sirf/sirf-audio-port.h > [2 Digital signature <application/pgp-signature (7bit)>] >

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[alsa-devel] [PATCH] ASoC: alc5623: Depend on I2C.
by Andrew Lunn 22 May '14

22 May '14

This is an i2c driver, so make the Kconfig entry depend on I2C. Signed-off-by: Andrew Lunn <andrew(a)lunn.ch> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c59943a19f94..397b66c2b999 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -274,6 +274,7 @@ config SND_SOC_AK5386 config SND_SOC_ALC5623 tristate "Realtek ALC5623 CODEC" + depends on I2C config SND_SOC_ALC5632 tristate -- 2.0.0.rc2

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[alsa-devel] [PATCH] ALSA: Replace DEFINE_PCI_DEVICE_TABLE macro use
by Benoit Taine 22 May '14

22 May '14

We should prefer `const struct pci_device_id` over `DEFINE_PCI_DEVICE_TABLE` to meet kernel coding style guidelines. This issue was reported by checkpatch. A simplified version of the semantic patch that makes this change is as follows (http://coccinelle.lip6.fr/): // <smpl> @@ identifier i; declarer name DEFINE_PCI_DEVICE_TABLE; initializer z; @@ - DEFINE_PCI_DEVICE_TABLE(i) + const struct pci_device_id i[] = z; // </smpl> It has been tested by compilation. Signed-off-by: Benoit Taine <benoit.taine(a)lip6.fr> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2c54629..ab0fd6d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1730,7 +1730,7 @@ static void azx_remove(struct pci_dev *pci) } /* PCI IDs */ -static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { +static const struct pci_device_id azx_ids[] = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },

2 1

[alsa-devel] ASoC: Two codecs on a single CPU DAI
by Petr Kulhavy 22 May '14

22 May '14

Hi, I'm writing a driver for a custom DaVinci board, which has two codecs (Wolfson and STA323 amplifier) connected to the same CPU DAI. The intention is that both codecs always play the same audio, only the volume/mixer settings are controlled separately. In addition to that the design is atypical by the fact, that the Wolfson is the clock master. So in order to play audio on the amplifier the Wolfson must be initialized as well. My question is: how to reflect this architecture in a SOC platform driver? As the first approach I tried to create two DAI links, one for each codec. That worked if audio was first played on the Wolfson link (clock master was initialized) and then on the second link, using the same sampling rate. But in fact I would need only one DAI link to both. So then I tried to make just one link to Wolfson and put the amplifier as an "aux_dev". However there I need to set hw_params and format for the amplifier codec DAI and those functions were not called. In fact there was no codec DAI created for the amplifier. I've also seen that one can define rtd_aux runtimes, but have nowhere found how to use it. Could someone please help me? Thanks a lot Petr -- Petr Kulhavy, MSc System Architect BARIX AG

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Re: [alsa-devel] [ALC1150] combined with Intel H97/Z97
by Takashi Iwai 22 May '14

22 May '14

At Thu, 22 May 2014 14:04:55 +0200, Arthur Borsboom wrote: > > No, I did not and it resolved the issue!! Awesome! > > What does this option do? Does it have any side effects? This disables snooping mode. It's rather a problem of controller, not the codec chip. There can be a slight performance decrease. > If this is the final solution, I will communicate to the other users this > solution. Looking at alsa-info.sh, it's a new chip, and the workarounds for Intel PCH & co aren't applied to it yet. That's the likely problem. Could you try the patch below without snoop=0 option? Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2c54629d62d1..36b98b89c4c0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1743,6 +1743,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + /* ?? New chip */ + { PCI_DEVICE(0x8086, 0x8ca0), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Wellsburg */ { PCI_DEVICE(0x8086, 0x8d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },

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Re: [alsa-devel] [ALC1150] combined with Intel H97/Z97
by Takashi Iwai 22 May '14

22 May '14

At Thu, 22 May 2014 13:30:50 +0200, Arthur Borsboom wrote: > > Hi Takashi, > > I have found 2 other users having the same audio issues all having the > recent Intel H97/Z97 chipset and ALC1150. > Also my other Linux distributions and a LiveDVD (Linux Mint) give the same > results. > > My ALSA info can be found here: > https://drive.google.com/file/d/0Bymxsci1WxB1VF9vVHVMSFpjSm8/edit?usp=shari… > >From another user his ALSA info: > http://www.alsa-project.org/db/?f=f9672df0b8d9150a06ca54a562ee7ef8d030d49f > > Some things I have already tried such as snd-hda-module options, kernel PM > parameters on the Arch forum: > https://bbs.archlinux.org/viewtopic.php?pid=1418093#p1418093 > I hear some static noise, repeating, skipping, etc. To give you a better > impression of the sound issues, I have recorded two samples. > > Audacious music player: > https://drive.google.com/file/d/0Bymxsci1WxB1RFBzSHVZTmotUWs/edit?usp=shari… > Chrome Youtube: > https://drive.google.com/file/d/0Bymxsci1WxB1RnBSZUEweGNuT2M/edit?usp=shari… > > I also get this dmesg: "hda-intel: IRQ timing workaround is activated for > card #0. Suggest a bigger bdl_pos_adj." This is often harmless, so you can ignore it for now. > Strangely enough, if the CPU load is high, for example compiling a Linux > kernel, the sound is seems to be perfect. > Although compiling the Linux kernel all day is a nice workaround, but not > the solution I am looking for. :D Did you try to pass snoop=0 option? Takashi > > On 22 May 2014 12:33, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Wed, 21 May 2014 20:08:45 +0200, > > Arthur Borsboom wrote: > > > > > > Hi guys, > > > > > > Another user and I are having sound issues (noise, repeating, skipping) > > > with the ALC1150. > > > I have the feeling it is related to the new Intel H97/Z97 chipset on our > > > motherboard. > > > > > > Does anybody know about other issues in relation to this chipset? > > > > Could you give a bit more details of the problem? > > Also, alsa-info.sh output is always good to have. (Attach the > > compressed file to avoid the max length of ML posts.) > > > > There can be many causes for strange playbacks. If it's a static > > noise, it's often from the analog loopback. If it happens during the > > stream playback, mostly either the missing irqs, the wrong buffer > > position, or coherency/snooping issue. > > > > > > Takashi > > > > > > -- > Arthur Borsboom > Polderweg 10 > 2493 ZA Den Haag > Mob: +31620529577 > Email: arthurborsboom(a)gmail.com > Skype: Arthur Borsboom, The Hague, The Netherlands > > [image: View Arthur's LinkedIn > profile]<http://uk.linkedin.com/in/arthurborsboom>

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[alsa-devel] [ALC1150] combined with Intel H97/Z97
by Arthur Borsboom 22 May '14

22 May '14

Hi guys, Another user and I are having sound issues (noise, repeating, skipping) with the ALC1150. I have the feeling it is related to the new Intel H97/Z97 chipset on our motherboard. Does anybody know about other issues in relation to this chipset? Cheers, Arthur.

2 1

[alsa-devel] Set eapd to follow stream on or off
by Kailang 22 May '14

22 May '14

Hi Takashi, HP machine had pop noise via speaker. If EAPD on or off, speaker will have pop noise. So, they want to set eapd to follow with stream on or off. Many Thanks. BR, Kailang

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Re: [alsa-devel] [BUG]ABit aa8xe sound card not selected
by Raymond Yau 22 May '14

22 May '14

> > i tried to force the install one of olds kernel like 3.12 but ubuntu crash on start. removed the 3.12 i reinstall the 3.13 same identical problem (no audio). > tried to install the lastest development kernel but not solve my problem. > > For the detailed ALC880 config pins , I reveived this by Raimond on ubuntu-launchpad ( https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1319291 ) but I dont know how to apply the fix that you have sayd me, sorry. Can you send me a link for a guide that could help me? > > Maybe I have to modify the config file into /sys/class/sound/hwC0D0/init_pin_configs folder ? tried with root but is impossible to save. http://voices.canonical.com/david.henningsson/2012/07/13/top-five-wrong-way… When your BIOS does not setup pin default for all pin comples The only way you get your alc880 work is to edit /etc/modprobe.d/alsa-base.conf and add the following line if your motherboard have six audio jacks and digital out at rear panel options snd-hda-intel model=6stack-digout or options snd-hda-intel model=6stack-automute If your computer chassis have a hda front audio panel which support jack detection >> > >> > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… >> >> have you try the model=6stack-digout or 6stack-automute ? >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> ALC880_FIXUP_6ST_BASE] = { >> + .type = ALC_FIXUP_PINS, >> + .v.pins = (const struct alc_pincfg[]) { >> + { 0x14, 0x01014010 }, /* front */ >> + { 0x15, 0x01016412 }, /* surr */ >> + { 0x16, 0x01011411 }, /* CLFE */ >> + { 0x17, 0x01012414 }, /* side */ >> + { 0x18, 0x01a19c30 }, /* mic-in */ >> + { 0x19, 0x02a19c40 }, /* front-mic */ >> + { 0x1a, 0x01813031 }, /* line-in */ >> + { 0x1b, 0x0121411f }, /* HP */ >> >> HP in ALC880_FIXUP_6ST_BASE explicitly disable jack detection , >> >> Do you need a new model which support headphone automute speaker with digital output since 6stack-automute does not have digital ouput ? >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> > >> > 00:1b.0 0403: 8086:2668 (rev 04) >> > >> > Subsystem: 147b:1045 >> > >> > send email to alsa-devel mailing list to add pci quirk since Bios does not setup pin default >> > Default 0x00000000: [Jack] Line Out at Ext N/A >> > >> > Sysfs Files >> > >> > !!----------- >> > >> > /sys/class/sound/hwC0D0/init_pin_configs: >> > >> > 0x14 0x00000000 >> > >> > 0x15 0x00000000 >> > >> > 0x16 0x00000000 >> > >> > 0x17 0x00000000 >> > >> > 0x18 0x00000000 >> > >> > 0x19 0x00000000 >> > >> > 0x1a 0x00000000 >> > >> > 0x1b 0x00000000 >> > >> > 0x1c 0x00000000 >> > >> > 0x1d 0x00000000 >> > >> > 0x1e 0x00000000 >> > >> > 0x1f 0x00000000 >> > ----- >> > >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> Do you mean the driver still does not provide any warning message when all your pins are ignored by the driver ? >> >

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[alsa-devel] [PATCH 0/4] ASoC: samsung: Add devres APIs
by Tushar Behera 22 May '14

22 May '14

Adding devres APIs to various files in sound/soc/samsung saves us close to 150 lines of code. The patches are based on next-20140519 and have been build tested. Tushar Behera (4): ASoC: samsung: Use devm_snd_soc_register_card ASoC: samsung: Use devm_snd_soc_register_platform ASoC: samsung: Use devm_snd_soc_register_component ASoC: samsung: Add devm_clk_get to pcm.c sound/soc/samsung/ac97.c | 9 ++------- sound/soc/samsung/bells.c | 16 +++------------- sound/soc/samsung/dma.c | 8 +------- sound/soc/samsung/dma.h | 1 - sound/soc/samsung/dmaengine.c | 13 ++++--------- sound/soc/samsung/goni_wm8994.c | 9 +++------ sound/soc/samsung/i2s.c | 2 -- sound/soc/samsung/idma.c | 9 +-------- sound/soc/samsung/littlemill.c | 18 +++--------------- sound/soc/samsung/lowland.c | 18 +++--------------- sound/soc/samsung/pcm.c | 17 ++++------------- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c2412-i2s.c | 15 +-------------- sound/soc/samsung/s3c24xx-i2s.c | 19 +++---------------- sound/soc/samsung/smdk_wm8580pcm.c | 15 +++------------ sound/soc/samsung/smdk_wm8994pcm.c | 15 +++------------ sound/soc/samsung/spdif.c | 11 +++-------- sound/soc/samsung/speyside.c | 18 +++--------------- sound/soc/samsung/tobermory.c | 18 +++--------------- 19 files changed, 44 insertions(+), 189 deletions(-) -- 1.7.9.5

2 5

[alsa-devel] [ALC1150] combined with Intel H97/Z97
by Arthur Borsboom 21 May '14

21 May '14

Hi guys, Another user and I are having sound issues (noise, repeating, skipping) with the ALC1150. I have the feeling it is related to the new Intel H97/Z97 chipset on our motherboard. Does anybody know about other issues in relation to this chipset? Cheers, Arthur.

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: samsung: s3c24{xx, 12}-i2s: port to use generic dmaengine API
by Vasily Khoruzhick 21 May '14

21 May '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] tinycompress: compilation fix
by Chandrakanth Gorantla 21 May '14

21 May '14

fix missing extern C closing statement. --- include/tinycompress/tinycompress.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 03c396f..3d94da6 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -270,4 +270,8 @@ int is_compress_ready(struct compress *compress); /* Returns a human readable reason for the last error */ const char *compress_get_error(struct compress *compress); +#if defined(__cplusplus) +} /* extern "C" */ +#endif + #endif -- 1.8.1.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Disable AA-mix on Sony Vaio S13
by Takashi Iwai 21 May '14

21 May '14

The analog-loopback causes the speaker noises even if it's set to zero volume. As a simple workaround, just get rid of the loopback mixer. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=873704 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 680283b23474..bf1e1a625086 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4196,6 +4196,7 @@ enum { ALC269_FIXUP_ASUS_G73JW, ALC269_FIXUP_LENOVO_EAPD, ALC275_FIXUP_SONY_HWEQ, + ALC275_FIXUP_SONY_DISABLE_AAMIX, ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, ALC269_FIXUP_STEREO_DMIC, @@ -4301,6 +4302,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2 }, + [ALC275_FIXUP_SONY_DISABLE_AAMIX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_SONY_VAIO + }, [ALC271_FIXUP_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc271_fixup_dmic, @@ -4811,6 +4818,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), + SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), -- 1.9.3

1 0

[alsa-devel] [PATCH] fix format type mismatch in sound/pci/lola/lola_proc.c
by Toralf Förster 21 May '14

21 May '14

Signed-off-by: Toralf Förster <toralf.foerster(a)gmx.de> --- sound/pci/lola/lola_proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c index 04df83d..c241dc0 100644 --- a/sound/pci/lola/lola_proc.c +++ b/sound/pci/lola/lola_proc.c @@ -151,7 +151,7 @@ static void lola_proc_codec_rw_write(struct snd_info_entry *entry, char line[64]; unsigned int id, verb, data, extdata; while (!snd_info_get_line(buffer, line, sizeof(line))) { - if (sscanf(line, "%i %i %i %i", &id, &verb, &data, &extdata) != 4) + if (sscanf(line, "%u %u %u %u", &id, &verb, &data, &extdata) != 4) continue; lola_codec_read(chip, id, verb, data, extdata, &chip->debug_res, -- 1.9.3

2 1

[alsa-devel] [PATCH] fix format type mismatch in sound/pci/hda/patch_sigmatel.c
by Toralf Förster 21 May '14

21 May '14

Signed-off-by: Toralf Förster <toralf.foerster(a)gmx.de> --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 75515b4..7f40a15 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -795,7 +795,7 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) } while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { - if (sscanf(dev->name, "HP_Mute_LED_%d_%x", + if (sscanf(dev->name, "HP_Mute_LED_%u_%x", &spec->gpio_led_polarity, &spec->gpio_led) == 2) { unsigned int max_gpio; @@ -808,7 +808,7 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) spec->vref_mute_led_nid = spec->gpio_led; return 1; } - if (sscanf(dev->name, "HP_Mute_LED_%d", + if (sscanf(dev->name, "HP_Mute_LED_%u", &spec->gpio_led_polarity) == 1) { set_hp_led_gpio(codec); return 1; -- 1.9.3

2 1

Re: [alsa-devel] [PATCH] ALSA: hda - Disable AA-loopback on ALC668
by Takashi Iwai 21 May '14

21 May '14

At Tue, 20 May 2014 18:58:26 +0200, Gabriele Mazzotta wrote: > > On Tuesday 20 May 2014 18:06:53, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 20 May 2014 17:59:30 +0200, > > Gabriele Mazzotta wrote: > > > > > > On Thursday 15 May 2014 21:56:23, Gabriele Mazzotta <gabriele.mzt(a)gmail.com> > > > wrote: > > > > On Wednesday 14 May 2014 16:31:13, Gabriele Mazzotta > > > <gabriele.mzt(a)gmail.com> wrote: > > > > I'm sorry, I was wrong. This patch does not help. I'm still trying to > > > > understand what temporarily fixed the problem, I was quite sure this patch > > > did > > > > it, but all of the sudden the noise came back. I'm still trying to find the > > > > real cause, but I doubt I'll succeed. > > > > > > > > Regards, > > > > Gabriele > > > I have to correct myself again, the first patch I submitted is working, I'm > > > sorry. However, it seems that disabling the AA-loopback path is not necessary, > > > muting the input 0x0b in node 0x0c (as it was suggested here: > > > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1252733/comments…) is enough to get rid of the white noise. > > > > This reminds me of a similar problem reported on Sony VAIO. > > https://bugzilla.novell.com/show_bug.cgi?id=873704 > > > > The codec is different, it's ALC275, though. > > > > > What is the proper way to fix this problem? > > > > Just to be sure: does the mic input itself work correctly? > > Unfortunately, I cannot test it. > > > Maybe the easiest fix would be disabling the aamix, indeed. > > But you shouldn't apply it generically to ALC668. I'm sure that it's > > pretty specific to machines, not codec chips. So, create a fixup > > function that matches with your machine instead. > > Yes, you are probably right. All I know is that this bug affect the > XPS13, XPS12 and XPS15. > > Is the following patch OK? > --- > > > ALSA: hda - White noise fix for XPS13 9333 > > Disable the AA-loopback path to get rid of the constant white noise > that can be heard when headphones are used. > > Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> Looks good, so applied now. Thanks. Takashi > --- > sound/pci/hda/patch_realtek.c | 19 ++++++++++++++++++- > 1 file changed, 18 insertions(+), 1 deletion(-) > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 49e884f..a2c4374 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3894,6 +3894,16 @@ static void alc_fixup_no_shutup(struct hda_codec *codec, > } > } > > +static void alc_fixup_disable_aamix(struct hda_codec *codec, > + const struct hda_fixup *fix, int action) > +{ > + if (action == HDA_FIXUP_ACT_PRE_PROBE) { > + struct alc_spec *spec = codec->spec; > + /* Disable AA-loopback as it causes white noise */ > + spec->gen.mixer_nid = 0; > + } > +} > + > static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > @@ -5313,6 +5323,7 @@ enum { > ALC662_FIXUP_BASS_1A, > ALC662_FIXUP_BASS_CHMAP, > ALC668_FIXUP_AUTO_MUTE, > + ALC668_FIXUP_DELL_DISABLE_AAMIX, > }; > > static const struct hda_fixup alc662_fixups[] = { > @@ -5479,6 +5490,12 @@ static const struct hda_fixup alc662_fixups[] = { > .type = HDA_FIXUP_FUNC, > .v.func = alc_fixup_inv_dmic_0x12, > }, > + [ALC668_FIXUP_DELL_DISABLE_AAMIX] = { > + .type = HDA_FIXUP_FUNC, > + .v.func = alc_fixup_disable_aamix, > + .chained = true, > + .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE > + }, > [ALC668_FIXUP_AUTO_MUTE] = { > .type = HDA_FIXUP_FUNC, > .v.func = alc_fixup_auto_mute_via_amp, > @@ -5539,7 +5556,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { > SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), > SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > - SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > + SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_DISABLE_AAMIX), > SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), > SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), > SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > -- > 2.0.0.rc2 >

1 0

[alsa-devel] [PATCH] ASoC: samsung: Add 'mclk' handling for Snow sound-card driver
by Tushar Behera 21 May '14

21 May '14

For audio to work on Snow board, we need the codec master clock 'mclk' to be properly configured. Currently XCLKOUT is configured as 'mclk' for codec chip and it is required to be clocked at 24MHz for the codec to work properly. Add appropriate clock handling within Snow sound-card driver to set 'mclk' to operate at 24MHz. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- Documentation/devicetree/bindings/sound/snow.txt | 7 ++++++ sound/soc/samsung/snow.c | 26 +++++++++++++++++++++- 2 files changed, 32 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 678b191..94dd29c 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -7,6 +7,10 @@ Required properties: - samsung,i2s-controller: The phandle of the Samsung I2S controller - samsung,audio-codec: The phandle of the audio codec +Optional properties: +- clocks: The phandle of the master clock to audio codec +- clock-names: Should be "mclk" + Example: sound { @@ -14,4 +18,7 @@ sound { samsung,i2s-controller = <&i2s0>; samsung,audio-codec = <&max98095>; + + clocks = <&pmu_system_controller>; + clock-names = "mclk"; }; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 014c177..c910739 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -15,13 +15,18 @@ #include <linux/platform_device.h> #include <linux/of.h> #include <linux/of_device.h> +#include <linux/clk.h> #include <sound/soc.h> #include "i2s.h" +/* Desired clock rate for codec mclk */ #define FIN_PLL_RATE 24000000 +static struct clk *mclk; +static unsigned long mclk_rate; + static struct snd_soc_dai_link snow_dai[] = { { .name = "Primary", @@ -41,7 +46,7 @@ static int snow_late_probe(struct snd_soc_card *card) /* Set the MCLK rate for the codec */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, - FIN_PLL_RATE, SND_SOC_CLOCK_IN); + mclk_rate, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -68,6 +73,16 @@ static int snow_probe(struct platform_device *pdev) struct device_node *i2s_node, *codec_node; int i, ret; + /* The codec MCLK should be clocked at 24MHz */ + mclk = devm_clk_get(&pdev->dev, "mclk"); + if (!IS_ERR(mclk)) { + clk_prepare_enable(mclk); + clk_set_rate(mclk, FIN_PLL_RATE); + mclk_rate = clk_get_rate(mclk); + } else { + mclk_rate = FIN_PLL_RATE; + } + i2s_node = of_parse_phandle(pdev->dev.of_node, "samsung,i2s-controller", 0); if (!i2s_node) { @@ -101,6 +116,14 @@ static int snow_probe(struct platform_device *pdev) return ret; } +static int snow_remove(struct platform_device *pdev) +{ + if (!IS_ERR(mclk)) + clk_disable_unprepare(mclk); + + return 0; +} + static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98090", }, { .compatible = "google,snow-audio-max98095", }, @@ -115,6 +138,7 @@ static struct platform_driver snow_driver = { .of_match_table = snow_of_match, }, .probe = snow_probe, + .remove = snow_remove, }; module_platform_driver(snow_driver); -- 1.7.9.5

2 2

[alsa-devel] Support for getting "manufacturer" from snd_seq_client_info and/or snd_ctl_card_info
by Adam Goode 21 May '14

21 May '14

Hi, I am doing a bit of work on the Alsa Web MIDI implementation in Chrome: https://code.google.com/p/chromium/issues/detail?id=344410 Web MIDI likes to know the "manufacturer" field as part of its API: http://webaudio.github.io/web-midi-api/#attributes-3 The current implementation in Chrome uses rawmidi, and then tries to parse out the manufacturer from the longname of the card. This works reasonably well, but is brittle and results in a lot of code. I am working on a seq implementation for Chrome, to replace the rawmidi implementation. I can retain the manufacturer extraction hack on the card, but seq requires yet another hack to guess the card for a particular seq client. Any suggestions on how to extract the manufacturer more easily? The best solution would be to add a manufacturer (or vendor) field to the snd_seq_client_info ioctl struct. This would likely require a new ioctl, since there is only 64 bytes reserved in the current struct. I am guessing this is not too likely to be accepted. (I would want to also add the same manufacturer field to the snd_ctl_card_info.) The next best solution would be to add a card field to snd_seq_client_info, which could be optionally populated with the card that is associated to the client. A completely different solution would be to add all this info to sysfs which is easier to extend dynamically and won't require any ioctl changes at all. (I kind of like this solution the best, but it would be a change from the ioctl-based mechanisms done so far.) Ultimately, if I am stuck guessing and parsing, that will work as long as the underlying implementation doesn't change. But I would rather have a more reliable solution going forward. Any thoughts? Thanks, Adam Goode

3 8

[alsa-devel] [PATCH] ASoC: neo1973_wm8753: Automatically disconnected non-connected pins
by Lars-Peter Clausen 21 May '14

21 May '14

The DAPM routes for this board are complete, hence we can let the core take care of disconnecting non-connected pins rather than doing it manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/neo1973_wm8753.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index b080033..9b4a09f 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -271,15 +271,8 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = rtd->card; - /* set up NC codec pins */ - snd_soc_dapm_nc_pin(&codec->dapm, "OUT3"); - snd_soc_dapm_nc_pin(&codec->dapm, "OUT4"); - snd_soc_dapm_nc_pin(&codec->dapm, "LINE1"); - snd_soc_dapm_nc_pin(&codec->dapm, "LINE2"); - /* set endpoints to default off mode */ snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out"); snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In"); @@ -355,6 +348,7 @@ static struct snd_soc_card neo1973 = { .num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets), .dapm_routes = neo1973_wm8753_routes, .num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes), + .fully_routed = true, }; static struct platform_device *neo1973_snd_device; -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: samsung: Allow setting OP_CLK of the IIS Multi Audio Interface
by Sylwester Nawrocki 21 May '14

21 May '14

This patch adds support for setting source clock of the "Core CLK" of the IIS Multi Audio Interface. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 4 ++++ sound/soc/samsung/i2s.h | 1 + 2 files changed, 5 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 048ead9..ae02811 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -451,6 +451,10 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, u32 mod = readl(i2s->addr + I2SMOD); switch (clk_id) { + case SAMSUNG_I2S_OPCLK: + mod &= ~MOD_OPCLK_MASK; + mod |= dir; + break; case SAMSUNG_I2S_CDCLK: /* Shouldn't matter in GATING(CLOCK_IN) mode */ if (dir == SND_SOC_CLOCK_IN) diff --git a/sound/soc/samsung/i2s.h b/sound/soc/samsung/i2s.h index 7966afc..21ff24e 100644 --- a/sound/soc/samsung/i2s.h +++ b/sound/soc/samsung/i2s.h @@ -18,5 +18,6 @@ #define SAMSUNG_I2S_RCLKSRC_0 0 #define SAMSUNG_I2S_RCLKSRC_1 1 #define SAMSUNG_I2S_CDCLK 2 +#define SAMSUNG_I2S_OPCLK 3 #endif /* __SND_SOC_SAMSUNG_I2S_H */ -- 1.7.9.5

3 5

[alsa-devel] [PATCH 15/15] ASoC: pxa: remove mach header dependency
by Xia Kaixu 21 May '14

21 May '14

From: Arnd Bergmann <arnd(a)arndb.de> As we are moving the mmp platform towards multiplatform support, we have to stop including platform header files. This changes the pxa-ssp sound driver file to no longer depend on mach/hardware.h and mach/dma.h. The code using the definitions from those headers is actually gone already, the only thing that was still being used was the pxa_dma_desc typedef, which we can easily work around by using the normal 'struct pxa_dma_desc' name. The pxa2xx-dma driver still uses this header, so we include it explicitly there, which is ok because that is only used on pxa, not on mmp. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Russell King <linux(a)arm.linux.org.uk> Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org --- sound/arm/pxa2xx-pcm.c | 2 ++ sound/arm/pxa2xx-pcm.h | 3 +-- sound/soc/pxa/pxa-ssp.c | 2 -- sound/soc/pxa/pxa2xx-pcm.c | 2 ++ 4 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index e6c727b..83be8e3 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -14,6 +14,8 @@ #include <linux/dma-mapping.h> #include <linux/dmaengine.h> +#include <mach/dma.h> + #include <sound/core.h> #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 2a8fc08..0033098 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -9,12 +9,11 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ -#include <mach/dma.h> struct pxa2xx_runtime_data { int dma_ch; struct snd_dmaengine_dai_dma_data *params; - pxa_dma_desc *dma_desc_array; + struct pxa_dma_desc *dma_desc_array; dma_addr_t dma_desc_array_phys; }; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a3119a0..9b19ee7 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -34,8 +34,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include <mach/hardware.h> - #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index d58b09f..42f2f01 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -15,6 +15,8 @@ #include <linux/dmaengine.h> #include <linux/of.h> +#include <mach/dma.h> + #include <sound/core.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> -- 1.7.9.5

2 1

[alsa-devel] [PATCH] Fix for possible null pointer dereference in dma.c
by Rickard Strandqvist 20 May '14

20 May '14

There is otherwise a risk of a possible null pointer dereference. Was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/samsung/dma.c | 10 ++++++---- 1 fil ändrad, 6 tillägg(+), 4 borttagningar(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index dc09b71..b1f6757 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -115,17 +115,19 @@ static void dma_enqueue(struct snd_pcm_substream *substream) static void audio_buffdone(void *data) { struct snd_pcm_substream *substream = data; - struct runtime_data *prtd = substream->runtime->private_data; + struct runtime_data *prtd = NULL; pr_debug("Entered %s\n", __func__); - if (prtd->state & ST_RUNNING) { + if(substream) + prtd = substream->runtime->private_data; + + if (prtd && prtd->state & ST_RUNNING) { prtd->dma_pos += prtd->dma_period; if (prtd->dma_pos >= prtd->dma_end) prtd->dma_pos = prtd->dma_start; - if (substream) - snd_pcm_period_elapsed(substream); + snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); if (!samsung_dma_has_circular()) { -- 1.7.10.4

6 8

[alsa-devel] [PATCH] ASoC: ad1980: Replace goto loop with do-while loop
by Lars-Peter Clausen 20 May '14

20 May '14

Using a proper do-while loop here instead of a open-coded goto loop is both cleaner and shorter. Also fixes the following warnings from smatch: sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement. sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code. sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code. While we are at it also change retry_cnt to unsigned int, using u16 for a on-stack loop counter doesn't make that much sense. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1980.c | 39 +++++++++++++++++++-------------------- 1 file changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 34d965a..304d300 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -189,28 +189,27 @@ static struct snd_soc_dai_driver ad1980_dai = { static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { - u16 retry_cnt = 0; + unsigned int retry_cnt = 0; -retry: - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(codec->ac97); - if (ac97_read(codec, AC97_RESET) == 0x0090) - return 1; - } - - soc_ac97_ops->reset(codec->ac97); - /* Set bit 16slot in register 74h, then every slot will has only 16 - * bits. This command is sent out in 20bit mode, in which case the - * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/ - ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900); - - if (ac97_read(codec, AC97_RESET) != 0x0090) - goto err; - return 0; + do { + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); + if (ac97_read(codec, AC97_RESET) == 0x0090) + return 1; + } -err: - while (retry_cnt++ < 10) - goto retry; + soc_ac97_ops->reset(codec->ac97); + /* + * Set bit 16slot in register 74h, then every slot will has only + * 16 bits. This command is sent out in 20bit mode, in which + * case the first nibble of data is eaten by the addr. (Tag is + * always 16 bit) + */ + ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900); + + if (ac97_read(codec, AC97_RESET) == 0x0090) + return 0; + } while (retry_cnt++ < 10); printk(KERN_ERR "AD1980 AC97 reset failed\n"); return -EIO; -- 1.8.0

2 1

[alsa-devel] [PATCH] ALSA: hda/tegra - Fix MODULE_DEVICE_TABLE typo.
by Dylan Reid 20 May '14

20 May '14

I missed a rename during the review process. Fix the MODULE_DEVICE_TABLE to match the structure. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- 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 e472bc8..086fd06 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -496,7 +496,7 @@ static const struct of_device_id hda_tegra_match[] = { { .compatible = "nvidia,tegra30-hda" }, {}, }; -MODULE_DEVICE_TABLE(of, tegra_platform_hda_match); +MODULE_DEVICE_TABLE(of, hda_tegra_match); static int hda_tegra_probe(struct platform_device *pdev) { -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCHv5 2/2] ALSA: hda - Add driver for Tegra SoC HDA
by Dylan Reid 20 May '14

20 May '14

This adds a driver for the HDA block in Tegra SoCs. The HDA bus is used to communicate with the HDMI codec on Tegra124. Most of the code is re-used from the Intel/PCI HDA driver. It brings over only two of the module params, power_save and probe_mask. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> Reviewed-by: Stephen Warren <swarren(a)nvidia.com> --- Changes since v4: Fix up binding doc, Kconfig depends and some style issue suggested by swarren. Test on Venice2 which now supports HDMI output. Changes since v3: Remove some runtime-pm only code from the regular pm path. Changes since v2: Remove runtime PM. The on-chip codec doesn't support D3 stop clock so the pm_runtime code was never run. Changes since v1: Kconfig depends on OF and ARCH_TEGRA. Fix warning on 64 bit build. Integrate a bunch of good suggestions from Thierry, including: Add resets property to device tree entry. More consistent/better naming. Remove probe mask that isn't needed. Cleaner/more readable handling of dword write restriction. --- .../bindings/sound/nvidia,tegra30-hda.txt | 28 + sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_tegra.c | 584 +++++++++++++++++++++ 4 files changed, 629 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt create mode 100644 sound/pci/hda/hda_tegra.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt new file mode 100644 index 0000000..c26f8ca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt @@ -0,0 +1,28 @@ +NVIDIA Tegra30 HDA controller + +Required properties: +- compatible : "nvidia,tegra30-hda" +- reg : Should contain the HDA registers location and length. +- interrupts : The interrupt from the HDA controller. +- clocks : Must contain an entry for each required entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi +- resets : Must contain an entry for each entry in reset-names. + See ../reset/reset.txt for details. +- reset-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi + +Example: + +hda@0,70030000 { + compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; + reg = <0x0 0x70030000 0x0 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hdacodec_2x"; + resets = <&tegra_car 125>, /* hda */ + <&tegra_car 128>; /* hda2hdmi */ + <&tegra_car 111>, /* hda2codec_2x */ + reset-names = "hda", "hda2hdmi", "hdacodec_2x"; +}; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ac17c3f..ebf4c2f 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -20,6 +20,21 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. +config SND_HDA_TEGRA + tristate "NVIDIA Tegra HD Audio" + depends on ARCH_TEGRA + select SND_HDA + help + Say Y here to support the HDA controller present in NVIDIA + Tegra SoCs + + This options enables support for the HD Audio controller + present in some NVIDIA Tegra SoCs, used to communicate audio + to the HDMI output. + + To compile this driver as a module, choose M here: the module + will be called snd-hda-tegra. + if SND_HDA config SND_HDA_DSP_LOADER diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index d0d0c19..194f3093 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,6 @@ snd-hda-intel-objs := hda_intel.o snd-hda-controller-objs := hda_controller.o +snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o @@ -47,3 +48,4 @@ obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o # otherwise the codec patches won't be hooked before the PCI probe # when built in kernel obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o +obj-$(CONFIG_SND_HDA_TEGRA) += snd-hda-tegra.o diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c new file mode 100644 index 0000000..e472bc8 --- /dev/null +++ b/sound/pci/hda/hda_tegra.c @@ -0,0 +1,584 @@ +/* + * + * Implementation of primary ALSA driver code base for NVIDIA Tegra HDA. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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, see <http://www.gnu.org/licenses/>. + * + */ + +#include <linux/clk.h> +#include <linux/clocksource.h> +#include <linux/completion.h> +#include <linux/delay.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/mutex.h> +#include <linux/of_device.h> +#include <linux/reboot.h> +#include <linux/slab.h> +#include <linux/time.h> + +#include <sound/core.h> +#include <sound/initval.h> + +#include "hda_codec.h" +#include "hda_controller.h" +#include "hda_priv.h" + +/* Defines for Nvidia Tegra HDA support */ +#define HDA_BAR0 0x8000 + +#define HDA_CFG_CMD 0x1004 +#define HDA_CFG_BAR0 0x1010 + +#define HDA_ENABLE_IO_SPACE (1 << 0) +#define HDA_ENABLE_MEM_SPACE (1 << 1) +#define HDA_ENABLE_BUS_MASTER (1 << 2) +#define HDA_ENABLE_SERR (1 << 8) +#define HDA_DISABLE_INTR (1 << 10) +#define HDA_BAR0_INIT_PROGRAM 0xFFFFFFFF +#define HDA_BAR0_FINAL_PROGRAM (1 << 14) + +/* IPFS */ +#define HDA_IPFS_CONFIG 0x180 +#define HDA_IPFS_EN_FPCI 0x1 + +#define HDA_IPFS_FPCI_BAR0 0x80 +#define HDA_FPCI_BAR0_START 0x40 + +#define HDA_IPFS_INTR_MASK 0x188 +#define HDA_IPFS_EN_INTR (1 << 16) + +/* max number of SDs */ +#define NUM_CAPTURE_SD 1 +#define NUM_PLAYBACK_SD 1 + +struct hda_tegra { + struct azx chip; + struct device *dev; + struct clk *hda_clk; + struct clk *hda2codec_2x_clk; + struct clk *hda2hdmi_clk; + void __iomem *regs; +}; + +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +module_param(power_save, bint, 0644); +MODULE_PARM_DESC(power_save, + "Automatic power-saving timeout (in seconds, 0 = disable)."); + +/* + * DMA page allocation ops. + */ +static int dma_alloc_pages(struct azx *chip, int type, size_t size, + struct snd_dma_buffer *buf) +{ + return snd_dma_alloc_pages(type, chip->card->dev, size, buf); +} + +static void dma_free_pages(struct azx *chip, struct snd_dma_buffer *buf) +{ + snd_dma_free_pages(buf); +} + +static int substream_alloc_pages(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size) +{ + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + return snd_pcm_lib_malloc_pages(substream, size); +} + +static int substream_free_pages(struct azx *chip, + struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* + * Register access ops. Tegra HDA register access is DWORD only. + */ +static void hda_tegra_writel(u32 value, u32 *addr) +{ + writel(value, addr); +} + +static u32 hda_tegra_readl(u32 *addr) +{ + return readl(addr); +} + +static void hda_tegra_writew(u16 value, u16 *addr) +{ + unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; + void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + v &= ~(0xffff << shift); + v |= value << shift; + writel(v, dword_addr); +} + +static u16 hda_tegra_readw(u16 *addr) +{ + unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; + void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + return (v >> shift) & 0xffff; +} + +static void hda_tegra_writeb(u8 value, u8 *addr) +{ + unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; + void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + v &= ~(0xff << shift); + v |= value << shift; + writel(v, dword_addr); +} + +static u8 hda_tegra_readb(u8 *addr) +{ + unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; + void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + return (v >> shift) & 0xff; +} + +static const struct hda_controller_ops hda_tegra_ops = { + .reg_writel = hda_tegra_writel, + .reg_readl = hda_tegra_readl, + .reg_writew = hda_tegra_writew, + .reg_readw = hda_tegra_readw, + .reg_writeb = hda_tegra_writeb, + .reg_readb = hda_tegra_readb, + .dma_alloc_pages = dma_alloc_pages, + .dma_free_pages = dma_free_pages, + .substream_alloc_pages = substream_alloc_pages, + .substream_free_pages = substream_free_pages, +}; + +static void hda_tegra_init(struct hda_tegra *hda) +{ + u32 v; + + /* Enable PCI access */ + v = readl(hda->regs + HDA_IPFS_CONFIG); + v |= HDA_IPFS_EN_FPCI; + writel(v, hda->regs + HDA_IPFS_CONFIG); + + /* Enable MEM/IO space and bus master */ + v = readl(hda->regs + HDA_CFG_CMD); + v &= ~HDA_DISABLE_INTR; + v |= HDA_ENABLE_MEM_SPACE | HDA_ENABLE_IO_SPACE | + HDA_ENABLE_BUS_MASTER | HDA_ENABLE_SERR; + writel(v, hda->regs + HDA_CFG_CMD); + + writel(HDA_BAR0_INIT_PROGRAM, hda->regs + HDA_CFG_BAR0); + writel(HDA_BAR0_FINAL_PROGRAM, hda->regs + HDA_CFG_BAR0); + writel(HDA_FPCI_BAR0_START, hda->regs + HDA_IPFS_FPCI_BAR0); + + v = readl(hda->regs + HDA_IPFS_INTR_MASK); + v |= HDA_IPFS_EN_INTR; + writel(v, hda->regs + HDA_IPFS_INTR_MASK); +} + +static int hda_tegra_enable_clocks(struct hda_tegra *data) +{ + int rc; + + rc = clk_prepare_enable(data->hda_clk); + if (rc) + return rc; + rc = clk_prepare_enable(data->hda2codec_2x_clk); + if (rc) + goto disable_hda; + rc = clk_prepare_enable(data->hda2hdmi_clk); + if (rc) + goto disable_codec_2x; + + return 0; + +disable_codec_2x: + clk_disable_unprepare(data->hda2codec_2x_clk); +disable_hda: + clk_disable_unprepare(data->hda_clk); + return rc; +} + +static void hda_tegra_disable_clocks(struct hda_tegra *data) +{ + clk_disable_unprepare(data->hda2hdmi_clk); + clk_disable_unprepare(data->hda2codec_2x_clk); + clk_disable_unprepare(data->hda_clk); +} + +#ifdef CONFIG_PM_SLEEP +/* + * power management + */ +static int hda_tegra_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + if (chip->initialized) + snd_hda_suspend(chip->bus); + + azx_stop_chip(chip); + azx_enter_link_reset(chip); + hda_tegra_disable_clocks(hda); + + return 0; +} + +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); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops hda_tegra_pm = { + SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) +}; + +/* + * reboot notifier for hang-up problem at power-down + */ +static int hda_tegra_halt(struct notifier_block *nb, unsigned long event, + void *buf) +{ + struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); + azx_stop_chip(chip); + return NOTIFY_OK; +} + +static void hda_tegra_notifier_register(struct azx *chip) +{ + chip->reboot_notifier.notifier_call = hda_tegra_halt; + register_reboot_notifier(&chip->reboot_notifier); +} + +static void hda_tegra_notifier_unregister(struct azx *chip) +{ + if (chip->reboot_notifier.notifier_call) + unregister_reboot_notifier(&chip->reboot_notifier); +} + +/* + * destructor + */ +static int hda_tegra_dev_free(struct snd_device *device) +{ + int i; + struct azx *chip = device->device_data; + + hda_tegra_notifier_unregister(chip); + + if (chip->initialized) { + for (i = 0; i < chip->num_streams; i++) + azx_stream_stop(chip, &chip->azx_dev[i]); + azx_stop_chip(chip); + } + + azx_free_stream_pages(chip); + + return 0; +} + +static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) +{ + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct device *dev = hda->dev; + struct resource *res; + int err; + + hda->hda_clk = devm_clk_get(dev, "hda"); + if (IS_ERR(hda->hda_clk)) + return PTR_ERR(hda->hda_clk); + hda->hda2codec_2x_clk = devm_clk_get(dev, "hda2codec_2x"); + if (IS_ERR(hda->hda2codec_2x_clk)) + return PTR_ERR(hda->hda2codec_2x_clk); + hda->hda2hdmi_clk = devm_clk_get(dev, "hda2hdmi"); + if (IS_ERR(hda->hda2hdmi_clk)) + return PTR_ERR(hda->hda2hdmi_clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + hda->regs = devm_ioremap_resource(dev, res); + if (IS_ERR(chip->remap_addr)) + return PTR_ERR(chip->remap_addr); + + chip->remap_addr = hda->regs + HDA_BAR0; + chip->addr = res->start + HDA_BAR0; + + err = hda_tegra_enable_clocks(hda); + if (err) + return err; + + hda_tegra_init(hda); + + return 0; +} + +/* + * The codecs were powered up in snd_hda_codec_new(). + * Now all initialization done, so turn them down if possible + */ +static void power_down_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + list_for_each_entry(codec, &chip->bus->codec_list, list) + snd_hda_power_down(codec); +} + +static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) +{ + struct snd_card *card = chip->card; + int err; + unsigned short gcap; + int irq_id = platform_get_irq(pdev, 0); + + err = hda_tegra_init_chip(chip, pdev); + if (err) + return err; + + err = devm_request_irq(chip->card->dev, irq_id, azx_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip); + if (err) { + dev_err(chip->card->dev, + "unable to request IRQ %d, disabling device\n", + irq_id); + return err; + } + chip->irq = irq_id; + + synchronize_irq(chip->irq); + + gcap = azx_readw(chip, GCAP); + dev_dbg(card->dev, "chipset global capabilities = 0x%x\n", gcap); + + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->capture_streams = (gcap >> 8) & 0x0f; + chip->playback_streams = (gcap >> 12) & 0x0f; + if (!chip->playback_streams && !chip->capture_streams) { + /* gcap didn't give any info, switching to old method */ + chip->playback_streams = NUM_PLAYBACK_SD; + chip->capture_streams = NUM_CAPTURE_SD; + } + chip->capture_index_offset = 0; + chip->playback_index_offset = chip->capture_streams; + chip->num_streams = chip->playback_streams + chip->capture_streams; + chip->azx_dev = devm_kcalloc(card->dev, chip->num_streams, + sizeof(*chip->azx_dev), GFP_KERNEL); + if (!chip->azx_dev) + return -ENOMEM; + + err = azx_alloc_stream_pages(chip); + if (err < 0) + return err; + + /* initialize streams */ + azx_init_stream(chip); + + /* initialize chip */ + azx_init_chip(chip, 1); + + /* codec detection */ + if (!chip->codec_mask) { + dev_err(card->dev, "no codecs found!\n"); + return -ENODEV; + } + + strcpy(card->driver, "tegra-hda"); + strcpy(card->shortname, "tegra-hda"); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->addr, chip->irq); + + return 0; +} + +/* + * constructor + */ +static int hda_tegra_create(struct snd_card *card, + unsigned int driver_caps, + const struct hda_controller_ops *hda_ops, + struct hda_tegra *hda) +{ + static struct snd_device_ops ops = { + .dev_free = hda_tegra_dev_free, + }; + struct azx *chip; + int err; + + chip = &hda->chip; + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->ops = hda_ops; + chip->irq = -1; + chip->driver_caps = driver_caps; + chip->driver_type = driver_caps & 0xff; + chip->dev_index = 0; + INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->list); + + chip->position_fix[0] = POS_FIX_AUTO; + chip->position_fix[1] = POS_FIX_AUTO; + chip->codec_probe_mask = -1; + + chip->single_cmd = false; + chip->snoop = true; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + dev_err(card->dev, "Error creating device\n"); + return err; + } + + return 0; +} + +static const struct of_device_id hda_tegra_match[] = { + { .compatible = "nvidia,tegra30-hda" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tegra_platform_hda_match); + +static int hda_tegra_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct azx *chip; + struct hda_tegra *hda; + int err; + const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY; + + hda = devm_kzalloc(&pdev->dev, sizeof(*hda), GFP_KERNEL); + if (!hda) + return -ENOMEM; + hda->dev = &pdev->dev; + chip = &hda->chip; + + err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0, &card); + if (err < 0) { + dev_err(&pdev->dev, "Error creating card!\n"); + return err; + } + + err = hda_tegra_create(card, driver_flags, &hda_tegra_ops, hda); + if (err < 0) + goto out_free; + card->private_data = chip; + + dev_set_drvdata(&pdev->dev, card); + + err = hda_tegra_first_init(chip, pdev); + if (err < 0) + goto out_free; + + /* create codec instances */ + err = azx_codec_create(chip, NULL, 0, &power_save); + if (err < 0) + goto out_free; + + err = azx_codec_configure(chip); + if (err < 0) + goto out_free; + + /* create PCM streams */ + err = snd_hda_build_pcms(chip->bus); + if (err < 0) + goto out_free; + + /* create mixer controls */ + err = azx_mixer_create(chip); + if (err < 0) + goto out_free; + + err = snd_card_register(chip->card); + if (err < 0) + goto out_free; + + chip->running = 1; + power_down_all_codecs(chip); + hda_tegra_notifier_register(chip); + + return 0; + +out_free: + snd_card_free(card); + return err; +} + +static int hda_tegra_remove(struct platform_device *pdev) +{ + return snd_card_free(dev_get_drvdata(&pdev->dev)); +} + +static struct platform_driver tegra_platform_hda = { + .driver = { + .name = "tegra-hda", + .pm = &hda_tegra_pm, + .of_match_table = hda_tegra_match, + }, + .probe = hda_tegra_probe, + .remove = hda_tegra_remove, +}; +module_platform_driver(tegra_platform_hda); + +MODULE_DESCRIPTION("Tegra HDA bus driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.3.605.g02339dd

3 3

[alsa-devel] [PATCH] ALSA: hda - Disable AA-loopback on ALC668
by Gabriele Mazzotta 20 May '14

20 May '14

Disable the AA-loopback path to get rid of the constant white noise that can be heard when headphones are used. This patch was tested only on a Dell XPS13 9333. Fix based on d2e92709e88d ("ALSA: hda - Disable AA-loopback on ALC283 Chromebook"). Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5f7c765..21c94e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5690,6 +5690,13 @@ static int patch_alc662(struct hda_codec *codec) if (has_cdefine_beep(codec)) spec->gen.beep_nid = 0x01; + switch (codec->vendor_id) { + case 0x10ec0668: + /* Disable AA-loopback as it causes white noise */ + spec->gen.mixer_nid = 0; + break; + } + if ((alc_get_coef0(codec) & (1 << 14)) && codec->bus->pci && codec->bus->pci->subsystem_vendor == 0x1025 && spec->cdefine.platform_type == 1) { -- 2.0.0.rc2

2 4

[alsa-devel] Force an update of the ELD info?
by VDR User 20 May '14

20 May '14

Hi. I was wondering if it's possible to force alsa/drivers to update the ELD info yet? If not, is this something that can be added? Reason being, htpc users may or may not have their receiver/surround sound on at any given time. The ELD info can be used to determine this, and thus audio playback can be appropriately set for surround/no-surround at any given time. As it is now, to even use surround, the receiver must be turned on before the system is booted. That may not always be convenient. Also, leaving a receiver on 24/7 is a waste of power, generates heat, and puts unnecessary wear & tear on equipment that's not being used at the time. I have been told one way to force an ELD update is to completely unload alsa/drivers, and reload them. However, that's very inconvenient as it requires plugging in a keyboard, or ssh'ing into the box, unloading everything, toggling the receivers power state, then reloading everything. That's a pretty disruptive process which, if can be avoided, should be. Also, could someone with the proper knowledge clarify is this is something that alsa needs to support, the sound drivers need to support, and/or the video drivers need to support? I've heard very different answers and would like some clear info on this if anyone has any to offer. In case anyone is wondering, I have multiple linux htpcs connected via hdmi to receivers, that are then connected to tvs. These boxes have nothing more than an IR receiver connected to them for navigating an OSD. All of the boxes are using Nvidia video cards with vdpau decoding, and audio-over-hdmi with the snd_hda_intel. Many thanks for any input! -DK

2 2

[alsa-devel] [RFC] A question regarding open ioctl for user space in ASoC driver.
by Nicolin Chen 20 May '14

20 May '14

Hi all, I have an ASoC driver in my own branch and I'm going to get it upstream in the months ahead. The driver is designed for Freescale ASRC module, Asynchronous Sample Rate Converter. It has the capability to convert audio data without involving software conversion like SRC function in ALSA-lib. But before I try to revise and send the driver, I've a few questions here. We currently have two main functions for ASRC: P2P's and M2M's. 1) P2P, peripheral to peripheral, regards ASRC as a FE (front end link) corresponding to a BE (back end link like ESAI<->CS42888). The P2P function resolves a problem of unsupported sample rates for BE that we can convert the unsupported audio format to a supported one before we send the data into BE. It also has better quality and less CPU loading comparing to ALSA-lib's SRC function. * For this function, I don't see any critical problem here. But... 2) M2M, memory to memory, can simply make ASRC as a sample rate converter without playback via any BE sound card, just like using any software to convert a WAV file in one sample rate to another WAV in a required sample rate. The driver has its self-designed application to compelte this function by sending the audio data from a wave file into a misc device via non-generic ioctl: static long asrc_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct asrc_pair_params *params = file->private_data; void __user *user = (void __user *)arg; long ret = 0; switch (cmd) { case ASRC_REQ_PAIR: ret = asrc_ioctl_req_pair(params, user); break; case ASRC_CONFIG_PAIR: ret = asrc_ioctl_config_pair(params, user); break; case ASRC_START_CONV: [...] So I'm wondering if this kind of ioctl would be allowed or there's any alternative approach? Not only having application in user space, but we also have an ASRC plugin for ALSA-lib so that it can involve M2M function by using the plugin before sending data via pcm_write(). And here's the another question -- to share the ioctl and the struct asrc_pair_params with user space, there's a header file need to be put into include/uapi/. Is it possible? There's a reference code (drivers/mxc/asrc/mxc_asrc.c) located in: http://git.freescale.com/git/cgit.cgi/imx/linux-2.6-imx.git/tree/?h=imx_3.1… In our internal branch, the ASRC driver currently is divided into two drivers for M2M and P2P separately. For upstream version, I'm trying to merge them into a single ASoC driver with both M2M and P2P, just not sure how to handle the M2M part appropriately. Thank you, Nicolin Chen

2 3

[alsa-devel] [PATCH 1/2] ASoC: samsung: s3c24{xx, 12}-i2s: port to use generic dmaengine API
by Vasily Khoruzhick 20 May '14

20 May '14

Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412 Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- 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 | 63 +++++++++++++++++++++--------------- sound/soc/samsung/s3c24xx-i2s.c | 72 +++++++++++++++++++++++------------------ 5 files changed, 84 insertions(+), 80 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..f672352 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,25 +33,17 @@ #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, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_out, + .ch_name = "tx", .dma_size = 4, }; static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_in, + .ch_name = "rx", .dma_size = 4, }; @@ -63,6 +55,9 @@ 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 +65,9 @@ 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 +88,6 @@ 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 +97,10 @@ 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 +153,33 @@ 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; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get DMA resource for playback.\n"); + return -ENOENT; + } + s3c2412_i2s_pcm_stereo_out.channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "Can't get DMA resource for capture.\n"); + return -ENOENT; + } + s3c2412_i2s_pcm_stereo_in.channel = res->start; 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..f854d45 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,25 +31,17 @@ #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, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_out, + .ch_name = "tx", .dma_size = 2, }; static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_in, + .ch_name = "rx", .dma_size = 2, }; @@ -231,18 +223,12 @@ 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 +237,11 @@ 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 +256,6 @@ 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 +274,6 @@ 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 +363,12 @@ 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 +455,33 @@ 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; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get DMA resource for playback.\n"); + return -ENOENT; + } + s3c24xx_i2s_pcm_stereo_out.channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "Can't get DMA resource for capture.\n"); + return -ENOENT; + } + s3c24xx_i2s_pcm_stereo_in.channel = res->start; ret = snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1); -- 1.9.3

2 5

[alsa-devel] [PATCHv5 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 20 May '14

20 May '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra12x HDA codec id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> Acked-by: Stephen Warren <swarren(a)nvidia.com> --- Changes since v4: Commit message only. --- 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 2a3050d..be0a9ee 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3320,6 +3320,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, @@ -3376,6 +3377,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); MODULE_ALIAS("snd-hda-codec-id:10de001b"); MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0028"); MODULE_ALIAS("snd-hda-codec-id:10de0040"); MODULE_ALIAS("snd-hda-codec-id:10de0041"); MODULE_ALIAS("snd-hda-codec-id:10de0042"); -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH] sound: soc: samsung: dma.c: Fix for possible null pointer dereference
by Rickard Strandqvist 20 May '14

20 May '14

There is otherwise a risk of a possible null pointer dereference. Was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/samsung/dma.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index dc09b71..1d9bcaa 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -115,17 +115,19 @@ static void dma_enqueue(struct snd_pcm_substream *substream) static void audio_buffdone(void *data) { struct snd_pcm_substream *substream = data; - struct runtime_data *prtd = substream->runtime->private_data; + struct runtime_data *prtd = NULL; pr_debug("Entered %s\n", __func__); - if (prtd->state & ST_RUNNING) { + if (substream) + prtd = substream->runtime->private_data; + + if (prtd && prtd->state & ST_RUNNING) { prtd->dma_pos += prtd->dma_period; if (prtd->dma_pos >= prtd->dma_end) prtd->dma_pos = prtd->dma_start; - if (substream) - snd_pcm_period_elapsed(substream); + snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); if (!samsung_dma_has_circular()) { -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix pcm stream context restore crash
by Jarkko Nikula 19 May '14

19 May '14

From: Kevin Strasser <kevin.strasser(a)intel.com> In some cases the pcm stream is closed while context has been scheduled to be restored, causing a null pointer deref panic. Cancel work to ensure stream does not get freed while work is still active/pending. Also, restoring the pcm context can be safely skipped after the stream has been stopped. Check if pcm stream is still running before restoring stream context to help pending work finish more quickly in stream close path. Signed-off-by: Kevin Strasser <kevin.strasser(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- For topic/intel branch since fixes issue in recently added suspend/resume code. --- sound/soc/intel/sst-baytrail-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 6242ccce2bb7..3af38576e91e 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -164,7 +164,8 @@ static void sst_byt_pcm_work(struct work_struct *work) struct sst_byt_pcm_data *pcm_data = container_of(work, struct sst_byt_pcm_data, work); - sst_byt_pcm_restore_stream_context(pcm_data->substream); + if (snd_pcm_running(pcm_data->substream)) + sst_byt_pcm_restore_stream_context(pcm_data->substream); } static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -277,6 +278,7 @@ static int sst_byt_pcm_close(struct snd_pcm_substream *substream) dev_dbg(rtd->dev, "PCM: close\n"); + cancel_work_sync(&pcm_data->work); mutex_lock(&pcm_data->mutex); ret = sst_byt_stream_free(byt, pcm_data->stream); if (ret < 0) { -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH RFT] ASoC: rt5645: Fix updating wrong register for T5645_AIF2 case
by Axel Lin 19 May '14

19 May '14

This looks like a copy-paste bug, fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have this h/w, so please test if this patch works. Axel sound/soc/codecs/rt5645.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 68923ec..ab97d72 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1932,8 +1932,8 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | RT5645_I2S_DF_MASK, reg_val); break; - case RT5645_AIF2: - snd_soc_update_bits(codec, RT5645_I2S1_SDP, + case RT5645_AIF2: + snd_soc_update_bits(codec, RT5645_I2S2_SDP, RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | RT5645_I2S_DF_MASK, reg_val); break; -- 1.8.3.2

3 2

[alsa-devel] [PATCH 1/2] ASoC: Remove needless snd_soc_dapm_sync() from machine driver inits
by Jarkko Nikula 19 May '14

19 May '14

ALSA SoC core takes care of calling snd_soc_dapm_sync() at the end snd_soc_instantiate_card() so there is no need to call it from machine driver init functions. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-rt5640.c | 1 - sound/soc/pxa/brownstone.c | 2 -- sound/soc/tegra/tegra_wm9712.c | 4 +--- 3 files changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index dbc63d09f424..eee001f70964 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -105,7 +105,6 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Internal Mic"); - snd_soc_dapm_sync(dapm); return ret; } diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 08acdc236bf8..9c128d4c8cbf 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -70,8 +70,6 @@ static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); snd_soc_dapm_nc_pin(dapm, "IN2LN"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 25a7f8211ecf..de087ee3458a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -50,9 +50,7 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); - - return snd_soc_dapm_sync(dapm); + return snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); } static struct snd_soc_dai_link tegra_wm9712_dai = { -- 2.0.0.rc2

2 2

[alsa-devel] [PATCH 1/2] ASoC: cs42xx8: Do not use rtd->codec
by Lars-Peter Clausen 19 May '14

19 May '14

rtd->codec does not necessarily point to the CODEC instance for which the callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use dai->codec instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs42xx8.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 8502032..a25bc60 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -248,8 +248,7 @@ static int cs42xx8_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_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 ratio = cs42xx8->sysclk / params_rate(params); -- 1.8.0

3 4

[alsa-devel] [PATCH] fm801: convert struct description to kernel-doc
by Andy Shevchenko 19 May '14

19 May '14

Just move field descriptions to the struct description in the kernel-doc format. There is no functional change. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/fm801.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index e8910d0..529f5f4 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -158,21 +158,27 @@ MODULE_PARM_DESC(radio_nr, "Radio device numbers"); #define FM801_GPIO_GS3 (1<<15) #define FM801_GPIO_GS(x) (1<<(12+(x))) -/* - +/** + * struct fm801 - describes FM801 chip + * @port: I/O port number + * @multichannel: multichannel support + * @secondary: secondary codec + * @secondary_addr: address of the secondary codec + * @tea575x_tuner: tuner access method & flags + * @ply_ctrl: playback control + * @cap_ctrl: capture control */ - struct fm801 { int irq; - unsigned long port; /* I/O port number */ - unsigned int multichannel: 1, /* multichannel support */ - secondary: 1; /* secondary codec */ - unsigned char secondary_addr; /* address of the secondary codec */ - unsigned int tea575x_tuner; /* tuner access method & flags */ + unsigned long port; + unsigned int multichannel: 1, + secondary: 1; + unsigned char secondary_addr; + unsigned int tea575x_tuner; - unsigned short ply_ctrl; /* playback control */ - unsigned short cap_ctrl; /* capture control */ + unsigned short ply_ctrl; + unsigned short cap_ctrl; unsigned long ply_buffer; unsigned int ply_buf; -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH V2] ASoC: SAMSUNG: Add sound card driver for Snow board
by Tushar Behera 19 May '14

19 May '14

Added machine driver to instantiate I2S based sound card on Snow board. It has MAX98095 audio codec on board. There are some other variants for Snow board which have MAX98090 audio codec. Hence support for MAX98090 is also added to this driver. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- Changes for V2: Addressed various comments from Mark Brown. 1. Moved DAIFMT flags to dai_link. Removed snd_soc_dai_set_fmt calls. 2. Removed BCLK settings. 3. Moved sysclk settings to late_probe. 4. Removed now empty snow_hw_params. 5. Removed snow_init 6. Used devm_snd_soc_register_card. Also removed empty remove function. 7. Removed non-DT handling as the driver is DT only. Additional nits: 1. Renamed snow_driver_probe to snow_probe 2. Removed unused pm_ops. 3. Updated clock settings for I2S block (Earlier it was I2S_CDCLK, whereas it should be I2S_RCLKSRC_0). 4. Removed 'card-name' DT property as it was not used in earlier sound-card driver DT bindings. Will be added later. Documentation/devicetree/bindings/sound/snow.txt | 17 +++ sound/soc/samsung/Kconfig | 10 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/snow.c | 129 ++++++++++++++++++++++ 4 files changed, 158 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/snow.txt create mode 100644 sound/soc/samsung/snow.c diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt new file mode 100644 index 0000000..6f3fd02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -0,0 +1,17 @@ +Audio Binding for Snow boards + +Required properties: +- compatible : Can be one of the following, + "google,snow-audio-max98090" or + "google,snow-audio-max98095" +- samsung,i2s-controller: The phandle of the Samsung I2S0 controller +- samsung,audio-codec: The phandle of the audio codec + +Example: + +sound { + compatible = "google,snow-audio-max98095"; + + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98095>; +}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f2e2891..50aa289 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -231,3 +231,13 @@ config SND_SOC_LITTLEMILL select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 + +config SND_SOC_SNOW + tristate "Audio support for Google Snow boards" + depends on SND_SOC_SAMSUNG + select SND_SOC_MAX98090 + select SND_SOC_MAX98095 + select SND_SAMSUNG_I2S + help + Say Y if you want to add audio support for various Snow + boards based on Exynos5 series of SoCs. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 86715d8..6d0212b 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -34,6 +34,7 @@ snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o snd-soc-smdk-wm8580-objs := smdk_wm8580.o snd-soc-smdk-wm8994-objs := smdk_wm8994.o +snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o @@ -58,6 +59,7 @@ obj-$(CONFIG_SND_SOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_SOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8994) += snd-soc-smdk-wm8994.o +obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c new file mode 100644 index 0000000..e1a4b45 --- /dev/null +++ b/sound/soc/samsung/snow.c @@ -0,0 +1,129 @@ +/* + * ASoC machine driver for Snow boards + * + * 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 <linux/of.h> +#include <linux/of_device.h> + +#include <sound/soc.h> + +#include "i2s.h" + +#define FIN_PLL_RATE 24000000 + +static struct snd_soc_dai_link snow_dai[] = { + { /* Playback DAI i/f */ + .name = "Playback", + .stream_name = "Playback", + .codec_dai_name = "HiFi", + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, { /* Capture DAI i/f */ + .name = "Capture", + .stream_name = "Capture", + .codec_dai_name = "HiFi", + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static int snow_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; + int ret; + + /* Set the MCLK rate for the codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + FIN_PLL_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Select I2S Bus clock to set RCLK and BCLK */ + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_card snow_snd = { + .name = "Snow-I2S", + .dai_link = snow_dai, + .num_links = ARRAY_SIZE(snow_dai), + + .late_probe = snow_late_probe, +}; + +static int snow_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snow_snd; + struct device_node *i2s_node, *codec_node; + int i, ret; + + i2s_node = of_parse_phandle(pdev->dev.of_node, + "samsung,i2s-controller", 0); + if (!i2s_node) { + dev_err(&pdev->dev, + "Property 'i2s-controller' missing or invalid\n"); + return -EINVAL; + } + + codec_node = of_parse_phandle(pdev->dev.of_node, + "samsung,audio-codec", 0); + if (!codec_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(snow_dai); i++) { + snow_dai[i].codec_of_node = codec_node; + snow_dai[i].cpu_of_node = i2s_node; + snow_dai[i].platform_of_node = i2s_node; + } + + card->dev = &pdev->dev; + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + return ret; + } + + return ret; +} + +static const struct of_device_id snow_of_match[] = { + { .compatible = "google,snow-audio-max98090", }, + { .compatible = "google,snow-audio-max98095", }, + {}, +}; + +static struct platform_driver snow_driver = { + .driver = { + .name = "snow-audio", + .owner = THIS_MODULE, + .of_match_table = snow_of_match, + }, + .probe = snow_probe, +}; + +module_platform_driver(snow_driver); + +MODULE_DESCRIPTION("ALSA SoC Audio machine driver for Snow"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

3 13

[alsa-devel] [PATCH] ALSA: pcm_dmaengine: Add check during device suspend
by Tushar Behera 19 May '14

19 May '14

Currently snd_dmaengine_pcm_trigger() calls dmaengine_pause() unconditinally during device suspend. In case where DMA controller doesn't support PAUSE/RESUME functionality, this call is not able to stop the DMA controller. In this scenario, audio playback doesn't resume after device resume. Calling dmaengine_pause/dmaengine_terminate_all conditionally fixes the issue. It has been tested with audio playback on Samsung platform having PL330 DMA controller which doesn't support PAUSE/RESUME. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- sound/core/pcm_dmaengine.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 94d0873..76cbb9e 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -182,6 +182,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; int ret; switch (cmd) { @@ -196,6 +197,11 @@ int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) dmaengine_resume(prtd->dma_chan); break; case SNDRV_PCM_TRIGGER_SUSPEND: + if (runtime->info & SNDRV_PCM_INFO_PAUSE) + dmaengine_pause(prtd->dma_chan); + else + dmaengine_terminate_all(prtd->dma_chan); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dmaengine_pause(prtd->dma_chan); break; -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH] dma: pl330: Add support for DMA_PAUSE command
by Lars-Peter Clausen 19 May '14

19 May '14

On 05/15/2014 02:01 PM, Tushar Behera wrote: > On 14 May 2014 17:54, Lars-Peter Clausen <lars(a)metafoo.de> wrote: >> On 05/14/2014 02:07 PM, Tushar Behera wrote: >>> >>> On 14 May 2014 17:29, Jassi Brar <jassisinghbrar(a)gmail.com> wrote: >>>> >>>> On Wed, May 14, 2014 at 8:53 AM, Tushar Behera <tushar.behera(a)linaro.org> >>>> wrote: >>>>> >>>>> While playing back audio, pmc_dmaengine requests the DMA channel to >>>>> stop DMA transmission through DMA_PAUSE command. >>>>> >>>>> Currently PL330 driver doesn't support DMA pause command, leaving >>>>> the DMA state inconsistent when the system resumes. Instead, it would >>>>> be better to terminate the DMA transfer during suspend and restart >>>>> again during resume. >>>>> >>>>> Tested with audio playback across a suspend-resume cycle. >>>>> >>>> What is pmc_dmaengine? How does DMA_PAUSE help, when there is no >>>> DMA_RESUME? >>>> >>> >>> Sorry, it is a typo. >>> >>> sound/core/pcm_dmaengine.c:snd_dmaengine_pcm_trigger() --> >>> dmaengine_pause() is called during system suspend. >> >> >> It is only called if the DMA driver has support for pausing and resuming DMA >> transfers. Or at least that is the intention. >> >> - Lars > > During suspend, snd_dmaengine_pcm_trigger():SNDRV_PCM_TRIGGER_SUSPEND > is called which unconditionally calls dmaengine_pause(). Should we > update snd_dmaengine_pcm_trigger() to check for DMA pause/resume > support and call dmaengine_pause() or dmaengine_terminate_all() > accordingly? As far as I understand it we do not have to do anything for TRIGGER_SUSPEND if we do not set the SNDRV_PCM_INFO_RESUME flag. It looks like TRIGGER_SUSPEND is called unconditionally during suspend. But since the error code is ignored it should be fine if we just call dmaengine_pause() and that return -ENOSYS or similar. Are you seeing an actual issue that you are trying to fix with your patch? - Lars

3 6

[alsa-devel] [PATCH 0/13] make return of 0 explicit
by Julia Lawall 19 May '14

19 May '14

Sometimes a local variable is used as a return value in a case where the return value is always 0. The result is more apparent if this variable is just replaced by 0. This is done by the following semantic patch, although some cleanups may be needed. (http://coccinelle.lip6.fr/) // <smpl> @r exists@ local idexpression ret; identifier reti; expression e; position p; @@ ret@reti = 0; ... when != $ret = e\|ret &= e\|ret |= e\|ret += e\|ret -= e\|ret *= e\|ret ^= e$ when != $++ret\|--ret\|ret++\|ret--$ when != &ret return ret;@p @bad1 exists@ expression e != 0; local idexpression r.ret; position r.p; @@ $ret = e\|ret &= e\|ret |= e\|ret += e\|ret -= e\|ret *= e\|ret ^= e\|++ret\|--ret\|ret++\|ret--\|&ret$ ... return ret;@p @bad2 exists@ identifier r.reti; position r.p; identifier f; type T; @@ f(...,T reti,...) { ... return reti;@p } @change depends on !bad1 && !bad2@ position r.p; local idexpression r.ret; identifier f; @@ f(...) { <+... return -ret +0 ;@p ...+> } @rewrite@ local idexpression r.ret; expression e; position p; identifier change.f; @@ f(...) { <+... $ret@p = e\|&ret@p$ ...+> } @depends on change@ local idexpression r.ret; position p != rewrite.p; identifier change.f; @@ f(...) { <+... ( break; | ret = 0; ... when exists ret@p ... when any | - ret = 0; ) ...+> } @depends on change@ identifier r.reti; type T; constant C; identifier change.f; @@ f(...) { ... when any -T reti = C; ... when != reti when strict } @depends on change@ identifier r.reti; type T; identifier change.f; @@ f(...) { ... when any -T reti; ... when != reti when strict } // </smpl> The first four rules detect cases where only 0 reaches a return. The remaining rules clean up any variable initializations or declarations that are made unnecessary by this change.

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[alsa-devel] [PATCH] ASoC: Intel: Add Baytrail byt-max98090 machine driver
by Jarkko Nikula 19 May '14

19 May '14

Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec. Jack detect code from Kevin Strasser <kevin.strasser(a)intel.com>, GPIO resolving from Mika Westerberg <mika.westerberg(a)linux.intel.com> and fixes and cleanups from Liam Girdwood <liam.r.girdwood(a)linux.intel.com>. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/Kconfig | 10 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/byt-max98090.c | 252 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-acpi.c | 1 + 4 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/byt-max98090.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 3c81b3891209..d6c3d840ed82 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -49,3 +49,13 @@ config SND_SOC_INTEL_BYT_RT5640_MACH help This adds audio driver for Intel Baytrail platform based boards with the RT5640 audio codec. + +config SND_SOC_INTEL_BYT_MAX98090_MACH + tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_BAYTRAIL + select SND_SOC_MAX98090 + select VLV2_PLAT_CLK + help + This adds audio driver for Intel Baytrail platform based boards + with the MAX98090 audio codec. diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 0db4e2f336dc..4bfca79a42ba 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -23,6 +23,8 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o # Machine support snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o +snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o obj-$(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 diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c new file mode 100644 index 000000000000..4246b2c81201 --- /dev/null +++ b/sound/soc/intel/byt-max98090.c @@ -0,0 +1,252 @@ +/* + * Intel Baytrail SST MAX98090 machine driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/max98090.h" + +struct byt_max98090_private { + struct snd_soc_jack jack; +}; + +static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { + {"IN34", NULL, "Headset Mic"}, + {"IN34", NULL, "MICBIAS"}, + {"MICBIAS", NULL, "Headset Mic"}, + {"DMICL", NULL, "Int Mic"}, + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKL"}, + {"Ext Spk", NULL, "SPKR"}, +}; + +static const struct snd_kcontrol_new byt_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Ext Spk", + .mask = SND_JACK_LINEOUT, + }, + { + .pin = "Int Mic", + .mask = SND_JACK_LINEIN, + }, +}; + +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, + .debounce_time = 200, + }, + { + .name = "mic-gpio", + .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, + .debounce_time = 200, + }, +}; + +static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = runtime->card; + struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); + struct snd_soc_jack *jack = &drv->jack; + struct gpio_desc *mic_desc; + struct gpio_desc *hp_desc; + + card->dapm.idle_bias_off = true; + + ret = snd_soc_dai_set_sysclk(runtime->codec_dai, + M98090_REG_SYSTEM_CLOCK, + 25000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Headphone"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Int Mic"); + + snd_soc_dapm_sync(dapm); + + /* + * ASoC still uses legacy GPIOs so we look both GPIOs using + * descriptors here, convert them to numbers and release the + * acquired descriptors. Once ASoC switches over to GPIO descriptor + * APIs we can pass them directly. + */ + hp_desc = gpiod_get_index(card->dev->parent, NULL, 0); + if (IS_ERR(hp_desc)) + return 0; + mic_desc = gpiod_get_index(card->dev->parent, NULL, 1); + if (IS_ERR(mic_desc)) { + gpiod_put(hp_desc); + return 0; + } + + hs_jack_gpios[0].gpio = desc_to_gpio(hp_desc); + hs_jack_gpios[1].gpio = desc_to_gpio(mic_desc); + + gpiod_put(mic_desc); + gpiod_put(hp_desc); + + /* Enable jack detection */ + ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, M98090_IJDET_MASK, + 1 << M98090_IJDET_SHIFT); + + snd_soc_jack_report(jack, SND_JACK_LINEOUT | SND_JACK_LINEIN, + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_LINEIN); + + ret = snd_soc_jack_add_gpios(jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return ret; +} + +static struct snd_soc_dai_link byt_max98090_dais[] = { + { + .name = "Baytrail Audio", + .stream_name = "Audio", + .cpu_dai_name = "baytrail-pcm-audio", + .codec_dai_name = "HiFi", + .codec_name = "i2c-193C9890:00", + .platform_name = "baytrail-pcm-audio", + .init = byt_max98090_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card byt_max98090_card = { + .name = "byt-max98090", + .dai_link = byt_max98090_dais, + .num_links = ARRAY_SIZE(byt_max98090_dais), + .dapm_widgets = byt_max98090_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_max98090_widgets), + .dapm_routes = byt_max98090_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), + .controls = byt_max98090_controls, + .num_controls = ARRAY_SIZE(byt_max98090_controls), +}; + +#ifdef CONFIG_PM_SLEEP +static const struct dev_pm_ops byt_max98090_pm_ops = { + .suspend = snd_soc_suspend, + .resume = snd_soc_resume, +}; + +#define BYT_MAX98090_PM_OPS (&byt_max98090_pm_ops) +#else +#define BYT_MAX98090_PM_OPS NULL +#endif + +static int byt_max98090_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct byt_max98090_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) { + dev_err(&pdev->dev, "allocation failed\n"); + return -ENOMEM; + } + + /* register the soc card */ + byt_max98090_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&byt_max98090_card, drv); + ret_val = snd_soc_register_card(&byt_max98090_card); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &byt_max98090_card); + + return ret_val; +} + +static int byt_max98090_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + struct byt_max98090_private *drv = snd_soc_card_get_drvdata(soc_card); + + snd_soc_jack_free_gpios(&drv->jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver byt_max98090_driver = { + .probe = byt_max98090_probe, + .remove = byt_max98090_remove, + .driver = { + .name = "byt-max98090", + .owner = THIS_MODULE, + .pm = BYT_MAX98090_PM_OPS, + }, +}; +module_platform_driver(byt_max98090_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); +MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:byt-max98090"); diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 18aee77f8d4a..42edc6f4fc4a 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -247,6 +247,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { static struct sst_acpi_mach baytrail_machines[] = { { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" }, {} }; -- 2.0.0.rc0

2 2

[alsa-devel] This patch is for hearing sound separate on channels
by asd 18 May '14

18 May '14

--- a/hda_codec.c +++ b/hda_codec.c @@ -5375,6 +5375,7 @@ int chs = substream->runtime->channels; struct hda_spdif_out *spdif; int i; + //mout->no_share_stream = 1; mout->no_share_stream = 1; mutex_lock(&codec->spdif_mutex);

1 0

[alsa-devel] [BUG]ABit aa8xe sound card not selected
by Stefano Statuti 18 May '14

18 May '14

hello, I have opened this ticket on ubuntu-alsa launchpad _https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1319291 Raymond (superquad-vortrex2) ask me to contact you. ----quote Try specify model=6stack-dig https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… 00:1b.0 0403: 8086:2668 (rev 04) Subsystem: 147b:1045 send email to alsa-devel mailing list to add pci quirk since Bios does not setup pin default Default 0x00000000: [Jack] Line Out at Ext N/A Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x00000000 0x15 0x00000000 0x16 0x00000000 0x17 0x00000000 0x18 0x00000000 0x19 0x00000000 0x1a 0x00000000 0x1b 0x00000000 0x1c 0x00000000 0x1d 0x00000000 0x1e 0x00000000 0x1f 0x00000000 ----- whit Ubuntu 12.04 I don't have this problem bye

2 1

[alsa-devel] Segfaults in snd_pcm_area_silence() when using softvol
by Tanu Kaskinen 18 May '14

18 May '14

Hi, A bug was reported to pulseaudio about a segfault in alsa-lib code when using softvol: https://bugs.freedesktop.org/show_bug.cgi?id=51060 . It has been observed by several people, and this mail was prompted by the last bug comment, which rightly recommends getting in touch with the alsa developers: > I've had this problem too and the problem only occurs with > -ftree-vectorize. > > It looks to me as if this is a gcc bug, as it produces a "movdqa" or > "vmovdqa" instruction which is used with an unaligned pointer. > > Now, I have no clue why gcc assumes the pointer would be an aligned > one, maybe it is even right and something else is wrong, but I haven't > spent too much time digging into this. > > (note that it still happens with gcc 4.7 and the current 4.8 > development one) > > Also, this bug should probably be brought up with the alsa guys as > well, as I don't think pulseaudio is at fault here. Here's the stack trace for your convenience (see the linked bug for more details) #0 snd_pcm_area_silence (dst_area=0x7fffe875bb00, dst_offset=<optimized out>, samples=2, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2549 #1 0x00007fffe87b4398 in snd_pcm_areas_silence (dst_areas=0x5555557fa100, dst_offset=891, channels=2, frames=439, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2662 #2 0x00007fffe87f9bdb in softvol_convert_stereo_vol (svol=0x555555813cf0, dst_areas=0x5555557fa0e0, dst_offset=891, src_areas=0x5555557fa0e0, src_offset=<optimized out>, channels=2, frames=439) at pcm_softvol.c:286 #3 0x00007fffe87fae58 in snd_pcm_softvol_write_areas (pcm=0x55555581db60, areas=0x5555557fa0e0, offset=891, size=439, slave_areas=0x5555557fa0e0, slave_offset=891, slave_sizep=0x7fffe875bcd0) at pcm_softvol.c:607 #4 0x00007fffe87c6fc7 in snd_pcm_plugin_mmap_commit (pcm=0x55555581db60, offset=<optimized out>, size=439) at pcm_plugin.c:424 #5 0x00007fffe87b9676 in snd_pcm_mmap_commit (pcm=<optimized out>, offset=<optimized out>, frames=<optimized out>) at pcm.c:6584 #6 0x00007fffe8a74876 in mmap_write (on_timeout=false, polled=false, sleep_usec=<synthetic pointer>, u=0x5555557e6d40) at modules/alsa/alsa-sink.c:636 #7 thread_func (userdata=0x5555557e6d40) at modules/alsa/alsa-sink.c:1694 #8 0x00007ffff722c039 in internal_thread_func (userdata=0x55555587ef80) at pulsecore/thread-posix.c:83 #9 0x00007ffff4a24e2c in ?? () from /lib64/libpthread.so.0 #10 0x00007ffff3ccc1dd in clone () from /lib64/libc.so.6 -- Tanu

2 1

[alsa-devel] [PATCH] ALSA: ASoC: Fix WM9713 mic 2 preamp volume
by Matt Reimer 17 May '14

17 May '14

Fix WM9713's mic 2 preamp volume. The previous definition specified the Mic A pre-amp input select bits (aka "Mic A Source") instead. Signed-off-by: Matt Reimer <mreimer(a)sdgsystems.com> --- sound/soc/codecs/wm9713.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index acea892..5f69b52 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -134,7 +134,7 @@ SOC_DOUBLE_TLV("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 1 Preamp Volume", AC97_3D_CONTROL, 10, 3, 0, mic_tlv), -SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 12, 3, 0, mic_tlv), +SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 8, 3, 0, mic_tlv), SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), -- 1.7.5.4

2 1

[alsa-devel] [PATCH 0/7] ASoC: max98090: Couple improvements and ACPI probing support
by Jarkko Nikula 16 May '14

16 May '14

Here are couple improvements to MAX98090 codec driver. I think only 1/7 is a real fix and rest 2-7/7 are regular updates that make codec resume working and add ACPI probing support on Intel Baytrail platform. Jarkko Nikula (2): ASoC: max98090: Move microphone bias voltage setting to probe function ASoC: max98090: Add ACPI probing support Liam Girdwood (5): ASoC: max98090: Fix digital sidetone gain TLV ASoC: max98090: Fix reset at resume time. ASoC: max98090: Reset codec on resume ASoC: max98090: Mark cache as dirty prior to restoring ASoC: max98090: Make sure we configure BCLK in one place sound/soc/codecs/max98090.c | 77 +++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98090.h | 1 + 2 files changed, 65 insertions(+), 13 deletions(-) -- 2.0.0.rc0

2 8

[alsa-devel] [PATCH v3 0/3] ASoC: Intel: rework the mfld platform driver
by Vinod Koul 16 May '14

16 May '14

This is the remaining bits of the v2 which were removed last time around. One patch was wrongly applied so reverting that and adding two other missed in the series. Also added Liam's ack Vinod Koul (3): Revert "ASoC: Intel: rename pcm dias to media dai" ASoC: Intel: add drain_notify support ASoC: Intel: remove codec memeber from codec structs sound/soc/intel/sst-mfld-dsp.h | 4 -- sound/soc/intel/sst-mfld-platform-compress.c | 13 +++++++- sound/soc/intel/sst-mfld-platform-pcm.c | 39 ++++++++++++------------- sound/soc/intel/sst-mfld-platform.h | 2 + 4 files changed, 32 insertions(+), 26 deletions(-)

2 4

[alsa-devel] [PATCH 1/2] ALSA: hda - Add fixup_forced flag
by David Henningsson 16 May '14

16 May '14

The "fixup_forced" flag will indicate whether a specific fixup (or nofixup) has been set by the user, to override the driver's default. This flag will help future patches. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 9 ++++++--- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 90d2fda..36961ab 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -852,15 +852,17 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { codec->fixup_list = NULL; codec->fixup_id = -1; + codec->fixup_forced = 1; return; } if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { - id = models->id; - name = models->name; - break; + codec->fixup_id = models->id; + codec->fixup_name = models->name; + codec->fixup_forced = 1; + return; } models++; } @@ -889,6 +891,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } + codec->fixup_forced = 0; codec->fixup_id = id; if (id >= 0) { codec->fixup_list = fixlist; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a423313..5825aa1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -402,6 +402,7 @@ struct hda_codec { /* fix-up list */ int fixup_id; + unsigned int fixup_forced:1; /* fixup explicitly set by user */ const struct hda_fixup *fixup_list; const char *fixup_name; -- 1.9.1

1 1

[alsa-devel] poor i4600 HDMI output under 1.0.27.2
by Dan MacDonald 15 May '14

15 May '14

Hi list I have got HDMI audio out to kinda work but the sound quality is awful and not really listenable. Its very choppy and is slightly more high-pitched than it should be. This is when using hdmi:CARD=HDMI,DEV=1 @48khz outputting via ALSA 1.0.27.2 / jackd 0.124.2 under Deb Jessie amd64 with kernel 3.13.10-1. Its a System76 Gazelle Pro laptop FWIW. $ lspci -v: 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) Subsystem: CLEVO/KAPOK Computer Device 0655 Flags: bus master, fast devsel, latency 0, IRQ 49 Memory at f7e14000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05) Subsystem: CLEVO/KAPOK Computer Device 0655 Flags: bus master, fast devsel, latency 0, IRQ 48 Memory at f7e10000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel $ cat /proc/asound/card0/codec#0 Codec: Intel Haswell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862807 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=02, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=03, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono

2 2

[alsa-devel] [PATCH] ak4642 codec updates
by Sascha Hauer 15 May '14

15 May '14

This series has some updates to the ak4642 codec driver like ALC controls, improved sysclk support for the ak4648. Sascha ---------------------------------------------------------------- Sascha Hauer (6): ASoC: ak4642: Fix typo zoro -> zero ASoC: ak4642: Add ALC controls ASoC: ak4642: Add driver data and driver private struct ASoC: ak4642: Add support for extended sysclk frequencies of the ak4648 ASoC: ak4642: Implement Microphone in pathes ASoC: ak4642: Add enable gpio support Documentation/devicetree/bindings/sound/ak4642.txt | 3 + sound/soc/codecs/ak4642.c | 230 +++++++++++++++++++-- 2 files changed, 221 insertions(+), 12 deletions(-)

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[alsa-devel] [PATCH] ASoC: rsnd: Fix warnings due to improper printk formats
by Laurent Pinchart 15 May '14

15 May '14

Use the %pap printk specifier to print resource_size_t variables. This fixes warnings on platforms where resource_size_t has a different size than int. Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas(a)ideasonboard.com> --- sound/soc/sh/rcar/gen.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) Would it be possible to get this patch in v3.16 ? diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 50a1ef3..1d5c025 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -252,13 +252,13 @@ static int rsnd_gen2_probe(struct platform_device *pdev, return ret; dev_dbg(dev, "Gen2 device probed\n"); - dev_dbg(dev, "SCU : %08x => %p\n", scu_res->start, + dev_dbg(dev, "SCU : %pap => %p\n", &scu_res->start, gen->base[RSND_GEN2_SCU]); - dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, gen->base[RSND_GEN2_ADG]); - dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start, + dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start, gen->base[RSND_GEN2_SSIU]); - dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, gen->base[RSND_GEN2_SSI]); return 0; @@ -345,11 +345,11 @@ static int rsnd_gen1_probe(struct platform_device *pdev, return ret; dev_dbg(dev, "Gen1 device probed\n"); - dev_dbg(dev, "SRU : %08x => %p\n", sru_res->start, + dev_dbg(dev, "SRU : %pap => %p\n", &sru_res->start, gen->base[RSND_GEN1_SRU]); - dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, gen->base[RSND_GEN1_ADG]); - dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, gen->base[RSND_GEN1_SSI]); return 0; -- Regards, Laurent Pinchart

5 5

[alsa-devel] [PATCH] ASoC: Intel: Fix simultaneous Baytrail SST capture and playback
by Jarkko Nikula 15 May '14

15 May '14

I managed to drop a change to stream ID setting from commit 49fee1781634 ("ASoC: Intel: Only export one Baytrail DAI") leading to non-working simultaneous capture-playback since after one DAI conversion rtd->cpu_dai->id + 1 will be the same for both playback and capture. Use substream->stream + 1 like it was in original Liam's patch. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 23a25c0627bd..6242ccce2bb7 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -254,7 +254,7 @@ static int sst_byt_pcm_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &sst_byt_pcm_hardware); - pcm_data->stream = sst_byt_stream_new(byt, rtd->cpu_dai->id + 1, + pcm_data->stream = sst_byt_stream_new(byt, substream->stream + 1, byt_notify_pointer, pcm_data); if (pcm_data->stream == NULL) { dev_err(rtd->dev, "failed to create stream\n"); -- 2.0.0.rc0

1 0

[alsa-devel] [Resend] Add the stream report feature. Could you review my idea?
by JongHo Kim 15 May '14

15 May '14

(Add the alsa-devel(a)alsa-project.org) Hi I am work on ASoC part and developing the smartphone(Android). we have been doing the stress and long runtime test. audio output is no problem but I sometimes face the "playback write error(DMA or IRQ trouble?)" error message. I know the meaning of the message and I can solve the error. the problem may be caused by AP(clk) or I2S or DMA or Codec. I would like to trace the hardware error at the printed message. Could you review my idea? Thanks --- include/sound/pcm.h | 2 ++ include/sound/soc-dai.h | 2 ++ include/sound/soc.h | 1 + sound/core/pcm_lib.c | 3 +++ sound/core/pcm_native.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 26 +++++++++++++++++++++++++ 6 files changed, 85 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index b4d6697..cd3093a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -82,6 +82,7 @@ struct snd_pcm_ops { unsigned long offset); int (*mmap)(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int (*ack)(struct snd_pcm_substream *substream); + void (*report)(struct snd_pcm_substream *substream); }; /* @@ -502,6 +503,7 @@ int snd_pcm_status(struct snd_pcm_substream *substream, int snd_pcm_start(struct snd_pcm_substream *substream); int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status); int snd_pcm_drain_done(struct snd_pcm_substream *substream); +void snd_pcm_report(struct snd_pcm_substream *substream); #ifdef CONFIG_PM int snd_pcm_suspend(struct snd_pcm_substream *substream); int snd_pcm_suspend_all(struct snd_pcm *pcm); diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index fad7676..c032393 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -184,6 +184,8 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*bespoke_trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); + void (*report)(struct snd_pcm_substream *, + struct snd_soc_dai *); /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..7685839 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -641,6 +641,7 @@ struct snd_soc_ops { int (*hw_free)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); int (*trigger)(struct snd_pcm_substream *, int); + void (*report)(struct snd_pcm_substream *); }; struct snd_soc_compr_ops { diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ce83def..0d6c9a3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1944,6 +1944,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream, "%s write error (DMA or IRQ trouble?)\n", is_playback ? "playback" : "capture"); err = -EIO; + snd_pcm_stream_unlock_irq(substream); + snd_pcm_report(substream); + snd_pcm_stream_lock_irq(substream); break; } } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b653ab0..0ccdc10 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -978,6 +978,57 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) SNDRV_PCM_STATE_SETUP); } +void snd_pcm_report(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + if (!runtime) { + snd_printd("runtime disappear\n"); + return; + } + + mutex_lock(&substream->pcm->open_mutex); + snd_printd("access: %i\n", runtime->access); + snd_printd("format: %i\n", runtime->format); + snd_printd("subformat: %d\n", runtime->subformat); + snd_printd("channels: %u\n", runtime->channels); + snd_printd("rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, + runtime->rate_den); + snd_printd("period_size: %lu\n", runtime->period_size); + snd_printd("buffer_size: %lu\n", runtime->buffer_size); + snd_printd("tstamp_mode: %i\n", runtime->tstamp_mode); + snd_printd("period_step: %u\n", runtime->period_step); + snd_printd("avail_min: %lu\n", runtime->control->avail_min); + snd_printd("start_threshold: %lu\n", runtime->start_threshold); + snd_printd("stop_threshold: %lu\n", runtime->stop_threshold); + snd_printd("silence_threshold: %lu\n", runtime->silence_threshold); + snd_printd("silence_size: %lu\n", runtime->silence_size); + snd_printd("boundary: %lu\n", runtime->boundary); + snd_printd("state: %i\n", runtime->status->state); + snd_printd("owner_pid: %d\n", pid_vnr(substream->pid)); + snd_printd("trigger_time: %ld.%09ld\n", + runtime->trigger_tstamp.tv_sec, + runtime->trigger_tstamp.tv_nsec); + snd_printd("tstamp: %ld.%09ld\n", + runtime->status->tstamp.tv_sec, + runtime->status->tstamp.tv_nsec); + snd_printd("delay: %ld\n", runtime->delay); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_printd("avail: %ld\n", snd_pcm_playback_avail(runtime)); + else + snd_printd("avail: %ld\n", snd_pcm_capture_avail(runtime)); + snd_printd("avail_max: %ld\n", runtime->avail_max); + snd_printd("hw_ptr: %ld\n", runtime->status->hw_ptr); + snd_printd("appl_ptr: %ld\n", runtime->control->appl_ptr); + + mutex_unlock(&substream->pcm->open_mutex); + + if (substream->ops->report) + substream->ops->report(substream); + + return; +} + /* * pause callbacks */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2cedf09..98a47b3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -883,6 +883,30 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } +static void soc_pcm_report(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + /* report the audio subsystem */ + if (rtd->dai_link->ops && rtd->dai_link->ops->report) + rtd->dai_link->ops->report(substream); + + if (cpu_dai->driver->ops->report) + cpu_dai->driver->ops->report(substream, cpu_dai); + + if (platform->driver->ops && platform->driver->ops->report) + platform->driver->ops->report(substream); + + if (codec_dai->driver->ops->report) + codec_dai->driver->ops->report(substream, codec_dai); + + mutex_unlock(&rtd->pcm_mutex); +} + /* connect a FE and BE */ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) @@ -2267,6 +2291,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->ops.close = dpcm_fe_dai_close; rtd->ops.pointer = soc_pcm_pointer; rtd->ops.ioctl = soc_pcm_ioctl; + rtd->ops.report = soc_pcm_report; } else { rtd->ops.open = soc_pcm_open; rtd->ops.hw_params = soc_pcm_hw_params; @@ -2276,6 +2301,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->ops.close = soc_pcm_close; rtd->ops.pointer = soc_pcm_pointer; rtd->ops.ioctl = soc_pcm_ioctl; + rtd->ops.report = soc_pcm_report; } if (platform->driver->ops) { -- 1.8.1.2

3 2

[alsa-devel] [PATCH 0/2] ASoC: Intel: Export Baytrail PCM streams under one DAI
by Jarkko Nikula 14 May '14

14 May '14

It wasn't necessary to export Baytrail SST PCM streams under separate DAI links and complicated needlessly basic ALSA command usage (for instance arecord). This set exports PCM playback/capture streams under one DAI. Change should be safe now since known users are already converted and to my understanding no other distribution is using this yet (mainly due missing firmware). Liam Girdwood (2): ASoC: Intel: Make Baytrail PCM data per stream rather than per DAI device ASoC: Intel: Only export one Baytrail DAI sound/soc/intel/byt-rt5640.c | 14 +------------- sound/soc/intel/sst-baytrail-pcm.c | 23 ++++++++++------------- 2 files changed, 11 insertions(+), 26 deletions(-) -- 2.0.0.rc0

2 3

[alsa-devel] [patch] ASoC: compress: indent an if statement
by Dan Carpenter 14 May '14

14 May '14

The return statement was not indented correctly. I lined up the condition a bit as well. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 91083e6..e4c7ad5 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -317,8 +317,9 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) cmd == SND_COMPR_TRIGGER_DRAIN) { if (platform->driver->compr_ops && - platform->driver->compr_ops->trigger) - return platform->driver->compr_ops->trigger(cstream, cmd); + platform->driver->compr_ops->trigger) + return platform->driver->compr_ops->trigger(cstream, + cmd); } if (cstream->direction == SND_COMPRESS_PLAYBACK)

3 3

[alsa-devel] [patch] ALSA: hda - if statement not indented
by Dan Carpenter 14 May '14

14 May '14

The "break;" should be indented. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1613388..589e47c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3722,7 +3722,7 @@ static void parse_digital(struct hda_codec *codec) } else { spec->multiout.slave_dig_outs = spec->slave_dig_outs; if (nums >= ARRAY_SIZE(spec->slave_dig_outs) - 1) - break; + break; spec->slave_dig_outs[nums - 1] = dig_nid; } nums++;

2 1

[alsa-devel] [patch] ALSA: sb_mixer: missing return statement
by Dan Carpenter 14 May '14

14 May '14

The if condition here was supposed to return on error but the return statement is missing. The effect is that the ->mixername is set to "???" instead of "DT019X". Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 6496822..1ff78ec 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -818,12 +818,14 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_DT019X: - if ((err = snd_sbmixer_init(chip, - snd_dt019x_controls, - ARRAY_SIZE(snd_dt019x_controls), - snd_dt019x_init_values, - ARRAY_SIZE(snd_dt019x_init_values), - "DT019X")) < 0) + err = snd_sbmixer_init(chip, + snd_dt019x_controls, + ARRAY_SIZE(snd_dt019x_controls), + snd_dt019x_init_values, + ARRAY_SIZE(snd_dt019x_init_values), + "DT019X"); + if (err < 0) + return err; break; default: strcpy(card->mixername, "???");

2 1

[alsa-devel] [PATCH v3 0/5] ALSA: hda - hdmi: ATI/AMD multi-channel and HBR support
by Anssi Hannula 14 May '14

14 May '14

Hi all! Here's another revision of the ATI/AMD multichannel+HBR patchset. Changes: - added customizable callbacks instead of polluting generic code - fixed wrong codec id in rev3+ check (as previously noted) - cosmetics I've only tested this on NVIDIA/Intel, but I'd guess at least Peter should be able to test this on AMD in a few days or so. The earlier revisions were tested to fully work. Combined patch can be found at: http://onse.fi/files/atihdmi5.patch However, this patchset is on top recent sound-next commits so it does not directly apply on older kernels. Olivier, I hope you will be able to test this at some point as well (same things as last time), even though it will be a bit more work :) Anssi Hannula (5): ALSA: hda - hdmi: Allow HDA patches to customize more operations ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support ALSA: hda - hdmi: Add ELD emulation for ATI/AMD codecs ALSA: hda - hdmi: Add HBR bitstreaming support for ATI/AMD HDMI codecs ALSA: hda - hdmi: Disable ramp-up/down for non-PCM on AMD codecs sound/pci/hda/hda_eld.c | 151 +++++++++++ sound/pci/hda/hda_local.h | 4 + sound/pci/hda/patch_hdmi.c | 643 +++++++++++++++++++++++++++++++++++++-------- 3 files changed, 689 insertions(+), 109 deletions(-) -- Anssi Hannula

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[alsa-devel] [GIT PULL] ASoC fixes for Intel for v3.15
by Mark Brown 14 May '14

14 May '14

The following changes since commit 95e9ee92e20162681e9f65c25962e0606db9d149: ASoC: Intel: Fix audio crash due to negative address offset (2014-04-22 19:22:53 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.15-rc5-intel for you to fetch changes up to cffd6665f57ed18f4be9185c4330c8c98c22e201: ASoC: Intel: Fix Baytrail SST DSP firmware loading (2014-05-13 18:21:02 +0100) ---------------------------------------------------------------- ASoC: Intel fixes for v3.15 This is a relatively large batch of fixes for the newly added Haswell/Baytrail drivers from Intel. It's a bit larger than is good for this point in the cycle but it's all for a newly added driver so not so worrying as it might otherwise be. Some of it's integration problems, some of it's the sort of problem usually turned up in stress tests. ---------------------------------------------------------------- Jarkko Nikula (2): ASoC: Intel: Cancel hsw_notification_work before freeing the stream ASoC: Intel: Fix Baytrail SST DSP firmware loading Liam Girdwood (8): ASoC: Intel: Fix block allocation so we only allocate blocks once. ASoC: Intel: Fix allocated block list usage when adding blocks. ASoC: Intel: Fix Haswell/Broadwell DSP page table creation. ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled. ASoC: Intel: Fix allow hw_params to be called more than once. ASoC: Intel: Fix stream position pointer. ASoC: Intel: Fix check for pdata usage before dereference. ASoC: Intel: Fix block offset calculations. Wenkai Du (1): ASoC: Intel: Fix audio crash due to race condition in stream deletion sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-baytrail-dsp.c | 2 +- sound/soc/intel/sst-baytrail-ipc.c | 8 +++ sound/soc/intel/sst-dsp-priv.h | 1 + sound/soc/intel/sst-dsp.c | 1 + sound/soc/intel/sst-dsp.h | 1 + sound/soc/intel/sst-firmware.c | 25 +++------ sound/soc/intel/sst-haswell-dsp.c | 4 +- sound/soc/intel/sst-haswell-ipc.c | 31 ++++++++++- sound/soc/intel/sst-haswell-ipc.h | 4 +- sound/soc/intel/sst-haswell-pcm.c | 107 ++++++++++++++++++++++++------------- 11 files changed, 126 insertions(+), 59 deletions(-)

2 1

[alsa-devel] [GIT PULL] ASoC driver fixes for v3.15
by Mark Brown 14 May '14

14 May '14

The following changes since commit 4e17d2d33abe4b523f79fec037ef66ba93643f65: ASoC: cs42l52: Convert to use devm_gpio_request_one (2014-04-18 15:14:39 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.15-rc5-drivers for you to fetch changes up to f9a405961eec4ee9a71a433a473b244ae2d74c3c: Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', 'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus (2014-05-14 12:49:10 +0100) ---------------------------------------------------------------- ASoC: Driver fixes for v3.15 A small set of driver fixes, nothing remarkable in itself or of any relevance outside of the driver. ---------------------------------------------------------------- Charles Keepax (1): ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile Joe Perches (1): ASoC: tlv320aic31xx: Convert /n to \n Kuninori Morimoto (1): ASoC: rsnd: call rsnd_dai_pointer_update() from outside of lock Lars-Peter Clausen (1): ASoC: imx-audmux: Fix section mismatch Mark Brown (1): Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', 'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (3): ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP ASoC: fsl_esai: Only bypass sck_div for EXTAL source ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params() .../devicetree/bindings/sound/tlv320aic31xx.txt | 6 +++--- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/wm8962.c | 15 ++++++++++++--- sound/soc/codecs/wm8962.h | 4 ++++ sound/soc/fsl/fsl_esai.c | 22 ++++++++++++++-------- sound/soc/fsl/fsl_spdif.h | 4 ++-- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/sh/rcar/core.c | 5 ++--- sound/soc/sh/rcar/src.c | 4 ++-- sound/soc/sh/rcar/ssi.c | 4 ++-- 10 files changed, 43 insertions(+), 25 deletions(-)

2 1

[alsa-devel] [GIT PULL] ASoC core fixes for v3.15
by Mark Brown 14 May '14

14 May '14

The following changes since commit 2697e4fb9209dfe1d1b24c92d254158f63d4bc8e: ASoC: dapm: Fix widget double free with auto-disable DAPM kcontrol (2014-04-21 11:49:53 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v3.15-rc5-core for you to fetch changes up to cf86197ec5655ed1f54f1f7ba34a01d576e3737d: Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus (2014-05-14 12:52:41 +0100) ---------------------------------------------------------------- ASoC: Core fixes for v3.15 A few things here: - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to have audio paths which shouldn't be present causing spurious powerups and potential audible issues for users. - Ensure the suspend->off transition doesn't have spurious transitions to prepare added to the sequence. - Fix incorrect skipping of PCM suspension for active audio streams. - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing this and Timur no longer has the boards that he was using. ---------------------------------------------------------------- Lars-Peter Clausen (2): ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets() ASoC: dapm: Fix SUSPEND -> OFF bias sequence Mark Brown (2): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Nicolin Chen (1): ASoC: pcm: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPEND Paul Handrigan (1): ASoC: Update Cirrus Logic CODEC maintainers. MAINTAINERS | 6 ------ sound/soc/soc-dapm.c | 14 ++++++++++---- sound/soc/soc-pcm.c | 2 +- 3 files changed, 11 insertions(+), 11 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: samsung: Add missing pm ops for Snow sound card driver
by Tushar Behera 14 May '14

14 May '14

Adding missing pm ops so that audio playback works across suspend and resume cycle. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- sound/soc/samsung/snow.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 0fa89a4..014c177 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -111,6 +111,7 @@ static struct platform_driver snow_driver = { .driver = { .name = "snow-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = snow_of_match, }, .probe = snow_probe, -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: Fix Baytrail SST DSP firmware loading
by Jarkko Nikula 14 May '14

14 May '14

Commit 10df350977b1 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.") caused following regression in Baytrail SST: baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with the same dma_dev device what is now used in sst_fw_new() when allocating the DMA buffer. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c index 4a5d489e520b..fc588764ffa3 100644 --- a/sound/soc/intel/sst-baytrail-dsp.c +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -324,7 +324,7 @@ static int sst_byt_init(struct sst_dsp *sst, struct sst_pdata *pdata) return ret; } - ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + ret = dma_coerce_mask_and_coherent(sst->dma_dev, DMA_BIT_MASK(32)); if (ret) return ret; -- 2.0.0.rc0

2 11

[alsa-devel] [PATCH 0/5] ASoC: rsnd: add DVC support
by Kuninori Morimoto 13 May '14

13 May '14

Hi Mark These patches add DVC support to Renesas sound driver. DVC is one of feature of CMD unit, it can control sound Volume. Kuninori Morimoto (5): ASoC: rsnd: add rsnd_path_parse() macro ASoC: rsnd: add rsnd_get_adinr() ASoC: rsnd: remove duplicate parameter from rsnd_mod_ops ASoC: rsnd: enable to use multi parameter on rsnd_dai_call/rsnd_mod_call ASoC: rsnd: add DVC support include/sound/rcar_snd.h | 10 ++ sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 18 +++ sound/soc/sh/rcar/core.c | 117 +++++++++++++------ sound/soc/sh/rcar/dvc.c | 273 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 11 ++ sound/soc/sh/rcar/rsnd.h | 65 ++++++++--- sound/soc/sh/rcar/src.c | 121 +++++++------------- sound/soc/sh/rcar/ssi.c | 34 +++--- 9 files changed, 502 insertions(+), 149 deletions(-)

2 6

[alsa-devel] [PATCH] ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile
by Charles Keepax 13 May '14

13 May '14

The register CLASS_D_CONTROL_1 is marked as volatile because it contains a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1 register. This causes problems for the "Speaker Switch" control, which will report an error if the CODEC is suspended because it relies on a volatile register. To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and manually keep the register cache in sync by updating both bits when changing the mute status. Reported-by: Shawn Guo <shawn.guo(a)linaro.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Hi Shawn, I am afraid I don't have access to hardware for 8962 at the moment, so have only build tested this, but I believe it should fix your issue. Please let me know how you get on. Thanks, Charles sound/soc/codecs/wm8962.c | 15 ++++++++++++--- sound/soc/codecs/wm8962.h | 4 ++++ 2 files changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5522d25..ecd26dd 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -154,6 +154,7 @@ static struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ + { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ { 56, 0x0506 }, /* R56 - Clocking 4 */ @@ -795,7 +796,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg) case WM8962_ALC2: case WM8962_THERMAL_SHUTDOWN_STATUS: case WM8962_ADDITIONAL_CONTROL_4: - case WM8962_CLASS_D_CONTROL_1: case WM8962_DC_SERVO_6: case WM8962_INTERRUPT_STATUS_1: case WM8962_INTERRUPT_STATUS_2: @@ -2929,13 +2929,22 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, static int wm8962_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - int val; + int val, ret; if (mute) - val = WM8962_DAC_MUTE; + val = WM8962_DAC_MUTE | WM8962_DAC_MUTE_ALT; else val = 0; + /** + * The DAC mute bit is mirrored in two registers, update both to keep + * the register cache consistent. + */ + ret = snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_1, + WM8962_DAC_MUTE_ALT, val); + if (ret < 0) + return ret; + return snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, WM8962_DAC_MUTE, val); } diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index a1a5d52..910aafd 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -1954,6 +1954,10 @@ #define WM8962_SPKOUTL_ENA_MASK 0x0040 /* SPKOUTL_ENA */ #define WM8962_SPKOUTL_ENA_SHIFT 6 /* SPKOUTL_ENA */ #define WM8962_SPKOUTL_ENA_WIDTH 1 /* SPKOUTL_ENA */ +#define WM8962_DAC_MUTE_ALT 0x0010 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_MASK 0x0010 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_SHIFT 4 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_WIDTH 1 /* DAC_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE 0x0002 /* SPKOUTL_PGA_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE_MASK 0x0002 /* SPKOUTL_PGA_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE_SHIFT 1 /* SPKOUTL_PGA_MUTE */ -- 1.7.2.5

3 2

[alsa-devel] Broken outmixer_event function in wm8400/wm8990/wm8991
by Lars-Peter Clausen 13 May '14

13 May '14

Hi, I was looking at the drivers that were using the SND_SOC_DAPM_PRE_REG event and I noticed that the outmixer_event function for the wm8400, wm8990 and wm8991 is broken. They do a switch statement to determine which kcontrol cause the event in the form of 'switch(kcontrol->private_value) { case (REG | (SHIFT << 8)): ...'. Long long time ago we used to store the register and the shift in the same integer which was stored in private_value. But this was changed in commit 4eaa9819 ("ALSA: ASoC: Convert bitfields in ASoC into full int width"). This was long before the wm8400 and wm8991 drivers were merged and just a month after the wm8990 driver was merged. So these functions have been dead code pretty much for as long as they existed. Do you want to fix this up? If not I'm going to send a patch that removes the functions. If you want to fix this up, this should be re-implemented as a custom put handler for the kcontrols rather than a event callback, which checks whether the change is valid before doing any DAPM changes. The event callback runs in the middle of the DAPM sequence, so half of the changes will already have been applied before the check is made whether the change should be done or not. - Lars

3 5

Re: [alsa-devel] Sound issue in omap-twl4030
by Chaitral Purkar 13 May '14

13 May '14

Can somebody provide any pointers? Thanks and regards, Chaitral Purkar DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.

1 0

[alsa-devel] [PATCH] ALSA: Add patchwork URL to MAINTAINERS
by Takashi Iwai 13 May '14

13 May '14

Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index e67ea2442041..dd400288f9fc 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8244,6 +8244,7 @@ L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://www.alsa-project.org/ T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git T: git git://git.alsa-project.org/alsa-kernel.git +Q: http://patchwork.kernel.org/project/alsa-devel/list/ S: Maintained F: Documentation/sound/ F: include/sound/ -- 1.9.2

2 2

[alsa-devel] [PATCH v2] ALSA: hda - mask buggy stream DMA0 for Broadwell display controller
by mengdong.lin＠intel.com 13 May '14

13 May '14

From: Mengdong Lin <mengdong.lin(a)intel.com> Broadwell display controller has 3 stream DMA engines. DMA0 cannot update DMA postion buffer properly while DMA1 and DMA2 can work well. So this patch masks the buggy DMA0 by keeping it as opened. This is a tentative workaround, so keep the change small 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 d6bca62..6309b5b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1366,6 +1366,12 @@ static int azx_first_init(struct azx *chip) /* initialize streams */ azx_init_stream(chip); + /* workaround for Broadwell HDMI: the first stream is broken, + * so mask it by keeping it as if opened + */ + if (pci->vendor == 0x8086 && pci->device == 0x160c) + chip->azx_dev[0].opened = 1; + /* initialize chip */ azx_init_pci(chip); azx_init_chip(chip, (probe_only[dev] & 2) == 0); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: remove duplicate headers
by Andy Shevchenko 13 May '14

13 May '14

A few files contain duplicate headers. This patch removes the second entry of duplicate in each file under question. There is no functional changes. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-ipc.c | 1 - sound/soc/intel/sst-haswell-ipc.c | 1 - sound/soc/intel/sst-haswell-pcm.c | 1 - 3 files changed, 3 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 0d31dbb..a9c58cd 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -22,7 +22,6 @@ #include <linux/export.h> #include <linux/slab.h> #include <linux/delay.h> -#include <linux/list.h> #include <linux/platform_device.h> #include <linux/kthread.h> #include <linux/firmware.h> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index e7996b3..2dfd070 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -25,7 +25,6 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/sched.h> -#include <linux/list.h> #include <linux/platform_device.h> #include <linux/kthread.h> #include <linux/firmware.h> diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index ce27b50..058efb1 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -17,7 +17,6 @@ #include <linux/module.h> #include <linux/dma-mapping.h> #include <linux/slab.h> -#include <linux/module.h> #include <linux/delay.h> #include <asm/page.h> #include <asm/pgtable.h> -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH] ALSA: hda - using POS_FIX_LPIB on Broadwell HDMI Audio
by libin.yang＠intel.com 13 May '14

13 May '14

From: Libin Yang <libin.yang(a)intel.com> Audio stream is not continuous if using position buffer on Broadwell HDMI audio. Sound is clipped. Fix this issue by using POS_FIX_LPIB. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b540ad7..9cd40c6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -993,6 +993,7 @@ static struct snd_pci_quirk position_fix_list[] = { SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB), + SND_PCI_QUIRK(0x8086, 0x2010, "Broadwell HDMI audio", POS_FIX_LPIB), {} }; -- 1.8.3.2

6 21

[alsa-devel] Too many snd_pcm_mmap_readi calls
by Jan Homann 13 May '14

13 May '14

Hi ALSA-Dev-List I am very new to all this ALSA-Dev-stuff. For a few decades I have been just an alsa user but now i try to develop a tiny sound recorder like millions or thousands did before. On the recorder there is a small while-loop for capturing the pcm data, encode them and send them to a server every 10ms. On my desktop-pc and on a Freescale i.mx 233 (ARMv5) the recorder is working quite well. Now i run into a problem where the recorder is using all the cpu-time on the arm-device for a few minutes. I tried to debug this behaviour without any luck. Next i tried to trace the problem and removed as many calls as possible and put a "gettimeofday" between every remaining call. Here is the remaining code: while(1) { sleep(1); gettimeofday(&time1,NULL); snd_pcm_wait( pcm_handle, REC_BUFFER_ELEMENT_SIZE_MS + 10); gettimeofday(&time2,NULL); snd_pcm_mmap_begin(pcm_handle, &my_areas, &offset, &frames); gettimeofday(&time2,NULL); snd_pcm_avail_update ( pcm_handle ); gettimeofday(&time2,NULL); snd_pcm_mmap_readi( pcm_handle, network_data.data_UC, ALSA_READ_FRAMES); gettimeofday(&time3,NULL); snd_pcm_mmap_commit(pcm_handle, offset, frames); gettimeofday(&time4,NULL); } This is the output from strace: 0.001062 [401b418c] nanosleep({1, 0}, 0xbebd017c) = 0 1.001190 [401a4dac] gettimeofday({1399380864, 459211}, NULL) = 0 0.001032 [401de46c] ioctl(3, SNDRV_PCM_IOCTL_SYNC_PTR, 0x4e1b0) = 0 0.002780 [401a4dac] gettimeofday({1399380864, 463023}, NULL) = 0 0.001063 [401a4dac] gettimeofday({1399380864, 464086}, NULL) = 0 0.001031 [401de46c] ioctl(3, SNDRV_PCM_IOCTL_SYNC_PTR, 0x4e1b0) = 0 0.000968 [401a4dac] gettimeofday({1399380864, 466085}, NULL) = 0 0.001032 [401de46c] ioctl(3, SNDRV_PCM_IOCTL_SYNC_PTR, 0x4e1b0) = 0 0.000968 [401de46c] ioctl(3, SNDRV_PCM_IOCTL_SYNC_PTR, 0x4e1b0) = 0 0.000969 [401de46c] ioctl(3, SNDRV_PCM_IOCTL_SYNC_PTR, 0x4e1b0) = 0 0.001000 [401de46c] ioctl(3, SNDRV_PCM_IOCTL_SYNC_PTR, 0x4e1b0) = 0 0.002249 [401a4dac] gettimeofday({1399380864, 472678}, NULL) = 0 0.001438 [401a4dac] gettimeofday({1399380864, 473740}, NULL) = 0 0.001031 [40149518] rt_sigprocmask(SIG_BLOCK, [CHLD], [], 8) = 0 0.001218 [40149400] rt_sigaction(SIGCHLD, NULL, {SIG_DFL, [], 0}, 8) = 0 0.001562 [40149518] rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 0.001032 [401b418c] nanosleep({1, 0}, ^CProcess 14219 detached As you can see, the output starts with the sleep. I just added the sleep to find my loop-start in the trace-outpus. Before every gettimeofday there is one call to a snd_pcm-function (ok, besides the sleep). There is a block of four ioctl calls between the fourth and the fifth gettimeofday. This sould be the sound_pcm_mmap_readi. In the while loop there is only one call to sound_pcm_mmap_read and on my desktop-pc trace even shows only one call to sound_pcm_mmap_readi. Maybe anyone can explain to me why the function is called that often on the arm-device? Maybe you can give some hints on what to check for the high cpu utilization? Any hints would be kindly appreciated. Thanks in advance, Jan Homann

3 5

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID to snd-hda
by Aaron Plattner 13 May '14

13 May '14

Vendor ID 0x10de0071 is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Stephen Warren <swarren(a)nvidia.com> --- 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 0cb5b89cd0c8..347a84b58c39 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3330,6 +3330,7 @@ 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 = 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 }, { .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3385,6 +3386,7 @@ 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:10de0071"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); MODULE_ALIAS("snd-hda-codec-id:11069f81"); -- 1.9.2

2 1

[alsa-devel] [PATCH 2/2] ALSA: hda - mask buggy stream DMA0 for Broadwell display controller
by mengdong.lin＠intel.com 13 May '14

13 May '14

From: Mengdong Lin <mengdong.lin(a)intel.com> Broadwell display controller has 3 stream DMA engines. DMA0 cannot update DMA postion buffer properly while DMA1 and DMA2 can work well. So this patch masks the buggy DMA0. 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 377c3a5..13b13a3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -281,6 +281,9 @@ static char *driver_short_names[] = { [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; +#define is_broadwell_controller(pci) \ + ((pci)->vendor == 0x8086 && (pci)->device == 0x160c) + #ifdef CONFIG_X86 static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on) { @@ -1249,6 +1252,7 @@ static int azx_first_init(struct azx *chip) struct snd_card *card = chip->card; int err; unsigned short gcap; + unsigned int stream_mask = 0; #if BITS_PER_LONG != 64 /* Fix up base address on ULI M5461 */ @@ -1364,7 +1368,10 @@ static int azx_first_init(struct azx *chip) return err; /* initialize streams */ - azx_init_stream(chip, 0); + if (is_broadwell_controller(pci)) + stream_mask = 0x1; /* mask buggy stream DMA0 */ + + azx_init_stream(chip, stream_mask); /* initialize chip */ azx_init_pci(chip); -- 1.8.1.2

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - allow to mask a buggy stream DMA by setting it as opened
by mengdong.lin＠intel.com 13 May '14

13 May '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The function azx_init_stream() can accept a bit mask parameter. If the bit for a stream DMA is set, that stream device will be masked by setting it as opened and so it will never be assigned to any PCM stream. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 248b90a..378fed7 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -2004,7 +2004,7 @@ EXPORT_SYMBOL_GPL(azx_mixer_create); /* initialize SD streams */ -int azx_init_stream(struct azx *chip) +int azx_init_stream(struct azx *chip, unsigned int mask) { int i; @@ -2022,6 +2022,9 @@ int azx_init_stream(struct azx *chip) /* stream tag: must be non-zero and unique */ azx_dev->index = i; azx_dev->stream_tag = i + 1; + /* mask a buggy stream DMA by setting it as opened */ + if (mask & (1 << i)) + azx_dev->opened = 1; } return 0; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index baf0e77..2fd8e6d 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -48,6 +48,6 @@ int azx_codec_create(struct azx *chip, const char *model, int *power_save_to); int azx_codec_configure(struct azx *chip); int azx_mixer_create(struct azx *chip); -int azx_init_stream(struct azx *chip); +int azx_init_stream(struct azx *chip, unsigned int mask); #endif /* __SOUND_HDA_CONTROLLER_H */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d6bca62..377c3a5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1364,7 +1364,7 @@ static int azx_first_init(struct azx *chip) return err; /* initialize streams */ - azx_init_stream(chip); + azx_init_stream(chip, 0); /* initialize chip */ azx_init_pci(chip); -- 1.8.1.2

1 0

[alsa-devel] [RFC PATCH 0/2] Add support for audio cards with multiple endpoints on the one interface
by Eduard Gilmutdinov 13 May '14

13 May '14

Devices of company Focusrite (Novation) contain multiple endpoints on a single interface. In particular (1235:0010) "Novation Saffire 6 USB" on one interface have endpoints for playback and capture. Functions "set_format" and "pcm_close" are generated when new stream is creating and when it is closing respectively. They contain a function call "usb_set_interface" which resets interface and endpoints on it. Thus it is imposible to capture and playback a sound at the same time. To avoid this, suggest to monitor the status of the interface and endpoint on it. Status of endpoints can be monitored by the field "enabled" of the structure "usb_host_endpoint". This approach should not affect for the audio cards with one endpoint on the interface. Please correct me if I'm wrong.

1 2

[alsa-devel] [PATCH v4 0/5] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 13 May '14

13 May '14

Changes since v3: * Removed the omap-hdmi compile fix that slipped in by accident. v3 cover letter: The only change in v3 is the renaming of the component IO mutex from 'mutex' to 'io_mutex'. v2 cover letter: This series contains the remaining patches from the previous series that have not been applied yet. The series has been rebased ontop of asoc/for-next and depends on topic/sta350, topic/core, topic/dapm, topic/cache and topic/component. Some minor updates were made to the patches and three new patches: * One new user of snd_kcontrol_chip() showed up, the patch converts it to snd_soc_kcontrol_codec() * Change return type of snd_soc_write() from unsigned int to int * Remove ASoC level tracing Original cover letter: This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. Lars-Peter Clausen (5): ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() ASoC: Remove ASoC level IO tracing include/sound/soc-dapm.h | 1 + include/sound/soc.h | 51 +++++-- include/trace/events/asoc.h | 92 ------------- sound/soc/soc-core.c | 285 +++++++++++++++++++++++++------------- sound/soc/soc-dapm.c | 91 ++---------- sound/soc/soc-io.c | 327 ++++++++++++++++++++++++++------------------ 6 files changed, 447 insertions(+), 400 deletions(-) -- 1.8.0

5 27

[alsa-devel] [PATCH] alsa-plugins: fix polling and recovering in alsa-jack plugin
by Sergey 13 May '14

13 May '14

Hello. This patch fixes polling in alsa-to-jack plugin. It makes poll()+_poll_revents() return correct values whenever there's something to read/write. Additionally jack_deactivate() code makes it survive snd_pcm_recover(). It works for the code example I posted before, aplay, arecord, mozilla, etc. Comments and suggestions are welcome. --- diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 7a8b24d..a39fea5 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -42,6 +42,7 @@ typedef struct { unsigned int num_ports; unsigned int hw_ptr; unsigned int sample_bits; + snd_pcm_uframes_t min_avail; unsigned int channels; snd_pcm_channel_area_t *areas; @@ -50,6 +51,41 @@ typedef struct { jack_client_t *client; } snd_pcm_jack_t; +static inline int poll_blocked(snd_pcm_ioplug_t *io) +{ + static char buf[65536]; + snd_pcm_sframes_t avail; + snd_pcm_jack_t *jack = io->private_data; + + if (io->state == SND_PCM_STATE_RUNNING) + { + avail = snd_pcm_avail_update(io->pcm); + if (avail >= 0 && avail < jack->min_avail) + { + read(io->poll_fd, &buf, sizeof buf); + return 1; + } + } + + return 0; +} + +static inline int poll_unblocked(snd_pcm_ioplug_t *io) +{ + static char buf[1]; + snd_pcm_sframes_t avail; + snd_pcm_jack_t *jack = io->private_data; + + avail = snd_pcm_avail_update(io->pcm); + if (avail < 0 || avail >= jack->min_avail) + { + write(jack->fd, &buf, 1); + return 1; + } + + return 0; +} + static void snd_pcm_jack_free(snd_pcm_jack_t *jack) { if (jack) { @@ -81,14 +117,10 @@ static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { - static char buf[1]; - assert(pfds && nfds == 1 && revents); - read(pfds[0].fd, buf, 1); - *revents = pfds[0].revents & ~(POLLIN | POLLOUT); - if (pfds[0].revents & POLLIN) + if (pfds[0].revents & POLLIN && !poll_blocked(io)) *revents |= (io->stream == SND_PCM_STREAM_PLAYBACK) ? POLLOUT : POLLIN; return 0; } @@ -105,7 +137,6 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) snd_pcm_jack_t *jack = io->private_data; const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t xfer = 0; - static char buf[1]; unsigned int channel; for (channel = 0; channel < io->channels; channel++) { @@ -145,7 +176,7 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) xfer += frames; } - write(jack->fd, buf, 1); /* for polling */ + poll_unblocked(io); /* unblock socket for polling if needed */ return 0; } @@ -154,9 +185,25 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; unsigned int i; + snd_pcm_sw_params_t *swparams; + int err; jack->hw_ptr = 0; + jack->min_avail = io->period_size; + snd_pcm_sw_params_alloca(&swparams); + err = snd_pcm_sw_params_current(io->pcm, swparams); + if (err == 0) { + snd_pcm_sw_params_get_avail_min(swparams, &jack->min_avail); + } + + poll_unblocked(io); /* unblock socket for initial polling if needed */ + + if (jack->activated) { + jack_deactivate(jack->client); + jack->activated = 0; + } + if (jack->ports) return 0; --

2 4

[alsa-devel] [PATCH RESEND 0/4] ASoC: fsl_esai: Fix some issues
by Nicolin Chen 13 May '14

13 May '14

Sorry for this resend. I just found my previous patches have not been delivered successfully to the alsa maillist. This series of patches fix the following problems: PATCH-1 for an incorrect ratio range check. PATCH-2 for an incorrect bypass settings of FSYS clock source. PATCH-3 for duplicated calculations and configurations of dividers PATCH-4 for mismatched init flow of configuration comparing to RM. Nicolin Chen (4): ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP ASoC: fsl_esai: Only bypass sck_div for EXTAL source ASoC: fsl_esai: Bypass divider settings if clock requirement is not changed ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params() sound/soc/fsl/fsl_esai.c | 45 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 34 insertions(+), 11 deletions(-) -- 1.8.4

2 5

[alsa-devel] [PATCH 2/2] ASoC: fsl_spdif: Improve coding style
by Nicolin Chen 13 May '14

13 May '14

1) Apply better indentations 2) Drop braces for single statement. 3) Use simpler ternary to reduce code. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 72 +++++++++++++++++++++++------------------------ sound/soc/fsl/fsl_spdif.h | 2 ++ 2 files changed, 37 insertions(+), 37 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9ea2dd4..9d35c5c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -32,10 +32,13 @@ #define FSL_SPDIF_TXFIFO_WML 0x8 #define FSL_SPDIF_RXFIFO_WML 0x8 -#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) -#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\ - INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\ - INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED) +#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) +#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL |\ + INT_URX_OV | INT_QRX_FUL | INT_QRX_OV |\ + INT_UQ_SYNC | INT_UQ_ERR | INT_RXFIFO_RESYNC |\ + INT_LOSS_LOCK | INT_DPLL_LOCKED) + +#define SIE_INTR_FOR(tx) (tx ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE) /* Index list for the values that has if (DPLL Locked) condition */ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; @@ -137,10 +140,9 @@ static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv) dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n"); - if (!spdif_priv->dpll_locked) { - /* DPLL unlocked seems no audio stream */ + /* Clear illegal symbol if DPLL unlocked since no audio stream */ + if (!spdif_priv->dpll_locked) regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0); - } } /* U/Q Channel receive register full */ @@ -335,8 +337,8 @@ static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv) u32 ch_status; ch_status = (bitrev8(ctrl->ch_status[0]) << 16) | - (bitrev8(ctrl->ch_status[1]) << 8) | - bitrev8(ctrl->ch_status[2]); + (bitrev8(ctrl->ch_status[1]) << 8) | + bitrev8(ctrl->ch_status[2]); regmap_write(regmap, REG_SPDIF_STCSCH, ch_status); dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status); @@ -433,13 +435,12 @@ clk_set_bypass: spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs); /* select clock source and divisor */ - stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DF(txclk_df); - mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DF_MASK; + stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | + STC_TXCLK_DF(txclk_df) | STC_SYSCLK_DF(sysclk_df); + mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | + STC_TXCLK_DF_MASK | STC_SYSCLK_DF_MASK; regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc); - regmap_update_bits(regmap, REG_SPDIF_STC, - STC_SYSCLK_DF_MASK, STC_SYSCLK_DF(sysclk_df)); - dev_dbg(&pdev->dev, "set sample rate to %dHz for %dHz playback\n", spdif_priv->txrate[rate], sample_rate); @@ -553,7 +554,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, return ret; } spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK, - IEC958_AES3_CON_CLOCK_1000PPM); + IEC958_AES3_CON_CLOCK_1000PPM); spdif_write_channel_status(spdif_priv); } else { /* Setup rx clock source */ @@ -569,9 +570,9 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct regmap *regmap = spdif_priv->regmap; - int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE; - u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 intr = SIE_INTR_FOR(tx); + u32 dmaen = SCR_DMA_xX_EN(tx); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -662,9 +663,8 @@ static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol, u32 cstatus, val; regmap_read(regmap, REG_SPDIF_SIS, &val); - if (!(val & INT_CNEW)) { + if (!(val & INT_CNEW)) return -EAGAIN; - } regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus); ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF; @@ -693,15 +693,14 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; unsigned long flags; - int ret = 0; + int ret = -EAGAIN; spin_lock_irqsave(&ctrl->ctl_lock, flags); if (ctrl->ready_buf) { int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE; memcpy(&ucontrol->value.iec958.subcode[0], &ctrl->subcode[idx], SPDIF_UBITS_SIZE); - } else { - ret = -EAGAIN; + ret = 0; } spin_unlock_irqrestore(&ctrl->ctl_lock, flags); @@ -726,15 +725,14 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; unsigned long flags; - int ret = 0; + int ret = -EAGAIN; spin_lock_irqsave(&ctrl->ctl_lock, flags); if (ctrl->ready_buf) { int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE; memcpy(&ucontrol->value.bytes.data[0], &ctrl->qsub[idx], SPDIF_QSUB_SIZE); - } else { - ret = -EAGAIN; + ret = 0; } spin_unlock_irqrestore(&ctrl->ctl_lock, flags); @@ -799,10 +797,10 @@ static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv, regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf); clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf; - if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) { - /* Get bus clock from system */ + + /* Get bus clock from system */ + if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) busclk_freq = clk_get_rate(spdif_priv->sysclk); - } /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */ tmpval64 = (u64) busclk_freq * freqmeas; @@ -826,12 +824,12 @@ static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); - int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + int rate = 0; if (spdif_priv->dpll_locked) - ucontrol->value.integer.value[0] = rate; - else - ucontrol->value.integer.value[0] = 0; + rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + + ucontrol->value.integer.value[0] = rate; return 0; } @@ -1238,12 +1236,12 @@ static int fsl_spdif_probe(struct platform_device *pdev) spin_lock_init(&ctrl->ctl_lock); /* Init tx channel status default value */ - ctrl->ch_status[0] = - IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015; + ctrl->ch_status[0] = IEC958_AES0_CON_NOT_COPYRIGHT | + IEC958_AES0_CON_EMPHASIS_5015; ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID; ctrl->ch_status[2] = 0x00; - ctrl->ch_status[3] = - IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM; + ctrl->ch_status[3] = IEC958_AES3_CON_FS_44100 | + IEC958_AES3_CON_CLOCK_1000PPM; spdif_priv->dpll_locked = false; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index 16fde4b..b13e30e 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -93,6 +93,8 @@ #define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET) #define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET) +#define SCR_DMA_xX_EN(tx) (tx ? SCR_DMA_TX_EN : SCR_DMA_RX_EN) + /* SPDIF CDText control */ #define SRCD_CD_USER_OFFSET 1 #define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET) -- 1.8.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl_spdif: Fix incorrect usage of regmap_read()
by Nicolin Chen 13 May '14

13 May '14

We should not copy the return value into this val since it's supposed to get the value of the register not the success result of regmap_read(). Thus fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index b912d45..9ea2dd4 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -762,7 +762,7 @@ static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, struct regmap *regmap = spdif_priv->regmap; u32 val; - val = regmap_read(regmap, REG_SPDIF_SIS, &val); + regmap_read(regmap, REG_SPDIF_SIS, &val); ucontrol->value.integer.value[0] = (val & INT_VAL_NOGOOD) != 0; regmap_write(regmap, REG_SPDIF_SIC, INT_VAL_NOGOOD); -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: rt5640: add default case for unexpected ID
by bardliao＠realtek.com 12 May '14

12 May '14

From: Bard Liao <bardliao(a)realtek.com> We may read an unexpected value when detemining which codec is attached. In that case, either a unsupported codec is attached or something wrong with I2C. The driver will not work properly on both cases. So we return an error for that. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b0967df..9e0d48f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2019,6 +2019,10 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5639_specific_dapm_routes, ARRAY_SIZE(rt5639_specific_dapm_routes)); break; + default: + dev_err(codec->dev, + "The driver is for RT5639 RT5640 or RT5642 only\n"); + return -ENODEV; } return 0; -- 1.8.1.1.439.g50a6b54

3 2

[alsa-devel] [PATCH 0/5] ASoC: Trivial bits and cleanups
by Lars-Peter Clausen 12 May '14

12 May '14

Hi, This series contains a few cleanup patches that came up while working on the componentization, but all also make sense on their own. The biggest change is admittedly the removal of the suspend/resume code from the atmel-pcm-pdc driver and may not seem trivial at first. But this code has been fully broken for over 4 years, was always partially broken and was never really needed in the first place. And right now it is blocking other cleanups. Everything else is just removal of unused fields and the consolidation of duplicated fields. Depends on: asoc/topic/component - Lars Lars-Peter Clausen (5): ASoC: Remove card's DAI list ASoC: Remove unused 'list' field form card ASoC: Remove unused num_dai field from CODEC ASoC: atmel-pcm-pdc: Remove broken suspend/resume code ASoC: Remove runtime field from DAI include/sound/soc-dai.h | 2 -- include/sound/soc.h | 4 --- sound/soc/atmel/atmel-pcm-pdc.c | 63 ----------------------------------------- sound/soc/soc-compress.c | 1 - sound/soc/soc-core.c | 7 ----- sound/soc/soc-pcm.c | 1 - 6 files changed, 78 deletions(-) -- 1.8.0

2 10

[alsa-devel] [PATCH v2] ASoC: dapm: Fix SUSPEND -> OFF bias sequence
by Lars-Peter Clausen 12 May '14

12 May '14

Currently when the DAPM context bias level is SUSPEND and the target bias level is OFF dapm_pre_sequence_async() will first transition to PREPARE and dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and then to OFF. This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE when either going to ON or away from ON. This avoids the extra unnecessary transitions. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Given that SUSPEND -> OFF transitions are rather rare and the code will still work correctly (but inefficent) without this patch, the patch probably does not need to go into stable. Changes since v1: * Change indention --- sound/soc/soc-dapm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5d54dfc..bfc46a9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1549,8 +1549,11 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) "ASoC: Failed to turn on bias: %d\n", ret); } - /* Prepare for a STADDBY->ON or ON->STANDBY transition */ - if (d->bias_level != d->target_bias_level) { + /* Prepare for a transition to ON or away from ON */ + if ((d->target_bias_level == SND_SOC_BIAS_ON && + d->bias_level != SND_SOC_BIAS_ON) || + (d->target_bias_level != SND_SOC_BIAS_ON && + d->bias_level == SND_SOC_BIAS_ON)) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, -- 1.8.0

2 1

[alsa-devel] [PATCH 0/7] ASoC: Intel: Baytrail SST suspend/resume support
by Jarkko Nikula 12 May '14

12 May '14

This set adds suspend/resume support to Baytrail SST. This is implemented by putting the DSP into reset and by releasing driver resources before suspend. In resume path the DSP is rebooted in normal boot state. Active streams are restored by restarting them from last known buffer position before suspend. Patches 1-5 are preparing for the suspend/resume and patch 6 implements the core logic that takes care of stopping the DSP before suspend and restarting it again after resume. Patch 7 removes ignore_suspend flag from byt-rt5640 machine driver since it doesn't have use at the moment and adds suspend/resume callbacks that are hooked to ASoC core. This takes care that active stream will be paused and resumed over suspend/resume cycle on byt-rt5640 machine. Jarkko Nikula (5): ASoC: Intel: Sample Baytrail DSP DMA pointer only after each period ASoC: Intel: Simplify Baytrail stream control IPC construction ASoC: Intel: Pass stream start position to sst_byt_stream_start() ASoC: Intel: Move Baytrail extended fw address saving to sst_byt_boot() ASoC: Intel: Allow byt-5640 machine driver and SST core go to suspend Liam Girdwood (2): ASoC: Intel: Allow Rx/Tx message list can be cleared prior to suspend ASoC: Intel: Add Baytrail suspend/resume support sound/soc/intel/byt-rt5640.c | 14 +++- sound/soc/intel/sst-baytrail-dsp.c | 14 ++-- sound/soc/intel/sst-baytrail-ipc.c | 124 ++++++++++++++++++++++++++++------ sound/soc/intel/sst-baytrail-ipc.h | 7 +- sound/soc/intel/sst-baytrail-pcm.c | 135 ++++++++++++++++++++++++++++++++----- 5 files changed, 249 insertions(+), 45 deletions(-) -- 2.0.0.rc0

2 8

[alsa-devel] [PATCH] ASoC: dapm: Handle SND_SOC_DAPM_REG() generically
by Lars-Peter Clausen 12 May '14

12 May '14

Commit commit de9ba98b6d ("ASoC: dapm: Make widget power register settings more flexible") added generic support for on_val/off_val in the DAPM core. With this in place there is no need anymore for having a special event callback for SND_SOC_DAPM_REG() widgets. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 7 ++----- sound/soc/soc-dapm.c | 20 -------------------- 2 files changed, 2 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 75020f5..5932d66 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -256,9 +256,8 @@ struct device; /* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ { .id = wid, .name = wname, .kcontrol_news = NULL, .num_kcontrols = 0, \ - .reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \ - .on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \ - .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} + .reg = wreg, .shift = wshift, .mask = wmask, \ + .on_val = won_val, .off_val = woff_val, } #define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \ { .id = snd_soc_dapm_supply, .name = wname, \ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ @@ -362,8 +361,6 @@ struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; -int dapm_reg_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event); int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fb6c7b7..56fcd6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1059,26 +1059,6 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } /* - * Handler for generic register modifier widget. - */ -int dapm_reg_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - unsigned int val; - - if (SND_SOC_DAPM_EVENT_ON(event)) - val = w->on_val; - else - val = w->off_val; - - soc_widget_update_bits(w, -(w->reg + 1), - w->mask << w->shift, val << w->shift); - - return 0; -} -EXPORT_SYMBOL_GPL(dapm_reg_event); - -/* * Handler for regulator supply widget. */ int dapm_regulator_event(struct snd_soc_dapm_widget *w, -- 1.8.0

2 1

[alsa-devel] [PATCH] sound: soc: intel: remove unneeded dependency from Makefile
by Lad Prabhakar 12 May '14

12 May '14

From: "Lad, Prabhakar" <prabhakar.csengg(a)gmail.com> this patch removes drops the entries form Makefile, fixing following issues when make clean was run, scripts/Makefile.clean:17: sound/soc/intel/board/Makefile: No such file or directory scripts/Makefile.clean:17: sound/soc/intel/sst/Makefile: No such file or directory Signed-off-by: Lad, Prabhakar <prabhakar.csengg(a)gmail.com> --- Found this issue on linux-next not sure if this is fixed in sound subsystem tree. sound/soc/intel/Makefile | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 57242c4..26544c5 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -32,9 +32,3 @@ PLATFORM_LIBS = platform-libs/controls_v2_dpcm.o snd-soc-sst-platform-objs := pcm.o compress.o $(PLATFORM_LIBS) obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o - -# Relevant Machine driver -obj-$(CONFIG_SND_SST_MACHINE) += board/ - -# IPC driver -obj-$(CONFIG_SND_INTEL_SST) += sst/ -- 1.7.9.5

3 2

[alsa-devel] [PATCH v2 1/5] ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets()
by Lars-Peter Clausen 12 May '14

12 May '14

For CODEC to CODEC DAI links the paths are created in snd_soc_dapm_new_pcm(). Also for CODEC to CODEC links the widgets are connected cross-over via a DAI link widget, meaning that the capture widget of one CODEC will be connected to the playback widget of the other and vice versa. Whereas snd_soc_dapm_connect_dai_link_widgets() directly connects the playback widget of the CPU DAI to the playback widget of the CODEC DAI and the capture widget of the CPU DAI to the capture widget of the CODEC DAI. So not skipping CODEC<->CODEC links in snd_soc_dapm_connect_dai_link_widgets() will create incorrect connections between the two CODECs which will cause DAPM to detect active paths where there are none and unnecessarily power up widgets. Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.") Cc: <stable(a)vger.kernel.org> (for 3.14+) Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fb6c7b7..142a738 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3410,8 +3410,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) cpu_dai = rtd->cpu_dai; codec_dai = rtd->codec_dai; - /* dynamic FE links have no fixed DAI mapping */ - if (rtd->dai_link->dynamic) + /* + * dynamic FE links have no fixed DAI mapping. + * CODEC<->CODEC links have no direct connection. + */ + if (rtd->dai_link->dynamic || rtd->dai_link->params) continue; /* there is no point in connecting BE DAI links with dummies */ -- 1.8.0

4 13

[alsa-devel] [PATCH] ASoC: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPEND
by Nicolin Chen 12 May '14

12 May '14

The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 88230ea..49930bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1666,7 +1666,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; case SNDRV_PCM_TRIGGER_SUSPEND: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 1.8.4

2 1

[alsa-devel] [PATCH RFT] ASoC: cs42l56: Fix update mute register bits in cs42l56_digital_mute
by Axel Lin 12 May '14

12 May '14

The new value in snd_soc_update_bits() needs to match the mask bit fields. Use CS42L56_MUTE/CS42L56_UNMUTE only works when the mask is 0x01. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- For example, *Both* below calls actually *clear* CS42L56_LO_MUTE_MASK bit (0x80). snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, CS42L56_LO_MUTE_MASK, CS42L56_MUTE); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); So I think use CS42L56_UNMUTE/CS42L56_MUTE is not good here. Hi Brian, I don't have this h/w, so this patch needs test. I'd appreciate if you can review and test this patch. Regards, Axel sound/soc/codecs/cs42l56.c | 65 ++++++++++++++++++---------------------------- sound/soc/codecs/cs42l56.h | 2 -- 2 files changed, 25 insertions(+), 42 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 5bb134b..ecb2fed 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -839,62 +839,47 @@ static int cs42l56_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + unsigned int mask; if (mute) { /* Hit the DSP Mixer first */ - snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, - CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_MUTE); + mask = CS42L56_ADCAMIX_MUTE_MASK | CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | CS42L56_MSTA_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, mask, mask); + /* Mute ADC's */ - snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_MUTE); + mask = CS42L56_ADCA_MUTE_MASK | CS42L56_ADCB_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, mask, mask); + /* HP And LO */ snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_HP_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_LO_MUTE_MASK); } else { - snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, - CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_UNMUTE); - snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_UNMUTE); + mask = CS42L56_ADCAMIX_MUTE_MASK | CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | CS42L56_MSTA_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, mask , 0); + + mask = CS42L56_ADCA_MUTE_MASK | CS42L56_ADCB_MUTE_MASK; + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, mask, 0); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, 0); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, 0); } return 0; } diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h index ad2b50a..4997587 100644 --- a/sound/soc/codecs/cs42l56.h +++ b/sound/soc/codecs/cs42l56.h @@ -116,8 +116,6 @@ #define CS42L56_DEEMPH_MASK 0x40 #define CS42L56_PLYBCK_GANG_MASK 0x10 #define CS42L56_PCM_INV_MASK 0x0c -#define CS42L56_MUTE 1 -#define CS42L56_UNMUTE 0 #define CS42L56_ADCAMIX_MUTE_MASK 0x40 #define CS42L56_ADCBMIX_MUTE_MASK 0x80 #define CS42L56_PCMAMIX_MUTE_MASK 0x10 -- 1.8.3.2

2 1

[alsa-devel] [REGRESSION] alsa PM regression in 3.15*, hda intel, realtek codec
by Knut Petersen 10 May '14

10 May '14

S3 suspend / resume cycle works on kernels up to 3.14.x as expected. With kernels 3.15.0-rc2 it's necessary to reload the alsa modules as sound stays silent otherwise. Hardware: AOpen i915GMm-hfs mobo, hda intel sound with realtek codec. 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 04) Subsystem: AOPEN Inc. Device 0560 Flags: bus master, fast devsel, latency 0, IRQ 46 Memory at d5440000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Capabilities: [130] Root Complex Link Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel Is this regression known or should I bisect? cu, Knut

2 1

[alsa-devel] [patch] ASoC: wm0010: set error code on allocation failure
by Dan Carpenter 09 May '14

09 May '14

We should return -ENOMEM if the kzalloc() fails instead of returning -ENOMEM. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 71ce315..8dac6c5 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -682,6 +682,7 @@ static int wm0010_boot(struct snd_soc_codec *codec) if (!img_swap) { dev_err(codec->dev, "Failed to allocate image buffer\n"); + ret = -ENOMEM; goto abort; }

2 2

Re: [alsa-devel] [PATCH 1/2] mfd: mc13xxx: Do not use platform data with devicetree
by Lee Jones 09 May '14

09 May '14

On Fri, 09 May 2014, Alexander Shiyan wrote: > Fri, 9 May 2014 12:30:30 +0100 от Lee Jones <lee.jones(a)linaro.org>: > > > This patch removes platform data usage for codec and touchscreen > > > when driver is used with devicetree. > > > This fixes possible "null pointer dereference" error if DTS uses > > > "fsl,mc13xxx-uses-codec" or "fsl,mc13xxx-uses-touch" options. > > > > > > Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> > > > --- > > > drivers/mfd/mc13xxx-core.c | 18 ++++++++++-------- > > > 1 file changed, 10 insertions(+), 8 deletions(-) > > > > Applied, thanks. > > This patch has been written as a way to separate MFD and ASoC changes of > the patch: (ASoC: mc13783: Add devicetree support) > https://git.kernel.org/cgit/linux/kernel/git/next/linux-next.git/commit/?id… > > So, it should produce conflict. Exactly. Mark, You need to revert (or remove) the original one and apply: "ASoC: mc13783: Add devicetree support" ... instead. -- Lee Jones Linaro STMicroelectronics Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

2 3

Re: [alsa-devel] Sound issue in omap-twl4030
by Chaitral Purkar 09 May '14

09 May '14

I wrongly configured McBSP channel. Now, I am getting an error at the end which is 'aplay: pcm_write:1710: write error: Input/output error'. Can anybody provide some pointers in this regard. I am copying the aplay output. ubuntu@arm:~$ aplay -v test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'Zoom2' 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 : 27560 period_size : 5512 period_time : 124988 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 27560 stop_threshold : 27560 silence_threshold: 0 silence_size : 0 boundary : 1806172160 appl_ptr : 0 hw_ptr : 0 aplay: pcm_write:1710: write error: Input/output error Regards, Chaitral Purkar DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.

1 0

[alsa-devel] [PATCH 1/2] mfd: mc13xxx: Do not use platform data with devicetree
by Alexander Shiyan 09 May '14

09 May '14

This patch removes platform data usage for codec and touchscreen when driver is used with devicetree. This fixes possible "null pointer dereference" error if DTS uses "fsl,mc13xxx-uses-codec" or "fsl,mc13xxx-uses-touch" options. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- drivers/mfd/mc13xxx-core.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 06e64b6..2ed22d7 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -673,17 +673,9 @@ int mc13xxx_common_init(struct device *dev) if (mc13xxx->flags & MC13XXX_USE_ADC) mc13xxx_add_subdevice(mc13xxx, "%s-adc"); - if (mc13xxx->flags & MC13XXX_USE_CODEC) - mc13xxx_add_subdevice_pdata(mc13xxx, "%s-codec", - pdata->codec, sizeof(*pdata->codec)); - if (mc13xxx->flags & MC13XXX_USE_RTC) mc13xxx_add_subdevice(mc13xxx, "%s-rtc"); - if (mc13xxx->flags & MC13XXX_USE_TOUCHSCREEN) - mc13xxx_add_subdevice_pdata(mc13xxx, "%s-ts", - &pdata->touch, sizeof(pdata->touch)); - if (pdata) { mc13xxx_add_subdevice_pdata(mc13xxx, "%s-regulator", &pdata->regulators, sizeof(pdata->regulators)); @@ -691,10 +683,20 @@ int mc13xxx_common_init(struct device *dev) pdata->leds, sizeof(*pdata->leds)); mc13xxx_add_subdevice_pdata(mc13xxx, "%s-pwrbutton", pdata->buttons, sizeof(*pdata->buttons)); + if (mc13xxx->flags & MC13XXX_USE_CODEC) + mc13xxx_add_subdevice_pdata(mc13xxx, "%s-codec", + pdata->codec, sizeof(*pdata->codec)); + if (mc13xxx->flags & MC13XXX_USE_TOUCHSCREEN) + mc13xxx_add_subdevice_pdata(mc13xxx, "%s-ts", + &pdata->touch, sizeof(pdata->touch)); } else { mc13xxx_add_subdevice(mc13xxx, "%s-regulator"); mc13xxx_add_subdevice(mc13xxx, "%s-led"); mc13xxx_add_subdevice(mc13xxx, "%s-pwrbutton"); + if (mc13xxx->flags & MC13XXX_USE_CODEC) + mc13xxx_add_subdevice(mc13xxx, "%s-codec"); + if (mc13xxx->flags & MC13XXX_USE_TOUCHSCREEN) + mc13xxx_add_subdevice(mc13xxx, "%s-ts"); } return 0; -- 1.8.3.2

2 1

[alsa-devel] Sound issue in omap-twl4030
by Chaitral Purkar 09 May '14

09 May '14

Hello, I am working on enabling audio support on a customized board which is similar to zoom2. I am not able to hear any sound (with aplay) except two click/pop sounds (at start and at end). I have ported Ubuntu kernel 3.7.10 on this board which has OMAP3530. On this board, I have TPS65930 chip which has a speaker connected over PreDriv.L and PreDriv.R. I have incorporated the changes suggested on http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/058119.h… and updated sound/soc/omap/omap-twl4030.c, sound/soc/codecs/twl4030.c and board-file under arch/arm/mach-omap2/. When I try to play audio, ubuntu@arm:~$ aplay -v test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo ALSA <-> PulseAudio PCM I/O Plugin 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 : 22050 period_size : 5512 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 22050 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 1445068800 ubuntu@arm:~$ However, no audio comes out from speaker. dmesg shows following- [ 1613.446319] omap-dma-engine omap-dma-engine: allocating channel for 33 [ 1613.502471] clock: dpll1_ck: set rate: locking rate to 600000000. [ 1613.502502] clock: configuring DPLL dpll1_ck for low-power bypass [ 1613.502563] clock: dpll1_ck transition to 'bypassed' in 0 loops [ 1613.502624] clock: locking DPLL dpll1_ck [ 1613.502777] clock: dpll1_ck transition to 'locked' in 94 loops [ 1613.822082] clock: dpll1_ck: set rate: locking rate to 500000000. [ 1613.822113] clock: configuring DPLL dpll1_ck for low-power bypass [ 1613.822143] clock: dpll1_ck transition to 'bypassed' in 0 loops [ 1613.822174] clock: locking DPLL dpll1_ck [ 1613.822326] clock: dpll1_ck transition to 'locked' in 29 loops [ 1614.126739] clock: dpll1_ck: set rate: locking rate to 250000000. [ 1614.126770] clock: configuring DPLL dpll1_ck for low-power bypass [ 1614.126800] clock: dpll1_ck transition to 'bypassed' in 0 loops [ 1614.126831] clock: locking DPLL dpll1_ck [ 1614.127014] clock: dpll1_ck transition to 'locked' in 34 loops [ 1614.783264] clock: dpll1_ck: set rate: locking rate to 125000000. [ 1614.783294] clock: configuring DPLL dpll1_ck for low-power bypass [ 1614.783325] clock: dpll1_ck transition to 'bypassed' in 0 loops [ 1614.783355] clock: locking DPLL dpll1_ck [ 1614.783538] clock: dpll1_ck transition to 'locked' in 58 loops [ 1617.033203] omap-dma-engine omap-dma-engine: freeing channel for 33 I have following struct defined in the board file and not using device tree support. static struct omap_tw4030_pdata omap_twl4030_audio_data = { .voice_connected = false, .custom_routing = true, .has_predriv = OMAP_TWL4030_LEFT | OMAP_TWL4030_RIGHT, }; Also, I have set the amixer controls. Please find the file attached herewith. However, I am still not able hear sound. Can anybody please provide any pointer in this regard. Thanks in advance. Regards, Chaitral Purkar DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.

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[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for three Dell laptops
by Hui Wang 09 May '14

09 May '14

When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255, SID: 0x1028065c; VID: 0x10ec0255, SID: 0x10280680; VID: 0x10ec0292, SID: 0x10280684), the headset mic can't be detected, after apply this patch, the headset mic can work well. And on the machine with SID 0x10280684, and the Lineout and external microphone should be routed to docking, this patch also fix this problem. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b60de0d..385b003 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4705,6 +4705,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0657, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x065c, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -4713,6 +4714,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0674, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x067e, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0680, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0684, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

Re: [alsa-devel] [PATCH] ASoC: Intel: Build Medfield compressed ops
by Koul, Vinod 09 May '14

09 May '14

On 9 May 2014 04:01, Mark Brown <broonie(a)kernel.org> wrote: > > From: Mark Brown <broonie(a)linaro.org> > > Since commit 4b68b4e1c564 (ASoC: Intel: split the pcm and compress to > different files) the compressed ops haven't been built causing link > failures on allyesconfig and making the driver unbuildable. Add the > object to the Makefile to fix that. > Thanks looks like makefile patch missed this. I will fix and resend the series next week when I am back > Signed-off-by: Mark Brown <broonie(a)linaro.org> Acked-by Vinod Koul <vinod.koul(a)intel.com> > --- > sound/soc/intel/Makefile | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile > index 5d3fb6f35a19..0db4e2f336dc 100644 > --- a/sound/soc/intel/Makefile > +++ b/sound/soc/intel/Makefile > @@ -2,7 +2,7 @@ > snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o > snd-soc-sst-acpi-objs := sst-acpi.o > > -snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o > +snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o sst-mfld-platform-compress.o > snd-soc-mfld-machine-objs := mfld_machine.o > > obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o > -- > 2.0.0.rc2 >

1 0

[alsa-devel] [PATCH] ASoC: Intel: Build Medfield compressed ops
by Mark Brown 09 May '14

09 May '14

From: Mark Brown <broonie(a)linaro.org> Since commit 4b68b4e1c564 (ASoC: Intel: split the pcm and compress to different files) the compressed ops haven't been built causing link failures on allyesconfig and making the driver unbuildable. Add the object to the Makefile to fix that. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/intel/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 5d3fb6f35a19..0db4e2f336dc 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -2,7 +2,7 @@ snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o snd-soc-sst-acpi-objs := sst-acpi.o -snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o sst-mfld-platform-compress.o snd-soc-mfld-machine-objs := mfld_machine.o obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o -- 2.0.0.rc2

1 0

[alsa-devel] [PATCH] rawmidi.c Set a name for a newly allocated rawmidi substream
by Ralf Beck 08 May '14

08 May '14

Rationale: Bitwig Studio (and possibly other rawmidi clients) enumerate the Midi I/Os by substream names. With the current rawmidi implementation, the name is an empty string and only the USB mididriver changes it to a meaningful value after allocation. This patch sets a default substream name of the form "somename x-y-z" --- linux-source-3.13.0/sound/core/rawmidi.c.orig 2014-05-06 20:25:33.683081389 +0200 +++ linux-source-3.13.0/sound/core/rawmidi.c 2014-05-06 17:42:35.058097185 +0200 @@ -34,6 +34,7 @@ #include <sound/control.h> #include <sound/minors.h> #include <sound/initval.h> +#include <linux/string.h> MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); MODULE_DESCRIPTION("Midlevel RawMidi code for ALSA."); @@ -1420,6 +1421,8 @@ static int snd_rawmidi_alloc_substreams( substream->number = idx; substream->rmidi = rmidi; substream->pstr = stream; + snprintf(substream->name, sizeof(substream->name), + "%s %d-%d-%d", rmidi->id, rmidi->card->number, rmidi->device, idx); list_add_tail(&substream->list, &stream->substreams); stream->substream_count++; }

2 3

[alsa-devel] [PATCH]: snd_seq_virmidi.c do not use running status for virmidi raw input
by Ralf Beck 08 May '14

08 May '14

Rationale: Not every rawmidi alsa client (e.q. Bitwig) support running status on input. There is also no need for running status inside a computer. This patch disables the removal of the status byte when decoding snd_midi_event structures to a rawmidi bytestream --- sound/core/seq/seq_virmidi.c.orig 2014-05-07 19:30:44.175914670 +0200 +++ sound/core/seq/seq_virmidi.c 2014-05-07 19:33:38.931906859 +0200 @@ -212,6 +212,7 @@ static int snd_virmidi_input_open(struct kfree(vmidi); return -ENOMEM; } + vmidi->parser->nostat = 1; vmidi->seq_mode = rdev->seq_mode; vmidi->client = rdev->client; vmidi->port = rdev->port;

2 3

[alsa-devel] [PATCH v2 0/8] ASoC: Intel: rework the mfld platform driver
by Vinod Koul 08 May '14

08 May '14

This patch series reworks and reformats the existing platform driver we use. This is stepping support for seconds series of patches which will support Merrifield platform with DPCM based implementation Not much logical changes in code except adding drain support for compressed audio. Most is mere modularization of the existing code Fixes in v2: - add two more patches - fix makefile addition is 4th patch - dropped name change in platform ops patch Vinod Koul (8): ASoC: Intel: remove FSF snail mail address ASoc: Intel: rename sst-mfld-platform.c ASoC: Intel: mark sst_set_stream_status as non static ASoC: Intel: split the pcm and compress to different files ASoC: Intel: remove unused sst-mfld platform dais ASoC: Intel: add drain_notify support ASoC: Intel: remove codec memeber from codec structs ASoC: Intel: move component registration blob sound/soc/intel/Makefile | 14 +- sound/soc/intel/sst-mfld-dsp.h | 8 - sound/soc/intel/sst-mfld-platform-compress.c | 237 +++++++++++++++++++ ...sst-mfld-platform.c => sst-mfld-platform-pcm.c} | 250 +------------------- sound/soc/intel/sst-mfld-platform.h | 11 +- 5 files changed, 263 insertions(+), 257 deletions(-) create mode 100644 sound/soc/intel/sst-mfld-platform-compress.c rename sound/soc/intel/{sst-mfld-platform.c => sst-mfld-platform-pcm.c} (67%)

3 11

[alsa-devel] [PATCH 0/7] ARM: shmobile: swtich audio settings to new style, and remove compatibility code from driver
by Kuninori Morimoto 08 May '14

08 May '14

Hi Simon, Mark Now, Linux v3.15-rc1 was released. 1) - 5) remove old style audio settings from platform, and 6) - 7) remove compatibility (= old style) code from driver. These are based on v3.15-rc1 Simon, can you please create v3.15-rc1 base branch/tag for Mark ? Kuninori Morimoto (7): 1) ARM: shmobile: r8a7778: remove old style audio clock 2) ARM: shmobile: r8a7790: remove old style audio clock 3) ARM: shmobile: bockw: remove old style audio clock 4) ARM: shmobile: bockw: switch to use dai info for R-Car sound 5) ARM: shmobile: lager: switch to use dai info for R-Car sound 6) ASoC: rsnd: remove old clock style support 7) ASoC: rsnd: remove compatibility code arch/arm/mach-shmobile/board-bockw.c | 63 +++++++++++++++++++---------- arch/arm/mach-shmobile/board-lager.c | 17 +++++--- arch/arm/mach-shmobile/clock-r8a7778.c | 22 +++++----- arch/arm/mach-shmobile/clock-r8a7790.c | 24 +++++------ include/sound/rcar_snd.h | 22 ++-------- sound/soc/sh/rcar/adg.c | 39 ++---------------- sound/soc/sh/rcar/core.c | 54 +++++-------------------- sound/soc/sh/rcar/rsnd.h | 3 -- sound/soc/sh/rcar/src.c | 69 ++++++++++---------------------- sound/soc/sh/rcar/ssi.c | 51 +---------------------- 10 files changed, 111 insertions(+), 253 deletions(-) Best regards --- Kuninori Morimoto

4 17

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Fix a serious typo
by Oder Chiou 08 May '14

08 May '14

The patch corrects a serious typo in struct reg_default. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 6db6f81..30410eb 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -50,10 +50,10 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_default init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, - {RT5645_PR_BASE + 0x1c, 0xfd20); - {RT5645_PR_BASE + 0x20, 0x611f); - {RT5645_PR_BASE + 0x21, 0x4040); - {RT5645_PR_BASE + 0x23, 0x0004); + {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x20, 0x611f}, + {RT5645_PR_BASE + 0x21, 0x4040}, + {RT5645_PR_BASE + 0x23, 0x0004}, }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH 3/4 v3] ASoC: rt5645: Move settings from probe() to reg_default struct
by Oder Chiou 08 May '14

08 May '14

The patch moves the private register settings from probe() to reg_default struct. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9d8c7c8..4be9000 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -50,6 +50,10 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_default init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, + {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x20, 0x611f}, + {RT5645_PR_BASE + 0x21, 0x4040}, + {RT5645_PR_BASE + 0x23, 0x0004}, }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) @@ -2205,10 +2209,6 @@ static int rt5645_probe(struct snd_soc_codec *codec) rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); - snd_soc_write(codec, RT5645_PR_BASE + 0x1c, 0xfd20); - snd_soc_write(codec, RT5645_PR_BASE + 0x20, 0x611f); - snd_soc_write(codec, RT5645_PR_BASE + 0x21, 0x4040); - snd_soc_write(codec, RT5645_PR_BASE + 0x23, 0x0004); return 0; } -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH 1/4] ASoC: rt5645: Remove the unused variable
by Oder Chiou 08 May '14

08 May '14

The patch is for removing the unused variable. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7d8cc16..37ec468 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1166,7 +1166,6 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH 3/4 v2] ASoC: rt5645: Move settings from probe() to reg_default struct
by Oder Chiou 08 May '14

08 May '14

The patch moves the private register settings from probe() to reg_default struct. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9d8c7c8..4be9000 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -50,6 +50,10 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_default init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, + {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x20, 0x611f}, + {RT5645_PR_BASE + 0x21, 0x4040}, + {RT5645_PR_BASE + 0x23, 0x0004}, }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) @@ -2205,10 +2209,6 @@ static int rt5645_probe(struct snd_soc_codec *codec) rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); - snd_soc_write(codec, RT5645_PR_BASE + 0x1c, 0xfd20); - snd_soc_write(codec, RT5645_PR_BASE + 0x20, 0x611f); - snd_soc_write(codec, RT5645_PR_BASE + 0x21, 0x4040); - snd_soc_write(codec, RT5645_PR_BASE + 0x23, 0x0004); return 0; } -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH v2] ASoC: rt5645: Add codec driver
by Oder Chiou 08 May '14

08 May '14

This patch adds the Realtek ALC5645 codec driver. It is the base version that because the jack detect function is not implemented to it, the headphone and AMIC1 are not workable. We will fill up the further functions later. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 25 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5645.c | 2476 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5645.h | 2188 +++++++++++++++++++++++++++++++++++++++ 5 files changed, 4695 insertions(+) create mode 100644 include/sound/rt5645.h create mode 100644 sound/soc/codecs/rt5645.c create mode 100644 sound/soc/codecs/rt5645.h diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h new file mode 100644 index 0000000..1de744c --- /dev/null +++ b/include/sound/rt5645.h @@ -0,0 +1,25 @@ +/* + * linux/sound/rt5645.h -- Platform data for RT5645 + * + * 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_RT5645_H +#define __LINUX_SND_RT5645_H + +struct rt5645_platform_data { + /* IN2 can optionally be differential */ + bool in2_diff; + + bool dmic_en; + unsigned int dmic1_data_pin; + /* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */ + unsigned int dmic2_data_pin; + /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5e0a4e2..f6ce2fe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,6 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE @@ -402,6 +403,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT5645 + tristate + config SND_SOC_RT5651 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8d1930d..3dd8c52 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -61,6 +61,7 @@ snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o 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-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -215,6 +216,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(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-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c new file mode 100644 index 0000000..7d8cc16 --- /dev/null +++ b/sound/soc/codecs/rt5645.c @@ -0,0 +1,2476 @@ +/* + * rt5645.c -- RT5645 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 <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 "rt5645.h" + +#define RT5645_DEVICE_ID 0x6308 + +#define RT5645_PR_RANGE_BASE (0xff + 1) +#define RT5645_PR_SPACING 0x100 + +#define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING)) + +static const struct regmap_range_cfg rt5645_ranges[] = { + { + .name = "PR", + .range_min = RT5645_PR_BASE, + .range_max = RT5645_PR_BASE + 0xf8, + .selector_reg = RT5645_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5645_PRIV_DATA, + .window_len = 0x1, + }, +}; + +static const struct reg_default init_list[] = { + {RT5645_PR_BASE + 0x3d, 0x3600}, +}; +#define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5645_reg[] = { + { 0x00, 0x0000 }, + { 0x01, 0xc8c8 }, + { 0x02, 0xc8c8 }, + { 0x03, 0xc8c8 }, + { 0x0a, 0x0002 }, + { 0x0b, 0x2827 }, + { 0x0c, 0xe000 }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x14, 0x3333 }, + { 0x16, 0x4b00 }, + { 0x18, 0x018b }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0001 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x20, 0x0000 }, + { 0x27, 0x7060 }, + { 0x28, 0x7070 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5656 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaaa0 }, + { 0x2f, 0x1002 }, + { 0x31, 0x5000 }, + { 0x32, 0x0000 }, + { 0x33, 0x0000 }, + { 0x34, 0x0000 }, + { 0x35, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x3f, 0x0000 }, + { 0x40, 0x001f }, + { 0x41, 0x0000 }, + { 0x42, 0x001f }, + { 0x45, 0x6000 }, + { 0x46, 0x003e }, + { 0x47, 0x003e }, + { 0x48, 0xf807 }, + { 0x4a, 0x0004 }, + { 0x4d, 0x0000 }, + { 0x4e, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x53, 0xf000 }, + { 0x56, 0x0111 }, + { 0x57, 0x0064 }, + { 0x58, 0xef0e }, + { 0x59, 0xf0f0 }, + { 0x5a, 0xef0e }, + { 0x5b, 0xf0f0 }, + { 0x5c, 0xef0e }, + { 0x5d, 0xf0f0 }, + { 0x5e, 0xf000 }, + { 0x5f, 0x0000 }, + { 0x61, 0x0300 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c2 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6c, 0x0aaa }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x7770 }, + { 0x74, 0x3e00 }, + { 0x75, 0x2409 }, + { 0x76, 0x000a }, + { 0x77, 0x0c00 }, + { 0x78, 0x0000 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c06 }, + { 0x93, 0x0000 }, + { 0x94, 0x0200 }, + { 0x95, 0x0000 }, + { 0x9a, 0x2184 }, + { 0x9b, 0x010a }, + { 0x9c, 0x0aea }, + { 0x9d, 0x000c }, + { 0x9e, 0x0400 }, + { 0xa0, 0xa0a8 }, + { 0xa1, 0x0059 }, + { 0xa2, 0x0001 }, + { 0xae, 0x6000 }, + { 0xaf, 0x0000 }, + { 0xb0, 0x6000 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x020c }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x3100 }, + { 0xc0, 0x0000 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xc3, 0x2000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x1813 }, + { 0xd0, 0x0690 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0003 }, + { 0xdc, 0x0049 }, + { 0xdd, 0x001b }, + { 0xe6, 0x8000 }, + { 0xe7, 0x0200 }, + { 0xec, 0xb300 }, + { 0xed, 0x0000 }, + { 0xf0, 0x001f }, + { 0xf1, 0x020c }, + { 0xf2, 0x1f00 }, + { 0xf3, 0x0000 }, + { 0xf4, 0x4000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x2060 }, + { 0xfb, 0x4040 }, + { 0xfc, 0x0000 }, + { 0xfd, 0x0002 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6308 }, +}; + +static int rt5645_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5645_RESET, 0); +} + +static bool rt5645_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5645_ranges); i++) { + if (reg >= rt5645_ranges[i].range_min && + reg <= rt5645_ranges[i].range_max) { + return true; + } + } + + switch (reg) { + case RT5645_RESET: + case RT5645_PRIV_DATA: + case RT5645_IN1_CTRL1: + case RT5645_IN1_CTRL2: + case RT5645_IN1_CTRL3: + case RT5645_A_JD_CTRL1: + case RT5645_ADC_EQ_CTRL1: + case RT5645_EQ_CTRL1: + case RT5645_ALC_CTRL_1: + case RT5645_IRQ_CTRL2: + case RT5645_IRQ_CTRL3: + case RT5645_INT_IRQ_ST: + case RT5645_IL_CMD: + case RT5645_VENDOR_ID: + case RT5645_VENDOR_ID1: + case RT5645_VENDOR_ID2: + return 1; + default: + return 0; + } +} + +static bool rt5645_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5645_ranges); i++) { + if (reg >= rt5645_ranges[i].range_min && + reg <= rt5645_ranges[i].range_max) { + return true; + } + } + + switch (reg) { + case RT5645_RESET: + case RT5645_SPK_VOL: + case RT5645_HP_VOL: + case RT5645_LOUT1: + case RT5645_IN1_CTRL1: + case RT5645_IN1_CTRL2: + case RT5645_IN1_CTRL3: + case RT5645_IN2_CTRL: + case RT5645_INL1_INR1_VOL: + case RT5645_SPK_FUNC_LIM: + case RT5645_ADJ_HPF_CTRL: + case RT5645_DAC1_DIG_VOL: + case RT5645_DAC2_DIG_VOL: + case RT5645_DAC_CTRL: + case RT5645_STO1_ADC_DIG_VOL: + case RT5645_MONO_ADC_DIG_VOL: + case RT5645_ADC_BST_VOL1: + case RT5645_ADC_BST_VOL2: + case RT5645_STO1_ADC_MIXER: + case RT5645_MONO_ADC_MIXER: + case RT5645_AD_DA_MIXER: + case RT5645_STO_DAC_MIXER: + case RT5645_MONO_DAC_MIXER: + case RT5645_DIG_MIXER: + case RT5645_DIG_INF1_DATA: + case RT5645_PDM_OUT_CTRL: + case RT5645_REC_L1_MIXER: + case RT5645_REC_L2_MIXER: + case RT5645_REC_R1_MIXER: + case RT5645_REC_R2_MIXER: + case RT5645_HPMIXL_CTRL: + case RT5645_HPOMIXL_CTRL: + case RT5645_HPMIXR_CTRL: + case RT5645_HPOMIXR_CTRL: + case RT5645_HPO_MIXER: + case RT5645_SPK_L_MIXER: + case RT5645_SPK_R_MIXER: + case RT5645_SPO_MIXER: + case RT5645_SPO_CLSD_RATIO: + case RT5645_OUT_L1_MIXER: + case RT5645_OUT_R1_MIXER: + case RT5645_OUT_L_GAIN1: + case RT5645_OUT_L_GAIN2: + case RT5645_OUT_R_GAIN1: + case RT5645_OUT_R_GAIN2: + case RT5645_LOUT_MIXER: + case RT5645_HAPTIC_CTRL1: + case RT5645_HAPTIC_CTRL2: + case RT5645_HAPTIC_CTRL3: + case RT5645_HAPTIC_CTRL4: + case RT5645_HAPTIC_CTRL5: + case RT5645_HAPTIC_CTRL6: + case RT5645_HAPTIC_CTRL7: + case RT5645_HAPTIC_CTRL8: + case RT5645_HAPTIC_CTRL9: + case RT5645_HAPTIC_CTRL10: + case RT5645_PWR_DIG1: + case RT5645_PWR_DIG2: + case RT5645_PWR_ANLG1: + case RT5645_PWR_ANLG2: + case RT5645_PWR_MIXER: + case RT5645_PWR_VOL: + case RT5645_PRIV_INDEX: + case RT5645_PRIV_DATA: + case RT5645_I2S1_SDP: + case RT5645_I2S2_SDP: + case RT5645_ADDA_CLK1: + case RT5645_ADDA_CLK2: + case RT5645_DMIC_CTRL1: + case RT5645_DMIC_CTRL2: + case RT5645_TDM_CTRL_1: + case RT5645_TDM_CTRL_2: + case RT5645_GLB_CLK: + case RT5645_PLL_CTRL1: + case RT5645_PLL_CTRL2: + case RT5645_ASRC_1: + case RT5645_ASRC_2: + case RT5645_ASRC_3: + case RT5645_ASRC_4: + case RT5645_DEPOP_M1: + case RT5645_DEPOP_M2: + case RT5645_DEPOP_M3: + case RT5645_MICBIAS: + case RT5645_A_JD_CTRL1: + case RT5645_VAD_CTRL4: + case RT5645_CLSD_OUT_CTRL: + case RT5645_ADC_EQ_CTRL1: + case RT5645_ADC_EQ_CTRL2: + case RT5645_EQ_CTRL1: + case RT5645_EQ_CTRL2: + case RT5645_ALC_CTRL_1: + case RT5645_ALC_CTRL_2: + case RT5645_ALC_CTRL_3: + case RT5645_ALC_CTRL_4: + case RT5645_ALC_CTRL_5: + case RT5645_JD_CTRL: + case RT5645_IRQ_CTRL1: + case RT5645_IRQ_CTRL2: + case RT5645_IRQ_CTRL3: + case RT5645_INT_IRQ_ST: + case RT5645_GPIO_CTRL1: + case RT5645_GPIO_CTRL2: + case RT5645_GPIO_CTRL3: + case RT5645_BASS_BACK: + case RT5645_MP3_PLUS1: + case RT5645_MP3_PLUS2: + case RT5645_ADJ_HPF1: + case RT5645_ADJ_HPF2: + case RT5645_HP_CALIB_AMP_DET: + case RT5645_SV_ZCD1: + case RT5645_SV_ZCD2: + case RT5645_IL_CMD: + case RT5645_IL_CMD2: + case RT5645_IL_CMD3: + case RT5645_DRC1_HL_CTRL1: + case RT5645_DRC2_HL_CTRL1: + case RT5645_ADC_MONO_HP_CTRL1: + case RT5645_ADC_MONO_HP_CTRL2: + case RT5645_DRC2_CTRL1: + case RT5645_DRC2_CTRL2: + case RT5645_DRC2_CTRL3: + case RT5645_DRC2_CTRL4: + case RT5645_DRC2_CTRL5: + case RT5645_JD_CTRL3: + case RT5645_JD_CTRL4: + case RT5645_GEN_CTRL1: + case RT5645_GEN_CTRL2: + case RT5645_GEN_CTRL3: + case RT5645_VENDOR_ID: + case RT5645_VENDOR_ID1: + case RT5645_VENDOR_ID2: + return 1; + default: + 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); +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), +}; + +static const char * const rt5645_tdm_data_swap_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_slot0_1_enum, + RT5645_TDM_CTRL_1, 6, rt5645_tdm_data_swap_select); + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_slot2_3_enum, + RT5645_TDM_CTRL_1, 4, rt5645_tdm_data_swap_select); + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_slot4_5_enum, + RT5645_TDM_CTRL_1, 2, rt5645_tdm_data_swap_select); + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_slot6_7_enum, + RT5645_TDM_CTRL_1, 0, rt5645_tdm_data_swap_select); + +static const char * const rt5645_tdm_adc_data_select[] = { + "1/2/R", "2/1/R", "R/1/2", "R/2/1" +}; + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, + RT5645_TDM_CTRL_1, 8, + rt5645_tdm_adc_data_select); + +static const struct snd_kcontrol_new rt5645_snd_controls[] = { + /* Speaker Output Volume */ + SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, + RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* Headphone Output Volume */ + SOC_DOUBLE("HP Channel Switch", RT5645_HP_VOL, + RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5645_HP_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5645_LOUT1, + RT5645_L_MUTE_SFT, RT5645_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5645_LOUT1, + RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5645_LOUT1, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5645_DAC_CTRL, + RT5645_M_DAC_L2_VOL_SFT, RT5645_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5645_DAC1_DIG_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5645_DAC2_DIG_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5645_IN1_CTRL1, + RT5645_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5645_IN2_CTRL, + RT5645_BST_SFT2, 8, 0, bst_tlv), + + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5645_INL1_INR1_VOL, + RT5645_INL_VOL_SFT, RT5645_INR_VOL_SFT, 31, 1, in_vol_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5645_STO1_ADC_DIG_VOL, + RT5645_L_MUTE_SFT, RT5645_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5645_STO1_ADC_DIG_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 127, 0, adc_vol_tlv), + SOC_DOUBLE("Mono ADC Capture Switch", RT5645_MONO_ADC_DIG_VOL, + RT5645_L_MUTE_SFT, RT5645_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5645_MONO_ADC_DIG_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1, + RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), + SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1, + RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), + + /* I2S2 function select */ + SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, + 1, 1), + + /* TDM */ + SOC_ENUM("TDM Adc Slot0 1 Data", rt5645_tdm_adc_slot0_1_enum), + SOC_ENUM("TDM Adc Slot2 3 Data", rt5645_tdm_adc_slot2_3_enum), + SOC_ENUM("TDM Adc Slot4 5 Data", rt5645_tdm_adc_slot4_5_enum), + SOC_ENUM("TDM Adc Slot6 7 Data", rt5645_tdm_adc_slot6_7_enum), + SOC_ENUM("TDM IF1 ADC DATA Sel", rt5645_tdm_adc_sel_enum), + SOC_SINGLE("TDM IF1_DAC1_L Sel", RT5645_TDM_CTRL_3, 12, 7, 0), + SOC_SINGLE("TDM IF1_DAC1_R Sel", RT5645_TDM_CTRL_3, 8, 7, 0), + SOC_SINGLE("TDM IF1_DAC2_L Sel", RT5645_TDM_CTRL_3, 4, 7, 0), + SOC_SINGLE("TDM IF1_DAC2_R Sel", RT5645_TDM_CTRL_3, 0, 7, 0), +}; + +/** + * 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 rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}; + int idx = -EINVAL, i; + int rate, red, bound, temp; + + rate = rt5645->sysclk; + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5645_DMIC_CTRL1, + RT5645_DMIC_CLK_MASK, idx << RT5645_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, RT5645_GLB_CLK); + val &= RT5645_SCLK_SRC_MASK; + if (val == RT5645_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, + RT5645_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5645_STO1_ADC_MIXER, + RT5645_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, + RT5645_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5645_STO1_ADC_MIXER, + RT5645_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5645_MONO_ADC_MIXER, + RT5645_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5645_MONO_ADC_MIXER, + RT5645_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5645_MONO_ADC_MIXER, + RT5645_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5645_MONO_ADC_MIXER, + RT5645_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER, + RT5645_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER, + RT5645_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER, + RT5645_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER, + RT5645_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_dig_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5645_DIG_MIXER, + RT5645_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_DIG_MIXER, + RT5645_M_DAC_L2_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_DIG_MIXER, + RT5645_M_DAC_R2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_dig_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5645_DIG_MIXER, + RT5645_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_DIG_MIXER, + RT5645_M_DAC_R2_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_DIG_MIXER, + RT5645_M_DAC_L2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5645_rec_l_mix[] = { + SOC_DAPM_SINGLE("HPOL Switch", RT5645_REC_L2_MIXER, + RT5645_M_HP_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5645_REC_L2_MIXER, + RT5645_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5645_REC_L2_MIXER, + RT5645_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5645_REC_L2_MIXER, + RT5645_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5645_REC_L2_MIXER, + RT5645_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_rec_r_mix[] = { + SOC_DAPM_SINGLE("HPOR Switch", RT5645_REC_R2_MIXER, + RT5645_M_HP_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5645_REC_R2_MIXER, + RT5645_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5645_REC_R2_MIXER, + RT5645_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5645_REC_R2_MIXER, + RT5645_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5645_REC_R2_MIXER, + RT5645_M_OM_R_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_spk_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_SPK_L_MIXER, + RT5645_M_DAC_L1_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_SPK_L_MIXER, + RT5645_M_DAC_L2_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5645_SPK_L_MIXER, + RT5645_M_IN_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5645_SPK_L_MIXER, + RT5645_M_BST1_L_SM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_spk_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_SPK_R_MIXER, + RT5645_M_DAC_R1_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_SPK_R_MIXER, + RT5645_M_DAC_R2_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5645_SPK_R_MIXER, + RT5645_M_IN_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5645_SPK_R_MIXER, + RT5645_M_BST2_R_SM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5645_OUT_L1_MIXER, + RT5645_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5645_OUT_L1_MIXER, + RT5645_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_OUT_L1_MIXER, + RT5645_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_OUT_L1_MIXER, + RT5645_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5645_OUT_R1_MIXER, + RT5645_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5645_OUT_R1_MIXER, + RT5645_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_OUT_R1_MIXER, + RT5645_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_OUT_R1_MIXER, + RT5645_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_spo_l_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_SPO_MIXER, + RT5645_M_DAC_R1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_SPO_MIXER, + RT5645_M_DAC_L1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5645_SPO_MIXER, + RT5645_M_SV_R_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL L Switch", RT5645_SPO_MIXER, + RT5645_M_SV_L_SPM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_spo_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_SPO_MIXER, + RT5645_M_DAC_R1_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5645_SPO_MIXER, + RT5645_M_SV_R_SPM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_HPO_MIXER, + RT5645_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5645_HPO_MIXER, + RT5645_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_hpvoll_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_HPOMIXL_CTRL, + RT5645_M_DAC1_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 Switch", RT5645_HPOMIXL_CTRL, + RT5645_M_DAC2_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5645_HPOMIXL_CTRL, + RT5645_M_IN_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5645_HPOMIXL_CTRL, + RT5645_M_BST1_HV_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_hpvolr_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_HPOMIXR_CTRL, + RT5645_M_DAC1_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 Switch", RT5645_HPOMIXR_CTRL, + RT5645_M_DAC2_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5645_HPOMIXR_CTRL, + RT5645_M_IN_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5645_HPOMIXR_CTRL, + RT5645_M_BST2_HV_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_LOUT_MIXER, + RT5645_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_LOUT_MIXER, + RT5645_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX L Switch", RT5645_LOUT_MIXER, + RT5645_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX R Switch", RT5645_LOUT_MIXER, + RT5645_M_OV_R_LM_SFT, 1, 1), +}; + +/*DAC1 L/R source*/ /* MX-29 [9:8] [11:10] */ +static const char * const rt5645_dac1_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac1l_enum, RT5645_AD_DA_MIXER, + RT5645_DAC1_L_SEL_SFT, rt5645_dac1_src); + +static const struct snd_kcontrol_new rt5645_dac1l_mux = + SOC_DAPM_ENUM("DAC1 L source", rt5645_dac1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac1r_enum, RT5645_AD_DA_MIXER, + RT5645_DAC1_R_SEL_SFT, rt5645_dac1_src); + +static const struct snd_kcontrol_new rt5645_dac1r_mux = + SOC_DAPM_ENUM("DAC1 R source", rt5645_dac1r_enum); + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +static const char * const rt5645_dac12_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "Mono ADC", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac2l_enum, RT5645_DAC_CTRL, + RT5645_DAC2_L_SEL_SFT, rt5645_dac12_src); + +static const struct snd_kcontrol_new rt5645_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5645_dac2l_enum); + +static const char * const rt5645_dacr2_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "Mono ADC", "Haptic" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac2r_enum, RT5645_DAC_CTRL, + RT5645_DAC2_R_SEL_SFT, rt5645_dacr2_src); + +static const struct snd_kcontrol_new rt5645_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5645_dac2r_enum); + + +/* INL/R source */ +static const char * const rt5645_inl_src[] = { + "IN2P", "MonoP" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_inl_enum, RT5645_INL1_INR1_VOL, + RT5645_INL_SEL_SFT, rt5645_inl_src); + +static const struct snd_kcontrol_new rt5645_inl_mux = + SOC_DAPM_ENUM("INL source", rt5645_inl_enum); + +static const char * const rt5645_inr_src[] = { + "IN2N", "MonoN" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_inr_enum, RT5645_INL1_INR1_VOL, + RT5645_INR_SEL_SFT, rt5645_inr_src); + +static const struct snd_kcontrol_new rt5645_inr_mux = + SOC_DAPM_ENUM("INR source", rt5645_inr_enum); + +/* Stereo1 ADC source */ +/* MX-27 [12] */ +static const char * const rt5645_stereo_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_stereo1_adc1_enum, RT5645_STO1_ADC_MIXER, + RT5645_ADC_1_SRC_SFT, rt5645_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5645_sto_adc1_mux = + SOC_DAPM_ENUM("Stereo1 ADC1 Mux", rt5645_stereo1_adc1_enum); + +/* MX-27 [11] */ +static const char * const rt5645_stereo_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_stereo1_adc2_enum, RT5645_STO1_ADC_MIXER, + RT5645_ADC_2_SRC_SFT, rt5645_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5645_sto_adc2_mux = + SOC_DAPM_ENUM("Stereo1 ADC2 Mux", rt5645_stereo1_adc2_enum); + +/* MX-27 [8] */ +static const char * const rt5645_stereo_dmic_src[] = { + "DMIC1", "DMIC2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_stereo1_dmic_enum, RT5645_STO1_ADC_MIXER, + RT5645_DMIC_SRC_SFT, rt5645_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5645_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5645_stereo1_dmic_enum); + +/* Mono ADC source */ +/* MX-28 [12] */ +static const char * const rt5645_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_adc_l1_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_ADC_L1_SRC_SFT, rt5645_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5645_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5645_mono_adc_l1_enum); +/* MX-28 [11] */ +static const char * const rt5645_mono_adc_l2_src[] = { + "Mono DAC MIXL", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_adc_l2_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_ADC_L2_SRC_SFT, rt5645_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5645_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5645_mono_adc_l2_enum); + +/* MX-28 [8] */ +static const char * const rt5645_mono_dmic_src[] = { + "DMIC1", "DMIC2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_dmic_l_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_DMIC_L_SRC_SFT, rt5645_mono_dmic_src); + +static const struct snd_kcontrol_new rt5645_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC left source", rt5645_mono_dmic_l_enum); +/* MX-28 [1:0] */ +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_dmic_r_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_DMIC_R_SRC_SFT, rt5645_mono_dmic_src); + +static const struct snd_kcontrol_new rt5645_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC Right source", rt5645_mono_dmic_r_enum); +/* MX-28 [4] */ +static const char * const rt5645_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_adc_r1_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_ADC_R1_SRC_SFT, rt5645_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5645_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5645_mono_adc_r1_enum); +/* MX-28 [3] */ +static const char * const rt5645_mono_adc_r2_src[] = { + "Mono DAC MIXR", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_adc_r2_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_ADC_R2_SRC_SFT, rt5645_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5645_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5645_mono_adc_r2_enum); + +/* MX-77 [9:8] */ +static const char * const rt5645_if1_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_if1_adc_in_enum, RT5645_TDM_CTRL_1, + RT5645_IF1_ADC_IN_SFT, rt5645_if1_adc_in_src); + +static const struct snd_kcontrol_new rt5645_if1_adc_in_mux = + SOC_DAPM_ENUM("IF1 ADC IN source", rt5645_if1_adc_in_enum); + +/* MX-2F [13:12] */ +static const char * const rt5645_if2_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_if2_adc_in_enum, RT5645_DIG_INF1_DATA, + RT5645_IF2_ADC_IN_SFT, rt5645_if2_adc_in_src); + +static const struct snd_kcontrol_new rt5645_if2_adc_in_mux = + SOC_DAPM_ENUM("IF2 ADC IN source", rt5645_if2_adc_in_enum); + +/* MX-2F [1:0] */ +static const char * const rt5645_if3_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_if3_adc_in_enum, RT5645_DIG_INF1_DATA, + RT5645_IF3_ADC_IN_SFT, rt5645_if3_adc_in_src); + +static const struct snd_kcontrol_new rt5645_if3_adc_in_mux = + SOC_DAPM_ENUM("IF3 ADC IN source", rt5645_if3_adc_in_enum); + +/* MX-31 [15] [13] [11] [9] */ +static const char * const rt5645_pdm_src[] = { + "Mono DAC", "Stereo DAC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_pdm1_l_enum, RT5645_PDM_OUT_CTRL, + RT5645_PDM1_L_SFT, rt5645_pdm_src); + +static const struct snd_kcontrol_new rt5645_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 L source", rt5645_pdm1_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5645_pdm1_r_enum, RT5645_PDM_OUT_CTRL, + RT5645_PDM1_R_SFT, rt5645_pdm_src); + +static const struct snd_kcontrol_new rt5645_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 R source", rt5645_pdm1_r_enum); + +/* MX-9D [9:8] */ +static const char * const rt5645_vad_adc_src[] = { + "Sto1 ADC L", "Mono ADC L", "Mono ADC R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_vad_adc_enum, RT5645_VAD_CTRL4, + RT5645_VAD_SEL_SFT, rt5645_vad_adc_src); + +static const struct snd_kcontrol_new rt5645_vad_adc_mux = + SOC_DAPM_ENUM("VAD ADC source", rt5645_vad_adc_enum); + +static const struct snd_kcontrol_new spk_l_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_SPK_VOL, + RT5645_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_SPK_VOL, + RT5645_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_HP_VOL, + RT5645_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_HP_VOL, + RT5645_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new pdm1_l_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_PDM_OUT_CTRL, + RT5645_M_PDM1_L, 1, 1); + +static const struct snd_kcontrol_new pdm1_r_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_PDM_OUT_CTRL, + RT5645_M_PDM1_R, 1, 1); + +static void hp_amp_power(struct snd_soc_codec *codec, int on) +{ + static int hp_amp_power_count; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + if (on) { + if (hp_amp_power_count <= 0) { + /* depop parameters */ + snd_soc_update_bits(codec, RT5645_DEPOP_M2, + RT5645_DEPOP_MASK, RT5645_DEPOP_MAN); + snd_soc_write(codec, RT5645_DEPOP_M1, 0x000d); + regmap_write(rt5645->regmap, RT5645_PR_BASE + + RT5645_HP_DCC_INT1, 0x9f01); + mdelay(150); + /* headphone amp power on */ + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2 , 0); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_HV_L | RT5645_PWR_HV_R, + RT5645_PWR_HV_L | RT5645_PWR_HV_R); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_HP_L | RT5645_PWR_HP_R | + RT5645_PWR_HA, + RT5645_PWR_HP_L | RT5645_PWR_HP_R | + RT5645_PWR_HA); + mdelay(5); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2, + RT5645_PWR_FV1 | RT5645_PWR_FV2); + + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_CO_MASK | RT5645_HP_SG_MASK, + RT5645_HP_CO_EN | RT5645_HP_SG_EN); + regmap_write(rt5645->regmap, RT5645_PR_BASE + + 0x14, 0x1aaa); + regmap_write(rt5645->regmap, RT5645_PR_BASE + + 0x24, 0x0430); + } + hp_amp_power_count++; + } else { + hp_amp_power_count--; + if (hp_amp_power_count <= 0) { + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_SG_MASK | RT5645_HP_L_SMT_MASK | + RT5645_HP_R_SMT_MASK, RT5645_HP_SG_DIS | + RT5645_HP_L_SMT_DIS | RT5645_HP_R_SMT_DIS); + /* headphone amp power down */ + snd_soc_write(codec, RT5645_DEPOP_M1, 0x0000); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_HP_L | RT5645_PWR_HP_R | + RT5645_PWR_HA, 0); + } + } +} + +static int rt5645_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power(codec, 1); + /* headphone unmute sequence */ + snd_soc_update_bits(codec, RT5645_DEPOP_M3, RT5645_CP_FQ1_MASK | + RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + regmap_write(rt5645->regmap, + RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_SMT_TRIG_MASK, RT5645_SMT_TRIG_EN); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_RSTN_MASK, RT5645_RSTN_EN); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_RSTN_MASK | RT5645_HP_L_SMT_MASK | + RT5645_HP_R_SMT_MASK, RT5645_RSTN_DIS | + RT5645_HP_L_SMT_EN | RT5645_HP_R_SMT_EN); + msleep(40); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_SG_MASK | RT5645_HP_L_SMT_MASK | + RT5645_HP_R_SMT_MASK, RT5645_HP_SG_DIS | + RT5645_HP_L_SMT_DIS | RT5645_HP_R_SMT_DIS); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* headphone mute sequence */ + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + regmap_write(rt5645->regmap, + RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_SG_MASK, RT5645_HP_SG_EN); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_RSTP_MASK, RT5645_RSTP_EN); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_RSTP_MASK | RT5645_HP_L_SMT_MASK | + RT5645_HP_R_SMT_MASK, RT5645_RSTP_DIS | + RT5645_HP_L_SMT_EN | RT5645_HP_R_SMT_EN); + msleep(30); + hp_amp_power(codec, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5645_PWR_DIG1, + RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | + RT5645_PWR_CLS_D_L, + RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | + RT5645_PWR_CLS_D_L); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_PWR_DIG1, + RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | + RT5645_PWR_CLS_D_L, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_lout_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: + hp_amp_power(codec, 1); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_LM, RT5645_PWR_LM); + snd_soc_update_bits(codec, RT5645_LOUT1, + RT5645_L_MUTE | RT5645_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_LOUT1, + RT5645_L_MUTE | RT5645_R_MUTE, + RT5645_L_MUTE | RT5645_R_MUTE); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_LM, 0); + hp_amp_power(codec, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_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, RT5645_PWR_ANLG2, + RT5645_PWR_BST2_P, RT5645_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5645_PWR_MIXER, + RT5645_PWR_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5645_PWR_ANLG2, + RT5645_PWR_PLL_BIT, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("JD Power", RT5645_PWR_ANLG2, + RT5645_PWR_JD1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5645_PWR_VOL, + RT5645_PWR_MIC_DET_BIT, 0, NULL, 0), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_MICBIAS("micbias1", RT5645_PWR_ANLG2, + RT5645_PWR_MB1_BIT, 0), + SND_SOC_DAPM_MICBIAS("micbias2", RT5645_PWR_ANLG2, + RT5645_PWR_MB2_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("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_INPUT("Haptic Generator"), + + 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_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5645_DMIC_CTRL1, + RT5645_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5645_DMIC_CTRL1, + RT5645_DMIC_2_EN_SFT, 0, NULL, 0), + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5645_PWR_ANLG2, + RT5645_PWR_BST1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("BST2", RT5645_PWR_ANLG2, + RT5645_PWR_BST2_BIT, 0, NULL, 0, rt5645_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5645_PWR_VOL, + RT5645_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5645_PWR_VOL, + RT5645_PWR_IN_R_BIT, 0, NULL, 0), + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5645_PWR_MIXER, RT5645_PWR_RM_L_BIT, + 0, rt5645_rec_l_mix, ARRAY_SIZE(rt5645_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5645_PWR_MIXER, RT5645_PWR_RM_R_BIT, + 0, rt5645_rec_r_mix, ARRAY_SIZE(rt5645_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC L power", RT5645_PWR_DIG1, + RT5645_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC R power", RT5645_PWR_DIG1, + RT5645_PWR_ADC_R_BIT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto_adc2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto_adc2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto_adc1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto_adc1_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_adc_r2_mux), + /* ADC Mixer */ + + SND_SOC_DAPM_SUPPLY_S("adc stereo1 filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("adc stereo2 filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_sto1_adc_l_mix, ARRAY_SIZE(rt5645_sto1_adc_l_mix), + NULL, 0), + SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_sto1_adc_r_mix, ARRAY_SIZE(rt5645_sto1_adc_r_mix), + NULL, 0), + SND_SOC_DAPM_SUPPLY_S("adc mono left filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_mono_adc_l_mix, ARRAY_SIZE(rt5645_mono_adc_l_mix), + NULL, 0), + SND_SOC_DAPM_SUPPLY_S("adc mono right filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_mono_adc_r_mix, ARRAY_SIZE(rt5645_mono_adc_r_mix), + NULL, 0), + + /* 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("Sto2 ADC LR 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("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), + + /* IF1 2 Mux */ + SND_SOC_DAPM_MUX("IF1 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc_in_mux), + SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if2_adc_in_mux), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5645_PWR_DIG1, + RT5645_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", RT5645_PWR_DIG1, + RT5645_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), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_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, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5645_dac_l_mix, ARRAY_SIZE(rt5645_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5645_dac_r_mix, ARRAY_SIZE(rt5645_dac_r_mix)), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5645_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5645_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", RT5645_PWR_DIG1, + RT5645_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", RT5645_PWR_DIG1, + RT5645_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5645_dac1l_mux), + SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, &rt5645_dac1r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY_S("dac stereo1 filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("dac mono left filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_DAC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("dac mono right filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_DAC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_sto_dac_l_mix, ARRAY_SIZE(rt5645_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_sto_dac_r_mix, ARRAY_SIZE(rt5645_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_mono_dac_l_mix, ARRAY_SIZE(rt5645_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_mono_dac_r_mix, ARRAY_SIZE(rt5645_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_dig_l_mix, ARRAY_SIZE(rt5645_dig_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_dig_r_mix, ARRAY_SIZE(rt5645_dig_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5645_PWR_DIG1, RT5645_PWR_DAC_L1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5645_PWR_DIG1, RT5645_PWR_DAC_L2_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5645_PWR_DIG1, RT5645_PWR_DAC_R1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5645_PWR_DIG1, RT5645_PWR_DAC_R2_BIT, + 0), + /* OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIXL", RT5645_PWR_MIXER, RT5645_PWR_SM_L_BIT, + 0, rt5645_spk_l_mix, ARRAY_SIZE(rt5645_spk_l_mix)), + SND_SOC_DAPM_MIXER("SPK MIXR", RT5645_PWR_MIXER, RT5645_PWR_SM_R_BIT, + 0, rt5645_spk_r_mix, ARRAY_SIZE(rt5645_spk_r_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5645_PWR_MIXER, RT5645_PWR_OM_L_BIT, + 0, rt5645_out_l_mix, ARRAY_SIZE(rt5645_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5645_PWR_MIXER, RT5645_PWR_OM_R_BIT, + 0, rt5645_out_r_mix, ARRAY_SIZE(rt5645_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_SWITCH("SPKVOL L", RT5645_PWR_VOL, RT5645_PWR_SV_L_BIT, 0, + &spk_l_vol_control), + SND_SOC_DAPM_SWITCH("SPKVOL R", RT5645_PWR_VOL, RT5645_PWR_SV_R_BIT, 0, + &spk_r_vol_control), + SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5645_PWR_VOL, RT5645_PWR_HV_L_BIT, + 0, rt5645_hpvoll_mix, ARRAY_SIZE(rt5645_hpvoll_mix)), + SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5645_PWR_VOL, RT5645_PWR_HV_R_BIT, + 0, rt5645_hpvolr_mix, ARRAY_SIZE(rt5645_hpvolr_mix)), + SND_SOC_DAPM_SUPPLY("HPOVOL MIXL Power", RT5645_PWR_MIXER, + RT5645_PWR_HM_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOVOL MIXR Power", RT5645_PWR_MIXER, + RT5645_PWR_HM_R_BIT, 0, NULL, 0), + 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), + SND_SOC_DAPM_SWITCH("HPOVOL L", SND_SOC_NOPM, 0, 0, &hp_l_vol_control), + SND_SOC_DAPM_SWITCH("HPOVOL R", SND_SOC_NOPM, 0, 0, &hp_r_vol_control), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPOL MIX", SND_SOC_NOPM, 0, 0, rt5645_spo_l_mix, + ARRAY_SIZE(rt5645_spo_l_mix)), + SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5645_spo_r_mix, + ARRAY_SIZE(rt5645_spo_r_mix)), + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, rt5645_hpo_mix, + ARRAY_SIZE(rt5645_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, rt5645_lout_mix, + ARRAY_SIZE(rt5645_lout_mix)), + + SND_SOC_DAPM_PGA_S("HP amp", 1, SND_SOC_NOPM, 0, 0, rt5645_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0, rt5645_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 2, SND_SOC_NOPM, 0, 0, rt5645_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5645_PWR_DIG2, RT5645_PWR_PDM1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_MUX("PDM1 L Mux", SND_SOC_NOPM, 0, 0, &rt5645_pdm1_l_mux), + SND_SOC_DAPM_MUX("PDM1 R Mux", SND_SOC_NOPM, 0, 0, &rt5645_pdm1_r_mux), + + SND_SOC_DAPM_SWITCH("PDM1 L", SND_SOC_NOPM, 0, 0, &pdm1_l_vol_control), + SND_SOC_DAPM_SWITCH("PDM1 R", SND_SOC_NOPM, 0, 0, &pdm1_r_vol_control), + + /* 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("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), +}; + +static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { + { "IN1P", NULL, "LDO2" }, + { "IN2P", NULL, "LDO2" }, + + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "JD Power" }, + { "BST1", NULL, "Mic Det Power" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "INL VOL", NULL, "IN2P" }, + { "INR VOL", NULL, "IN2N" }, + + { "RECMIXL", "HPOL Switch", "HPOL" }, + { "RECMIXL", "INL Switch", "INL VOL" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + { "RECMIXL", "OUT MIXL Switch", "OUT MIXL" }, + + { "RECMIXR", "HPOR Switch", "HPOR" }, + { "RECMIXR", "INR Switch", "INR VOL" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + { "RECMIXR", "OUT MIXR Switch", "OUT MIXR" }, + + { "ADC L", NULL, "RECMIXL" }, + { "ADC L", NULL, "ADC L power" }, + { "ADC R", NULL, "RECMIXR" }, + { "ADC R", NULL, "ADC R power" }, + + {"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"}, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + + { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo1 ADC L1 Mux", "ADC", "ADC L" }, + { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo1 ADC R1 Mux", "ADC", "ADC R" }, + { "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", "ADC", "ADC L" }, + + { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + { "Mono ADC R1 Mux", "ADC", "ADC R" }, + { "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 }, + + { "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" }, + + { "IF_ADC1", NULL, "Stereo1 ADC MIXL" }, + { "IF_ADC1", NULL, "Stereo1 ADC MIXR" }, + { "IF_ADC2", NULL, "Mono ADC MIXL" }, + { "IF_ADC2", NULL, "Mono ADC MIXR" }, + { "VAD_ADC", NULL, "VAD ADC Mux" }, + + { "IF1 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF1 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF1 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF1 ADC", NULL, "I2S1" }, + { "IF1 ADC", NULL, "IF1 ADC Mux" }, + { "IF2 ADC", NULL, "I2S2" }, + { "IF2 ADC", NULL, "IF2 ADC Mux" }, + + { "AIF1TX", NULL, "IF1 ADC" }, + { "AIF1TX", NULL, "IF2 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 L2 Mux", "IF1 DAC", "IF1 DAC2 L" }, + { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" }, + { "DAC L2 Mux", "Mono ADC", "Mono ADC MIXL" }, + { "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", "Mono ADC", "Mono ADC MIXR" }, + { "DAC R2 Mux", "Haptic", "Haptic Generator" }, + { "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 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" }, + + { "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, "Stereo DAC MIXL" }, + { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, + { "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 }, + + { "SPK MIXL", "BST1 Switch", "BST1" }, + { "SPK MIXL", "INL Switch", "INL VOL" }, + { "SPK MIXL", "DAC L1 Switch", "DAC L1" }, + { "SPK MIXL", "DAC L2 Switch", "DAC L2" }, + { "SPK MIXR", "BST2 Switch", "BST2" }, + { "SPK MIXR", "INR Switch", "INR VOL" }, + { "SPK MIXR", "DAC R1 Switch", "DAC R1" }, + { "SPK MIXR", "DAC R2 Switch", "DAC R2" }, + + { "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", "DAC2 Switch", "DAC L2" }, + { "HPOVOL MIXL", "INL Switch", "INL VOL" }, + { "HPOVOL MIXL", "BST1 Switch", "BST1" }, + { "HPOVOL MIXL", NULL, "HPOVOL MIXL Power" }, + { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" }, + { "HPOVOL MIXR", "DAC2 Switch", "DAC R2" }, + { "HPOVOL MIXR", "INR Switch", "INR VOL" }, + { "HPOVOL MIXR", "BST2 Switch", "BST2" }, + { "HPOVOL MIXR", NULL, "HPOVOL MIXR Power" }, + + { "DAC 2", NULL, "DAC L2" }, + { "DAC 2", NULL, "DAC R2" }, + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + { "HPOVOL L", "Switch", "HPOVOL MIXL" }, + { "HPOVOL R", "Switch", "HPOVOL MIXR" }, + { "HPOVOL", NULL, "HPOVOL L" }, + { "HPOVOL", NULL, "HPOVOL R" }, + { "HPO MIX", "DAC1 Switch", "DAC 1" }, + { "HPO MIX", "HPVOL Switch", "HPOVOL" }, + + { "SPKVOL L", "Switch", "SPK MIXL" }, + { "SPKVOL R", "Switch", "SPK MIXR" }, + + { "SPOL MIX", "DAC R1 Switch", "DAC R1" }, + { "SPOL MIX", "DAC L1 Switch", "DAC L1" }, + { "SPOL MIX", "SPKVOL R Switch", "SPKVOL R" }, + { "SPOL MIX", "SPKVOL L Switch", "SPKVOL L" }, + { "SPOR MIX", "DAC R1 Switch", "DAC R1" }, + { "SPOR MIX", "SPKVOL R Switch", "SPKVOL R" }, + + { "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" }, + + { "HP amp", NULL, "HPO MIX" }, + { "HP amp", NULL, "JD Power" }, + { "HP amp", NULL, "Mic Det Power" }, + { "HP amp", NULL, "LDO2" }, + { "HPOL", NULL, "HP amp" }, + { "HPOR", NULL, "HP amp" }, + + { "LOUT amp", NULL, "LOUT MIX" }, + { "LOUTL", NULL, "LOUT amp" }, + { "LOUTR", NULL, "LOUT amp" }, + + { "PDM1 L", "Switch", "PDM1 L Mux" }, + { "PDM1 R", "Switch", "PDM1 R Mux" }, + + { "PDM1L", NULL, "PDM1 L" }, + { "PDM1R", NULL, "PDM1 R" }, + + { "SPK amp", NULL, "SPOL MIX" }, + { "SPK amp", NULL, "SPOR MIX" }, + { "SPOL", NULL, "SPK amp" }, + { "SPOR", NULL, "SPK amp" }, +}; + +static int get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5645_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 rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5645->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5645->sysclk, rt5645->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + 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; + rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5645->bclk[dai->id], rt5645->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 |= RT5645_I2S_DL_20; + break; + case 24: + val_len |= RT5645_I2S_DL_24; + break; + case 8: + val_len |= RT5645_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5645_AIF1: + mask_clk = RT5645_I2S_BCLK_MS1_MASK | RT5645_I2S_PD1_MASK; + val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | + pre_div << RT5645_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5645_I2S1_SDP, + RT5645_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5645_AIF2: + mask_clk = RT5645_I2S_BCLK_MS2_MASK | RT5645_I2S_PD2_MASK; + val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | + pre_div << RT5645_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5645_I2S2_SDP, + RT5645_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5645_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 rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5645->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5645_I2S_MS_S; + rt5645->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 |= RT5645_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 |= RT5645_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5645_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5645_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + switch (dai->id) { + case RT5645_AIF1: + snd_soc_update_bits(codec, RT5645_I2S1_SDP, + RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_DF_MASK, reg_val); + break; + case RT5645_AIF2: + snd_soc_update_bits(codec, RT5645_I2S1_SDP, + RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5645_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5645->sysclk && clk_id == rt5645->sysclk_src) + return 0; + + switch (clk_id) { + case RT5645_SCLK_S_MCLK: + reg_val |= RT5645_SCLK_SRC_MCLK; + break; + case RT5645_SCLK_S_PLL1: + reg_val |= RT5645_SCLK_SRC_PLL1; + break; + case RT5645_SCLK_S_RCCLK: + reg_val |= RT5645_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_SCLK_SRC_MASK, reg_val); + rt5645->sysclk = freq; + rt5645->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +/** + * rt5645_pll_calc - Calcualte PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calcualte M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5645_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5645_pll_code *pll_code) +{ + int max_n = RT5645_PLL_N_MAX, max_m = RT5645_PLL_M_MAX; + int k, n = 0, m = 0, red, n_t, m_t, pll_out, in_t, out_t; + int red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5645_PLL_INP_MAX < freq_in || RT5645_PLL_INP_MIN > freq_in) + return -EINVAL; + + k = 100000000 / freq_out - 2; + if (k > RT5645_PLL_K_MAX) + k = RT5645_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) { + bypass = true; + n = n_t; + goto code_find; + } + red = abs(in_t - pll_out); + if (red < red_t) { + bypass = 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) { + bypass = 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 = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = k; + return 0; +} + +static int rt5645_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 rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + struct rt5645_pll_code pll_code; + int ret; + + if (source == rt5645->pll_src && freq_in == rt5645->pll_in && + freq_out == rt5645->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5645->pll_in = 0; + rt5645->pll_out = 0; + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_SCLK_SRC_MASK, RT5645_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5645_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_PLL1_SRC_MASK, RT5645_PLL1_SRC_MCLK); + break; + case RT5645_PLL1_S_BCLK1: + case RT5645_PLL1_S_BCLK2: + switch (dai->id) { + case RT5645_AIF1: + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_PLL1_SRC_MASK, RT5645_PLL1_SRC_BCLK1); + break; + case RT5645_AIF2: + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_PLL1_SRC_MASK, RT5645_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 = rt5645_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, RT5645_PLL_CTRL1, + pll_code.n_code << RT5645_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5645_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5645_PLL_M_SFT | + pll_code.m_bp << RT5645_PLL_M_BP_SFT); + + rt5645->pll_in = freq_in; + rt5645->pll_out = freq_out; + rt5645->pll_src = source; + + return 0; +} + +static int rt5645_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, RT5645_TDM_CTRL_1, 0x7c00, val); + + return 0; +} + +static int rt5645_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2, + RT5645_PWR_FV1 | RT5645_PWR_FV2); + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); + snd_soc_write(codec, RT5645_GEN_CTRL1, 0x0128); + snd_soc_write(codec, RT5645_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5645_PWR_DIG2, 0x0000); + snd_soc_write(codec, RT5645_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5645_PWR_MIXER, 0x0000); + snd_soc_write(codec, RT5645_PWR_ANLG1, 0x0000); + snd_soc_write(codec, RT5645_PWR_ANLG2, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5645_probe(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + rt5645->codec = codec; + + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); + snd_soc_write(codec, RT5645_PR_BASE + 0x1c, 0xfd20); + snd_soc_write(codec, RT5645_PR_BASE + 0x20, 0x611f); + snd_soc_write(codec, RT5645_PR_BASE + 0x21, 0x4040); + snd_soc_write(codec, RT5645_PR_BASE + 0x23, 0x0004); + + return 0; +} + +static int rt5645_remove(struct snd_soc_codec *codec) +{ + rt5645_reset(codec); + return 0; +} + +#ifdef CONFIG_PM +static int rt5645_suspend(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5645->regmap, true); + regcache_mark_dirty(rt5645->regmap); + + return 0; +} + +static int rt5645_resume(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5645->regmap, false); + snd_soc_cache_sync(codec); + + return 0; +} +#else +#define rt5645_suspend NULL +#define rt5645_resume NULL +#endif + +#define RT5645_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5645_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 rt5645_aif_dai_ops = { + .hw_params = rt5645_hw_params, + .set_fmt = rt5645_set_dai_fmt, + .set_sysclk = rt5645_set_dai_sysclk, + .set_tdm_slot = rt5645_set_tdm_slot, + .set_pll = rt5645_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5645_dai[] = { + { + .name = "rt5645-aif1", + .id = RT5645_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5645_STEREO_RATES, + .formats = RT5645_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5645_STEREO_RATES, + .formats = RT5645_FORMATS, + }, + .ops = &rt5645_aif_dai_ops, + }, + { + .name = "rt5645-aif2", + .id = RT5645_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5645_STEREO_RATES, + .formats = RT5645_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5645_STEREO_RATES, + .formats = RT5645_FORMATS, + }, + .ops = &rt5645_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5645 = { + .probe = rt5645_probe, + .remove = rt5645_remove, + .suspend = rt5645_suspend, + .resume = rt5645_resume, + .set_bias_level = rt5645_set_bias_level, + .idle_bias_off = true, + .controls = rt5645_snd_controls, + .num_controls = ARRAY_SIZE(rt5645_snd_controls), + .dapm_widgets = rt5645_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5645_dapm_widgets), + .dapm_routes = rt5645_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5645_dapm_routes), +}; + +static const struct regmap_config rt5645_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5645_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5645_ranges) * + RT5645_PR_SPACING), + .volatile_reg = rt5645_volatile_register, + .readable_reg = rt5645_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5645_reg, + .num_reg_defaults = ARRAY_SIZE(rt5645_reg), + .ranges = rt5645_ranges, + .num_ranges = ARRAY_SIZE(rt5645_ranges), +}; + +static const struct i2c_device_id rt5645_i2c_id[] = { + { "rt5645", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); + +static int rt5645_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5645_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5645_priv *rt5645; + int ret; + unsigned int val; + + rt5645 = devm_kzalloc(&i2c->dev, sizeof(struct rt5645_priv), + GFP_KERNEL); + if (rt5645 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5645); + + if (pdata) + rt5645->pdata = *pdata; + + rt5645->regmap = devm_regmap_init_i2c(i2c, &rt5645_regmap); + if (IS_ERR(rt5645->regmap)) { + ret = PTR_ERR(rt5645->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val); + if (val != RT5645_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5645\n", val); + return -ENODEV; + } + + regmap_write(rt5645->regmap, RT5645_RESET, 0); + + ret = regmap_register_patch(rt5645->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5645->pdata.in2_diff) + regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, + RT5645_IN_DF2, RT5645_IN_DF2); + + if (rt5645->pdata.dmic_en) { + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP2_PIN_MASK, RT5645_GP2_PIN_DMIC1_SCL); + + switch (rt5645->pdata.dmic1_data_pin) { + case RT5645_DMIC_DATA_IN2N: + regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, + RT5645_DMIC_1_DP_MASK, RT5645_DMIC_1_DP_IN2N); + break; + + case RT5645_DMIC_DATA_GPIO5: + regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, + RT5645_DMIC_1_DP_MASK, RT5645_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP5_PIN_MASK, RT5645_GP5_PIN_DMIC1_SDA); + break; + + case RT5645_DMIC_DATA_GPIO11: + regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, + RT5645_DMIC_1_DP_MASK, RT5645_DMIC_1_DP_GPIO11); + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP11_PIN_MASK, + RT5645_GP11_PIN_DMIC1_SDA); + break; + + default: + break; + } + + switch (rt5645->pdata.dmic2_data_pin) { + case RT5645_DMIC_DATA_IN2P: + regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, + RT5645_DMIC_2_DP_MASK, RT5645_DMIC_2_DP_IN2P); + break; + + case RT5645_DMIC_DATA_GPIO6: + regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, + RT5645_DMIC_2_DP_MASK, RT5645_DMIC_2_DP_GPIO6); + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP6_PIN_MASK, RT5645_GP6_PIN_DMIC2_SDA); + break; + + case RT5645_DMIC_DATA_GPIO10: + regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, + RT5645_DMIC_2_DP_MASK, RT5645_DMIC_2_DP_GPIO10); + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP10_PIN_MASK, + RT5645_GP10_PIN_DMIC2_SDA); + break; + + case RT5645_DMIC_DATA_GPIO12: + regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, + RT5645_DMIC_1_DP_MASK, RT5645_DMIC_2_DP_GPIO12); + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP12_PIN_MASK, + RT5645_GP12_PIN_DMIC2_SDA); + break; + + default: + break; + } + + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, + rt5645_dai, ARRAY_SIZE(rt5645_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5645_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt5645_i2c_driver = { + .driver = { + .name = "rt5645", + .owner = THIS_MODULE, + }, + .probe = rt5645_i2c_probe, + .remove = rt5645_i2c_remove, + .id_table = rt5645_i2c_id, +}; +module_i2c_driver(rt5645_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5645 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h new file mode 100644 index 0000000..345aa3f --- /dev/null +++ b/sound/soc/codecs/rt5645.h @@ -0,0 +1,2188 @@ +/* + * rt5645.h -- RT5645 ALSA SoC audio driver + * + * Copyright 2013 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 __RT5645_H__ +#define __RT5645_H__ + +#include <sound/rt5645.h> + +/* Info */ +#define RT5645_RESET 0x00 +#define RT5645_VENDOR_ID 0xfd +#define RT5645_VENDOR_ID1 0xfe +#define RT5645_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5645_SPK_VOL 0x01 +#define RT5645_HP_VOL 0x02 +#define RT5645_LOUT1 0x03 +#define RT5645_LOUT_CTRL 0x05 +/* I/O - Input */ +#define RT5645_IN1_CTRL1 0x0a +#define RT5645_IN1_CTRL2 0x0b +#define RT5645_IN1_CTRL3 0x0c +#define RT5645_IN2_CTRL 0x0d +#define RT5645_INL1_INR1_VOL 0x0f +#define RT5645_SPK_FUNC_LIM 0x14 +#define RT5645_ADJ_HPF_CTRL 0x16 +/* I/O - ADC/DAC/DMIC */ +#define RT5645_DAC1_DIG_VOL 0x19 +#define RT5645_DAC2_DIG_VOL 0x1a +#define RT5645_DAC_CTRL 0x1b +#define RT5645_STO1_ADC_DIG_VOL 0x1c +#define RT5645_MONO_ADC_DIG_VOL 0x1d +#define RT5645_ADC_BST_VOL1 0x1e +/* Mixer - D-D */ +#define RT5645_ADC_BST_VOL2 0x20 +#define RT5645_STO1_ADC_MIXER 0x27 +#define RT5645_MONO_ADC_MIXER 0x28 +#define RT5645_AD_DA_MIXER 0x29 +#define RT5645_STO_DAC_MIXER 0x2a +#define RT5645_MONO_DAC_MIXER 0x2b +#define RT5645_DIG_MIXER 0x2c +#define RT5645_DIG_INF1_DATA 0x2f +/* Mixer - PDM */ +#define RT5645_PDM_OUT_CTRL 0x31 +/* Mixer - ADC */ +#define RT5645_REC_L1_MIXER 0x3b +#define RT5645_REC_L2_MIXER 0x3c +#define RT5645_REC_R1_MIXER 0x3d +#define RT5645_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5645_HPMIXL_CTRL 0x3f +#define RT5645_HPOMIXL_CTRL 0x40 +#define RT5645_HPMIXR_CTRL 0x41 +#define RT5645_HPOMIXR_CTRL 0x42 +#define RT5645_HPO_MIXER 0x45 +#define RT5645_SPK_L_MIXER 0x46 +#define RT5645_SPK_R_MIXER 0x47 +#define RT5645_SPO_MIXER 0x48 +#define RT5645_SPO_CLSD_RATIO 0x4a +#define RT5645_OUT_L_GAIN1 0x4d +#define RT5645_OUT_L_GAIN2 0x4e +#define RT5645_OUT_L1_MIXER 0x4f +#define RT5645_OUT_R_GAIN1 0x50 +#define RT5645_OUT_R_GAIN2 0x51 +#define RT5645_OUT_R1_MIXER 0x52 +#define RT5645_LOUT_MIXER 0x53 +/* Haptic */ +#define RT5645_HAPTIC_CTRL1 0x56 +#define RT5645_HAPTIC_CTRL2 0x57 +#define RT5645_HAPTIC_CTRL3 0x58 +#define RT5645_HAPTIC_CTRL4 0x59 +#define RT5645_HAPTIC_CTRL5 0x5a +#define RT5645_HAPTIC_CTRL6 0x5b +#define RT5645_HAPTIC_CTRL7 0x5c +#define RT5645_HAPTIC_CTRL8 0x5d +#define RT5645_HAPTIC_CTRL9 0x5e +#define RT5645_HAPTIC_CTRL10 0x5f +/* Power */ +#define RT5645_PWR_DIG1 0x61 +#define RT5645_PWR_DIG2 0x62 +#define RT5645_PWR_ANLG1 0x63 +#define RT5645_PWR_ANLG2 0x64 +#define RT5645_PWR_MIXER 0x65 +#define RT5645_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5645_PRIV_INDEX 0x6a +#define RT5645_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5645_I2S1_SDP 0x70 +#define RT5645_I2S2_SDP 0x71 +#define RT5645_ADDA_CLK1 0x73 +#define RT5645_ADDA_CLK2 0x74 +#define RT5645_DMIC_CTRL1 0x75 +#define RT5645_DMIC_CTRL2 0x76 +/* Format - TDM Control */ +#define RT5645_TDM_CTRL_1 0x77 +#define RT5645_TDM_CTRL_2 0x78 +#define RT5645_TDM_CTRL_3 0x79 + +/* Function - Analog */ +#define RT5645_GLB_CLK 0x80 +#define RT5645_PLL_CTRL1 0x81 +#define RT5645_PLL_CTRL2 0x82 +#define RT5645_ASRC_1 0x83 +#define RT5645_ASRC_2 0x84 +#define RT5645_ASRC_3 0x85 +#define RT5645_ASRC_4 0x8a +#define RT5645_DEPOP_M1 0x8e +#define RT5645_DEPOP_M2 0x8f +#define RT5645_DEPOP_M3 0x90 +#define RT5645_CHARGE_PUMP 0x91 +#define RT5645_MICBIAS 0x93 +#define RT5645_A_JD_CTRL1 0x94 +#define RT5645_VAD_CTRL4 0x9d +#define RT5645_CLSD_OUT_CTRL 0xa0 +/* Function - Digital */ +#define RT5645_ADC_EQ_CTRL1 0xae +#define RT5645_ADC_EQ_CTRL2 0xaf +#define RT5645_EQ_CTRL1 0xb0 +#define RT5645_EQ_CTRL2 0xb1 +#define RT5645_ALC_CTRL_1 0xb3 +#define RT5645_ALC_CTRL_2 0xb4 +#define RT5645_ALC_CTRL_3 0xb5 +#define RT5645_ALC_CTRL_4 0xb6 +#define RT5645_ALC_CTRL_5 0xb7 +#define RT5645_JD_CTRL 0xbb +#define RT5645_IRQ_CTRL1 0xbc +#define RT5645_IRQ_CTRL2 0xbd +#define RT5645_IRQ_CTRL3 0xbe +#define RT5645_INT_IRQ_ST 0xbf +#define RT5645_GPIO_CTRL1 0xc0 +#define RT5645_GPIO_CTRL2 0xc1 +#define RT5645_GPIO_CTRL3 0xc2 +#define RT5645_BASS_BACK 0xcf +#define RT5645_MP3_PLUS1 0xd0 +#define RT5645_MP3_PLUS2 0xd1 +#define RT5645_ADJ_HPF1 0xd3 +#define RT5645_ADJ_HPF2 0xd4 +#define RT5645_HP_CALIB_AMP_DET 0xd6 +#define RT5645_SV_ZCD1 0xd9 +#define RT5645_SV_ZCD2 0xda +#define RT5645_IL_CMD 0xdb +#define RT5645_IL_CMD2 0xdc +#define RT5645_IL_CMD3 0xdd +#define RT5645_DRC1_HL_CTRL1 0xe7 +#define RT5645_DRC2_HL_CTRL1 0xe9 +#define RT5645_MUTI_DRC_CTRL1 0xea +#define RT5645_ADC_MONO_HP_CTRL1 0xec +#define RT5645_ADC_MONO_HP_CTRL2 0xed +#define RT5645_DRC2_CTRL1 0xf0 +#define RT5645_DRC2_CTRL2 0xf1 +#define RT5645_DRC2_CTRL3 0xf2 +#define RT5645_DRC2_CTRL4 0xf3 +#define RT5645_DRC2_CTRL5 0xf4 +#define RT5645_JD_CTRL3 0xf8 +#define RT5645_JD_CTRL4 0xf9 +/* General Control */ +#define RT5645_GEN_CTRL1 0xfa +#define RT5645_GEN_CTRL2 0xfb +#define RT5645_GEN_CTRL3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5645_DIG_VOL 0x00 +#define RT5645_PR_ALC_CTRL_1 0x01 +#define RT5645_PR_ALC_CTRL_2 0x02 +#define RT5645_PR_ALC_CTRL_3 0x03 +#define RT5645_PR_ALC_CTRL_4 0x04 +#define RT5645_PR_ALC_CTRL_5 0x05 +#define RT5645_PR_ALC_CTRL_6 0x06 +#define RT5645_BIAS_CUR1 0x12 +#define RT5645_BIAS_CUR3 0x14 +#define RT5645_CLSD_INT_REG1 0x1c +#define RT5645_MAMP_INT_REG2 0x37 +#define RT5645_CHOP_DAC_ADC 0x3d +#define RT5645_MIXER_INT_REG 0x3f +#define RT5645_3D_SPK 0x63 +#define RT5645_WND_1 0x6c +#define RT5645_WND_2 0x6d +#define RT5645_WND_3 0x6e +#define RT5645_WND_4 0x6f +#define RT5645_WND_5 0x70 +#define RT5645_WND_8 0x73 +#define RT5645_DIP_SPK_INF 0x75 +#define RT5645_HP_DCC_INT1 0x77 +#define RT5645_EQ_BW_LOP 0xa0 +#define RT5645_EQ_GN_LOP 0xa1 +#define RT5645_EQ_FC_BP1 0xa2 +#define RT5645_EQ_BW_BP1 0xa3 +#define RT5645_EQ_GN_BP1 0xa4 +#define RT5645_EQ_FC_BP2 0xa5 +#define RT5645_EQ_BW_BP2 0xa6 +#define RT5645_EQ_GN_BP2 0xa7 +#define RT5645_EQ_FC_BP3 0xa8 +#define RT5645_EQ_BW_BP3 0xa9 +#define RT5645_EQ_GN_BP3 0xaa +#define RT5645_EQ_FC_BP4 0xab +#define RT5645_EQ_BW_BP4 0xac +#define RT5645_EQ_GN_BP4 0xad +#define RT5645_EQ_FC_HIP1 0xae +#define RT5645_EQ_GN_HIP1 0xaf +#define RT5645_EQ_FC_HIP2 0xb0 +#define RT5645_EQ_BW_HIP2 0xb1 +#define RT5645_EQ_GN_HIP2 0xb2 +#define RT5645_EQ_PRE_VOL 0xb3 +#define RT5645_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5645_L_MUTE (0x1 << 15) +#define RT5645_L_MUTE_SFT 15 +#define RT5645_VOL_L_MUTE (0x1 << 14) +#define RT5645_VOL_L_SFT 14 +#define RT5645_R_MUTE (0x1 << 7) +#define RT5645_R_MUTE_SFT 7 +#define RT5645_VOL_R_MUTE (0x1 << 6) +#define RT5645_VOL_R_SFT 6 +#define RT5645_L_VOL_MASK (0x3f << 8) +#define RT5645_L_VOL_SFT 8 +#define RT5645_R_VOL_MASK (0x3f) +#define RT5645_R_VOL_SFT 0 + +/* IN1 Control 1 (0x0a) */ +#define RT5645_CBJ_BST1_MASK (0xf << 12) +#define RT5645_CBJ_BST1_SFT (12) +#define RT5645_CBJ_JD_HP_EN (0x1 << 9) +#define RT5645_CBJ_JD_MIC_EN (0x1 << 8) +#define RT5645_CBJ_JD_MIC_SW_EN (0x1 << 7) +#define RT5645_CBJ_MIC_SEL_R (0x1 << 6) +#define RT5645_CBJ_MIC_SEL_L (0x1 << 5) +#define RT5645_CBJ_MIC_SW (0x1 << 4) +#define RT5645_CBJ_BST1_EN (0x1 << 2) + +/* IN1 Control 2 (0x0b) */ +#define RT5645_CBJ_MN_JD (0x1 << 12) +#define RT5645_CAPLESS_EN (0x1 << 11) +#define RT5645_CBJ_DET_MODE (0x1 << 7) + +/* IN1 Control 3 (0x0c) */ +#define RT5645_CBJ_TIE_G_L (0x1 << 15) +#define RT5645_CBJ_TIE_G_R (0x1 << 14) + +/* IN2 Control (0x0d) */ +#define RT5645_BST_MASK1 (0xf<<12) +#define RT5645_BST_SFT1 12 +#define RT5645_BST_MASK2 (0xf<<8) +#define RT5645_BST_SFT2 8 +#define RT5645_IN_DF2 (0x1 << 6) +#define RT5645_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5645_INL_SEL_MASK (0x1 << 15) +#define RT5645_INL_SEL_SFT 15 +#define RT5645_INL_SEL_IN4P (0x0 << 15) +#define RT5645_INL_SEL_MONOP (0x1 << 15) +#define RT5645_INL_VOL_MASK (0x1f << 8) +#define RT5645_INL_VOL_SFT 8 +#define RT5645_INR_SEL_MASK (0x1 << 7) +#define RT5645_INR_SEL_SFT 7 +#define RT5645_INR_SEL_IN4N (0x0 << 7) +#define RT5645_INR_SEL_MONON (0x1 << 7) +#define RT5645_INR_VOL_MASK (0x1f) +#define RT5645_INR_VOL_SFT 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5645_DAC_L1_VOL_MASK (0xff << 8) +#define RT5645_DAC_L1_VOL_SFT 8 +#define RT5645_DAC_R1_VOL_MASK (0xff) +#define RT5645_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5645_DAC_L2_VOL_MASK (0xff << 8) +#define RT5645_DAC_L2_VOL_SFT 8 +#define RT5645_DAC_R2_VOL_MASK (0xff) +#define RT5645_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5645_M_DAC_L2_VOL (0x1 << 13) +#define RT5645_M_DAC_L2_VOL_SFT 13 +#define RT5645_M_DAC_R2_VOL (0x1 << 12) +#define RT5645_M_DAC_R2_VOL_SFT 12 +#define RT5645_DAC2_L_SEL_MASK (0x7 << 4) +#define RT5645_DAC2_L_SEL_SFT 4 +#define RT5645_DAC2_R_SEL_MASK (0x7 << 0) +#define RT5645_DAC2_R_SEL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5645_ADC_L_VOL_MASK (0x7f << 8) +#define RT5645_ADC_L_VOL_SFT 8 +#define RT5645_ADC_R_VOL_MASK (0x7f) +#define RT5645_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5645_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5645_MONO_ADC_L_VOL_SFT 8 +#define RT5645_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5645_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5645_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5645_STO1_ADC_L_BST_SFT 14 +#define RT5645_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5645_STO1_ADC_R_BST_SFT 12 +#define RT5645_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5645_STO1_ADC_COMP_SFT 10 +#define RT5645_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5645_STO2_ADC_L_BST_SFT 8 +#define RT5645_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5645_STO2_ADC_R_BST_SFT 6 +#define RT5645_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5645_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5645_STO2_ADC_SRC_MASK (0x1 << 15) +#define RT5645_STO2_ADC_SRC_SFT 15 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5645_M_ADC_L1 (0x1 << 14) +#define RT5645_M_ADC_L1_SFT 14 +#define RT5645_M_ADC_L2 (0x1 << 13) +#define RT5645_M_ADC_L2_SFT 13 +#define RT5645_ADC_1_SRC_MASK (0x1 << 12) +#define RT5645_ADC_1_SRC_SFT 12 +#define RT5645_ADC_1_SRC_ADC (0x1 << 12) +#define RT5645_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5645_ADC_2_SRC_MASK (0x1 << 11) +#define RT5645_ADC_2_SRC_SFT 11 +#define RT5645_DMIC_SRC_MASK (0x1 << 8) +#define RT5645_DMIC_SRC_SFT 8 +#define RT5645_M_ADC_R1 (0x1 << 6) +#define RT5645_M_ADC_R1_SFT 6 +#define RT5645_M_ADC_R2 (0x1 << 5) +#define RT5645_M_ADC_R2_SFT 5 +#define RT5645_DMIC3_SRC_MASK (0x1 << 1) +#define RT5645_DMIC3_SRC_SFT 0 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5645_M_MONO_ADC_L1 (0x1 << 14) +#define RT5645_M_MONO_ADC_L1_SFT 14 +#define RT5645_M_MONO_ADC_L2 (0x1 << 13) +#define RT5645_M_MONO_ADC_L2_SFT 13 +#define RT5645_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5645_MONO_ADC_L1_SRC_SFT 12 +#define RT5645_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5645_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5645_MONO_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5645_MONO_ADC_L2_SRC_SFT 11 +#define RT5645_MONO_DMIC_L_SRC_MASK (0x1 << 8) +#define RT5645_MONO_DMIC_L_SRC_SFT 8 +#define RT5645_M_MONO_ADC_R1 (0x1 << 6) +#define RT5645_M_MONO_ADC_R1_SFT 6 +#define RT5645_M_MONO_ADC_R2 (0x1 << 5) +#define RT5645_M_MONO_ADC_R2_SFT 5 +#define RT5645_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5645_MONO_ADC_R1_SRC_SFT 4 +#define RT5645_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5645_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5645_MONO_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5645_MONO_ADC_R2_SRC_SFT 3 +#define RT5645_MONO_DMIC_R_SRC_MASK (0x3) +#define RT5645_MONO_DMIC_R_SRC_SFT 0 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5645_M_ADCMIX_L (0x1 << 15) +#define RT5645_M_ADCMIX_L_SFT 15 +#define RT5645_M_DAC1_L (0x1 << 14) +#define RT5645_M_DAC1_L_SFT 14 +#define RT5645_DAC1_R_SEL_MASK (0x3 << 10) +#define RT5645_DAC1_R_SEL_SFT 10 +#define RT5645_DAC1_R_SEL_IF1 (0x0 << 10) +#define RT5645_DAC1_R_SEL_IF2 (0x1 << 10) +#define RT5645_DAC1_R_SEL_IF3 (0x2 << 10) +#define RT5645_DAC1_R_SEL_IF4 (0x3 << 10) +#define RT5645_DAC1_L_SEL_MASK (0x3 << 8) +#define RT5645_DAC1_L_SEL_SFT 8 +#define RT5645_DAC1_L_SEL_IF1 (0x0 << 8) +#define RT5645_DAC1_L_SEL_IF2 (0x1 << 8) +#define RT5645_DAC1_L_SEL_IF3 (0x2 << 8) +#define RT5645_DAC1_L_SEL_IF4 (0x3 << 8) +#define RT5645_M_ADCMIX_R (0x1 << 7) +#define RT5645_M_ADCMIX_R_SFT 7 +#define RT5645_M_DAC1_R (0x1 << 6) +#define RT5645_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5645_M_DAC_L1 (0x1 << 14) +#define RT5645_M_DAC_L1_SFT 14 +#define RT5645_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5645_DAC_L1_STO_L_VOL_SFT 13 +#define RT5645_M_DAC_L2 (0x1 << 12) +#define RT5645_M_DAC_L2_SFT 12 +#define RT5645_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5645_DAC_L2_STO_L_VOL_SFT 11 +#define RT5645_M_ANC_DAC_L (0x1 << 10) +#define RT5645_M_ANC_DAC_L_SFT 10 +#define RT5645_M_DAC_R1_STO_L (0x1 << 9) +#define RT5645_M_DAC_R1_STO_L_SFT 9 +#define RT5645_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5645_DAC_R1_STO_L_VOL_SFT 8 +#define RT5645_M_DAC_R1 (0x1 << 6) +#define RT5645_M_DAC_R1_SFT 6 +#define RT5645_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5645_DAC_R1_STO_R_VOL_SFT 5 +#define RT5645_M_DAC_R2 (0x1 << 4) +#define RT5645_M_DAC_R2_SFT 4 +#define RT5645_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5645_DAC_R2_STO_R_VOL_SFT 3 +#define RT5645_M_ANC_DAC_R (0x1 << 2) +#define RT5645_M_ANC_DAC_R_SFT 2 +#define RT5645_M_DAC_L1_STO_R (0x1 << 1) +#define RT5645_M_DAC_L1_STO_R_SFT 1 +#define RT5645_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5645_DAC_L1_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5645_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5645_M_DAC_L1_MONO_L_SFT 14 +#define RT5645_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5645_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5645_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5645_M_DAC_L2_MONO_L_SFT 12 +#define RT5645_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5645_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5645_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5645_M_DAC_R2_MONO_L_SFT 10 +#define RT5645_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5645_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5645_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5645_M_DAC_R1_MONO_R_SFT 6 +#define RT5645_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5645_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5645_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5645_M_DAC_R2_MONO_R_SFT 4 +#define RT5645_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5645_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5645_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5645_M_DAC_L2_MONO_R_SFT 2 +#define RT5645_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5645_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5645_M_STO_L_DAC_L (0x1 << 15) +#define RT5645_M_STO_L_DAC_L_SFT 15 +#define RT5645_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5645_STO_L_DAC_L_VOL_SFT 14 +#define RT5645_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5645_M_DAC_L2_DAC_L_SFT 13 +#define RT5645_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5645_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5645_M_STO_R_DAC_R (0x1 << 11) +#define RT5645_M_STO_R_DAC_R_SFT 11 +#define RT5645_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5645_STO_R_DAC_R_VOL_SFT 10 +#define RT5645_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5645_M_DAC_R2_DAC_R_SFT 9 +#define RT5645_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5645_DAC_R2_DAC_R_VOL_SFT 8 +#define RT5645_M_DAC_R2_DAC_L (0x1 << 7) +#define RT5645_M_DAC_R2_DAC_L_SFT 7 +#define RT5645_DAC_R2_DAC_L_VOL_MASK (0x1 << 6) +#define RT5645_DAC_R2_DAC_L_VOL_SFT 6 +#define RT5645_M_DAC_L2_DAC_R (0x1 << 5) +#define RT5645_M_DAC_L2_DAC_R_SFT 5 +#define RT5645_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) +#define RT5645_DAC_L2_DAC_R_VOL_SFT 4 + +/* Digital Interface Data Control (0x2f) */ +#define RT5645_IF1_ADC2_IN_SEL (0x1 << 15) +#define RT5645_IF1_ADC2_IN_SFT 15 +#define RT5645_IF2_ADC_IN_MASK (0x7 << 12) +#define RT5645_IF2_ADC_IN_SFT 12 +#define RT5645_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5645_IF2_DAC_SEL_SFT 10 +#define RT5645_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5645_IF2_ADC_SEL_SFT 8 +#define RT5645_IF3_DAC_SEL_MASK (0x3 << 6) +#define RT5645_IF3_DAC_SEL_SFT 6 +#define RT5645_IF3_ADC_SEL_MASK (0x3 << 4) +#define RT5645_IF3_ADC_SEL_SFT 4 +#define RT5645_IF3_ADC_IN_MASK (0x7) +#define RT5645_IF3_ADC_IN_SFT 0 + +/* PDM Output Control (0x31) */ +#define RT5645_PDM1_L_MASK (0x1 << 15) +#define RT5645_PDM1_L_SFT 15 +#define RT5645_M_PDM1_L (0x1 << 14) +#define RT5645_M_PDM1_L_SFT 14 +#define RT5645_PDM1_R_MASK (0x1 << 13) +#define RT5645_PDM1_R_SFT 13 +#define RT5645_M_PDM1_R (0x1 << 12) +#define RT5645_M_PDM1_R_SFT 12 +#define RT5645_PDM2_L_MASK (0x1 << 11) +#define RT5645_PDM2_L_SFT 11 +#define RT5645_M_PDM2_L (0x1 << 10) +#define RT5645_M_PDM2_L_SFT 10 +#define RT5645_PDM2_R_MASK (0x1 << 9) +#define RT5645_PDM2_R_SFT 9 +#define RT5645_M_PDM2_R (0x1 << 8) +#define RT5645_M_PDM2_R_SFT 8 +#define RT5645_PDM2_BUSY (0x1 << 7) +#define RT5645_PDM1_BUSY (0x1 << 6) +#define RT5645_PDM_PATTERN (0x1 << 5) +#define RT5645_PDM_GAIN (0x1 << 4) +#define RT5645_PDM_DIV_MASK (0x3) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5645_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5645_G_HP_L_RM_L_SFT 13 +#define RT5645_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5645_G_IN_L_RM_L_SFT 10 +#define RT5645_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5645_G_BST4_RM_L_SFT 7 +#define RT5645_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5645_G_BST3_RM_L_SFT 4 +#define RT5645_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5645_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5645_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5645_G_BST1_RM_L_SFT 13 +#define RT5645_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5645_G_OM_L_RM_L_SFT 10 +#define RT5645_M_MM_L_RM_L (0x1 << 6) +#define RT5645_M_MM_L_RM_L_SFT 6 +#define RT5645_M_IN_L_RM_L (0x1 << 5) +#define RT5645_M_IN_L_RM_L_SFT 5 +#define RT5645_M_HP_L_RM_L (0x1 << 4) +#define RT5645_M_HP_L_RM_L_SFT 4 +#define RT5645_M_BST3_RM_L (0x1 << 3) +#define RT5645_M_BST3_RM_L_SFT 3 +#define RT5645_M_BST2_RM_L (0x1 << 2) +#define RT5645_M_BST2_RM_L_SFT 2 +#define RT5645_M_BST1_RM_L (0x1 << 1) +#define RT5645_M_BST1_RM_L_SFT 1 +#define RT5645_M_OM_L_RM_L (0x1) +#define RT5645_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5645_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5645_G_HP_R_RM_R_SFT 13 +#define RT5645_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5645_G_IN_R_RM_R_SFT 10 +#define RT5645_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5645_G_BST4_RM_R_SFT 7 +#define RT5645_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5645_G_BST3_RM_R_SFT 4 +#define RT5645_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5645_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5645_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5645_G_BST1_RM_R_SFT 13 +#define RT5645_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5645_G_OM_R_RM_R_SFT 10 +#define RT5645_M_MM_R_RM_R (0x1 << 6) +#define RT5645_M_MM_R_RM_R_SFT 6 +#define RT5645_M_IN_R_RM_R (0x1 << 5) +#define RT5645_M_IN_R_RM_R_SFT 5 +#define RT5645_M_HP_R_RM_R (0x1 << 4) +#define RT5645_M_HP_R_RM_R_SFT 4 +#define RT5645_M_BST3_RM_R (0x1 << 3) +#define RT5645_M_BST3_RM_R_SFT 3 +#define RT5645_M_BST2_RM_R (0x1 << 2) +#define RT5645_M_BST2_RM_R_SFT 2 +#define RT5645_M_BST1_RM_R (0x1 << 1) +#define RT5645_M_BST1_RM_R_SFT 1 +#define RT5645_M_OM_R_RM_R (0x1) +#define RT5645_M_OM_R_RM_R_SFT 0 + +/* HPOMIX Control (0x40) (0x42) */ +#define RT5645_M_BST1_HV (0x1 << 4) +#define RT5645_M_BST1_HV_SFT 4 +#define RT5645_M_BST2_HV (0x1 << 4) +#define RT5645_M_BST2_HV_SFT 4 +#define RT5645_M_BST3_HV (0x1 << 3) +#define RT5645_M_BST3_HV_SFT 3 +#define RT5645_M_IN_HV (0x1 << 2) +#define RT5645_M_IN_HV_SFT 2 +#define RT5645_M_DAC2_HV (0x1 << 1) +#define RT5645_M_DAC2_HV_SFT 1 +#define RT5645_M_DAC1_HV (0x1 << 0) +#define RT5645_M_DAC1_HV_SFT 0 + +/* HPMIX Control (0x45) */ +#define RT5645_M_DAC1_HM (0x1 << 14) +#define RT5645_M_DAC1_HM_SFT 14 +#define RT5645_M_HPVOL_HM (0x1 << 13) +#define RT5645_M_HPVOL_HM_SFT 13 + +/* SPK Left Mixer Control (0x46) */ +#define RT5645_G_RM_L_SM_L_MASK (0x3 << 14) +#define RT5645_G_RM_L_SM_L_SFT 14 +#define RT5645_G_IN_L_SM_L_MASK (0x3 << 12) +#define RT5645_G_IN_L_SM_L_SFT 12 +#define RT5645_G_DAC_L1_SM_L_MASK (0x3 << 10) +#define RT5645_G_DAC_L1_SM_L_SFT 10 +#define RT5645_G_DAC_L2_SM_L_MASK (0x3 << 8) +#define RT5645_G_DAC_L2_SM_L_SFT 8 +#define RT5645_G_OM_L_SM_L_MASK (0x3 << 6) +#define RT5645_G_OM_L_SM_L_SFT 6 +#define RT5645_M_BST1_L_SM_L (0x1 << 5) +#define RT5645_M_BST1_L_SM_L_SFT 5 +#define RT5645_M_IN_L_SM_L (0x1 << 3) +#define RT5645_M_IN_L_SM_L_SFT 3 +#define RT5645_M_DAC_L1_SM_L (0x1 << 1) +#define RT5645_M_DAC_L1_SM_L_SFT 1 +#define RT5645_M_DAC_L2_SM_L (0x1 << 2) +#define RT5645_M_DAC_L2_SM_L_SFT 2 +#define RT5645_M_BST3_L_SM_L (0x1 << 4) +#define RT5645_M_BST3_L_SM_L_SFT 4 + +/* SPK Right Mixer Control (0x47) */ +#define RT5645_G_RM_R_SM_R_MASK (0x3 << 14) +#define RT5645_G_RM_R_SM_R_SFT 14 +#define RT5645_G_IN_R_SM_R_MASK (0x3 << 12) +#define RT5645_G_IN_R_SM_R_SFT 12 +#define RT5645_G_DAC_R1_SM_R_MASK (0x3 << 10) +#define RT5645_G_DAC_R1_SM_R_SFT 10 +#define RT5645_G_DAC_R2_SM_R_MASK (0x3 << 8) +#define RT5645_G_DAC_R2_SM_R_SFT 8 +#define RT5645_G_OM_R_SM_R_MASK (0x3 << 6) +#define RT5645_G_OM_R_SM_R_SFT 6 +#define RT5645_M_BST2_R_SM_R (0x1 << 5) +#define RT5645_M_BST2_R_SM_R_SFT 5 +#define RT5645_M_IN_R_SM_R (0x1 << 3) +#define RT5645_M_IN_R_SM_R_SFT 3 +#define RT5645_M_DAC_R1_SM_R (0x1 << 1) +#define RT5645_M_DAC_R1_SM_R_SFT 1 +#define RT5645_M_DAC_R2_SM_R (0x1 << 2) +#define RT5645_M_DAC_R2_SM_R_SFT 2 +#define RT5645_M_BST3_R_SM_R (0x1 << 4) +#define RT5645_M_BST3_R_SM_R_SFT 4 + +/* SPOLMIX Control (0x48) */ +#define RT5645_M_DAC_L1_SPM_L (0x1 << 15) +#define RT5645_M_DAC_L1_SPM_L_SFT 15 +#define RT5645_M_DAC_R1_SPM_L (0x1 << 14) +#define RT5645_M_DAC_R1_SPM_L_SFT 14 +#define RT5645_M_SV_L_SPM_L (0x1 << 13) +#define RT5645_M_SV_L_SPM_L_SFT 13 +#define RT5645_M_SV_R_SPM_L (0x1 << 12) +#define RT5645_M_SV_R_SPM_L_SFT 12 +#define RT5645_M_BST3_SPM_L (0x1 << 11) +#define RT5645_M_BST3_SPM_L_SFT 11 +#define RT5645_M_DAC_R1_SPM_R (0x1 << 2) +#define RT5645_M_DAC_R1_SPM_R_SFT 2 +#define RT5645_M_BST3_SPM_R (0x1 << 1) +#define RT5645_M_BST3_SPM_R_SFT 1 +#define RT5645_M_SV_R_SPM_R (0x1 << 0) +#define RT5645_M_SV_R_SPM_R_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5645_M_OV_L_MM (0x1 << 9) +#define RT5645_M_OV_L_MM_SFT 9 +#define RT5645_M_DAC_L2_MA (0x1 << 8) +#define RT5645_M_DAC_L2_MA_SFT 8 +#define RT5645_G_MONOMIX_MASK (0x1 << 10) +#define RT5645_G_MONOMIX_SFT 10 +#define RT5645_M_BST2_MM (0x1 << 4) +#define RT5645_M_BST2_MM_SFT 4 +#define RT5645_M_DAC_R1_MM (0x1 << 3) +#define RT5645_M_DAC_R1_MM_SFT 3 +#define RT5645_M_DAC_R2_MM (0x1 << 2) +#define RT5645_M_DAC_R2_MM_SFT 2 +#define RT5645_M_DAC_L2_MM (0x1 << 1) +#define RT5645_M_DAC_L2_MM_SFT 1 +#define RT5645_M_BST3_MM (0x1 << 0) +#define RT5645_M_BST3_MM_SFT 0 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5645_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5645_G_BST3_OM_L_SFT 13 +#define RT5645_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5645_G_BST2_OM_L_SFT 10 +#define RT5645_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5645_G_BST1_OM_L_SFT 7 +#define RT5645_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5645_G_IN_L_OM_L_SFT 4 +#define RT5645_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5645_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5645_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5645_G_DAC_R2_OM_L_SFT 13 +#define RT5645_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5645_G_DAC_L2_OM_L_SFT 10 +#define RT5645_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5645_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5645_M_BST3_OM_L (0x1 << 4) +#define RT5645_M_BST3_OM_L_SFT 4 +#define RT5645_M_BST1_OM_L (0x1 << 3) +#define RT5645_M_BST1_OM_L_SFT 3 +#define RT5645_M_IN_L_OM_L (0x1 << 2) +#define RT5645_M_IN_L_OM_L_SFT 2 +#define RT5645_M_DAC_L2_OM_L (0x1 << 1) +#define RT5645_M_DAC_L2_OM_L_SFT 1 +#define RT5645_M_DAC_L1_OM_L (0x1) +#define RT5645_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5645_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5645_G_BST4_OM_R_SFT 13 +#define RT5645_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5645_G_BST2_OM_R_SFT 10 +#define RT5645_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5645_G_BST1_OM_R_SFT 7 +#define RT5645_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5645_G_IN_R_OM_R_SFT 4 +#define RT5645_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5645_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5645_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5645_G_DAC_L2_OM_R_SFT 13 +#define RT5645_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5645_G_DAC_R2_OM_R_SFT 10 +#define RT5645_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5645_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5645_M_BST3_OM_R (0x1 << 4) +#define RT5645_M_BST3_OM_R_SFT 4 +#define RT5645_M_BST2_OM_R (0x1 << 3) +#define RT5645_M_BST2_OM_R_SFT 3 +#define RT5645_M_IN_R_OM_R (0x1 << 2) +#define RT5645_M_IN_R_OM_R_SFT 2 +#define RT5645_M_DAC_R2_OM_R (0x1 << 1) +#define RT5645_M_DAC_R2_OM_R_SFT 1 +#define RT5645_M_DAC_R1_OM_R (0x1) +#define RT5645_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5645_M_DAC_L1_LM (0x1 << 15) +#define RT5645_M_DAC_L1_LM_SFT 15 +#define RT5645_M_DAC_R1_LM (0x1 << 14) +#define RT5645_M_DAC_R1_LM_SFT 14 +#define RT5645_M_OV_L_LM (0x1 << 13) +#define RT5645_M_OV_L_LM_SFT 13 +#define RT5645_M_OV_R_LM (0x1 << 12) +#define RT5645_M_OV_R_LM_SFT 12 +#define RT5645_G_LOUTMIX_MASK (0x1 << 11) +#define RT5645_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5645_PWR_I2S1 (0x1 << 15) +#define RT5645_PWR_I2S1_BIT 15 +#define RT5645_PWR_I2S2 (0x1 << 14) +#define RT5645_PWR_I2S2_BIT 14 +#define RT5645_PWR_I2S3 (0x1 << 13) +#define RT5645_PWR_I2S3_BIT 13 +#define RT5645_PWR_DAC_L1 (0x1 << 12) +#define RT5645_PWR_DAC_L1_BIT 12 +#define RT5645_PWR_DAC_R1 (0x1 << 11) +#define RT5645_PWR_DAC_R1_BIT 11 +#define RT5645_PWR_CLS_D_R (0x1 << 9) +#define RT5645_PWR_CLS_D_R_BIT 9 +#define RT5645_PWR_CLS_D_L (0x1 << 8) +#define RT5645_PWR_CLS_D_L_BIT 8 +#define RT5645_PWR_ADC_R (0x1 << 1) +#define RT5645_PWR_ADC_R_BIT 1 +#define RT5645_PWR_DAC_L2 (0x1 << 7) +#define RT5645_PWR_DAC_L2_BIT 7 +#define RT5645_PWR_DAC_R2 (0x1 << 6) +#define RT5645_PWR_DAC_R2_BIT 6 +#define RT5645_PWR_ADC_L (0x1 << 2) +#define RT5645_PWR_ADC_L_BIT 2 +#define RT5645_PWR_ADC_R (0x1 << 1) +#define RT5645_PWR_ADC_R_BIT 1 +#define RT5645_PWR_CLS_D (0x1) +#define RT5645_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5645_PWR_ADC_S1F (0x1 << 15) +#define RT5645_PWR_ADC_S1F_BIT 15 +#define RT5645_PWR_ADC_MF_L (0x1 << 14) +#define RT5645_PWR_ADC_MF_L_BIT 14 +#define RT5645_PWR_ADC_MF_R (0x1 << 13) +#define RT5645_PWR_ADC_MF_R_BIT 13 +#define RT5645_PWR_I2S_DSP (0x1 << 12) +#define RT5645_PWR_I2S_DSP_BIT 12 +#define RT5645_PWR_DAC_S1F (0x1 << 11) +#define RT5645_PWR_DAC_S1F_BIT 11 +#define RT5645_PWR_DAC_MF_L (0x1 << 10) +#define RT5645_PWR_DAC_MF_L_BIT 10 +#define RT5645_PWR_DAC_MF_R (0x1 << 9) +#define RT5645_PWR_DAC_MF_R_BIT 9 +#define RT5645_PWR_ADC_S2F (0x1 << 8) +#define RT5645_PWR_ADC_S2F_BIT 8 +#define RT5645_PWR_PDM1 (0x1 << 7) +#define RT5645_PWR_PDM1_BIT 7 +#define RT5645_PWR_PDM2 (0x1 << 6) +#define RT5645_PWR_PDM2_BIT 6 +#define RT5645_PWR_IPTV (0x1 << 1) +#define RT5645_PWR_IPTV_BIT 1 +#define RT5645_PWR_PAD (0x1) +#define RT5645_PWR_PAD_BIT 0 + +/* Power Management for Analog 1 (0x63) */ +#define RT5645_PWR_VREF1 (0x1 << 15) +#define RT5645_PWR_VREF1_BIT 15 +#define RT5645_PWR_FV1 (0x1 << 14) +#define RT5645_PWR_FV1_BIT 14 +#define RT5645_PWR_MB (0x1 << 13) +#define RT5645_PWR_MB_BIT 13 +#define RT5645_PWR_LM (0x1 << 12) +#define RT5645_PWR_LM_BIT 12 +#define RT5645_PWR_BG (0x1 << 11) +#define RT5645_PWR_BG_BIT 11 +#define RT5645_PWR_MA (0x1 << 10) +#define RT5645_PWR_MA_BIT 10 +#define RT5645_PWR_HP_L (0x1 << 7) +#define RT5645_PWR_HP_L_BIT 7 +#define RT5645_PWR_HP_R (0x1 << 6) +#define RT5645_PWR_HP_R_BIT 6 +#define RT5645_PWR_HA (0x1 << 5) +#define RT5645_PWR_HA_BIT 5 +#define RT5645_PWR_VREF2 (0x1 << 4) +#define RT5645_PWR_VREF2_BIT 4 +#define RT5645_PWR_FV2 (0x1 << 3) +#define RT5645_PWR_FV2_BIT 3 +#define RT5645_LDO_SEL_MASK (0x3) +#define RT5645_LDO_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5645_PWR_BST1 (0x1 << 15) +#define RT5645_PWR_BST1_BIT 15 +#define RT5645_PWR_BST2 (0x1 << 14) +#define RT5645_PWR_BST2_BIT 14 +#define RT5645_PWR_BST3 (0x1 << 13) +#define RT5645_PWR_BST3_BIT 13 +#define RT5645_PWR_BST4 (0x1 << 12) +#define RT5645_PWR_BST4_BIT 12 +#define RT5645_PWR_MB1 (0x1 << 11) +#define RT5645_PWR_MB1_BIT 11 +#define RT5645_PWR_MB2 (0x1 << 10) +#define RT5645_PWR_MB2_BIT 10 +#define RT5645_PWR_PLL (0x1 << 9) +#define RT5645_PWR_PLL_BIT 9 +#define RT5645_PWR_BST2_P (0x1 << 5) +#define RT5645_PWR_BST2_P_BIT 5 +#define RT5645_PWR_BST3_P (0x1 << 4) +#define RT5645_PWR_BST3_P_BIT 4 +#define RT5645_PWR_BST4_P (0x1 << 3) +#define RT5645_PWR_BST4_P_BIT 3 +#define RT5645_PWR_JD1 (0x1 << 2) +#define RT5645_PWR_JD1_BIT 2 +#define RT5645_PWR_JD (0x1 << 1) +#define RT5645_PWR_JD_BIT 1 + +/* Power Management for Mixer (0x65) */ +#define RT5645_PWR_OM_L (0x1 << 15) +#define RT5645_PWR_OM_L_BIT 15 +#define RT5645_PWR_OM_R (0x1 << 14) +#define RT5645_PWR_OM_R_BIT 14 +#define RT5645_PWR_SM_L (0x1 << 13) +#define RT5645_PWR_SM_L_BIT 13 +#define RT5645_PWR_SM_R (0x1 << 12) +#define RT5645_PWR_SM_R_BIT 12 +#define RT5645_PWR_RM_L (0x1 << 11) +#define RT5645_PWR_RM_L_BIT 11 +#define RT5645_PWR_RM_R (0x1 << 10) +#define RT5645_PWR_RM_R_BIT 10 +#define RT5645_PWR_MM (0x1 << 8) +#define RT5645_PWR_MM_BIT 8 +#define RT5645_PWR_HM_L (0x1 << 7) +#define RT5645_PWR_HM_L_BIT 7 +#define RT5645_PWR_HM_R (0x1 << 6) +#define RT5645_PWR_HM_R_BIT 6 +#define RT5645_PWR_LDO2 (0x1 << 1) +#define RT5645_PWR_LDO2_BIT 1 + +/* Power Management for Volume (0x66) */ +#define RT5645_PWR_SV_L (0x1 << 15) +#define RT5645_PWR_SV_L_BIT 15 +#define RT5645_PWR_SV_R (0x1 << 14) +#define RT5645_PWR_SV_R_BIT 14 +#define RT5645_PWR_HV_L (0x1 << 11) +#define RT5645_PWR_HV_L_BIT 11 +#define RT5645_PWR_HV_R (0x1 << 10) +#define RT5645_PWR_HV_R_BIT 10 +#define RT5645_PWR_IN_L (0x1 << 9) +#define RT5645_PWR_IN_L_BIT 9 +#define RT5645_PWR_IN_R (0x1 << 8) +#define RT5645_PWR_IN_R_BIT 8 +#define RT5645_PWR_MIC_DET (0x1 << 5) +#define RT5645_PWR_MIC_DET_BIT 5 + +/* I2S1/2 Audio Serial Data Port Control (0x70 0x71) */ +#define RT5645_I2S_MS_MASK (0x1 << 15) +#define RT5645_I2S_MS_SFT 15 +#define RT5645_I2S_MS_M (0x0 << 15) +#define RT5645_I2S_MS_S (0x1 << 15) +#define RT5645_I2S_O_CP_MASK (0x3 << 10) +#define RT5645_I2S_O_CP_SFT 10 +#define RT5645_I2S_O_CP_OFF (0x0 << 10) +#define RT5645_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5645_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5645_I2S_I_CP_MASK (0x3 << 8) +#define RT5645_I2S_I_CP_SFT 8 +#define RT5645_I2S_I_CP_OFF (0x0 << 8) +#define RT5645_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5645_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5645_I2S_BP_MASK (0x1 << 7) +#define RT5645_I2S_BP_SFT 7 +#define RT5645_I2S_BP_NOR (0x0 << 7) +#define RT5645_I2S_BP_INV (0x1 << 7) +#define RT5645_I2S_DL_MASK (0x3 << 2) +#define RT5645_I2S_DL_SFT 2 +#define RT5645_I2S_DL_16 (0x0 << 2) +#define RT5645_I2S_DL_20 (0x1 << 2) +#define RT5645_I2S_DL_24 (0x2 << 2) +#define RT5645_I2S_DL_8 (0x3 << 2) +#define RT5645_I2S_DF_MASK (0x3) +#define RT5645_I2S_DF_SFT 0 +#define RT5645_I2S_DF_I2S (0x0) +#define RT5645_I2S_DF_LEFT (0x1) +#define RT5645_I2S_DF_PCM_A (0x2) +#define RT5645_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5645_I2S2_SDI_MASK (0x1 << 6) +#define RT5645_I2S2_SDI_SFT 6 +#define RT5645_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5645_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5645_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5645_I2S_BCLK_MS1_SFT 15 +#define RT5645_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5645_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5645_I2S_PD1_MASK (0x7 << 12) +#define RT5645_I2S_PD1_SFT 12 +#define RT5645_I2S_PD1_1 (0x0 << 12) +#define RT5645_I2S_PD1_2 (0x1 << 12) +#define RT5645_I2S_PD1_3 (0x2 << 12) +#define RT5645_I2S_PD1_4 (0x3 << 12) +#define RT5645_I2S_PD1_6 (0x4 << 12) +#define RT5645_I2S_PD1_8 (0x5 << 12) +#define RT5645_I2S_PD1_12 (0x6 << 12) +#define RT5645_I2S_PD1_16 (0x7 << 12) +#define RT5645_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5645_I2S_BCLK_MS2_SFT 11 +#define RT5645_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5645_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5645_I2S_PD2_MASK (0x7 << 8) +#define RT5645_I2S_PD2_SFT 8 +#define RT5645_I2S_PD2_1 (0x0 << 8) +#define RT5645_I2S_PD2_2 (0x1 << 8) +#define RT5645_I2S_PD2_3 (0x2 << 8) +#define RT5645_I2S_PD2_4 (0x3 << 8) +#define RT5645_I2S_PD2_6 (0x4 << 8) +#define RT5645_I2S_PD2_8 (0x5 << 8) +#define RT5645_I2S_PD2_12 (0x6 << 8) +#define RT5645_I2S_PD2_16 (0x7 << 8) +#define RT5645_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5645_I2S_BCLK_MS3_SFT 7 +#define RT5645_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5645_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5645_I2S_PD3_MASK (0x7 << 4) +#define RT5645_I2S_PD3_SFT 4 +#define RT5645_I2S_PD3_1 (0x0 << 4) +#define RT5645_I2S_PD3_2 (0x1 << 4) +#define RT5645_I2S_PD3_3 (0x2 << 4) +#define RT5645_I2S_PD3_4 (0x3 << 4) +#define RT5645_I2S_PD3_6 (0x4 << 4) +#define RT5645_I2S_PD3_8 (0x5 << 4) +#define RT5645_I2S_PD3_12 (0x6 << 4) +#define RT5645_I2S_PD3_16 (0x7 << 4) +#define RT5645_DAC_OSR_MASK (0x3 << 2) +#define RT5645_DAC_OSR_SFT 2 +#define RT5645_DAC_OSR_128 (0x0 << 2) +#define RT5645_DAC_OSR_64 (0x1 << 2) +#define RT5645_DAC_OSR_32 (0x2 << 2) +#define RT5645_DAC_OSR_16 (0x3 << 2) +#define RT5645_ADC_OSR_MASK (0x3) +#define RT5645_ADC_OSR_SFT 0 +#define RT5645_ADC_OSR_128 (0x0) +#define RT5645_ADC_OSR_64 (0x1) +#define RT5645_ADC_OSR_32 (0x2) +#define RT5645_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5645_DAC_L_OSR_MASK (0x3 << 14) +#define RT5645_DAC_L_OSR_SFT 14 +#define RT5645_DAC_L_OSR_128 (0x0 << 14) +#define RT5645_DAC_L_OSR_64 (0x1 << 14) +#define RT5645_DAC_L_OSR_32 (0x2 << 14) +#define RT5645_DAC_L_OSR_16 (0x3 << 14) +#define RT5645_ADC_R_OSR_MASK (0x3 << 12) +#define RT5645_ADC_R_OSR_SFT 12 +#define RT5645_ADC_R_OSR_128 (0x0 << 12) +#define RT5645_ADC_R_OSR_64 (0x1 << 12) +#define RT5645_ADC_R_OSR_32 (0x2 << 12) +#define RT5645_ADC_R_OSR_16 (0x3 << 12) +#define RT5645_DAHPF_EN (0x1 << 11) +#define RT5645_DAHPF_EN_SFT 11 +#define RT5645_ADHPF_EN (0x1 << 10) +#define RT5645_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5645_DMIC_1_EN_MASK (0x1 << 15) +#define RT5645_DMIC_1_EN_SFT 15 +#define RT5645_DMIC_1_DIS (0x0 << 15) +#define RT5645_DMIC_1_EN (0x1 << 15) +#define RT5645_DMIC_2_EN_MASK (0x1 << 14) +#define RT5645_DMIC_2_EN_SFT 14 +#define RT5645_DMIC_2_DIS (0x0 << 14) +#define RT5645_DMIC_2_EN (0x1 << 14) +#define RT5645_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5645_DMIC_1L_LH_SFT 13 +#define RT5645_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5645_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5645_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5645_DMIC_1R_LH_SFT 12 +#define RT5645_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5645_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5645_DMIC_2_DP_MASK (0x3 << 10) +#define RT5645_DMIC_2_DP_SFT 10 +#define RT5645_DMIC_2_DP_GPIO6 (0x0 << 10) +#define RT5645_DMIC_2_DP_GPIO10 (0x1 << 10) +#define RT5645_DMIC_2_DP_GPIO12 (0x2 << 10) +#define RT5645_DMIC_2_DP_IN2P (0x3 << 10) +#define RT5645_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5645_DMIC_2L_LH_SFT 9 +#define RT5645_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5645_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5645_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5645_DMIC_2R_LH_SFT 8 +#define RT5645_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5645_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5645_DMIC_CLK_MASK (0x7 << 5) +#define RT5645_DMIC_CLK_SFT 5 +#define RT5645_DMIC_3_EN_MASK (0x1 << 4) +#define RT5645_DMIC_3_EN_SFT 4 +#define RT5645_DMIC_3_DIS (0x0 << 4) +#define RT5645_DMIC_3_EN (0x1 << 4) +#define RT5645_DMIC_1_DP_MASK (0x3 << 0) +#define RT5645_DMIC_1_DP_SFT 0 +#define RT5645_DMIC_1_DP_GPIO5 (0x0 << 0) +#define RT5645_DMIC_1_DP_IN2N (0x1 << 0) +#define RT5645_DMIC_1_DP_GPIO11 (0x2 << 0) + +/* TDM Control 1 (0x77) */ +#define RT5645_IF1_ADC_IN_MASK (0x3 << 8) +#define RT5645_IF1_ADC_IN_SFT 8 + +/* Global Clock Control (0x80) */ +#define RT5645_SCLK_SRC_MASK (0x3 << 14) +#define RT5645_SCLK_SRC_SFT 14 +#define RT5645_SCLK_SRC_MCLK (0x0 << 14) +#define RT5645_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5645_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ +#define RT5645_PLL1_SRC_MASK (0x3 << 12) +#define RT5645_PLL1_SRC_SFT 12 +#define RT5645_PLL1_SRC_MCLK (0x0 << 12) +#define RT5645_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5645_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5645_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5645_PLL1_PD_MASK (0x1 << 3) +#define RT5645_PLL1_PD_SFT 3 +#define RT5645_PLL1_PD_1 (0x0 << 3) +#define RT5645_PLL1_PD_2 (0x1 << 3) + +#define RT5645_PLL_INP_MAX 40000000 +#define RT5645_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5645_PLL_N_MAX 0x1ff +#define RT5645_PLL_N_MASK (RT5645_PLL_N_MAX << 7) +#define RT5645_PLL_N_SFT 7 +#define RT5645_PLL_K_MAX 0x1f +#define RT5645_PLL_K_MASK (RT5645_PLL_K_MAX) +#define RT5645_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5645_PLL_M_MAX 0xf +#define RT5645_PLL_M_MASK (RT5645_PLL_M_MAX << 12) +#define RT5645_PLL_M_SFT 12 +#define RT5645_PLL_M_BP (0x1 << 11) +#define RT5645_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5645_STO_T_MASK (0x1 << 15) +#define RT5645_STO_T_SFT 15 +#define RT5645_STO_T_SCLK (0x0 << 15) +#define RT5645_STO_T_LRCK1 (0x1 << 15) +#define RT5645_M1_T_MASK (0x1 << 14) +#define RT5645_M1_T_SFT 14 +#define RT5645_M1_T_I2S2 (0x0 << 14) +#define RT5645_M1_T_I2S2_D3 (0x1 << 14) +#define RT5645_I2S2_F_MASK (0x1 << 12) +#define RT5645_I2S2_F_SFT 12 +#define RT5645_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5645_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5645_DMIC_1_M_MASK (0x1 << 9) +#define RT5645_DMIC_1_M_SFT 9 +#define RT5645_DMIC_1_M_NOR (0x0 << 9) +#define RT5645_DMIC_1_M_ASYN (0x1 << 9) +#define RT5645_DMIC_2_M_MASK (0x1 << 8) +#define RT5645_DMIC_2_M_SFT 8 +#define RT5645_DMIC_2_M_NOR (0x0 << 8) +#define RT5645_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5645_MDA_L_M_MASK (0x1 << 15) +#define RT5645_MDA_L_M_SFT 15 +#define RT5645_MDA_L_M_NOR (0x0 << 15) +#define RT5645_MDA_L_M_ASYN (0x1 << 15) +#define RT5645_MDA_R_M_MASK (0x1 << 14) +#define RT5645_MDA_R_M_SFT 14 +#define RT5645_MDA_R_M_NOR (0x0 << 14) +#define RT5645_MDA_R_M_ASYN (0x1 << 14) +#define RT5645_MAD_L_M_MASK (0x1 << 13) +#define RT5645_MAD_L_M_SFT 13 +#define RT5645_MAD_L_M_NOR (0x0 << 13) +#define RT5645_MAD_L_M_ASYN (0x1 << 13) +#define RT5645_MAD_R_M_MASK (0x1 << 12) +#define RT5645_MAD_R_M_SFT 12 +#define RT5645_MAD_R_M_NOR (0x0 << 12) +#define RT5645_MAD_R_M_ASYN (0x1 << 12) +#define RT5645_ADC_M_MASK (0x1 << 11) +#define RT5645_ADC_M_SFT 11 +#define RT5645_ADC_M_NOR (0x0 << 11) +#define RT5645_ADC_M_ASYN (0x1 << 11) +#define RT5645_STO_DAC_M_MASK (0x1 << 5) +#define RT5645_STO_DAC_M_SFT 5 +#define RT5645_STO_DAC_M_NOR (0x0 << 5) +#define RT5645_STO_DAC_M_ASYN (0x1 << 5) +#define RT5645_I2S1_R_D_MASK (0x1 << 4) +#define RT5645_I2S1_R_D_SFT 4 +#define RT5645_I2S1_R_D_DIS (0x0 << 4) +#define RT5645_I2S1_R_D_EN (0x1 << 4) +#define RT5645_I2S2_R_D_MASK (0x1 << 3) +#define RT5645_I2S2_R_D_SFT 3 +#define RT5645_I2S2_R_D_DIS (0x0 << 3) +#define RT5645_I2S2_R_D_EN (0x1 << 3) +#define RT5645_PRE_SCLK_MASK (0x3) +#define RT5645_PRE_SCLK_SFT 0 +#define RT5645_PRE_SCLK_512 (0x0) +#define RT5645_PRE_SCLK_1024 (0x1) +#define RT5645_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5645_I2S1_RATE_MASK (0xf << 12) +#define RT5645_I2S1_RATE_SFT 12 +#define RT5645_I2S2_RATE_MASK (0xf << 8) +#define RT5645_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5645_I2S1_PD_MASK (0x7 << 12) +#define RT5645_I2S1_PD_SFT 12 +#define RT5645_I2S2_PD_MASK (0x7 << 8) +#define RT5645_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5645_HP_OVCD_MASK (0x1 << 10) +#define RT5645_HP_OVCD_SFT 10 +#define RT5645_HP_OVCD_DIS (0x0 << 10) +#define RT5645_HP_OVCD_EN (0x1 << 10) +#define RT5645_HP_OC_TH_MASK (0x3 << 8) +#define RT5645_HP_OC_TH_SFT 8 +#define RT5645_HP_OC_TH_90 (0x0 << 8) +#define RT5645_HP_OC_TH_105 (0x1 << 8) +#define RT5645_HP_OC_TH_120 (0x2 << 8) +#define RT5645_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5645_CLSD_OC_MASK (0x1 << 9) +#define RT5645_CLSD_OC_SFT 9 +#define RT5645_CLSD_OC_PU (0x0 << 9) +#define RT5645_CLSD_OC_PD (0x1 << 9) +#define RT5645_AUTO_PD_MASK (0x1 << 8) +#define RT5645_AUTO_PD_SFT 8 +#define RT5645_AUTO_PD_DIS (0x0 << 8) +#define RT5645_AUTO_PD_EN (0x1 << 8) +#define RT5645_CLSD_OC_TH_MASK (0x3f) +#define RT5645_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5645_CLSD_RATIO_MASK (0xf << 12) +#define RT5645_CLSD_RATIO_SFT 12 +#define RT5645_CLSD_OM_MASK (0x1 << 11) +#define RT5645_CLSD_OM_SFT 11 +#define RT5645_CLSD_OM_MONO (0x0 << 11) +#define RT5645_CLSD_OM_STO (0x1 << 11) +#define RT5645_CLSD_SCH_MASK (0x1 << 10) +#define RT5645_CLSD_SCH_SFT 10 +#define RT5645_CLSD_SCH_L (0x0 << 10) +#define RT5645_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5645_SMT_TRIG_MASK (0x1 << 15) +#define RT5645_SMT_TRIG_SFT 15 +#define RT5645_SMT_TRIG_DIS (0x0 << 15) +#define RT5645_SMT_TRIG_EN (0x1 << 15) +#define RT5645_HP_L_SMT_MASK (0x1 << 9) +#define RT5645_HP_L_SMT_SFT 9 +#define RT5645_HP_L_SMT_DIS (0x0 << 9) +#define RT5645_HP_L_SMT_EN (0x1 << 9) +#define RT5645_HP_R_SMT_MASK (0x1 << 8) +#define RT5645_HP_R_SMT_SFT 8 +#define RT5645_HP_R_SMT_DIS (0x0 << 8) +#define RT5645_HP_R_SMT_EN (0x1 << 8) +#define RT5645_HP_CD_PD_MASK (0x1 << 7) +#define RT5645_HP_CD_PD_SFT 7 +#define RT5645_HP_CD_PD_DIS (0x0 << 7) +#define RT5645_HP_CD_PD_EN (0x1 << 7) +#define RT5645_RSTN_MASK (0x1 << 6) +#define RT5645_RSTN_SFT 6 +#define RT5645_RSTN_DIS (0x0 << 6) +#define RT5645_RSTN_EN (0x1 << 6) +#define RT5645_RSTP_MASK (0x1 << 5) +#define RT5645_RSTP_SFT 5 +#define RT5645_RSTP_DIS (0x0 << 5) +#define RT5645_RSTP_EN (0x1 << 5) +#define RT5645_HP_CO_MASK (0x1 << 4) +#define RT5645_HP_CO_SFT 4 +#define RT5645_HP_CO_DIS (0x0 << 4) +#define RT5645_HP_CO_EN (0x1 << 4) +#define RT5645_HP_CP_MASK (0x1 << 3) +#define RT5645_HP_CP_SFT 3 +#define RT5645_HP_CP_PD (0x0 << 3) +#define RT5645_HP_CP_PU (0x1 << 3) +#define RT5645_HP_SG_MASK (0x1 << 2) +#define RT5645_HP_SG_SFT 2 +#define RT5645_HP_SG_DIS (0x0 << 2) +#define RT5645_HP_SG_EN (0x1 << 2) +#define RT5645_HP_DP_MASK (0x1 << 1) +#define RT5645_HP_DP_SFT 1 +#define RT5645_HP_DP_PD (0x0 << 1) +#define RT5645_HP_DP_PU (0x1 << 1) +#define RT5645_HP_CB_MASK (0x1) +#define RT5645_HP_CB_SFT 0 +#define RT5645_HP_CB_PD (0x0) +#define RT5645_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5645_DEPOP_MASK (0x1 << 13) +#define RT5645_DEPOP_SFT 13 +#define RT5645_DEPOP_AUTO (0x0 << 13) +#define RT5645_DEPOP_MAN (0x1 << 13) +#define RT5645_RAMP_MASK (0x1 << 12) +#define RT5645_RAMP_SFT 12 +#define RT5645_RAMP_DIS (0x0 << 12) +#define RT5645_RAMP_EN (0x1 << 12) +#define RT5645_BPS_MASK (0x1 << 11) +#define RT5645_BPS_SFT 11 +#define RT5645_BPS_DIS (0x0 << 11) +#define RT5645_BPS_EN (0x1 << 11) +#define RT5645_FAST_UPDN_MASK (0x1 << 10) +#define RT5645_FAST_UPDN_SFT 10 +#define RT5645_FAST_UPDN_DIS (0x0 << 10) +#define RT5645_FAST_UPDN_EN (0x1 << 10) +#define RT5645_MRES_MASK (0x3 << 8) +#define RT5645_MRES_SFT 8 +#define RT5645_MRES_15MO (0x0 << 8) +#define RT5645_MRES_25MO (0x1 << 8) +#define RT5645_MRES_35MO (0x2 << 8) +#define RT5645_MRES_45MO (0x3 << 8) +#define RT5645_VLO_MASK (0x1 << 7) +#define RT5645_VLO_SFT 7 +#define RT5645_VLO_3V (0x0 << 7) +#define RT5645_VLO_32V (0x1 << 7) +#define RT5645_DIG_DP_MASK (0x1 << 6) +#define RT5645_DIG_DP_SFT 6 +#define RT5645_DIG_DP_DIS (0x0 << 6) +#define RT5645_DIG_DP_EN (0x1 << 6) +#define RT5645_DP_TH_MASK (0x3 << 4) +#define RT5645_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5645_CP_SYS_MASK (0x7 << 12) +#define RT5645_CP_SYS_SFT 12 +#define RT5645_CP_FQ1_MASK (0x7 << 8) +#define RT5645_CP_FQ1_SFT 8 +#define RT5645_CP_FQ2_MASK (0x7 << 4) +#define RT5645_CP_FQ2_SFT 4 +#define RT5645_CP_FQ3_MASK (0x7) +#define RT5645_CP_FQ3_SFT 0 +#define RT5645_CP_FQ_1_5_KHZ 0 +#define RT5645_CP_FQ_3_KHZ 1 +#define RT5645_CP_FQ_6_KHZ 2 +#define RT5645_CP_FQ_12_KHZ 3 +#define RT5645_CP_FQ_24_KHZ 4 +#define RT5645_CP_FQ_48_KHZ 5 +#define RT5645_CP_FQ_96_KHZ 6 +#define RT5645_CP_FQ_192_KHZ 7 + +/* PV detection and SPK gain control (0x92) */ +#define RT5645_PVDD_DET_MASK (0x1 << 15) +#define RT5645_PVDD_DET_SFT 15 +#define RT5645_PVDD_DET_DIS (0x0 << 15) +#define RT5645_PVDD_DET_EN (0x1 << 15) +#define RT5645_SPK_AG_MASK (0x1 << 14) +#define RT5645_SPK_AG_SFT 14 +#define RT5645_SPK_AG_DIS (0x0 << 14) +#define RT5645_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5645_MIC1_BS_MASK (0x1 << 15) +#define RT5645_MIC1_BS_SFT 15 +#define RT5645_MIC1_BS_9AV (0x0 << 15) +#define RT5645_MIC1_BS_75AV (0x1 << 15) +#define RT5645_MIC2_BS_MASK (0x1 << 14) +#define RT5645_MIC2_BS_SFT 14 +#define RT5645_MIC2_BS_9AV (0x0 << 14) +#define RT5645_MIC2_BS_75AV (0x1 << 14) +#define RT5645_MIC1_CLK_MASK (0x1 << 13) +#define RT5645_MIC1_CLK_SFT 13 +#define RT5645_MIC1_CLK_DIS (0x0 << 13) +#define RT5645_MIC1_CLK_EN (0x1 << 13) +#define RT5645_MIC2_CLK_MASK (0x1 << 12) +#define RT5645_MIC2_CLK_SFT 12 +#define RT5645_MIC2_CLK_DIS (0x0 << 12) +#define RT5645_MIC2_CLK_EN (0x1 << 12) +#define RT5645_MIC1_OVCD_MASK (0x1 << 11) +#define RT5645_MIC1_OVCD_SFT 11 +#define RT5645_MIC1_OVCD_DIS (0x0 << 11) +#define RT5645_MIC1_OVCD_EN (0x1 << 11) +#define RT5645_MIC1_OVTH_MASK (0x3 << 9) +#define RT5645_MIC1_OVTH_SFT 9 +#define RT5645_MIC1_OVTH_600UA (0x0 << 9) +#define RT5645_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5645_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5645_MIC2_OVCD_MASK (0x1 << 8) +#define RT5645_MIC2_OVCD_SFT 8 +#define RT5645_MIC2_OVCD_DIS (0x0 << 8) +#define RT5645_MIC2_OVCD_EN (0x1 << 8) +#define RT5645_MIC2_OVTH_MASK (0x3 << 6) +#define RT5645_MIC2_OVTH_SFT 6 +#define RT5645_MIC2_OVTH_600UA (0x0 << 6) +#define RT5645_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5645_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5645_PWR_MB_MASK (0x1 << 5) +#define RT5645_PWR_MB_SFT 5 +#define RT5645_PWR_MB_PD (0x0 << 5) +#define RT5645_PWR_MB_PU (0x1 << 5) +#define RT5645_PWR_CLK25M_MASK (0x1 << 4) +#define RT5645_PWR_CLK25M_SFT 4 +#define RT5645_PWR_CLK25M_PD (0x0 << 4) +#define RT5645_PWR_CLK25M_PU (0x1 << 4) + +/* VAD Control 4 (0x9d) */ +#define RT5645_VAD_SEL_MASK (0x3 << 8) +#define RT5645_VAD_SEL_SFT 8 + +/* EQ Control 1 (0xb0) */ +#define RT5645_EQ_SRC_MASK (0x1 << 15) +#define RT5645_EQ_SRC_SFT 15 +#define RT5645_EQ_SRC_DAC (0x0 << 15) +#define RT5645_EQ_SRC_ADC (0x1 << 15) +#define RT5645_EQ_UPD (0x1 << 14) +#define RT5645_EQ_UPD_BIT 14 +#define RT5645_EQ_CD_MASK (0x1 << 13) +#define RT5645_EQ_CD_SFT 13 +#define RT5645_EQ_CD_DIS (0x0 << 13) +#define RT5645_EQ_CD_EN (0x1 << 13) +#define RT5645_EQ_DITH_MASK (0x3 << 8) +#define RT5645_EQ_DITH_SFT 8 +#define RT5645_EQ_DITH_NOR (0x0 << 8) +#define RT5645_EQ_DITH_LSB (0x1 << 8) +#define RT5645_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5645_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5645_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5645_EQ_HPF1_M_SFT 8 +#define RT5645_EQ_HPF1_M_HI (0x0 << 8) +#define RT5645_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5645_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5645_EQ_LPF1_M_SFT 7 +#define RT5645_EQ_LPF1_M_LO (0x0 << 7) +#define RT5645_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5645_EQ_HPF2_MASK (0x1 << 6) +#define RT5645_EQ_HPF2_SFT 6 +#define RT5645_EQ_HPF2_DIS (0x0 << 6) +#define RT5645_EQ_HPF2_EN (0x1 << 6) +#define RT5645_EQ_HPF1_MASK (0x1 << 5) +#define RT5645_EQ_HPF1_SFT 5 +#define RT5645_EQ_HPF1_DIS (0x0 << 5) +#define RT5645_EQ_HPF1_EN (0x1 << 5) +#define RT5645_EQ_BPF4_MASK (0x1 << 4) +#define RT5645_EQ_BPF4_SFT 4 +#define RT5645_EQ_BPF4_DIS (0x0 << 4) +#define RT5645_EQ_BPF4_EN (0x1 << 4) +#define RT5645_EQ_BPF3_MASK (0x1 << 3) +#define RT5645_EQ_BPF3_SFT 3 +#define RT5645_EQ_BPF3_DIS (0x0 << 3) +#define RT5645_EQ_BPF3_EN (0x1 << 3) +#define RT5645_EQ_BPF2_MASK (0x1 << 2) +#define RT5645_EQ_BPF2_SFT 2 +#define RT5645_EQ_BPF2_DIS (0x0 << 2) +#define RT5645_EQ_BPF2_EN (0x1 << 2) +#define RT5645_EQ_BPF1_MASK (0x1 << 1) +#define RT5645_EQ_BPF1_SFT 1 +#define RT5645_EQ_BPF1_DIS (0x0 << 1) +#define RT5645_EQ_BPF1_EN (0x1 << 1) +#define RT5645_EQ_LPF_MASK (0x1) +#define RT5645_EQ_LPF_SFT 0 +#define RT5645_EQ_LPF_DIS (0x0) +#define RT5645_EQ_LPF_EN (0x1) +#define RT5645_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5645_MT_MASK (0x1 << 15) +#define RT5645_MT_SFT 15 +#define RT5645_MT_DIS (0x0 << 15) +#define RT5645_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5645_DRC_AGC_P_MASK (0x1 << 15) +#define RT5645_DRC_AGC_P_SFT 15 +#define RT5645_DRC_AGC_P_DAC (0x0 << 15) +#define RT5645_DRC_AGC_P_ADC (0x1 << 15) +#define RT5645_DRC_AGC_MASK (0x1 << 14) +#define RT5645_DRC_AGC_SFT 14 +#define RT5645_DRC_AGC_DIS (0x0 << 14) +#define RT5645_DRC_AGC_EN (0x1 << 14) +#define RT5645_DRC_AGC_UPD (0x1 << 13) +#define RT5645_DRC_AGC_UPD_BIT 13 +#define RT5645_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5645_DRC_AGC_AR_SFT 8 +#define RT5645_DRC_AGC_R_MASK (0x7 << 5) +#define RT5645_DRC_AGC_R_SFT 5 +#define RT5645_DRC_AGC_R_48K (0x1 << 5) +#define RT5645_DRC_AGC_R_96K (0x2 << 5) +#define RT5645_DRC_AGC_R_192K (0x3 << 5) +#define RT5645_DRC_AGC_R_441K (0x5 << 5) +#define RT5645_DRC_AGC_R_882K (0x6 << 5) +#define RT5645_DRC_AGC_R_1764K (0x7 << 5) +#define RT5645_DRC_AGC_RC_MASK (0x1f) +#define RT5645_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5645_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5645_DRC_AGC_POB_SFT 8 +#define RT5645_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5645_DRC_AGC_CP_SFT 7 +#define RT5645_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5645_DRC_AGC_CP_EN (0x1 << 7) +#define RT5645_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5645_DRC_AGC_CPR_SFT 5 +#define RT5645_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5645_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5645_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5645_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5645_DRC_AGC_PRB_MASK (0x1f) +#define RT5645_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5645_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5645_DRC_AGC_NGB_SFT 12 +#define RT5645_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5645_DRC_AGC_TAR_SFT 7 +#define RT5645_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5645_DRC_AGC_NG_SFT 6 +#define RT5645_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5645_DRC_AGC_NG_EN (0x1 << 6) +#define RT5645_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5645_DRC_AGC_NGH_SFT 5 +#define RT5645_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5645_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5645_DRC_AGC_NGT_MASK (0x1f) +#define RT5645_DRC_AGC_NGT_SFT 0 + +/* ANC Control 1 (0xb8) */ +#define RT5645_ANC_M_MASK (0x1 << 15) +#define RT5645_ANC_M_SFT 15 +#define RT5645_ANC_M_NOR (0x0 << 15) +#define RT5645_ANC_M_REV (0x1 << 15) +#define RT5645_ANC_MASK (0x1 << 14) +#define RT5645_ANC_SFT 14 +#define RT5645_ANC_DIS (0x0 << 14) +#define RT5645_ANC_EN (0x1 << 14) +#define RT5645_ANC_MD_MASK (0x3 << 12) +#define RT5645_ANC_MD_SFT 12 +#define RT5645_ANC_MD_DIS (0x0 << 12) +#define RT5645_ANC_MD_67MS (0x1 << 12) +#define RT5645_ANC_MD_267MS (0x2 << 12) +#define RT5645_ANC_MD_1067MS (0x3 << 12) +#define RT5645_ANC_SN_MASK (0x1 << 11) +#define RT5645_ANC_SN_SFT 11 +#define RT5645_ANC_SN_DIS (0x0 << 11) +#define RT5645_ANC_SN_EN (0x1 << 11) +#define RT5645_ANC_CLK_MASK (0x1 << 10) +#define RT5645_ANC_CLK_SFT 10 +#define RT5645_ANC_CLK_ANC (0x0 << 10) +#define RT5645_ANC_CLK_REG (0x1 << 10) +#define RT5645_ANC_ZCD_MASK (0x3 << 8) +#define RT5645_ANC_ZCD_SFT 8 +#define RT5645_ANC_ZCD_DIS (0x0 << 8) +#define RT5645_ANC_ZCD_T1 (0x1 << 8) +#define RT5645_ANC_ZCD_T2 (0x2 << 8) +#define RT5645_ANC_ZCD_WT (0x3 << 8) +#define RT5645_ANC_CS_MASK (0x1 << 7) +#define RT5645_ANC_CS_SFT 7 +#define RT5645_ANC_CS_DIS (0x0 << 7) +#define RT5645_ANC_CS_EN (0x1 << 7) +#define RT5645_ANC_SW_MASK (0x1 << 6) +#define RT5645_ANC_SW_SFT 6 +#define RT5645_ANC_SW_NOR (0x0 << 6) +#define RT5645_ANC_SW_AUTO (0x1 << 6) +#define RT5645_ANC_CO_L_MASK (0x3f) +#define RT5645_ANC_CO_L_SFT 0 + +/* ANC Control 2 (0xb6) */ +#define RT5645_ANC_FG_R_MASK (0xf << 12) +#define RT5645_ANC_FG_R_SFT 12 +#define RT5645_ANC_FG_L_MASK (0xf << 8) +#define RT5645_ANC_FG_L_SFT 8 +#define RT5645_ANC_CG_R_MASK (0xf << 4) +#define RT5645_ANC_CG_R_SFT 4 +#define RT5645_ANC_CG_L_MASK (0xf) +#define RT5645_ANC_CG_L_SFT 0 + +/* ANC Control 3 (0xb6) */ +#define RT5645_ANC_CD_MASK (0x1 << 6) +#define RT5645_ANC_CD_SFT 6 +#define RT5645_ANC_CD_BOTH (0x0 << 6) +#define RT5645_ANC_CD_IND (0x1 << 6) +#define RT5645_ANC_CO_R_MASK (0x3f) +#define RT5645_ANC_CO_R_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5645_JD_MASK (0x7 << 13) +#define RT5645_JD_SFT 13 +#define RT5645_JD_DIS (0x0 << 13) +#define RT5645_JD_GPIO1 (0x1 << 13) +#define RT5645_JD_JD1_IN4P (0x2 << 13) +#define RT5645_JD_JD2_IN4N (0x3 << 13) +#define RT5645_JD_GPIO2 (0x4 << 13) +#define RT5645_JD_GPIO3 (0x5 << 13) +#define RT5645_JD_GPIO4 (0x6 << 13) +#define RT5645_JD_HP_MASK (0x1 << 11) +#define RT5645_JD_HP_SFT 11 +#define RT5645_JD_HP_DIS (0x0 << 11) +#define RT5645_JD_HP_EN (0x1 << 11) +#define RT5645_JD_HP_TRG_MASK (0x1 << 10) +#define RT5645_JD_HP_TRG_SFT 10 +#define RT5645_JD_HP_TRG_LO (0x0 << 10) +#define RT5645_JD_HP_TRG_HI (0x1 << 10) +#define RT5645_JD_SPL_MASK (0x1 << 9) +#define RT5645_JD_SPL_SFT 9 +#define RT5645_JD_SPL_DIS (0x0 << 9) +#define RT5645_JD_SPL_EN (0x1 << 9) +#define RT5645_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5645_JD_SPL_TRG_SFT 8 +#define RT5645_JD_SPL_TRG_LO (0x0 << 8) +#define RT5645_JD_SPL_TRG_HI (0x1 << 8) +#define RT5645_JD_SPR_MASK (0x1 << 7) +#define RT5645_JD_SPR_SFT 7 +#define RT5645_JD_SPR_DIS (0x0 << 7) +#define RT5645_JD_SPR_EN (0x1 << 7) +#define RT5645_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5645_JD_SPR_TRG_SFT 6 +#define RT5645_JD_SPR_TRG_LO (0x0 << 6) +#define RT5645_JD_SPR_TRG_HI (0x1 << 6) +#define RT5645_JD_MO_MASK (0x1 << 5) +#define RT5645_JD_MO_SFT 5 +#define RT5645_JD_MO_DIS (0x0 << 5) +#define RT5645_JD_MO_EN (0x1 << 5) +#define RT5645_JD_MO_TRG_MASK (0x1 << 4) +#define RT5645_JD_MO_TRG_SFT 4 +#define RT5645_JD_MO_TRG_LO (0x0 << 4) +#define RT5645_JD_MO_TRG_HI (0x1 << 4) +#define RT5645_JD_LO_MASK (0x1 << 3) +#define RT5645_JD_LO_SFT 3 +#define RT5645_JD_LO_DIS (0x0 << 3) +#define RT5645_JD_LO_EN (0x1 << 3) +#define RT5645_JD_LO_TRG_MASK (0x1 << 2) +#define RT5645_JD_LO_TRG_SFT 2 +#define RT5645_JD_LO_TRG_LO (0x0 << 2) +#define RT5645_JD_LO_TRG_HI (0x1 << 2) +#define RT5645_JD1_IN4P_MASK (0x1 << 1) +#define RT5645_JD1_IN4P_SFT 1 +#define RT5645_JD1_IN4P_DIS (0x0 << 1) +#define RT5645_JD1_IN4P_EN (0x1 << 1) +#define RT5645_JD2_IN4N_MASK (0x1) +#define RT5645_JD2_IN4N_SFT 0 +#define RT5645_JD2_IN4N_DIS (0x0) +#define RT5645_JD2_IN4N_EN (0x1) + +/* Jack detect for ANC (0xbc) */ +#define RT5645_ANC_DET_MASK (0x3 << 4) +#define RT5645_ANC_DET_SFT 4 +#define RT5645_ANC_DET_DIS (0x0 << 4) +#define RT5645_ANC_DET_MB1 (0x1 << 4) +#define RT5645_ANC_DET_MB2 (0x2 << 4) +#define RT5645_ANC_DET_JD (0x3 << 4) +#define RT5645_AD_TRG_MASK (0x1 << 3) +#define RT5645_AD_TRG_SFT 3 +#define RT5645_AD_TRG_LO (0x0 << 3) +#define RT5645_AD_TRG_HI (0x1 << 3) +#define RT5645_ANCM_DET_MASK (0x3 << 4) +#define RT5645_ANCM_DET_SFT 4 +#define RT5645_ANCM_DET_DIS (0x0 << 4) +#define RT5645_ANCM_DET_MB1 (0x1 << 4) +#define RT5645_ANCM_DET_MB2 (0x2 << 4) +#define RT5645_ANCM_DET_JD (0x3 << 4) +#define RT5645_AMD_TRG_MASK (0x1 << 3) +#define RT5645_AMD_TRG_SFT 3 +#define RT5645_AMD_TRG_LO (0x0 << 3) +#define RT5645_AMD_TRG_HI (0x1 << 3) + +/* IRQ Control 1 (0xbd) */ +#define RT5645_IRQ_JD_MASK (0x1 << 15) +#define RT5645_IRQ_JD_SFT 15 +#define RT5645_IRQ_JD_BP (0x0 << 15) +#define RT5645_IRQ_JD_NOR (0x1 << 15) +#define RT5645_IRQ_OT_MASK (0x1 << 14) +#define RT5645_IRQ_OT_SFT 14 +#define RT5645_IRQ_OT_BP (0x0 << 14) +#define RT5645_IRQ_OT_NOR (0x1 << 14) +#define RT5645_JD_STKY_MASK (0x1 << 13) +#define RT5645_JD_STKY_SFT 13 +#define RT5645_JD_STKY_DIS (0x0 << 13) +#define RT5645_JD_STKY_EN (0x1 << 13) +#define RT5645_OT_STKY_MASK (0x1 << 12) +#define RT5645_OT_STKY_SFT 12 +#define RT5645_OT_STKY_DIS (0x0 << 12) +#define RT5645_OT_STKY_EN (0x1 << 12) +#define RT5645_JD_P_MASK (0x1 << 11) +#define RT5645_JD_P_SFT 11 +#define RT5645_JD_P_NOR (0x0 << 11) +#define RT5645_JD_P_INV (0x1 << 11) +#define RT5645_OT_P_MASK (0x1 << 10) +#define RT5645_OT_P_SFT 10 +#define RT5645_OT_P_NOR (0x0 << 10) +#define RT5645_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5645_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5645_IRQ_MB1_OC_SFT 15 +#define RT5645_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5645_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5645_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5645_IRQ_MB2_OC_SFT 14 +#define RT5645_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5645_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5645_MB1_OC_STKY_MASK (0x1 << 13) +#define RT5645_MB1_OC_STKY_SFT 13 +#define RT5645_MB1_OC_STKY_DIS (0x0 << 13) +#define RT5645_MB1_OC_STKY_EN (0x1 << 13) +#define RT5645_MB2_OC_STKY_MASK (0x1 << 12) +#define RT5645_MB2_OC_STKY_SFT 12 +#define RT5645_MB2_OC_STKY_DIS (0x0 << 12) +#define RT5645_MB2_OC_STKY_EN (0x1 << 12) +#define RT5645_MB1_OC_P_MASK (0x1 << 7) +#define RT5645_MB1_OC_P_SFT 7 +#define RT5645_MB1_OC_P_NOR (0x0 << 7) +#define RT5645_MB1_OC_P_INV (0x1 << 7) +#define RT5645_MB2_OC_P_MASK (0x1 << 6) +#define RT5645_MB2_OC_P_SFT 6 +#define RT5645_MB2_OC_P_NOR (0x0 << 6) +#define RT5645_MB2_OC_P_INV (0x1 << 6) +#define RT5645_MB1_OC_CLR (0x1 << 3) +#define RT5645_MB1_OC_CLR_SFT 3 +#define RT5645_MB2_OC_CLR (0x1 << 2) +#define RT5645_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5645_GP1_PIN_MASK (0x1 << 15) +#define RT5645_GP1_PIN_SFT 15 +#define RT5645_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5645_GP1_PIN_IRQ (0x1 << 15) +#define RT5645_GP2_PIN_MASK (0x1 << 14) +#define RT5645_GP2_PIN_SFT 14 +#define RT5645_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5645_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5645_GP3_PIN_MASK (0x3 << 12) +#define RT5645_GP3_PIN_SFT 12 +#define RT5645_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5645_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5645_GP3_PIN_IRQ (0x2 << 12) +#define RT5645_GP4_PIN_MASK (0x1 << 11) +#define RT5645_GP4_PIN_SFT 11 +#define RT5645_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5645_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5645_DP_SIG_MASK (0x1 << 10) +#define RT5645_DP_SIG_SFT 10 +#define RT5645_DP_SIG_TEST (0x0 << 10) +#define RT5645_DP_SIG_AP (0x1 << 10) +#define RT5645_GPIO_M_MASK (0x1 << 9) +#define RT5645_GPIO_M_SFT 9 +#define RT5645_GPIO_M_FLT (0x0 << 9) +#define RT5645_GPIO_M_PH (0x1 << 9) +#define RT5645_I2S2_SEL (0x1 << 8) +#define RT5645_I2S2_SEL_SFT 8 +#define RT5645_GP5_PIN_MASK (0x1 << 7) +#define RT5645_GP5_PIN_SFT 7 +#define RT5645_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5645_GP5_PIN_DMIC1_SDA (0x1 << 7) +#define RT5645_GP6_PIN_MASK (0x1 << 6) +#define RT5645_GP6_PIN_SFT 6 +#define RT5645_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5645_GP6_PIN_DMIC2_SDA (0x1 << 6) +#define RT5645_GP8_PIN_MASK (0x1 << 3) +#define RT5645_GP8_PIN_SFT 3 +#define RT5645_GP8_PIN_GPIO8 (0x0 << 3) +#define RT5645_GP8_PIN_DMIC2_SDA (0x1 << 3) +#define RT5645_GP12_PIN_MASK (0x1 << 2) +#define RT5645_GP12_PIN_SFT 2 +#define RT5645_GP12_PIN_GPIO12 (0x0 << 2) +#define RT5645_GP12_PIN_DMIC2_SDA (0x1 << 2) +#define RT5645_GP11_PIN_MASK (0x1 << 1) +#define RT5645_GP11_PIN_SFT 1 +#define RT5645_GP11_PIN_GPIO11 (0x0 << 1) +#define RT5645_GP11_PIN_DMIC1_SDA (0x1 << 1) +#define RT5645_GP10_PIN_MASK (0x1) +#define RT5645_GP10_PIN_SFT 0 +#define RT5645_GP10_PIN_GPIO10 (0x0) +#define RT5645_GP10_PIN_DMIC2_SDA (0x1) + +/* GPIO Control 3 (0xc2) */ +#define RT5645_GP4_PF_MASK (0x1 << 11) +#define RT5645_GP4_PF_SFT 11 +#define RT5645_GP4_PF_IN (0x0 << 11) +#define RT5645_GP4_PF_OUT (0x1 << 11) +#define RT5645_GP4_OUT_MASK (0x1 << 10) +#define RT5645_GP4_OUT_SFT 10 +#define RT5645_GP4_OUT_LO (0x0 << 10) +#define RT5645_GP4_OUT_HI (0x1 << 10) +#define RT5645_GP4_P_MASK (0x1 << 9) +#define RT5645_GP4_P_SFT 9 +#define RT5645_GP4_P_NOR (0x0 << 9) +#define RT5645_GP4_P_INV (0x1 << 9) +#define RT5645_GP3_PF_MASK (0x1 << 8) +#define RT5645_GP3_PF_SFT 8 +#define RT5645_GP3_PF_IN (0x0 << 8) +#define RT5645_GP3_PF_OUT (0x1 << 8) +#define RT5645_GP3_OUT_MASK (0x1 << 7) +#define RT5645_GP3_OUT_SFT 7 +#define RT5645_GP3_OUT_LO (0x0 << 7) +#define RT5645_GP3_OUT_HI (0x1 << 7) +#define RT5645_GP3_P_MASK (0x1 << 6) +#define RT5645_GP3_P_SFT 6 +#define RT5645_GP3_P_NOR (0x0 << 6) +#define RT5645_GP3_P_INV (0x1 << 6) +#define RT5645_GP2_PF_MASK (0x1 << 5) +#define RT5645_GP2_PF_SFT 5 +#define RT5645_GP2_PF_IN (0x0 << 5) +#define RT5645_GP2_PF_OUT (0x1 << 5) +#define RT5645_GP2_OUT_MASK (0x1 << 4) +#define RT5645_GP2_OUT_SFT 4 +#define RT5645_GP2_OUT_LO (0x0 << 4) +#define RT5645_GP2_OUT_HI (0x1 << 4) +#define RT5645_GP2_P_MASK (0x1 << 3) +#define RT5645_GP2_P_SFT 3 +#define RT5645_GP2_P_NOR (0x0 << 3) +#define RT5645_GP2_P_INV (0x1 << 3) +#define RT5645_GP1_PF_MASK (0x1 << 2) +#define RT5645_GP1_PF_SFT 2 +#define RT5645_GP1_PF_IN (0x0 << 2) +#define RT5645_GP1_PF_OUT (0x1 << 2) +#define RT5645_GP1_OUT_MASK (0x1 << 1) +#define RT5645_GP1_OUT_SFT 1 +#define RT5645_GP1_OUT_LO (0x0 << 1) +#define RT5645_GP1_OUT_HI (0x1 << 1) +#define RT5645_GP1_P_MASK (0x1) +#define RT5645_GP1_P_SFT 0 +#define RT5645_GP1_P_NOR (0x0) +#define RT5645_GP1_P_INV (0x1) + +/* Programmable Register Array Control 1 (0xc8) */ +#define RT5645_REG_SEQ_MASK (0xf << 12) +#define RT5645_REG_SEQ_SFT 12 +#define RT5645_SEQ1_ST_MASK (0x1 << 11) /*RO*/ +#define RT5645_SEQ1_ST_SFT 11 +#define RT5645_SEQ1_ST_RUN (0x0 << 11) +#define RT5645_SEQ1_ST_FIN (0x1 << 11) +#define RT5645_SEQ2_ST_MASK (0x1 << 10) /*RO*/ +#define RT5645_SEQ2_ST_SFT 10 +#define RT5645_SEQ2_ST_RUN (0x0 << 10) +#define RT5645_SEQ2_ST_FIN (0x1 << 10) +#define RT5645_REG_LV_MASK (0x1 << 9) +#define RT5645_REG_LV_SFT 9 +#define RT5645_REG_LV_MX (0x0 << 9) +#define RT5645_REG_LV_PR (0x1 << 9) +#define RT5645_SEQ_2_PT_MASK (0x1 << 8) +#define RT5645_SEQ_2_PT_BIT 8 +#define RT5645_REG_IDX_MASK (0xff) +#define RT5645_REG_IDX_SFT 0 + +/* Programmable Register Array Control 2 (0xc9) */ +#define RT5645_REG_DAT_MASK (0xffff) +#define RT5645_REG_DAT_SFT 0 + +/* Programmable Register Array Control 3 (0xca) */ +#define RT5645_SEQ_DLY_MASK (0xff << 8) +#define RT5645_SEQ_DLY_SFT 8 +#define RT5645_PROG_MASK (0x1 << 7) +#define RT5645_PROG_SFT 7 +#define RT5645_PROG_DIS (0x0 << 7) +#define RT5645_PROG_EN (0x1 << 7) +#define RT5645_SEQ1_PT_RUN (0x1 << 6) +#define RT5645_SEQ1_PT_RUN_BIT 6 +#define RT5645_SEQ2_PT_RUN (0x1 << 5) +#define RT5645_SEQ2_PT_RUN_BIT 5 + +/* Programmable Register Array Control 4 (0xcb) */ +#define RT5645_SEQ1_START_MASK (0xf << 8) +#define RT5645_SEQ1_START_SFT 8 +#define RT5645_SEQ1_END_MASK (0xf) +#define RT5645_SEQ1_END_SFT 0 + +/* Programmable Register Array Control 5 (0xcc) */ +#define RT5645_SEQ2_START_MASK (0xf << 8) +#define RT5645_SEQ2_START_SFT 8 +#define RT5645_SEQ2_END_MASK (0xf) +#define RT5645_SEQ2_END_SFT 0 + +/* Scramble Function (0xcd) */ +#define RT5645_SCB_KEY_MASK (0xff) +#define RT5645_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5645_SCB_SWAP_MASK (0x1 << 15) +#define RT5645_SCB_SWAP_SFT 15 +#define RT5645_SCB_SWAP_DIS (0x0 << 15) +#define RT5645_SCB_SWAP_EN (0x1 << 15) +#define RT5645_SCB_MASK (0x1 << 14) +#define RT5645_SCB_SFT 14 +#define RT5645_SCB_DIS (0x0 << 14) +#define RT5645_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5645_BB_MASK (0x1 << 15) +#define RT5645_BB_SFT 15 +#define RT5645_BB_DIS (0x0 << 15) +#define RT5645_BB_EN (0x1 << 15) +#define RT5645_BB_CT_MASK (0x7 << 12) +#define RT5645_BB_CT_SFT 12 +#define RT5645_BB_CT_A (0x0 << 12) +#define RT5645_BB_CT_B (0x1 << 12) +#define RT5645_BB_CT_C (0x2 << 12) +#define RT5645_BB_CT_D (0x3 << 12) +#define RT5645_M_BB_L_MASK (0x1 << 9) +#define RT5645_M_BB_L_SFT 9 +#define RT5645_M_BB_R_MASK (0x1 << 8) +#define RT5645_M_BB_R_SFT 8 +#define RT5645_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5645_M_BB_HPF_L_SFT 7 +#define RT5645_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5645_M_BB_HPF_R_SFT 6 +#define RT5645_G_BB_BST_MASK (0x3f) +#define RT5645_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5645_M_MP3_L_MASK (0x1 << 15) +#define RT5645_M_MP3_L_SFT 15 +#define RT5645_M_MP3_R_MASK (0x1 << 14) +#define RT5645_M_MP3_R_SFT 14 +#define RT5645_M_MP3_MASK (0x1 << 13) +#define RT5645_M_MP3_SFT 13 +#define RT5645_M_MP3_DIS (0x0 << 13) +#define RT5645_M_MP3_EN (0x1 << 13) +#define RT5645_EG_MP3_MASK (0x1f << 8) +#define RT5645_EG_MP3_SFT 8 +#define RT5645_MP3_HLP_MASK (0x1 << 7) +#define RT5645_MP3_HLP_SFT 7 +#define RT5645_MP3_HLP_DIS (0x0 << 7) +#define RT5645_MP3_HLP_EN (0x1 << 7) +#define RT5645_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5645_M_MP3_ORG_L_SFT 6 +#define RT5645_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5645_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5645_MP3_WT_MASK (0x1 << 13) +#define RT5645_MP3_WT_SFT 13 +#define RT5645_MP3_WT_1_4 (0x0 << 13) +#define RT5645_MP3_WT_1_2 (0x1 << 13) +#define RT5645_OG_MP3_MASK (0x1f << 8) +#define RT5645_OG_MP3_SFT 8 +#define RT5645_HG_MP3_MASK (0x3f) +#define RT5645_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5645_3D_CF_MASK (0x1 << 15) +#define RT5645_3D_CF_SFT 15 +#define RT5645_3D_CF_DIS (0x0 << 15) +#define RT5645_3D_CF_EN (0x1 << 15) +#define RT5645_3D_HP_MASK (0x1 << 14) +#define RT5645_3D_HP_SFT 14 +#define RT5645_3D_HP_DIS (0x0 << 14) +#define RT5645_3D_HP_EN (0x1 << 14) +#define RT5645_3D_BT_MASK (0x1 << 13) +#define RT5645_3D_BT_SFT 13 +#define RT5645_3D_BT_DIS (0x0 << 13) +#define RT5645_3D_BT_EN (0x1 << 13) +#define RT5645_3D_1F_MIX_MASK (0x3 << 11) +#define RT5645_3D_1F_MIX_SFT 11 +#define RT5645_3D_HP_M_MASK (0x1 << 10) +#define RT5645_3D_HP_M_SFT 10 +#define RT5645_3D_HP_M_SUR (0x0 << 10) +#define RT5645_3D_HP_M_FRO (0x1 << 10) +#define RT5645_M_3D_HRTF_MASK (0x1 << 9) +#define RT5645_M_3D_HRTF_SFT 9 +#define RT5645_M_3D_D2H_MASK (0x1 << 8) +#define RT5645_M_3D_D2H_SFT 8 +#define RT5645_M_3D_D2R_MASK (0x1 << 7) +#define RT5645_M_3D_D2R_SFT 7 +#define RT5645_M_3D_REVB_MASK (0x1 << 6) +#define RT5645_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5645_2ND_HPF_MASK (0x1 << 15) +#define RT5645_2ND_HPF_SFT 15 +#define RT5645_2ND_HPF_DIS (0x0 << 15) +#define RT5645_2ND_HPF_EN (0x1 << 15) +#define RT5645_HPF_CF_L_MASK (0x7 << 12) +#define RT5645_HPF_CF_L_SFT 12 +#define RT5645_1ST_HPF_MASK (0x1 << 11) +#define RT5645_1ST_HPF_SFT 11 +#define RT5645_1ST_HPF_DIS (0x0 << 11) +#define RT5645_1ST_HPF_EN (0x1 << 11) +#define RT5645_HPF_CF_R_MASK (0x7 << 8) +#define RT5645_HPF_CF_R_SFT 8 +#define RT5645_ZD_T_MASK (0x3 << 6) +#define RT5645_ZD_T_SFT 6 +#define RT5645_ZD_F_MASK (0x3 << 4) +#define RT5645_ZD_F_SFT 4 +#define RT5645_ZD_F_IM (0x0 << 4) +#define RT5645_ZD_F_ZC_IM (0x1 << 4) +#define RT5645_ZD_F_ZC_IOD (0x2 << 4) +#define RT5645_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5645_SI_DAC_MASK (0x1 << 11) +#define RT5645_SI_DAC_SFT 11 +#define RT5645_SI_DAC_AUTO (0x0 << 11) +#define RT5645_SI_DAC_TEST (0x1 << 11) +#define RT5645_DC_CAL_M_MASK (0x1 << 10) +#define RT5645_DC_CAL_M_SFT 10 +#define RT5645_DC_CAL_M_CAL (0x0 << 10) +#define RT5645_DC_CAL_M_NOR (0x1 << 10) +#define RT5645_DC_CAL_MASK (0x1 << 9) +#define RT5645_DC_CAL_SFT 9 +#define RT5645_DC_CAL_DIS (0x0 << 9) +#define RT5645_DC_CAL_EN (0x1 << 9) +#define RT5645_HPD_RCV_MASK (0x7 << 6) +#define RT5645_HPD_RCV_SFT 6 +#define RT5645_HPD_PS_MASK (0x1 << 5) +#define RT5645_HPD_PS_SFT 5 +#define RT5645_HPD_PS_DIS (0x0 << 5) +#define RT5645_HPD_PS_EN (0x1 << 5) +#define RT5645_CAL_M_MASK (0x1 << 4) +#define RT5645_CAL_M_SFT 4 +#define RT5645_CAL_M_DEP (0x0 << 4) +#define RT5645_CAL_M_CAL (0x1 << 4) +#define RT5645_CAL_MASK (0x1 << 3) +#define RT5645_CAL_SFT 3 +#define RT5645_CAL_DIS (0x0 << 3) +#define RT5645_CAL_EN (0x1 << 3) +#define RT5645_CAL_TEST_MASK (0x1 << 2) +#define RT5645_CAL_TEST_SFT 2 +#define RT5645_CAL_TEST_DIS (0x0 << 2) +#define RT5645_CAL_TEST_EN (0x1 << 2) +#define RT5645_CAL_P_MASK (0x3) +#define RT5645_CAL_P_SFT 0 +#define RT5645_CAL_P_NONE (0x0) +#define RT5645_CAL_P_CAL (0x1) +#define RT5645_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5645_SV_MASK (0x1 << 15) +#define RT5645_SV_SFT 15 +#define RT5645_SV_DIS (0x0 << 15) +#define RT5645_SV_EN (0x1 << 15) +#define RT5645_SPO_SV_MASK (0x1 << 14) +#define RT5645_SPO_SV_SFT 14 +#define RT5645_SPO_SV_DIS (0x0 << 14) +#define RT5645_SPO_SV_EN (0x1 << 14) +#define RT5645_OUT_SV_MASK (0x1 << 13) +#define RT5645_OUT_SV_SFT 13 +#define RT5645_OUT_SV_DIS (0x0 << 13) +#define RT5645_OUT_SV_EN (0x1 << 13) +#define RT5645_HP_SV_MASK (0x1 << 12) +#define RT5645_HP_SV_SFT 12 +#define RT5645_HP_SV_DIS (0x0 << 12) +#define RT5645_HP_SV_EN (0x1 << 12) +#define RT5645_ZCD_DIG_MASK (0x1 << 11) +#define RT5645_ZCD_DIG_SFT 11 +#define RT5645_ZCD_DIG_DIS (0x0 << 11) +#define RT5645_ZCD_DIG_EN (0x1 << 11) +#define RT5645_ZCD_MASK (0x1 << 10) +#define RT5645_ZCD_SFT 10 +#define RT5645_ZCD_PD (0x0 << 10) +#define RT5645_ZCD_PU (0x1 << 10) +#define RT5645_M_ZCD_MASK (0x3f << 4) +#define RT5645_M_ZCD_SFT 4 +#define RT5645_M_ZCD_RM_L (0x1 << 9) +#define RT5645_M_ZCD_RM_R (0x1 << 8) +#define RT5645_M_ZCD_SM_L (0x1 << 7) +#define RT5645_M_ZCD_SM_R (0x1 << 6) +#define RT5645_M_ZCD_OM_L (0x1 << 5) +#define RT5645_M_ZCD_OM_R (0x1 << 4) +#define RT5645_SV_DLY_MASK (0xf) +#define RT5645_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5645_ZCD_HP_MASK (0x1 << 15) +#define RT5645_ZCD_HP_SFT 15 +#define RT5645_ZCD_HP_DIS (0x0 << 15) +#define RT5645_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5645_3D_SPK_MASK (0x1 << 15) +#define RT5645_3D_SPK_SFT 15 +#define RT5645_3D_SPK_DIS (0x0 << 15) +#define RT5645_3D_SPK_EN (0x1 << 15) +#define RT5645_3D_SPK_M_MASK (0x3 << 13) +#define RT5645_3D_SPK_M_SFT 13 +#define RT5645_3D_SPK_CG_MASK (0x1f << 8) +#define RT5645_3D_SPK_CG_SFT 8 +#define RT5645_3D_SPK_SG_MASK (0x1f) +#define RT5645_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5645_WND_MASK (0x1 << 15) +#define RT5645_WND_SFT 15 +#define RT5645_WND_DIS (0x0 << 15) +#define RT5645_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5645_WND_FC_NW_MASK (0x3f << 10) +#define RT5645_WND_FC_NW_SFT 10 +#define RT5645_WND_FC_WK_MASK (0x3f << 4) +#define RT5645_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5645_HPF_FC_MASK (0x3f << 6) +#define RT5645_HPF_FC_SFT 6 +#define RT5645_WND_FC_ST_MASK (0x3f) +#define RT5645_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5645_WND_TH_LO_MASK (0x3ff) +#define RT5645_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5645_WND_TH_HI_MASK (0x3ff) +#define RT5645_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5645_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5645_WND_WIND_SFT 13 +#define RT5645_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5645_WND_STRONG_SFT 12 +enum { + RT5645_NO_WIND, + RT5645_BREEZE, + RT5645_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5645_DP_ATT_MASK (0x3 << 14) +#define RT5645_DP_ATT_SFT 14 +#define RT5645_DP_SPK_MASK (0x1 << 10) +#define RT5645_DP_SPK_SFT 10 +#define RT5645_DP_SPK_DIS (0x0 << 10) +#define RT5645_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5645_EQ_PRE_VOL_MASK (0xffff) +#define RT5645_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5645_EQ_PST_VOL_MASK (0xffff) +#define RT5645_EQ_PST_VOL_SFT 0 + +/* Jack Detect Control 3 (0xf8) */ +#define RT5645_CMP_MIC_IN_DET_MASK (0x7 << 12) +#define RT5645_JD_CBJ_EN (0x1 << 7) +#define RT5645_JD_CBJ_POL (0x1 << 6) +#define RT5645_JD_TRI_CBJ_SEL_MASK (0x7 << 3) +#define RT5645_JD_TRI_CBJ_SEL_SFT (3) +#define RT5645_JD_TRI_HPO_SEL_MASK (0x7) +#define RT5645_JD_TRI_HPO_SEL_SFT (0) +#define RT5645_JD_F_GPIO_JD1 (0x0) +#define RT5645_JD_F_JD1_1 (0x1) +#define RT5645_JD_F_JD1_2 (0x2) +#define RT5645_JD_F_JD2 (0x3) +#define RT5645_JD_F_JD3 (0x4) +#define RT5645_JD_F_GPIO_JD2 (0x5) +#define RT5645_JD_F_MX0B_12 (0x6) + +/* Digital Misc Control (0xfa) */ +#define RT5645_RST_DSP (0x1 << 13) +#define RT5645_IF1_ADC1_IN1_SEL (0x1 << 12) +#define RT5645_IF1_ADC1_IN1_SFT 12 +#define RT5645_IF1_ADC1_IN2_SEL (0x1 << 11) +#define RT5645_IF1_ADC1_IN2_SFT 11 +#define RT5645_IF1_ADC2_IN1_SEL (0x1 << 10) +#define RT5645_IF1_ADC2_IN1_SFT 10 +#define RT5645_DIG_GATE_CTRL 0x1 + +/* General Control2 (0xfb) */ +#define RT5645_RXDC_SRC_MASK (0x1 << 7) +#define RT5645_RXDC_SRC_STO (0x0 << 7) +#define RT5645_RXDC_SRC_MONO (0x1 << 7) +#define RT5645_RXDC_SRC_SFT (7) +#define RT5645_RXDP2_SEL_MASK (0x1 << 3) +#define RT5645_RXDP2_SEL_IF2 (0x0 << 3) +#define RT5645_RXDP2_SEL_ADC (0x1 << 3) +#define RT5645_RXDP2_SEL_SFT (3) + + +/* Vendor ID (0xfd) */ +#define RT5645_VER_C 0x2 +#define RT5645_VER_D 0x3 + + +/* Volume Rescale */ +#define RT5645_VOL_RSCL_MAX 0x27 +#define RT5645_VOL_RSCL_RANGE 0x1F +/* Debug String Length */ +#define RT5645_REG_DISP_LEN 23 + + +/* System Clock Source */ +enum { + RT5645_SCLK_S_MCLK, + RT5645_SCLK_S_PLL1, + RT5645_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5645_PLL1_S_MCLK, + RT5645_PLL1_S_BCLK1, + RT5645_PLL1_S_BCLK2, +}; + +enum { + RT5645_AIF1, + RT5645_AIF2, + RT5645_AIFS, +}; + +enum { + RT5645_DMIC_DATA_IN2P, + RT5645_DMIC_DATA_GPIO6, + RT5645_DMIC_DATA_GPIO10, + RT5645_DMIC_DATA_GPIO12, +}; + +enum { + RT5645_DMIC_DATA_IN2N, + RT5645_DMIC_DATA_GPIO5, + RT5645_DMIC_DATA_GPIO11, +}; + +struct rt5645_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5645_priv { + struct snd_soc_codec *codec; + struct rt5645_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5645_AIFS]; + int bclk[RT5645_AIFS]; + int master[RT5645_AIFS]; + + int pll_src; + int pll_in; + int pll_out; +}; + +#endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

3 5

[alsa-devel] How does UCM's api integrate with ALSA legacy api?
by Kamil Wekalski 07 May '14

07 May '14

Hello All, Please let me ask one question regarding one basic topic (I am newcomer in using ALSA in user space application). The ALSA native api and thus the handshaking between ALSA and its client seems to be based on stages as listed below. snd_pcm_open snd_pcm_hw_params_alloca snd_pcm_hw_params_any snd_pcm_hw_params_set_... snd_pcm_hw_params snd_pcm_hw_params_get_... snd_pcm_hw_params_is_... snd_pcm_hw_params_can_... snd_pcm_close How the api of UCM integrate with ALSA-client handshaking? Any document elaborating the topic in question available? For any hints special thanks in advance. kaweka

1 0

[alsa-devel] [PATCH] ASoC: Update Cirrus Logic CODEC maintainers.
by Paul Handrigan 07 May '14

07 May '14

Remove Timur Tabi as maintainer for CS4270 codec. I would like to thank Timur for his work on maintaining this driver. Signed-off-by: Paul Handrigan <paul.handrigan(a)cirrus.com> --- MAINTAINERS | 6 ------ 1 file changed, 6 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 1066264..5f2dd76 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -2245,12 +2245,6 @@ L: linux-usb(a)vger.kernel.org S: Maintained F: drivers/usb/host/ohci-ep93xx.c -CIRRUS LOGIC CS4270 SOUND DRIVER -M: Timur Tabi <timur(a)tabi.org> -L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Odd Fixes -F: sound/soc/codecs/cs4270* - CIRRUS LOGIC AUDIO CODEC DRIVERS M: Brian Austin <brian.austin(a)cirrus.com> M: Paul Handrigan <Paul.Handrigan(a)cirrus.com> -- 1.9.1

2 1

Re: [alsa-devel] Opening (even read-only) of /dev/snd/pcmC0D0p says Operation not permitted
by Sathish Kumar Balasubramaniam -ERS, HCL Tech 07 May '14

07 May '14

Ok. I found the issue. Since the playback routing configuration is not set, it failed. Once I set the playback routing configuration using amixer commands, aplay worked. Regards, B.Sathish Kumar From: Sathish Kumar Balasubramaniam -ERS, HCL Tech Sent: Tuesday, May 06, 2014 10:11 PM To: 'alsa-devel(a)alsa-project.org' Subject: Opening (even read-only) of /dev/snd/pcmC0D0p says Operation not permitted Hi, Please can anyone let me know how to debug more or fix the following issue I tried to play a wav file using aplay. Its giving the following error root@host# strace aplay a2002011001-e02.wav execve("/usr/bin/aplay", ["aplay", "a2002011001-e02.wav"], [/* 15 vars */]) = 0 ... ... open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1 EPERM (Operation not permitted) close(3) = 0 write(2, "aplay: main:722: ", 17aplay: main:722: ) = 17 write(2, "audio open error: Operation not "..., 41audio open error: Operation not permitted) = 41 write(2, "\n", 1 ) = 1 I tried to write a small program like the following err = open("/dev/snd/pcmC0D0p", O_RDONLY); (I tried O_RDWR|O_NONBLOCK|O_CLOEXEC also) printf("err = %d, errno = %s.\n", err, strerror(errno)); Even the above failed saying "err = -1, errno = Operation not permitted." I am running these as user "root". The following are my snd devices root@host:~# ls -l /dev/snd/ total 0 drwxr-xr-x 2 root root 60 Jan 1 1970 by-path crw-rw---- 1 root audio 116, 0 Jan 1 1970 controlC0 crw-rw---- 1 root audio 116, 24 Jan 1 1970 pcmC0D0c crw-rw---- 1 root audio 116, 16 Jan 1 1970 pcmC0D0p crw-rw---- 1 root audio 116, 33 Jan 1 1970 timer root@host:~# There is nothing present in my /etc/asound.conf The following are few parameters in /usr/share/alsa/alsa.conf defaults.ctl.card 0 defaults.pcm.card 0 defaults.pcm.device 0 defaults.pcm.subdevice -1 defaults.pcm.nonblock 1 defaults.pcm.compat 0 defaults.pcm.minperiodtime 5000 # in us defaults.pcm.ipc_key 5678293 defaults.pcm.ipc_gid audio defaults.pcm.ipc_perm 0660 ... ... Regards, B.Sathish Kumar ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

1 0

[alsa-devel] [PATCH 1/5] ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets()
by Lars-Peter Clausen 07 May '14

07 May '14

For CODEC to CODEC DAI links the paths are already created in snd_soc_dapm_new_pcm(). Make sure to not create additional connections in snd_soc_dapm_connect_dai_link_widgets() for CODEC to CODEC links. Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.") Cc: stable(a)vger.kernel.org (for 3.14+) Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- The series applies on topic/dapm --- sound/soc/soc-dapm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fb6c7b7..142a738 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3410,8 +3410,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) cpu_dai = rtd->cpu_dai; codec_dai = rtd->codec_dai; - /* dynamic FE links have no fixed DAI mapping */ - if (rtd->dai_link->dynamic) + /* + * dynamic FE links have no fixed DAI mapping. + * CODEC<->CODEC links have no direct connection. + */ + if (rtd->dai_link->dynamic || rtd->dai_link->params) continue; /* there is no point in connecting BE DAI links with dummies */ -- 1.8.0

2 6

[alsa-devel] [PATCH v2] alsactl: Store lockfile in /tmp
by Julian Scheel 07 May '14

07 May '14

It can not be generally assumed that the directories in which asound.state resides are writable. Instead using /tmp as location for lock files seems more reliable. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- alsactl/alsactl.c | 7 +++++++ alsactl/alsactl.h | 1 + alsactl/lock.c | 13 ++++++++++--- 3 files changed, 18 insertions(+), 3 deletions(-) diff --git a/alsactl/alsactl.c b/alsactl/alsactl.c index 6bc013f..415dfb8 100644 --- a/alsactl/alsactl.c +++ b/alsactl/alsactl.c @@ -38,6 +38,9 @@ #ifndef SYS_PIDFILE #define SYS_PIDFILE "/var/run/alsactl.pid" #endif +#ifndef SYS_LOCKPATH +#define SYS_LOCKPATH "/var/lock" +#endif int debugflag = 0; int force_restore = 1; @@ -46,6 +49,7 @@ int do_lock = 0; int use_syslog = 0; char *command; char *statefile = NULL; +char *lockpath = SYS_LOCKPATH; #define TITLE 0x0100 #define HEADER 0x0200 @@ -71,6 +75,7 @@ static struct arg args[] = { { HEADER, NULL, "Available state options:" }, { FILEARG | 'f', "file", "configuration file (default " SYS_ASOUNDRC ")" }, { 'l', "lock", "use file locking to serialize concurrent access" }, +{ FILEARG | 'D', "lock-dir", "directory to use for lock files (default " SYS_LOCKPATH ")" }, { 'F', "force", "try to restore the matching controls as much as possible" }, { 0, NULL, " (default mode)" }, { 'g', "ignore", "ignore 'No soundcards found' error" }, @@ -232,6 +237,8 @@ int main(int argc, char *argv[]) case 'l': do_lock = 1; break; + case 'D': + lockpath = optarg; case 'F': force_restore = 1; break; diff --git a/alsactl/alsactl.h b/alsactl/alsactl.h index 9109a70..6c6bee5 100644 --- a/alsactl/alsactl.h +++ b/alsactl/alsactl.h @@ -5,6 +5,7 @@ extern int do_lock; extern int use_syslog; extern char *command; extern char *statefile; +extern char *lockpath; void info_(const char *fcn, long line, const char *fmt, ...); void error_(const char *fcn, long line, const char *fmt, ...); diff --git a/alsactl/lock.c b/alsactl/lock.c index 587a109..c69e285 100644 --- a/alsactl/lock.c +++ b/alsactl/lock.c @@ -36,17 +36,24 @@ static int state_lock_(const char *file, int lock, int timeout) struct flock lck; struct stat st; char lcktxt[12]; + char *filename; char *nfile; if (!do_lock) return 0; - nfile = malloc(strlen(file) + 6); + + /* only use the actual filename, not the path */ + filename = strrchr(file, '/'); + if (!filename) + filename = file; + + nfile = malloc(strlen(lockpath) + strlen(filename) + 7); if (nfile == NULL) { error("No enough memory..."); return -ENOMEM; } - strcpy(nfile, file); - strcat(nfile, ".lock"); + + sprintf(nfile, "%s/%s.lock", lockpath, filename); lck.l_type = lock ? F_WRLCK : F_UNLCK; lck.l_whence = SEEK_SET; lck.l_start = 0; -- 1.9.2

3 5

[alsa-devel] No sound on Thinkpad T440s in Docking Station
by Joschi Brauchle 07 May '14

07 May '14

Dear all, using openSUSE 13.1 on an Thinkpad T440s, I am not getting sound once the laptop is docked to its docking station. I suspect an additional SND_PCI_QUIRK is needed for this model. "pactl list" returns ---------------- alsa.mixer_name = "Realtek ALC292" alsa.components = "HDA:10ec0292,17aa220c,00100001" ---------------- for the analog audio output sink in docked state. Hence I assume a new line *similar* to this ---------------- SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC269_FIXUP_LENOVO_DOCK), ---------------- is needed? What about the ALC292 instead of 269 Realtek chip...? Please find alsa-info.sh output attached. -- Dipl.-Ing. Joschi Brauchle, M.S. Institute for Communications Engineering (LNT) Technische Universitaet Muenchen (TUM) 80290 Munich, Germany Tel (work): +49 89 289-23474 Fax (work): +49 89 289-23490 E-mail: joschi.brauchle(a)tum.de Web: http://www.lnt.ei.tum.de/

2 15

[alsa-devel] Alsa IPV6 support
by Surabhi Mishra 07 May '14

07 May '14

Does any version of ALSA library supports IPV6 feature or is there any future plan for its release. I also need to know if there is any work in progress related to IPV6. I am not able to find any information regarding this in mail archive or change logs. Thanks And Regards, Surabhi Mishra

2 1

[alsa-devel] [RFC 0/4] Add support for merrfield audio
by Vinod Koul 07 May '14

07 May '14

This RFC patch series is dependent upon the "ASoC: Intel: rework the mfld platform driver" series This add support for Merrfield machine based on WM8958 codec and Intels Merrfield SoC. The SoC implementation is based on DPCM. At last the SST DSP driver has been added which provides the low level IPC functionality and power management. Vinod Koul (4): ASoC: Intel: add SSP BE DAIs ASoC: Intel: Add merrifield machine driver ASoC: Intel: add the low level dsp driver for mrfld ASoC: Intel: add support for mrfld DPCM platform sound/soc/intel/Kconfig | 20 + sound/soc/intel/board/Makefile | 4 + sound/soc/intel/board/merr_dpcm_wm8958.c | 619 ++++++++++ sound/soc/intel/platform-libs/controls_v2.h | 754 +++++++++++++ sound/soc/intel/platform-libs/controls_v2_dpcm.c | 1315 ++++++++++++++++++++++ sound/soc/intel/platform-libs/sst_widgets.h | 378 +++++++ sound/soc/intel/platform_ipc_v2.h | 694 ++++++++++++ sound/soc/intel/sst-mfld-platform-compress.c | 8 +- sound/soc/intel/sst-mfld-platform-pcm.c | 490 +++++++-- sound/soc/intel/sst-mfld-platform.h | 2 + sound/soc/intel/sst/Makefile | 5 + sound/soc/intel/sst/sst.c | 631 +++++++++++ sound/soc/intel/sst/sst.h | 686 +++++++++++ sound/soc/intel/sst/sst_drv_interface.c | 812 +++++++++++++ sound/soc/intel/sst/sst_ipc.c | 388 +++++++ sound/soc/intel/sst/sst_loader.c | 947 ++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 203 ++++ sound/soc/intel/sst/sst_stream.c | 529 +++++++++ 18 files changed, 8401 insertions(+), 84 deletions(-) create mode 100644 sound/soc/intel/board/Makefile create mode 100644 sound/soc/intel/board/merr_dpcm_wm8958.c create mode 100644 sound/soc/intel/platform-libs/controls_v2.h create mode 100644 sound/soc/intel/platform-libs/controls_v2_dpcm.c create mode 100644 sound/soc/intel/platform-libs/sst_widgets.h create mode 100644 sound/soc/intel/platform_ipc_v2.h create mode 100644 sound/soc/intel/sst/Makefile create mode 100644 sound/soc/intel/sst/sst.c create mode 100644 sound/soc/intel/sst/sst.h create mode 100644 sound/soc/intel/sst/sst_drv_interface.c create mode 100644 sound/soc/intel/sst/sst_ipc.c create mode 100644 sound/soc/intel/sst/sst_loader.c create mode 100644 sound/soc/intel/sst/sst_pvt.c create mode 100644 sound/soc/intel/sst/sst_stream.c

4 17

[alsa-devel] [PATCH - aconnect 1/1] fix aconnect -l option
by Karl Bongers 07 May '14

07 May '14

aconnect -l would just print usage. This fix makes it do what usage says it should do. Signed-off-by: Karl Bongers <karlbongers(a)gmail.com> diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 8c66cfd..8d6cebb 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -292,6 +292,7 @@ int main(int argc, char **argv) convert_real = 0; break; case 'l': + command = LIST; list_subs = 1; break; case 'x': -- 1.8.3.2

2 1

[alsa-devel] [PATCH] alsactl: Store lockfile in /tmp
by Julian Scheel 06 May '14

06 May '14

It can not be generally assumed that the directories in which asound.state resides are writable. Instead using /tmp as location for lock files seems more reliable. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- alsactl/lock.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/alsactl/lock.c b/alsactl/lock.c index 587a109..7ca3a09 100644 --- a/alsactl/lock.c +++ b/alsactl/lock.c @@ -36,17 +36,24 @@ static int state_lock_(const char *file, int lock, int timeout) struct flock lck; struct stat st; char lcktxt[12]; + char *filename; char *nfile; if (!do_lock) return 0; - nfile = malloc(strlen(file) + 6); + + /* only use the actual filename, not the path */ + filename = strrchr(file, '/'); + if (!filename) + filename = file; + + nfile = malloc(strlen(filename) + 10); if (nfile == NULL) { error("No enough memory..."); return -ENOMEM; } - strcpy(nfile, file); - strcat(nfile, ".lock"); + + sprintf(nfile, "/tmp/%s.lock", filename); lck.l_type = lock ? F_WRLCK : F_UNLCK; lck.l_whence = SEEK_SET; lck.l_start = 0; -- 1.9.2

3 6

Re: [alsa-devel] Opening (even read-only) of /dev/snd/pcmC0D0p says Operation not permitted
by Sathish Kumar Balasubramaniam -ERS, HCL Tech 06 May '14

06 May '14

Info: Linux kernel version is 3.4.25-ltsi ALSA version is 1.0.27.2 Audio CODEC is AK4643 Regards, B.Sathish Kumar From: Sathish Kumar Balasubramaniam -ERS, HCL Tech Sent: Tuesday, May 06, 2014 10:11 PM To: 'alsa-devel(a)alsa-project.org' Subject: Opening (even read-only) of /dev/snd/pcmC0D0p says Operation not permitted Hi, Please can anyone let me know how to debug more or fix the following issue I tried to play a wav file using aplay. Its giving the following error root@host# strace aplay a2002011001-e02.wav execve("/usr/bin/aplay", ["aplay", "a2002011001-e02.wav"], [/* 15 vars */]) = 0 ... ... open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1 EPERM (Operation not permitted) close(3) = 0 write(2, "aplay: main:722: ", 17aplay: main:722: ) = 17 write(2, "audio open error: Operation not "..., 41audio open error: Operation not permitted) = 41 write(2, "\n", 1 ) = 1 I tried to write a small program like the following err = open("/dev/snd/pcmC0D0p", O_RDONLY); (I tried O_RDWR|O_NONBLOCK|O_CLOEXEC also) printf("err = %d, errno = %s.\n", err, strerror(errno)); Even the above failed saying "err = -1, errno = Operation not permitted." I am running these as user "root". The following are my snd devices root@host:~# ls -l /dev/snd/ total 0 drwxr-xr-x 2 root root 60 Jan 1 1970 by-path crw-rw---- 1 root audio 116, 0 Jan 1 1970 controlC0 crw-rw---- 1 root audio 116, 24 Jan 1 1970 pcmC0D0c crw-rw---- 1 root audio 116, 16 Jan 1 1970 pcmC0D0p crw-rw---- 1 root audio 116, 33 Jan 1 1970 timer root@host:~# There is nothing present in my /etc/asound.conf The following are few parameters in /usr/share/alsa/alsa.conf defaults.ctl.card 0 defaults.pcm.card 0 defaults.pcm.device 0 defaults.pcm.subdevice -1 defaults.pcm.nonblock 1 defaults.pcm.compat 0 defaults.pcm.minperiodtime 5000 # in us defaults.pcm.ipc_key 5678293 defaults.pcm.ipc_gid audio defaults.pcm.ipc_perm 0660 ... ... Regards, B.Sathish Kumar ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

1 0

[alsa-devel] Opening (even read-only) of /dev/snd/pcmC0D0p says Operation not permitted
by Sathish Kumar Balasubramaniam -ERS, HCL Tech 06 May '14

06 May '14

Hi, Please can anyone let me know how to debug more or fix the following issue I tried to play a wav file using aplay. Its giving the following error root@host# strace aplay a2002011001-e02.wav execve("/usr/bin/aplay", ["aplay", "a2002011001-e02.wav"], [/* 15 vars */]) = 0 ... ... open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1 EPERM (Operation not permitted) close(3) = 0 write(2, "aplay: main:722: ", 17aplay: main:722: ) = 17 write(2, "audio open error: Operation not "..., 41audio open error: Operation not permitted) = 41 write(2, "\n", 1 ) = 1 I tried to write a small program like the following err = open("/dev/snd/pcmC0D0p", O_RDONLY); (I tried O_RDWR|O_NONBLOCK|O_CLOEXEC also) printf("err = %d, errno = %s.\n", err, strerror(errno)); Even the above failed saying "err = -1, errno = Operation not permitted." I am running these as user "root". The following are my snd devices root@host:~# ls -l /dev/snd/ total 0 drwxr-xr-x 2 root root 60 Jan 1 1970 by-path crw-rw---- 1 root audio 116, 0 Jan 1 1970 controlC0 crw-rw---- 1 root audio 116, 24 Jan 1 1970 pcmC0D0c crw-rw---- 1 root audio 116, 16 Jan 1 1970 pcmC0D0p crw-rw---- 1 root audio 116, 33 Jan 1 1970 timer root@host:~# There is nothing present in my /etc/asound.conf The following are few parameters in /usr/share/alsa/alsa.conf defaults.ctl.card 0 defaults.pcm.card 0 defaults.pcm.device 0 defaults.pcm.subdevice -1 defaults.pcm.nonblock 1 defaults.pcm.compat 0 defaults.pcm.minperiodtime 5000 # in us defaults.pcm.ipc_key 5678293 defaults.pcm.ipc_gid audio defaults.pcm.ipc_perm 0660 ... ... Regards, B.Sathish Kumar ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

1 0

[alsa-devel] No sound for a rather old laptop (ALC260).
by Fidel Leon 06 May '14

06 May '14

Hello, list. I own a rather old laptop which features a snd_hda_intel soundcard, with a ALC260 chipset. I know it works when using rather old kernels (Debian 7, for example), and when I was using Ubuntu 12.04 I could even download the "driver" from Realtek and compile from sources. However, after updating to newer releases (kernels 3.9 and beyond), I lost the ability to compile the module by hand (VERSION.H missing). For older kernels I could try some "model=xxx" combinations but as far as I see from hda_codec.c, ALC260 doesn't feature such parameter anymore (N/A). When my laptop starts up it make a few clicks and cracks (just when my Fedora initializes the module) but no sound afterwards. I have cranked every alsamixer slider to maximum but the most I get is a "very far far away sound, with a lot of skews and clicks). I already have an alsa-info.txt file to debug, may I post it here? Thanks, -- Fidel Leon fidelleon(a)mykolab.com GPG: AD74FB97 "Developers have the attention spans of slightly moronic woodland creatures."

2 2

[alsa-devel] [PATCH 1/2] ASoC: Add support for CS42L56 CODEC
by Brian Austin 06 May '14

06 May '14

This patch adds support for the Cirrus Logic Low Power Stereo I2C CODEC Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs42l56.h | 48 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l56.c | 1427 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l56.h | 175 ++++++ 5 files changed, 1657 insertions(+) create mode 100644 include/sound/cs42l56.h create mode 100644 sound/soc/codecs/cs42l56.c create mode 100644 sound/soc/codecs/cs42l56.h diff --git a/include/sound/cs42l56.h b/include/sound/cs42l56.h new file mode 100644 index 0000000..2467c8f --- /dev/null +++ b/include/sound/cs42l56.h @@ -0,0 +1,48 @@ +/* + * linux/sound/cs42l56.h -- Platform data for CS42L56 + * + * Copyright (c) 2014 Cirrus Logic Inc. + * + * 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 __CS42L56_H +#define __CS42L56_H + +struct cs42l56_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + + /* MICBIAS Level. Check datasheet Pg48 */ + unsigned int micbias_lvl; + + /* Analog Input 1A Reference 0=Single 1=Pseudo-Differential */ + unsigned int ain1a_ref_cfg; + + /* Analog Input 2A Reference 0=Single 1=Pseudo-Differential */ + unsigned int ain2a_ref_cfg; + + /* Analog Input 1B Reference 0=Single 1=Pseudo-Differential */ + unsigned int ain1b_ref_cfg; + + /* Analog Input 2B Reference 0=Single 1=Pseudo-Differential */ + unsigned int ain2b_ref_cfg; + + /* Charge Pump Freq. Check datasheet Pg62 */ + unsigned int chgfreq; + + /* HighPass Filter Right Channel Corner Frequency */ + unsigned int hpfb_freq; + + /* HighPass Filter Left Channel Corner Frequency */ + unsigned int hpfa_freq; + + /* Adaptive Power Control for LO/HP */ + unsigned int adaptive_pwr; + +}; + +#endif /* __CS42L56_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4711e50..7440e3a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT + select SND_SOC_CS42L56 if I2C && INPUT select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI @@ -290,6 +291,10 @@ config SND_SOC_CS42L52 tristate "Cirrus Logic CS42L52 CODEC" depends on I2C && INPUT +config SND_SOC_CS42L56 + tristate "Cirrus Logic CS42L56 CODEC" + depends on I2C && INPUT + config SND_SOC_CS42L73 tristate "Cirrus Logic CS42L73 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8d1930d..fef05b1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o +snd-soc-cs42l56-objs := cs42l56.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o @@ -182,6 +183,7 @@ obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o +obj-$(CONFIG_SND_SOC_CS42L56) += snd-soc-cs42l56.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c new file mode 100644 index 0000000..a18b332 --- /dev/null +++ b/sound/soc/codecs/cs42l56.c @@ -0,0 +1,1427 @@ +/* + * cs42l56.c -- CS42L56 ALSA SoC audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin <brian.austin(a)cirrus.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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/of_device.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/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs42l56.h> +#include "cs42l56.h" + +#define CS42L56_NUM_SUPPLIES 3 +static const char *const cs42l56_supply_names[CS42L56_NUM_SUPPLIES] = { + "VA", + "VCP", + "VLDO", +}; + +struct cs42l56_private { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct device *dev; + struct cs42l56_platform_data pdata; + struct regulator_bulk_data supplies[CS42L56_NUM_SUPPLIES]; + u32 mclk; + u8 mclk_prediv; + u8 mclk_div2; + u8 mclk_ratio; + u8 iface; + u8 iface_fmt; + u8 iface_inv; +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) + struct input_dev *beep; + struct work_struct beep_work; + int beep_rate; +#endif +}; + +static const struct reg_default cs42l56_reg_defaults[] = { + { 1, 0x56 }, /* r01 - ID 1 */ + { 2, 0x04 }, /* r02 - ID 2 */ + { 3, 0x7f }, /* r03 - Power Ctl 1 */ + { 4, 0xff }, /* r04 - Power Ctl 2 */ + { 5, 0x00 }, /* ro5 - Clocking Ctl 1 */ + { 6, 0x0b }, /* r06 - Clocking Ctl 2 */ + { 7, 0x00 }, /* r07 - Serial Format */ + { 8, 0x05 }, /* r08 - Class H Ctl */ + { 9, 0x0c }, /* r09 - Misc Ctl */ + { 10, 0x80 }, /* r0a - INT Status */ + { 11, 0x00 }, /* r0b - Playback Ctl */ + { 12, 0x0c }, /* r0c - DSP Mute Ctl */ + { 13, 0x00 }, /* r0d - ADCA Mixer Volume */ + { 14, 0x00 }, /* r0e - ADCB Mixer Volume */ + { 15, 0x00 }, /* r0f - PCMA Mixer Volume */ + { 16, 0x00 }, /* r10 - PCMB Mixer Volume */ + { 17, 0x00 }, /* r11 - Analog Input Advisory Volume */ + { 18, 0x00 }, /* r12 - Digital Input Advisory Volume */ + { 19, 0x00 }, /* r13 - Master A Volume */ + { 20, 0x00 }, /* r14 - Master B Volume */ + { 21, 0x00 }, /* r15 - Beep Freq / On Time */ + { 22, 0x00 }, /* r16 - Beep Volume / Off Time */ + { 23, 0x00 }, /* r17 - Beep Tone Ctl */ + { 24, 0x88 }, /* r18 - Tone Ctl */ + { 25, 0x00 }, /* r19 - Channel Mixer & Swap */ + { 26, 0x00 }, /* r1a - AIN Ref Config / ADC Mux */ + { 27, 0xa0 }, /* r1b - High-Pass Filter Ctl */ + { 28, 0x00 }, /* r1c - Misc ADC Ctl */ + { 29, 0x00 }, /* r1d - Gain & Bias Ctl */ + { 30, 0x00 }, /* r1e - PGAA Mux & Volume */ + { 31, 0x00 }, /* r1f - PGAB Mux & Volume */ + { 32, 0x00 }, /* r20 - ADCA Attenuator */ + { 33, 0x00 }, /* r21 - ADCB Attenuator */ + { 34, 0x00 }, /* r22 - ALC Enable & Attack Rate */ + { 35, 0xbf }, /* r23 - ALC Release Rate */ + { 36, 0x00 }, /* r24 - ALC Threshold */ + { 37, 0x00 }, /* r25 - Noise Gate Ctl */ + { 38, 0x00 }, /* r26 - ALC, Limiter, SFT, ZeroCross */ + { 39, 0x00 }, /* r27 - Analog Mute, LO & HP Mux */ + { 40, 0x00 }, /* r28 - HP A Volume */ + { 41, 0x00 }, /* r29 - HP B Volume */ + { 42, 0x00 }, /* r2a - LINEOUT A Volume */ + { 43, 0x00 }, /* r2b - LINEOUT B Volume */ + { 44, 0x00 }, /* r2c - Limit Threshold Ctl */ + { 45, 0x7f }, /* r2d - Limiter Ctl & Release Rate */ + { 46, 0x00 }, /* r2e - Limiter Attack Rate */ +}; + +static bool cs42l56_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L56_CHIP_ID_1: + case CS42L56_CHIP_ID_2: + case CS42L56_PWRCTL_1: + case CS42L56_PWRCTL_2: + case CS42L56_CLKCTL_1: + case CS42L56_CLKCTL_2: + case CS42L56_SERIAL_FMT: + case CS42L56_CLASSH_CTL: + case CS42L56_MISC_CTL: + case CS42L56_INT_STATUS: + case CS42L56_PLAYBACK_CTL: + case CS42L56_DSP_MUTE_CTL: + case CS42L56_ADCA_MIX_VOLUME: + case CS42L56_ADCB_MIX_VOLUME: + case CS42L56_PCMA_MIX_VOLUME: + case CS42L56_PCMB_MIX_VOLUME: + case CS42L56_ANAINPUT_ADV_VOLUME: + case CS42L56_DIGINPUT_ADV_VOLUME: + case CS42L56_MASTER_A_VOLUME: + case CS42L56_MASTER_B_VOLUME: + case CS42L56_BEEP_FREQ_ONTIME: + case CS42L56_BEEP_FREQ_OFFTIME: + case CS42L56_BEEP_TONE_CFG: + case CS42L56_TONE_CTL: + case CS42L56_CHAN_MIX_SWAP: + case CS42L56_AIN_REFCFG_ADC_MUX: + case CS42L56_HPF_CTL: + case CS42L56_MISC_ADC_CTL: + case CS42L56_GAIN_BIAS_CTL: + case CS42L56_PGAA_MUX_VOLUME: + case CS42L56_PGAB_MUX_VOLUME: + case CS42L56_ADCA_ATTENUATOR: + case CS42L56_ADCB_ATTENUATOR: + case CS42L56_ALC_EN_ATTACK_RATE: + case CS42L56_ALC_RELEASE_RATE: + case CS42L56_ALC_THRESHOLD: + case CS42L56_NOISE_GATE_CTL: + case CS42L56_ALC_LIM_SFT_ZC: + case CS42L56_AMUTE_HPLO_MUX: + case CS42L56_HPA_VOLUME: + case CS42L56_HPB_VOLUME: + case CS42L56_LOA_VOLUME: + case CS42L56_LOB_VOLUME: + case CS42L56_LIM_THRESHOLD_CTL: + case CS42L56_LIM_CTL_RELEASE_RATE: + case CS42L56_LIM_ATTACK_RATE: + return true; + default: + return false; + } +} + +static bool cs42l56_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L56_INT_STATUS: + return 1; + default: + return 0; + } +} + +static DECLARE_TLV_DB_SCALE(beep_tlv, -5000, 200, 0); +static DECLARE_TLV_DB_SCALE(hl_tlv, -6000, 50, 0); +static DECLARE_TLV_DB_SCALE(adv_tlv, -10200, 50, 0); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, 0); +static DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); +static DECLARE_TLV_DB_SCALE(preamp_tlv, 0, 1000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static const unsigned int ngnb_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(-8200, 600, 0), + 2, 5, TLV_DB_SCALE_ITEM(-7600, 300, 0), +}; +static const unsigned int ngb_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-6400, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-4600, 300, 0), +}; +static const unsigned int alc_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const char * const beep_config_text[] = { + "Off", "Single", "Multiple", "Continuous" +}; + +static const struct soc_enum beep_config_enum = + SOC_ENUM_SINGLE(CS42L56_BEEP_TONE_CFG, 6, + ARRAY_SIZE(beep_config_text), beep_config_text); + +static const char * const beep_pitch_text[] = { + "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", + "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" +}; + +static const struct soc_enum beep_pitch_enum = + SOC_ENUM_SINGLE(CS42L56_BEEP_FREQ_ONTIME, 4, + ARRAY_SIZE(beep_pitch_text), beep_pitch_text); + +static const char * const beep_ontime_text[] = { + "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", + "1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s", + "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" +}; + +static const struct soc_enum beep_ontime_enum = + SOC_ENUM_SINGLE(CS42L56_BEEP_FREQ_ONTIME, 0, + ARRAY_SIZE(beep_ontime_text), beep_ontime_text); + +static const char * const beep_offtime_text[] = { + "1.23 s", "2.58 s", "3.90 s", "5.20 s", + "6.60 s", "8.05 s", "9.35 s", "10.80 s" +}; + +static const struct soc_enum beep_offtime_enum = + SOC_ENUM_SINGLE(CS42L56_BEEP_FREQ_OFFTIME, 5, + ARRAY_SIZE(beep_offtime_text), beep_offtime_text); + +static const char * const beep_treble_text[] = { + "5kHz", "7kHz", "10kHz", "15kHz" +}; + +static const struct soc_enum beep_treble_enum = + SOC_ENUM_SINGLE(CS42L56_BEEP_TONE_CFG, 3, + ARRAY_SIZE(beep_treble_text), beep_treble_text); + +static const char * const beep_bass_text[] = { + "50Hz", "100Hz", "200Hz", "250Hz" +}; + +static const struct soc_enum beep_bass_enum = + SOC_ENUM_SINGLE(CS42L56_BEEP_TONE_CFG, 1, + ARRAY_SIZE(beep_bass_text), beep_bass_text); + +static const char * const adc_swap_text[] = { + "None", "A+B/2", "A-B/2", "Swap" +}; + +static const struct soc_enum adc_swap_enum = + SOC_ENUM_SINGLE(CS42L56_MISC_ADC_CTL, 3, + ARRAY_SIZE(adc_swap_text), adc_swap_text); + +static const char * const pgaa_mux_text[] = { + "AIN1A", "AIN2A", "AIN3A"}; + +static const struct soc_enum pgaa_mux_enum = + SOC_ENUM_SINGLE(CS42L56_PGAA_MUX_VOLUME, 0, + ARRAY_SIZE(pgaa_mux_text), + pgaa_mux_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Route", pgaa_mux_enum); + +static const char * const pgab_mux_text[] = { + "AIN1B", "AIN2B", "AIN3B"}; + +static const struct soc_enum pgab_mux_enum = + SOC_ENUM_SINGLE(CS42L56_PGAB_MUX_VOLUME, 0, + ARRAY_SIZE(pgab_mux_text), + pgab_mux_text); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Route", pgab_mux_enum); + +static const char * const adca_mux_text[] = { + "PGAA", "AIN1A", "AIN2A", "AIN3A"}; + +static const struct soc_enum adca_mux_enum = + SOC_ENUM_SINGLE(CS42L56_AIN_REFCFG_ADC_MUX, 0, + ARRAY_SIZE(adca_mux_text), + adca_mux_text); + +static const struct snd_kcontrol_new adca_mux = + SOC_DAPM_ENUM("Route", adca_mux_enum); + +static const char * const adcb_mux_text[] = { + "PGAB", "AIN1B", "AIN2B", "AIN3B"}; + +static const struct soc_enum adcb_mux_enum = + SOC_ENUM_SINGLE(CS42L56_AIN_REFCFG_ADC_MUX, 2, + ARRAY_SIZE(adcb_mux_text), + adcb_mux_text); + +static const struct snd_kcontrol_new adcb_mux = + SOC_DAPM_ENUM("Route", adcb_mux_enum); + +static const char * const left_swap_text[] = { + "Left", "LR 2", "Right"}; + +static const char * const right_swap_text[] = { + "Right", "LR 2", "Left"}; + +static const unsigned int swap_values[] = { 0, 1, 3 }; + +static const struct soc_enum adca_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 0, 3, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +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 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 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 hpa_switch = + SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 6, 1, 1); + +static const struct snd_kcontrol_new hpb_switch = + SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 4, 1, 1); + +static const struct snd_kcontrol_new loa_switch = + SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 2, 1, 1); + +static const struct snd_kcontrol_new lob_switch = + SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 0, 1, 1); + +static const char * const hploa_input_text[] = { + "DACA", "PGAA"}; + +static const struct soc_enum lineouta_input_enum = + SOC_ENUM_SINGLE(CS42L56_AMUTE_HPLO_MUX, 2, + ARRAY_SIZE(hploa_input_text), + hploa_input_text); + +static const struct snd_kcontrol_new lineouta_input = + SOC_DAPM_ENUM("Route", lineouta_input_enum); + +static const struct soc_enum hpa_input_enum = + SOC_ENUM_SINGLE(CS42L56_AMUTE_HPLO_MUX, 0, + ARRAY_SIZE(hploa_input_text), + hploa_input_text); + +static const struct snd_kcontrol_new hpa_input = + SOC_DAPM_ENUM("Route", hpa_input_enum); + +static const char * const hplob_input_text[] = { + "DACB", "PGAB"}; + +static const struct soc_enum lineoutb_input_enum = + SOC_ENUM_SINGLE(CS42L56_AMUTE_HPLO_MUX, 3, + ARRAY_SIZE(hplob_input_text), + hplob_input_text); + +static const struct snd_kcontrol_new lineoutb_input = + SOC_DAPM_ENUM("Route", lineoutb_input_enum); + +static const struct soc_enum hpb_input_enum = + SOC_ENUM_SINGLE(CS42L56_AMUTE_HPLO_MUX, 1, + ARRAY_SIZE(hplob_input_text), + hplob_input_text); + +static const struct snd_kcontrol_new hpb_input = + SOC_DAPM_ENUM("Route", hpb_input_enum); + +static const char * const dig_mux_text[] = { + "ADC", "DSP"}; + +static const struct soc_enum dig_mux_enum = + SOC_ENUM_SINGLE(CS42L56_MISC_CTL, 7, + ARRAY_SIZE(dig_mux_text), + dig_mux_text); + +static const struct snd_kcontrol_new dig_mux = + SOC_DAPM_ENUM("Route", dig_mux_enum); + +static const char * const hpf_freq_text[] = { + "1.8Hz", "119Hz", "236Hz", "464Hz" +}; + +static const struct soc_enum hpfa_freq_enum = + SOC_ENUM_SINGLE(CS42L56_HPF_CTL, 0, + ARRAY_SIZE(hpf_freq_text), hpf_freq_text); + +static const struct soc_enum hpfb_freq_enum = + SOC_ENUM_SINGLE(CS42L56_HPF_CTL, 2, + ARRAY_SIZE(hpf_freq_text), hpf_freq_text); + +static const char * const ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" +}; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L56_NOISE_GATE_CTL, 0, + ARRAY_SIZE(ng_delay_text), ng_delay_text); + +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), + 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), + 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), + SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1), + + SOC_SINGLE_TLV("Analog Advisory Volume", + CS42L56_ANAINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv), + SOC_SINGLE_TLV("Digital Advisory Volume", + 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), + 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_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv), + SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, + CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv), + + SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, + 0, 0x00, 1, tone_tlv), + SOC_SINGLE_TLV("Treble Shelving Volume", CS42L56_TONE_CTL, + 4, 0x00, 1, tone_tlv), + + SOC_DOUBLE_TLV("PGA Preamp Volume", CS42L56_GAIN_BIAS_CTL, + 4, 6, 0x02, 1, preamp_tlv), + + SOC_SINGLE("DSP Switch", CS42L56_PLAYBACK_CTL, 7, 1, 1), + SOC_SINGLE("Gang Playback Switch", CS42L56_PLAYBACK_CTL, 4, 1, 1), + SOC_SINGLE("Gang ADC Switch", CS42L56_MISC_ADC_CTL, 7, 1, 1), + SOC_SINGLE("Gang PGA Switch", CS42L56_MISC_ADC_CTL, 6, 1, 1), + + SOC_SINGLE("PCMA Invert", CS42L56_PLAYBACK_CTL, 2, 1, 1), + SOC_SINGLE("PCMB Invert", CS42L56_PLAYBACK_CTL, 3, 1, 1), + 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), + SOC_ENUM("HPFB Corner Freq", hpfb_freq_enum), + + SOC_SINGLE("Analog Soft Ramp", CS42L56_MISC_CTL, 4, 1, 1), + SOC_DOUBLE("Analog Soft Ramp Disable", CS42L56_ALC_LIM_SFT_ZC, + 7, 5, 1, 1), + SOC_SINGLE("Analog Zero Cross", CS42L56_MISC_CTL, 3, 1, 1), + SOC_DOUBLE("Analog Zero Cross Disable", CS42L56_ALC_LIM_SFT_ZC, + 6, 4, 1, 1), + SOC_SINGLE("Digital Soft Ramp", CS42L56_MISC_CTL, 2, 1, 1), + SOC_SINGLE("Digital Soft Ramp Disable", CS42L56_ALC_LIM_SFT_ZC, + 3, 1, 1), + + SOC_SINGLE("HL Deemphasis", CS42L56_PLAYBACK_CTL, 6, 1, 1), + + SOC_SINGLE("ALC Switch", CS42L56_ALC_EN_ATTACK_RATE, 6, 1, 1), + SOC_SINGLE("ALC Limit All Switch", CS42L56_ALC_RELEASE_RATE, 7, 1, 1), + SOC_SINGLE_RANGE("ALC Attack", CS42L56_ALC_EN_ATTACK_RATE, + 0, 0, 0x3f, 0), + SOC_SINGLE_RANGE("ALC Release", CS42L56_ALC_RELEASE_RATE, + 0, 0x3f, 0, 0), + SOC_SINGLE_TLV("ALC MAX", CS42L56_ALC_THRESHOLD, + 5, 0x07, 1, alc_tlv), + SOC_SINGLE_TLV("ALC MIN", CS42L56_ALC_THRESHOLD, + 2, 0x07, 1, alc_tlv), + + SOC_SINGLE("Limiter Switch", CS42L56_LIM_CTL_RELEASE_RATE, 7, 1, 1), + SOC_SINGLE("Limit All Switch", CS42L56_LIM_CTL_RELEASE_RATE, 6, 1, 1), + SOC_SINGLE_RANGE("Limiter Attack", CS42L56_LIM_ATTACK_RATE, + 0, 0, 0x3f, 0), + SOC_SINGLE_RANGE("Limiter Release", CS42L56_LIM_CTL_RELEASE_RATE, + 0, 0x3f, 0, 0), + SOC_SINGLE_TLV("Limiter MAX", CS42L56_LIM_THRESHOLD_CTL, + 5, 0x07, 1, alc_tlv), + SOC_SINGLE_TLV("Limiter Cushion", CS42L56_ALC_THRESHOLD, + 2, 0x07, 1, alc_tlv), + + SOC_SINGLE("NG Switch", CS42L56_NOISE_GATE_CTL, 6, 1, 1), + SOC_SINGLE("NG All Switch", CS42L56_NOISE_GATE_CTL, 7, 1, 1), + SOC_SINGLE("NG Boost Switch", CS42L56_NOISE_GATE_CTL, 5, 1, 1), + SOC_SINGLE_TLV("NG Unboost Threshold", CS42L56_NOISE_GATE_CTL, + 2, 0x07, 1, ngnb_tlv), + SOC_SINGLE_TLV("NG Boost Threshold", CS42L56_NOISE_GATE_CTL, + 2, 0x07, 1, ngb_tlv), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_ENUM("Beep Config", beep_config_enum), + SOC_ENUM("Beep Pitch", beep_pitch_enum), + SOC_ENUM("Beep on Time", beep_ontime_enum), + SOC_ENUM("Beep off Time", beep_offtime_enum), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L56_BEEP_FREQ_OFFTIME, + 0, 0x07, 0x23, beep_tlv), + SOC_SINGLE("Beep Tone Ctl Switch", CS42L56_BEEP_TONE_CFG, 0, 1, 1), + SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), + SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), + +}; + +static const struct snd_soc_dapm_widget cs42l56_dapm_widgets[] = { + + SND_SOC_DAPM_SIGGEN("Beep"), + SND_SOC_DAPM_SUPPLY("VBUF", CS42L56_PWRCTL_1, 5, 1, NULL, 0), + SND_SOC_DAPM_MICBIAS("MIC1 Bias", CS42L56_PWRCTL_1, 4, 1), + SND_SOC_DAPM_SUPPLY("Charge Pump", CS42L56_PWRCTL_1, 3, 1, NULL, 0), + + SND_SOC_DAPM_INPUT("AIN1A"), + SND_SOC_DAPM_INPUT("AIN2A"), + SND_SOC_DAPM_INPUT("AIN1B"), + SND_SOC_DAPM_INPUT("AIN2B"), + SND_SOC_DAPM_INPUT("AIN3A"), + SND_SOC_DAPM_INPUT("AIN3B"), + + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("Digital Output Mux", SND_SOC_NOPM, + 0, 0, &dig_mux), + + SND_SOC_DAPM_PGA("PGAA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGAB", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("PGAA Input Mux", + SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGAB Input Mux", + SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_MUX("ADCA Mux", SND_SOC_NOPM, + 0, 0, &adca_mux), + SND_SOC_DAPM_MUX("ADCB Mux", SND_SOC_NOPM, + 0, 0, &adcb_mux), + + 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_DAC("DACA", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DACB", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("HPA"), + SND_SOC_DAPM_OUTPUT("LOA"), + SND_SOC_DAPM_OUTPUT("HPB"), + SND_SOC_DAPM_OUTPUT("LOB"), + + SND_SOC_DAPM_SWITCH("Headphone Right", + CS42L56_PWRCTL_2, 4, 1, &hpb_switch), + SND_SOC_DAPM_SWITCH("Headphone Left", + CS42L56_PWRCTL_2, 6, 1, &hpa_switch), + + SND_SOC_DAPM_SWITCH("Lineout Right", + CS42L56_PWRCTL_2, 0, 1, &lob_switch), + SND_SOC_DAPM_SWITCH("Lineout Left", + CS42L56_PWRCTL_2, 2, 1, &loa_switch), + + SND_SOC_DAPM_MUX("LINEOUTA Input Mux", SND_SOC_NOPM, + 0, 0, &lineouta_input), + SND_SOC_DAPM_MUX("LINEOUTB Input Mux", SND_SOC_NOPM, + 0, 0, &lineoutb_input), + SND_SOC_DAPM_MUX("HPA Input Mux", SND_SOC_NOPM, + 0, 0, &hpa_input), + SND_SOC_DAPM_MUX("HPB Input Mux", SND_SOC_NOPM, + 0, 0, &hpb_input), + +}; + +static const struct snd_soc_dapm_route cs42l56_audio_map[] = { + + {"HiFi Capture", "DSP", "Digital Output Mux"}, + {"HiFi Capture", "ADC", "Digital Output Mux"}, + + {"Digital Output Mux", NULL, "ADCA"}, + {"Digital Output Mux", NULL, "ADCB"}, + + {"ADCB", NULL, "ADCB Mux"}, + {"ADCA", NULL, "ADCA Mux"}, + + {"ADCA Mux", NULL, "AIN3A"}, + {"ADCA Mux", NULL, "AIN2A"}, + {"ADCA Mux", NULL, "AIN1A"}, + {"ADCA Mux", NULL, "PGAA"}, + {"ADCB Mux", NULL, "AIN3B"}, + {"ADCB Mux", NULL, "AIN2B"}, + {"ADCB Mux", NULL, "AIN1B"}, + {"ADCB Mux", NULL, "PGAB"}, + + {"PGAA", "AIN1A", "PGAA Input Mux"}, + {"PGAA", "AIN2A", "PGAA Input Mux"}, + {"PGAA", "AIN3A", "PGAA Input Mux"}, + {"PGAB", "AIN1B", "PGAB Input Mux"}, + {"PGAB", "AIN2B", "PGAB Input Mux"}, + {"PGAB", "AIN3B", "PGAB Input Mux"}, + + {"PGAA Input Mux", NULL, "AIN1A"}, + {"PGAA Input Mux", NULL, "AIN2A"}, + {"PGAA Input Mux", NULL, "AIN3A"}, + {"PGAB Input Mux", NULL, "AIN1B"}, + {"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"}, + + {"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 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"}, + +}; + +struct cs42l56_clk_para { + u32 mclk; + u32 srate; + u8 ratio; +}; + +static const struct cs42l56_clk_para clk_ratio_table[] = { + /* 8k */ + { 6000000, 8000, CS42L56_MCLK_LRCLK_768 }, + { 6144000, 8000, CS42L56_MCLK_LRCLK_750 }, + { 12000000, 8000, CS42L56_MCLK_LRCLK_768 }, + { 12288000, 8000, CS42L56_MCLK_LRCLK_750 }, + { 24000000, 8000, CS42L56_MCLK_LRCLK_768 }, + { 24576000, 8000, CS42L56_MCLK_LRCLK_750 }, + /* 11.025k */ + { 5644800, 11025, CS42L56_MCLK_LRCLK_512}, + { 11289600, 11025, CS42L56_MCLK_LRCLK_512}, + { 22579200, 11025, CS42L56_MCLK_LRCLK_512 }, + /* 11.0294k */ + { 6000000, 110294, CS42L56_MCLK_LRCLK_544 }, + { 12000000, 110294, CS42L56_MCLK_LRCLK_544 }, + { 24000000, 110294, CS42L56_MCLK_LRCLK_544 }, + /* 12k */ + { 6000000, 12000, CS42L56_MCLK_LRCLK_500 }, + { 6144000, 12000, CS42L56_MCLK_LRCLK_512 }, + { 12000000, 12000, CS42L56_MCLK_LRCLK_500 }, + { 12288000, 12000, CS42L56_MCLK_LRCLK_512 }, + { 24000000, 12000, CS42L56_MCLK_LRCLK_500 }, + { 24576000, 12000, CS42L56_MCLK_LRCLK_512 }, + /* 16k */ + { 6000000, 16000, CS42L56_MCLK_LRCLK_375 }, + { 6144000, 16000, CS42L56_MCLK_LRCLK_384 }, + { 12000000, 16000, CS42L56_MCLK_LRCLK_375 }, + { 12288000, 16000, CS42L56_MCLK_LRCLK_384 }, + { 24000000, 16000, CS42L56_MCLK_LRCLK_375 }, + { 24576000, 16000, CS42L56_MCLK_LRCLK_384 }, + /* 22.050k */ + { 5644800, 22050, CS42L56_MCLK_LRCLK_256 }, + { 11289600, 22050, CS42L56_MCLK_LRCLK_256 }, + { 22579200, 22050, CS42L56_MCLK_LRCLK_256 }, + /* 22.0588k */ + { 6000000, 220588, CS42L56_MCLK_LRCLK_272 }, + { 12000000, 220588, CS42L56_MCLK_LRCLK_272 }, + { 24000000, 220588, CS42L56_MCLK_LRCLK_272 }, + /* 24k */ + { 6000000, 24000, CS42L56_MCLK_LRCLK_250 }, + { 6144000, 24000, CS42L56_MCLK_LRCLK_256 }, + { 12000000, 24000, CS42L56_MCLK_LRCLK_250 }, + { 12288000, 24000, CS42L56_MCLK_LRCLK_256 }, + { 24000000, 24000, CS42L56_MCLK_LRCLK_250 }, + { 24576000, 24000, CS42L56_MCLK_LRCLK_256 }, + /* 32k */ + { 6000000, 32000, CS42L56_MCLK_LRCLK_187P5 }, + { 6144000, 32000, CS42L56_MCLK_LRCLK_192 }, + { 12000000, 32000, CS42L56_MCLK_LRCLK_187P5 }, + { 12288000, 32000, CS42L56_MCLK_LRCLK_192 }, + { 24000000, 32000, CS42L56_MCLK_LRCLK_187P5 }, + { 24576000, 32000, CS42L56_MCLK_LRCLK_192 }, + /* 44.118k */ + { 6000000, 44118, CS42L56_MCLK_LRCLK_136 }, + { 12000000, 44118, CS42L56_MCLK_LRCLK_136 }, + { 24000000, 44118, CS42L56_MCLK_LRCLK_136 }, + /* 44.1k */ + { 5644800, 44100, CS42L56_MCLK_LRCLK_128 }, + { 11289600, 44100, CS42L56_MCLK_LRCLK_128 }, + { 22579200, 44100, CS42L56_MCLK_LRCLK_128 }, + /* 48k */ + { 6000000, 48000, CS42L56_MCLK_LRCLK_125 }, + { 6144000, 48000, CS42L56_MCLK_LRCLK_128 }, + { 12000000, 48000, CS42L56_MCLK_LRCLK_125 }, + { 12288000, 48000, CS42L56_MCLK_LRCLK_128 }, + { 24000000, 48000, CS42L56_MCLK_LRCLK_125 }, + { 24576000, 48000, CS42L56_MCLK_LRCLK_128 }, +}; + +static int cs42l56_get_mclk_ratio(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(clk_ratio_table); i++) { + if (clk_ratio_table[i].mclk == mclk && + clk_ratio_table[i].srate == rate) + return clk_ratio_table[i].ratio; + } + return -EINVAL; +} + +static int cs42l56_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS42L56_MCLK_5P6448MHZ: + case CS42L56_MCLK_6MHZ: + case CS42L56_MCLK_6P144MHZ: + cs42l56->mclk_div2 = 0; + cs42l56->mclk_prediv = 0; + break; + case CS42L56_MCLK_11P2896MHZ: + case CS42L56_MCLK_12MHZ: + case CS42L56_MCLK_12P288MHZ: + cs42l56->mclk_div2 = 1; + cs42l56->mclk_prediv = 0; + break; + case CS42L56_MCLK_22P5792MHZ: + case CS42L56_MCLK_24MHZ: + case CS42L56_MCLK_24P576MHZ: + cs42l56->mclk_div2 = 1; + cs42l56->mclk_prediv = 1; + break; + default: + return -EINVAL; + } + cs42l56->mclk = freq; + + snd_soc_update_bits(codec, CS42L56_CLKCTL_1, + CS42L56_MCLK_PREDIV_MASK, + cs42l56->mclk_prediv); + snd_soc_update_bits(codec, CS42L56_CLKCTL_1, + CS42L56_MCLK_DIV2_MASK, + cs42l56->mclk_div2); + + return 0; +} + +static int cs42l56_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cs42l56->iface = CS42L56_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + cs42l56->iface = CS42L56_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs42l56->iface_fmt = CS42L56_DIG_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + cs42l56->iface_fmt = CS42L56_DIG_FMT_LEFT_J; + break; + default: + return -EINVAL; + } + + /* sclk inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + cs42l56->iface_inv = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + cs42l56->iface_inv = CS42L56_SCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, CS42L56_CLKCTL_1, + CS42L56_MS_MODE_MASK, cs42l56->iface); + snd_soc_update_bits(codec, CS42L56_SERIAL_FMT, + CS42L56_DIG_FMT_MASK, cs42l56->iface_fmt); + snd_soc_update_bits(codec, CS42L56_CLKCTL_1, + CS42L56_SCLK_INV_MASK, cs42l56->iface_inv); + return 0; +} + +static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + /* Hit the DSP Mixer first */ + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, + CS42L56_ADCAMIX_MUTE_MASK | + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_MUTE); + /* Mute ADC's */ + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_MUTE); + /* HP And LO */ + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, + CS42L56_HP_MUTE_MASK, + CS42L56_MUTE); + snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, + CS42L56_HP_MUTE_MASK, + CS42L56_MUTE); + snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, + CS42L56_LO_MUTE_MASK, + CS42L56_MUTE); + snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, + CS42L56_LO_MUTE_MASK, + CS42L56_MUTE); + + + } else { + snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, + CS42L56_ADCAMIX_MUTE_MASK | + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, + CS42L56_HP_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, + CS42L56_HP_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, + CS42L56_LO_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, + CS42L56_LO_MUTE_MASK, + CS42L56_UNMUTE); + } + return 0; +} + +static int cs42l56_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 cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); + int ratio; + + ratio = cs42l56_get_mclk_ratio(cs42l56->mclk, params_rate(params)); + if (ratio >= 0) { + snd_soc_update_bits(codec, CS42L56_CLKCTL_2, + CS42L56_CLK_RATIO_MASK, ratio); + } else { + dev_err(codec->dev, "unsupported mclk/sclk/lrclk ratio\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42l56_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, CS42L56_CLKCTL_1, + CS42L56_MCLK_DIS_MASK, 0); + snd_soc_update_bits(codec, CS42L56_PWRCTL_1, + CS42L56_PDN_ALL_MASK, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(cs42l56->regmap, false); + regcache_sync(cs42l56->regmap); + ret = regulator_bulk_enable(ARRAY_SIZE(cs42l56->supplies), + cs42l56->supplies); + if (ret != 0) { + dev_err(cs42l56->dev, + "Failed to enable regulators: %d\n", + ret); + return ret; + } + } + snd_soc_update_bits(codec, CS42L56_PWRCTL_1, + CS42L56_PDN_ALL_MASK, 1); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L56_PWRCTL_1, + CS42L56_PDN_ALL_MASK, 1); + snd_soc_update_bits(codec, CS42L56_CLKCTL_1, + CS42L56_MCLK_DIS_MASK, 1); + regcache_cache_only(cs42l56->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(cs42l56->supplies), + cs42l56->supplies); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +#define CS42L56_RATES (SNDRV_PCM_RATE_8000_48000) + +#define CS42L56_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + + +static struct snd_soc_dai_ops cs42l56_ops = { + .hw_params = cs42l56_pcm_hw_params, + .digital_mute = cs42l56_digital_mute, + .set_fmt = cs42l56_set_dai_fmt, + .set_sysclk = cs42l56_set_sysclk, +}; + +static struct snd_soc_dai_driver cs42l56_dai = { + .name = "cs42l56", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L56_RATES, + .formats = CS42L56_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L56_RATES, + .formats = CS42L56_FORMATS, + }, + .ops = &cs42l56_ops, +}; + +static int cs42l56_suspend(struct snd_soc_codec *codec) +{ + cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l56_resume(struct snd_soc_codec *codec) +{ + cs42l56_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int beep_freq[] = { + 261, 522, 585, 667, 706, 774, 889, 1000, + 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 +}; + +static void cs42l56_beep_work(struct work_struct *work) +{ + struct cs42l56_private *cs42l56 = + container_of(work, struct cs42l56_private, beep_work); + struct snd_soc_codec *codec = cs42l56->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int i; + int val = 0; + int best = 0; + + if (cs42l56->beep_rate) { + for (i = 0; i < ARRAY_SIZE(beep_freq); i++) { + if (abs(cs42l56->beep_rate - beep_freq[i]) < + abs(cs42l56->beep_rate - beep_freq[best])) + best = i; + } + + dev_dbg(codec->dev, "Set beep rate %dHz for requested %dHz\n", + beep_freq[best], cs42l56->beep_rate); + + val = (best << CS42L56_BEEP_RATE_SHIFT); + + snd_soc_dapm_enable_pin(dapm, "Beep"); + } else { + dev_dbg(codec->dev, "Disabling beep\n"); + snd_soc_dapm_disable_pin(dapm, "Beep"); + } + + snd_soc_update_bits(codec, CS42L56_BEEP_FREQ_ONTIME, + CS42L56_BEEP_FREQ_MASK, val); + + snd_soc_dapm_sync(dapm); +} + +/* For usability define a way of injecting beep events for the device - + * many systems will not have a keyboard. + */ +static int cs42l56_beep_event(struct input_dev *dev, unsigned int type, + unsigned int code, int hz) +{ + struct snd_soc_codec *codec = input_get_drvdata(dev); + struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "Beep event %x %x\n", code, hz); + + switch (code) { + case SND_BELL: + if (hz) + hz = 261; + case SND_TONE: + break; + default: + return -1; + } + + /* Kick the beep from a workqueue */ + cs42l56->beep_rate = hz; + schedule_work(&cs42l56->beep_work); + return 0; +} + +static ssize_t cs42l56_beep_set(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct cs42l56_private *cs42l56 = dev_get_drvdata(dev); + long int time; + int ret; + + ret = kstrtol(buf, 10, &time); + if (ret != 0) + return ret; + + input_event(cs42l56->beep, EV_SND, SND_TONE, time); + + return count; +} + +static DEVICE_ATTR(beep, 0200, NULL, cs42l56_beep_set); + +static void cs42l56_init_beep(struct snd_soc_codec *codec) +{ + struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); + int ret; + + cs42l56->beep = devm_input_allocate_device(codec->dev); + if (!cs42l56->beep) { + dev_err(codec->dev, "Failed to allocate beep device\n"); + return; + } + + INIT_WORK(&cs42l56->beep_work, cs42l56_beep_work); + cs42l56->beep_rate = 0; + + cs42l56->beep->name = "CS42L56 Beep Generator"; + cs42l56->beep->phys = dev_name(codec->dev); + cs42l56->beep->id.bustype = BUS_I2C; + + cs42l56->beep->evbit[0] = BIT_MASK(EV_SND); + cs42l56->beep->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + cs42l56->beep->event = cs42l56_beep_event; + cs42l56->beep->dev.parent = codec->dev; + input_set_drvdata(cs42l56->beep, codec); + + ret = input_register_device(cs42l56->beep); + if (ret != 0) { + cs42l56->beep = NULL; + dev_err(codec->dev, "Failed to register beep device\n"); + } + + ret = device_create_file(codec->dev, &dev_attr_beep); + if (ret != 0) { + dev_err(codec->dev, "Failed to create keyclick file: %d\n", + ret); + } +} + +static void cs42l56_free_beep(struct snd_soc_codec *codec) +{ + struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); + + device_remove_file(codec->dev, &dev_attr_beep); + cancel_work_sync(&cs42l56->beep_work); + cs42l56->beep = NULL; + + snd_soc_update_bits(codec, CS42L56_BEEP_TONE_CFG, + CS42L56_BEEP_EN_MASK, 0); +} + +static int cs42l56_probe(struct snd_soc_codec *codec) +{ + cs42l56_init_beep(codec); + + cs42l56_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l56_remove(struct snd_soc_codec *codec) +{ + struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); + + cs42l56_free_beep(codec); + cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(cs42l56->supplies), cs42l56->supplies); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l56 = { + .probe = cs42l56_probe, + .remove = cs42l56_remove, + .suspend = cs42l56_suspend, + .resume = cs42l56_resume, + .set_bias_level = cs42l56_set_bias_level, + + .dapm_widgets = cs42l56_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l56_dapm_widgets), + .dapm_routes = cs42l56_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l56_audio_map), + + .controls = cs42l56_snd_controls, + .num_controls = ARRAY_SIZE(cs42l56_snd_controls), +}; + +static struct regmap_config cs42l56_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L56_MAX_REGISTER, + .reg_defaults = cs42l56_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l56_reg_defaults), + .readable_reg = cs42l56_readable_register, + .volatile_reg = cs42l56_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs42l56_handle_of_data(struct i2c_client *i2c_client, + struct cs42l56_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + u32 val32; + + if (of_property_read_bool(np, "cirrus,ain1a-reference-cfg")) + pdata->ain1a_ref_cfg = true; + + if (of_property_read_bool(np, "cirrus,ain2a-reference-cfg")) + pdata->ain2a_ref_cfg = true; + + if (of_property_read_bool(np, "cirrus,ain1b-reference-cfg")) + pdata->ain1b_ref_cfg = true; + + if (of_property_read_bool(np, "cirrus,ain2b-reference-cfg")) + pdata->ain2b_ref_cfg = true; + + if (of_property_read_u32(np, "cirrus,micbias-lvl", &val32) >= 0) + pdata->micbias_lvl = val32; + + if (of_property_read_u32(np, "cirrus,chgfreq-divisor", &val32) >= 0) + pdata->chgfreq = val32; + + if (of_property_read_u32(np, "cirrus,adaptive-pwr-cfg", &val32) >= 0) + pdata->adaptive_pwr = val32; + + if (of_property_read_u32(np, "cirrus,hpf-left-freq", &val32) >= 0) + pdata->hpfa_freq = val32; + + if (of_property_read_u32(np, "cirrus,hpf-left-freq", &val32) >= 0) + pdata->hpfb_freq = val32; + + pdata->gpio_nreset = of_get_named_gpio(np, "cirrus,gpio-nreset", 0); + + return 0; +} + +static int cs42l56_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l56_private *cs42l56; + struct cs42l56_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int ret, i; + unsigned int devid = 0; + unsigned int alpha_rev, metal_rev; + unsigned int reg; + + cs42l56 = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l56_private), + GFP_KERNEL); + if (cs42l56 == NULL) + return -ENOMEM; + cs42l56->dev = &i2c_client->dev; + + cs42l56->regmap = devm_regmap_init_i2c(i2c_client, &cs42l56_regmap); + if (IS_ERR(cs42l56->regmap)) { + ret = PTR_ERR(cs42l56->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) { + cs42l56->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l56_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, + "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + ret = cs42l56_handle_of_data(i2c_client, + &cs42l56->pdata); + if (ret != 0) + return ret; + } + cs42l56->pdata = *pdata; + } + + if (cs42l56->pdata.gpio_nreset) { + ret = gpio_request_one(cs42l56->pdata.gpio_nreset, + GPIOF_OUT_INIT_HIGH, "CS42L56 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Failed to request /RST %d: %d\n", + cs42l56->pdata.gpio_nreset, ret); + return ret; + } + gpio_set_value_cansleep(cs42l56->pdata.gpio_nreset, 0); + gpio_set_value_cansleep(cs42l56->pdata.gpio_nreset, 1); + } + + + i2c_set_clientdata(i2c_client, cs42l56); + + for (i = 0; i < ARRAY_SIZE(cs42l56->supplies); i++) + cs42l56->supplies[i].supply = cs42l56_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c_client->dev, + ARRAY_SIZE(cs42l56->supplies), + cs42l56->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42l56->supplies), + cs42l56->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_bypass(cs42l56->regmap, true); + + ret = regmap_read(cs42l56->regmap, CS42L56_CHIP_ID_1, &reg); + devid = reg & CS42L56_CHIP_ID_MASK; + if (devid != CS42L56_DEVID) { + dev_err(&i2c_client->dev, + "CS42L56 Device ID (%X). Expected %X\n", + devid, CS42L56_DEVID); + goto err_enable; + } + alpha_rev = reg & CS42L56_AREV_MASK; + metal_rev = reg & CS42L56_MTLREV_MASK; + + dev_info(&i2c_client->dev, "Cirrus Logic CS42L56 "); + dev_info(&i2c_client->dev, "Alpha Rev %X Metal Rev %X\n", + alpha_rev, metal_rev); + + regcache_cache_bypass(cs42l56->regmap, false); + + if (cs42l56->pdata.ain1a_ref_cfg) + regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, + CS42L56_AIN1A_REF_MASK, 1); + + if (cs42l56->pdata.ain1b_ref_cfg) + regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, + CS42L56_AIN1B_REF_MASK, 1); + + if (cs42l56->pdata.ain2a_ref_cfg) + regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, + CS42L56_AIN2A_REF_MASK, 1); + + if (cs42l56->pdata.ain2b_ref_cfg) + regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, + CS42L56_AIN2B_REF_MASK, 1); + + if (cs42l56->pdata.micbias_lvl) + regmap_update_bits(cs42l56->regmap, CS42L56_GAIN_BIAS_CTL, + CS42L56_MIC_BIAS_MASK, + cs42l56->pdata.micbias_lvl); + + if (cs42l56->pdata.chgfreq) + regmap_update_bits(cs42l56->regmap, CS42L56_CLASSH_CTL, + CS42L56_CHRG_FREQ_MASK, + cs42l56->pdata.chgfreq); + + if (cs42l56->pdata.hpfb_freq) + regmap_update_bits(cs42l56->regmap, CS42L56_HPF_CTL, + CS42L56_HPFB_FREQ_MASK, + cs42l56->pdata.hpfb_freq); + + if (cs42l56->pdata.hpfa_freq) + regmap_update_bits(cs42l56->regmap, CS42L56_HPF_CTL, + CS42L56_HPFA_FREQ_MASK, + cs42l56->pdata.hpfa_freq); + + if (cs42l56->pdata.adaptive_pwr) + regmap_update_bits(cs42l56->regmap, CS42L56_CLASSH_CTL, + CS42L56_ADAPT_PWR_MASK, + cs42l56->pdata.adaptive_pwr); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l56, &cs42l56_dai, 1); + if (ret < 0) + return ret; + + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42l56->supplies), + cs42l56->supplies); + return ret; +} + +static int cs42l56_i2c_remove(struct i2c_client *client) +{ + struct cs42l56_private *cs42l56 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regulator_bulk_disable(ARRAY_SIZE(cs42l56->supplies), + cs42l56->supplies); + return 0; +} + +static const struct of_device_id cs42l56_of_match[] = { + { .compatible = "cirrus,cs42l56", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs42l56_of_match); + + +static const struct i2c_device_id cs42l56_id[] = { + { "cs42l56", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs42l56_id); + +static struct i2c_driver cs42l56_i2c_driver = { + .driver = { + .name = "cs42l56", + .owner = THIS_MODULE, + .of_match_table = cs42l56_of_match, + }, + .id_table = cs42l56_id, + .probe = cs42l56_i2c_probe, + .remove = cs42l56_i2c_remove, +}; + +module_i2c_driver(cs42l56_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS42L56 driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h new file mode 100644 index 0000000..ad2b50a --- /dev/null +++ b/sound/soc/codecs/cs42l56.h @@ -0,0 +1,175 @@ +/* + * cs42l52.h -- CS42L56 ALSA SoC audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin <brian.austin(a)cirrus.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 __CS42L56_H__ +#define __CS42L56_H__ + +#define CS42L56_CHIP_ID_1 0x01 +#define CS42L56_CHIP_ID_2 0x02 +#define CS42L56_PWRCTL_1 0x03 +#define CS42L56_PWRCTL_2 0x04 +#define CS42L56_CLKCTL_1 0x05 +#define CS42L56_CLKCTL_2 0x06 +#define CS42L56_SERIAL_FMT 0x07 +#define CS42L56_CLASSH_CTL 0x08 +#define CS42L56_MISC_CTL 0x09 +#define CS42L56_INT_STATUS 0x0a +#define CS42L56_PLAYBACK_CTL 0x0b +#define CS42L56_DSP_MUTE_CTL 0x0c +#define CS42L56_ADCA_MIX_VOLUME 0x0d +#define CS42L56_ADCB_MIX_VOLUME 0x0e +#define CS42L56_PCMA_MIX_VOLUME 0x0f +#define CS42L56_PCMB_MIX_VOLUME 0x10 +#define CS42L56_ANAINPUT_ADV_VOLUME 0x11 +#define CS42L56_DIGINPUT_ADV_VOLUME 0x12 +#define CS42L56_MASTER_A_VOLUME 0x13 +#define CS42L56_MASTER_B_VOLUME 0x14 +#define CS42L56_BEEP_FREQ_ONTIME 0x15 +#define CS42L56_BEEP_FREQ_OFFTIME 0x16 +#define CS42L56_BEEP_TONE_CFG 0x17 +#define CS42L56_TONE_CTL 0x18 +#define CS42L56_CHAN_MIX_SWAP 0x19 +#define CS42L56_AIN_REFCFG_ADC_MUX 0x1a +#define CS42L56_HPF_CTL 0x1b +#define CS42L56_MISC_ADC_CTL 0x1c +#define CS42L56_GAIN_BIAS_CTL 0x1d +#define CS42L56_PGAA_MUX_VOLUME 0x1e +#define CS42L56_PGAB_MUX_VOLUME 0x1f +#define CS42L56_ADCA_ATTENUATOR 0x20 +#define CS42L56_ADCB_ATTENUATOR 0x21 +#define CS42L56_ALC_EN_ATTACK_RATE 0x22 +#define CS42L56_ALC_RELEASE_RATE 0x23 +#define CS42L56_ALC_THRESHOLD 0x24 +#define CS42L56_NOISE_GATE_CTL 0x25 +#define CS42L56_ALC_LIM_SFT_ZC 0x26 +#define CS42L56_AMUTE_HPLO_MUX 0x27 +#define CS42L56_HPA_VOLUME 0x28 +#define CS42L56_HPB_VOLUME 0x29 +#define CS42L56_LOA_VOLUME 0x2a +#define CS42L56_LOB_VOLUME 0x2b +#define CS42L56_LIM_THRESHOLD_CTL 0x2c +#define CS42L56_LIM_CTL_RELEASE_RATE 0x2d +#define CS42L56_LIM_ATTACK_RATE 0x2e + +/* Device ID and Rev ID Masks */ +#define CS42L56_DEVID 0x56 +#define CS42L56_CHIP_ID_MASK 0xff +#define CS42L56_AREV_MASK 0x1c +#define CS42L56_MTLREV_MASK 0x03 + +/* Power bit masks */ +#define CS42L56_PDN_ALL_MASK 0x01 +#define CS42L56_PDN_ADCA_MASK 0x02 +#define CS42L56_PDN_ADCB_MASK 0x04 +#define CS42L56_PDN_CHRG_MASK 0x08 +#define CS42L56_PDN_BIAS_MASK 0x10 +#define CS42L56_PDN_VBUF_MASK 0x20 +#define CS42L56_PDN_LOA_MASK 0x03 +#define CS42L56_PDN_LOB_MASK 0x0c +#define CS42L56_PDN_HPA_MASK 0x30 +#define CS42L56_PDN_HPB_MASK 0xc0 + +/* serial port and clk masks */ +#define CS42L56_MASTER_MODE 1 +#define CS42L56_SLAVE_MODE 0 +#define CS42L56_MS_MODE_MASK 0x40 +#define CS42L56_SCLK_INV 1 +#define CS42L56_SCLK_INV_MASK 0x20 +#define CS42L56_SCLK_MCLK_MASK 0x18 +#define CS42L56_MCLK_PREDIV_MASK 0x04 +#define CS42L56_MCLK_DIV2_MASK 0x02 +#define CS42L56_MCLK_DIS_MASK 0x01 +#define CS42L56_CLK_AUTO_MASK 0x20 +#define CS42L56_CLK_RATIO_MASK 0x1f +#define CS42L56_DIG_FMT_I2S 0 +#define CS42L56_DIG_FMT_LEFT_J 1 +#define CS42L56_DIG_FMT_MASK 0x08 + +/* Class H and misc ctl masks */ +#define CS42L56_ADAPT_PWR_MASK 0xc0 +#define CS42L56_CHRG_FREQ_MASK 0x0f +#define CS42L56_DIG_MUX_MASK 0x80 +#define CS42L56_ANLGSFT_MASK 0x10 +#define CS42L56_ANLGZC_MASK 0x08 +#define CS42L56_DIGSFT_MASK 0x04 +#define CS42L56_FREEZE_MASK 0x01 +#define CS42L56_MIC_BIAS_MASK 0x03 +#define CS42L56_HPFA_FREQ_MASK 0x03 +#define CS42L56_HPFB_FREQ_MASK 0xc0 +#define CS42L56_AIN1A_REF_MASK 0x10 +#define CS42L56_AIN2A_REF_MASK 0x40 +#define CS42L56_AIN1B_REF_MASK 0x20 +#define CS42L56_AIN2B_REF_MASK 0x80 + +/* Playback Capture ctl masks */ +#define CS42L56_PDN_DSP_MASK 0x80 +#define CS42L56_DEEMPH_MASK 0x40 +#define CS42L56_PLYBCK_GANG_MASK 0x10 +#define CS42L56_PCM_INV_MASK 0x0c +#define CS42L56_MUTE 1 +#define CS42L56_UNMUTE 0 +#define CS42L56_ADCAMIX_MUTE_MASK 0x40 +#define CS42L56_ADCBMIX_MUTE_MASK 0x80 +#define CS42L56_PCMAMIX_MUTE_MASK 0x10 +#define CS42L56_PCMBMIX_MUTE_MASK 0x20 +#define CS42L56_MSTB_MUTE_MASK 0x02 +#define CS42L56_MSTA_MUTE_MASK 0x01 +#define CS42L56_ADCA_MUTE_MASK 0x01 +#define CS42L56_ADCB_MUTE_MASK 0x02 +#define CS42L56_HP_MUTE_MASK 0x80 +#define CS42L56_LO_MUTE_MASK 0x80 + +/* Beep masks */ +#define CS42L56_BEEP_FREQ_MASK 0xf0 +#define CS42L56_BEEP_ONTIME_MASK 0x0f +#define CS42L56_BEEP_OFFTIME_MASK 0xe0 +#define CS42L56_BEEP_CFG_MASK 0xc0 +#define CS42L56_BEEP_TREBCF_MASK 0x18 +#define CS42L56_BEEP_BASSCF_MASK 0x06 +#define CS42L56_BEEP_TCEN_MASK 0x01 +#define CS42L56_BEEP_RATE_SHIFT 4 +#define CS42L56_BEEP_EN_MASK 0x3f + + +/* Supported MCLKS */ +#define CS42L56_MCLK_5P6448MHZ 5644800 +#define CS42L56_MCLK_6MHZ 6000000 +#define CS42L56_MCLK_6P144MHZ 6144000 +#define CS42L56_MCLK_11P2896MHZ 11289600 +#define CS42L56_MCLK_12MHZ 12000000 +#define CS42L56_MCLK_12P288MHZ 12288000 +#define CS42L56_MCLK_22P5792MHZ 22579200 +#define CS42L56_MCLK_24MHZ 24000000 +#define CS42L56_MCLK_24P576MHZ 24576000 + +/* Clock ratios */ +#define CS42L56_MCLK_LRCLK_128 0x08 +#define CS42L56_MCLK_LRCLK_125 0x09 +#define CS42L56_MCLK_LRCLK_136 0x0b +#define CS42L56_MCLK_LRCLK_192 0x0c +#define CS42L56_MCLK_LRCLK_187P5 0x0d +#define CS42L56_MCLK_LRCLK_256 0x10 +#define CS42L56_MCLK_LRCLK_250 0x11 +#define CS42L56_MCLK_LRCLK_272 0x13 +#define CS42L56_MCLK_LRCLK_384 0x14 +#define CS42L56_MCLK_LRCLK_375 0x15 +#define CS42L56_MCLK_LRCLK_512 0x18 +#define CS42L56_MCLK_LRCLK_500 0x19 +#define CS42L56_MCLK_LRCLK_544 0x1b +#define CS42L56_MCLK_LRCLK_750 0x1c +#define CS42L56_MCLK_LRCLK_768 0x1d + + +#define CS42L56_MAX_REGISTER 0x34 + +#endif -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: sta350: fix DT bindings document
by Daniel Mack 05 May '14

05 May '14

Fix a misleading property description, and denote the fact that st,output-conf and st,ch*-output-mapping have to be passed as /bits/ 8. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/st,sta350.txt | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt index 9501888..ecd7a62 100644 --- a/Documentation/devicetree/bindings/sound/st,sta350.txt +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -25,6 +25,7 @@ Optional properties: 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX 3: 1 Channel Mono-Parallel If parameter is missing, mode 0 will be enabled. + This property has to be specified as '/bits/ 8' value. - st,ch1-output-mapping: Channel 1 output mapping - st,ch2-output-mapping: Channel 2 output mapping @@ -33,6 +34,7 @@ Optional properties: 1: Channel 2 2: Channel 3 If parameter is missing, channel 1 is choosen. + This properties have to be specified as '/bits/ 8' values. - st,thermal-warning-recover: If present, thermal warning recovery is enabled. @@ -82,7 +84,7 @@ Optional properties: If not present, preset DC coefficient is used. - st,invalid-input-detect-mute: - If not present, automatic invalid input detect mute is enabled. + If present, automatic invalid input detect mute is enabled. @@ -93,12 +95,12 @@ codec: sta350@38 { reg = <0x1c>; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; - st,output-conf = <0x3>; // set output to 2-channel + st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel // (full-bridge) power, // 2-channel data-out - st,ch1-output-mapping = <0>; // set channel 1 output ch 1 - st,ch2-output-mapping = <0>; // set channel 2 output ch 1 - st,ch3-output-mapping = <0>; // set channel 3 output ch 1 + st,ch1-output-mapping = /bits/ 8 <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = /bits/ 8 <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = /bits/ 8 <0>; // set channel 3 output ch 1 st,max-power-correction; // enables power bridge // correction for THD reduction // near maximum power output -- 1.9.0

2 3

[alsa-devel] [PATCH] ASoC: Intel: Add support to unload/reload firmware modules.
by Liam Girdwood 05 May '14

05 May '14

Add some SST API calls to unload and reload firmware modules. This can be used by PM code to restore state and also allow modular FW to unload and release memory blocks. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp-priv.h | 2 ++ sound/soc/intel/sst-firmware.c | 38 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 40 insertions(+) diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index 4012134..ffb308b 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -284,6 +284,8 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, const struct firmware *fw, void *private); void sst_fw_free(struct sst_fw *sst_fw); void sst_fw_free_all(struct sst_dsp *dsp); +int sst_fw_reload(struct sst_fw *sst_fw); +void sst_fw_unload(struct sst_fw *sst_fw); /* Create/Free firmware modules */ struct sst_module *sst_module_new(struct sst_fw *sst_fw, diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 928f228..5693d9a 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,6 +30,8 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" +static void block_module_remove(struct sst_module *module); + static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { u32 i; @@ -91,6 +93,42 @@ parse_err: } EXPORT_SYMBOL_GPL(sst_fw_new); +int sst_fw_reload(struct sst_fw *sst_fw) +{ + struct sst_dsp *dsp = sst_fw->dsp; + int ret; + + dev_dbg(dsp->dev, "reloading firmware\n"); + + /* call core specific FW paser to load FW data into DSP */ + ret = dsp->ops->parse_fw(sst_fw); + if (ret < 0) + dev_err(dsp->dev, "error: parse fw failed %d\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(sst_fw_reload); + +void sst_fw_unload(struct sst_fw *sst_fw) +{ + struct sst_dsp *dsp = sst_fw->dsp; + struct sst_module *module, *tmp; + + dev_dbg(dsp->dev, "unloading firmware\n"); + + mutex_lock(&dsp->mutex); + list_for_each_entry_safe(module, tmp, &dsp->module_list, list) { + if (module->sst_fw == sst_fw) { + block_module_remove(module); + list_del(&module->list); + kfree(module); + } + } + + mutex_unlock(&dsp->mutex); +} +EXPORT_SYMBOL_GPL(sst_fw_unload); + /* free single firmware object */ void sst_fw_free(struct sst_fw *sst_fw) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix check for pdata usage before dereference.
by Liam Girdwood 05 May '14

05 May '14

This patch fixes the following dereference check ordering. sound/soc/intel/sst-haswell-pcm.c:749 hsw_pcm_probe() warn: variable dereferenced before check 'pdata' (see line 746) git remote add asoc git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout 0b708c87f66a15190fb43661c2320fd48c4dc6c8 vim +/pdata +749 sound/soc/intel/sst-haswell-pcm.c a4b12990 Mark Brown 2014-03-12 740 }; a4b12990 Mark Brown 2014-03-12 741 a4b12990 Mark Brown 2014-03-12 742 static int hsw_pcm_probe(struct snd_soc_platform *platform) a4b12990 Mark Brown 2014-03-12 743 { a4b12990 Mark Brown 2014-03-12 744 struct sst_pdata *pdata = dev_get_platdata(platform->dev); a4b12990 Mark Brown 2014-03-12 745 struct hsw_priv_data *priv_data; 0b708c87 Liam Girdwood 2014-05-02 @746 struct device *dma_dev = pdata->dma_dev; 0b708c87 Liam Girdwood 2014-05-02 747 int i, ret = 0; a4b12990 Mark Brown 2014-03-12 748 a4b12990 Mark Brown 2014-03-12 @749 if (!pdata) a4b12990 Mark Brown 2014-03-12 750 return -ENODEV; a4b12990 Mark Brown 2014-03-12 751 a4b12990 Mark Brown 2014-03-12 752 priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 8c6bd33..9d5f64a 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -772,12 +772,14 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) { struct sst_pdata *pdata = dev_get_platdata(platform->dev); struct hsw_priv_data *priv_data; - struct device *dma_dev = pdata->dma_dev; + struct device *dma_dev; int i, ret = 0; if (!pdata) return -ENODEV; + dma_dev = pdata->dma_dev; + priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); priv_data->hsw = pdata->dsp; snd_soc_platform_set_drvdata(platform, priv_data); -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: snd_soc_dapm_put_volsw: Make sure to always update the DAPM graph
by Lars-Peter Clausen 05 May '14

05 May '14

When using auto-muted controls it may happen that the register value will not change when changing a control from enabled to disabled (since the control might be physically disabled due to the auto-muting). We have to make sure to still update the DAPM graph and disconnect the mixer input. Fixes: commit 5729507 ("ASoC: dapm: Implement mixer input auto-disable") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1871dfc..5d54dfc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2791,22 +2791,19 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = dapm_kcontrol_set_value(kcontrol, val); - - if (reg != SND_SOC_NOPM) { - mask = mask << shift; - val = val << shift; - - change = snd_soc_test_bits(codec, reg, mask, val); - } - if (change) { if (reg != SND_SOC_NOPM) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + mask = mask << shift; + val = val << shift; + + if (snd_soc_test_bits(codec, reg, mask, val)) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; + card->update = &update; + } - card->update = &update; } ret = soc_dapm_mixer_update_power(card, kcontrol, connect); -- 1.8.0

2 3

[alsa-devel] [PATCH] ASoC: Make soc_find_matching_codec() static
by Lars-Peter Clausen 05 May '14

05 May '14

The function is only used locally, make it static. Fixes the following warning from sparse: sound/soc/soc-core.c:1644:22: warning: symbol 'soc_find_matching_codec' was not declared. Should it be static? Fixes: 3ca041ed ("ASoC: dt: Allow Aux Codecs to be specified using DT") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ff2b32f..e14bc6c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1641,7 +1641,8 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) } #endif -struct snd_soc_codec *soc_find_matching_codec(struct snd_soc_card *card, int num) +static struct snd_soc_codec *soc_find_matching_codec(struct snd_soc_card *card, + int num) { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; struct snd_soc_codec *codec; -- 1.8.0

3 2

[alsa-devel] [PATCH 0/5] ASoC: fsl_spdif: Add sysclk df support
by Nicolin Chen 05 May '14

05 May '14

This series of patches mainly add sysclk_df support to derive tx clock from sysclk, and meanwhile, accordingly do some additional improvements. Nicolin Chen (5): ASoC: fsl_spdif: Use clk_set_rate() for spdif root clock only ASoC: fsl_spdif: Rename all _div to _df ASoC: fsl_spdif: Add sysclk df support to derive txclk from sysclk ASoC: fsl_spdif: Print actual sample rate for debug ASoc: fsl_spdif: Add descriptions for fsl_spdif_priv sound/soc/fsl/fsl_spdif.c | 145 ++++++++++++++++++++++++++++++++-------------- sound/soc/fsl/fsl_spdif.h | 14 +++-- 2 files changed, 109 insertions(+), 50 deletions(-) -- 1.8.4

2 10

[alsa-devel] [PATCH] ASoC: rt5640: correct 5640's device ID
by bardliao＠realtek.com 05 May '14

05 May '14

From: Bard Liao <bardliao(a)realtek.com> This patch correct rt5640's device ID Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- We will send another patch to fix the issue about the "Failed to add route" error. --- sound/soc/codecs/rt5640.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index ded2059..895ca14 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -195,7 +195,7 @@ /* SW Reset & Device ID (0x00) */ #define RT5640_ID_MASK (0x3 << 1) #define RT5640_ID_5639 (0x0 << 1) -#define RT5640_ID_5640 (0x1 << 1) +#define RT5640_ID_5640 (0x2 << 1) #define RT5640_ID_5642 (0x3 << 1) -- 1.8.1.1.439.g50a6b54

4 4

[alsa-devel] [PATCH 1/2] ALSA: hda - hdmi: Set converter channel count even without sink
by Anssi Hannula 05 May '14

05 May '14

Since commit 1df5a06a ("ALSA: hda - hdmi: Fix programmed active channel count") channel count is no longer being set if monitor_present is 0. This is because setting the count was moved after the CA value is determined, which is only after the monitor_present check in hdmi_setup_audio_infoframe(). Unfortunately, in some cases, such as with a non-spec-compliant codec or with a problematic video driver, monitor_present is always 0. As a specific example, this seems to happen with gen1 ATV (SiI1390 codec), causing left-channel-only stereo playback (multi-channel playback has apparently never worked with this codec despite it reporting 8 channels, reason unknown). Simply setting converter channel count without setting the pin infoframe and channel mapping as well does not theoretically make much sense as this will just mean they are out-of-sync and multichannel playback will have a wrong channel mapping. However, adding back just setting the converter channel count even in no-monitor case is the safest change which at least fixes the stereo playback regression on SiI1390 codec. Do that. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Reported-by: Stephan Raue <stephan(a)openelec.tv> Tested-by: Stephan Raue <stephan(a)openelec.tv> Cc: <stable(a)vger.kernel.org> # 3.12+ --- sound/pci/hda/patch_hdmi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89cd0c8..1edbb9c47c2d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1127,8 +1127,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, AMP_OUT_UNMUTE); eld = &per_pin->sink_eld; - if (!eld->monitor_present) + if (!eld->monitor_present) { + hdmi_set_channel_count(codec, per_pin->cvt_nid, channels); return; + } if (!non_pcm && per_pin->chmap_set) ca = hdmi_manual_channel_allocation(channels, per_pin->chmap); -- 1.8.4.5

2 4

[alsa-devel] [RFC PATCH] ALSA: hda - hdmi: Disallow 3/4/5-channel playback to Statement D2
by Anssi Hannula 05 May '14

05 May '14

Anthem Statement D2 receiver plays only front channels when a 3/4/5-channel HDMI CA is selected. Only 2 and 6 channel maps work properly. Disallow 3/4/5-channel maps based on sink name, allowing userspace to retry with a 6-channel mode with additional silent channels. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Reported-by: Grant Warecki <gjwaudio1(a)hotmail.com> Tested-by: Grant Warecki <gjwaudio1(a)hotmail.com> --- Hi Takashi, I got a report about an issue concerning a fairly obscure high-end Anthem Statement D2 sink, and wrote this patch for it. However, I'm now having second thoughts about whether the kernel is the correct place for this, since (a) this is the only case I've encountered so far (and is obscure), and (b) these kind of issues might get fixed with fw upgrades, and (c) cross-platform media players will have to handle (possibly with manual configuration, though) these cases anyway on non-Linux... WDYT? sound/pci/hda/patch_hdmi.c | 74 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 72 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89cd0c8..0a9c19cdb511 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -64,6 +64,9 @@ struct hdmi_spec_per_cvt { /* max. connections to a widget */ #define HDA_MAX_CONNECTIONS 32 +/* sink supports only CAs with 2/6 channels */ +#define SINK_QUIRK_2_6_CHANNELS 0x0001 + struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; @@ -83,6 +86,8 @@ struct hdmi_spec_per_pin { bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ char pcm_name[8]; /* filled in build_pcm callbacks */ + int sink_quirks; /* sink-specific quirks */ + #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif @@ -342,6 +347,13 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, }; +static struct hdmi_sink_quirks { + int mfg_id; + char name[ELD_MAX_MNL + 1]; + int quirks; +} sink_quirks[] = { + { .mfg_id = 0xed40, .name = "Statement D2", .quirks = SINK_QUIRK_2_6_CHANNELS }, +}; /* * HDMI routines @@ -640,6 +652,39 @@ static int get_channel_allocation_order(int ca) return i; } +static unsigned int channels_2_6[] = { + 2, 6 +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_2_6_channels = { + .count = ARRAY_SIZE(channels_2_6), + .list = channels_2_6, + .mask = 0, +}; + +static void set_constraints_from_sink_quirks(struct snd_pcm_runtime *runtime, + int sink_quirks) +{ + if (sink_quirks & SINK_QUIRK_2_6_CHANNELS) { + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &hw_constraints_2_6_channels); + } +} + +static bool ca_allowed_by_sink_quirks(int ca, int sink_quirks) +{ + if (sink_quirks & SINK_QUIRK_2_6_CHANNELS) { + int ordered_ca = get_channel_allocation_order(ca); + if (channel_allocations[ordered_ca].channels != 2 && + channel_allocations[ordered_ca].channels != 6) + return false; + } + + return true; +} + + /* * The transformation takes two steps: * @@ -1482,6 +1527,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pin_idx); return -ENODEV; } + + set_constraints_from_sink_quirks(substream->runtime, per_pin->sink_quirks); } /* Store the updated parameters */ @@ -1518,6 +1565,24 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } +static void hdmi_set_sink_quirks(struct hdmi_spec_per_pin *per_pin) +{ + int i; + + per_pin->sink_quirks = 0; + + if (!per_pin->sink_eld.eld_valid) + return; + + for (i = 0; i < ARRAY_SIZE(sink_quirks); i++) { + if (per_pin->sink_eld.info.manufacture_id == sink_quirks[i].mfg_id && + !strcmp(per_pin->sink_eld.info.monitor_name, sink_quirks[i].name)) { + per_pin->sink_quirks = sink_quirks[i].quirks; + break; + } + } +} + static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_jack_tbl *jack; @@ -1592,6 +1657,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; + hdmi_set_sink_quirks(per_pin); + /* * Re-setup pin and infoframe. This is needed e.g. when * - sink is first plugged-in (infoframe is not set up if !monitor_present) @@ -1916,6 +1983,8 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; + int pin_idx = kcontrol->private_value; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); unsigned int __user *dst; int chs, count = 0; @@ -1934,7 +2003,8 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, int type = spec->ops.chmap_cea_alloc_validate_get_type(cap, chs); unsigned int tlv_chmap[8]; - if (type < 0) + if (type < 0 || + !ca_allowed_by_sink_quirks(cap->ca_index, per_pin->sink_quirks)) continue; if (size < 8) return -ENOMEM; @@ -2007,7 +2077,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, if (!memcmp(chmap, per_pin->chmap, sizeof(chmap))) return 0; ca = hdmi_manual_channel_allocation(ARRAY_SIZE(chmap), chmap); - if (ca < 0) + if (ca < 0 || !ca_allowed_by_sink_quirks(ca, per_pin->sink_quirks)) return -EINVAL; if (spec->ops.chmap_validate) { err = spec->ops.chmap_validate(ca, ARRAY_SIZE(chmap), chmap); -- 1.8.4.5

2 1

[alsa-devel] [Query] Play two audios simultaneously on two channels
by Prabhakar Lad 05 May '14

05 May '14

Hi folks, Is it possible to play two different audio on two separate channels ? For example a.wav on left channel and b.wav on the right channel. Any inputs on this would be helpful. Thanks, --Prabhakar

1 0

[alsa-devel] arecord - bug with big file?
by korgman 05 May '14

05 May '14

Bugtracker is down -> https://bugtrack.alsa-project.org/alsa-bug from main page -> http://alsa-project.org So, excuse my reporting here. I can't record with arecord for long periods. time arecord -f dat -D hw:0,0 | aplay Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo real 186m25.375s user 1m26.204s sys 0m20.772s It seems that this bug was also reported there. https://bugzilla.redhat.com/show_bug.cgi?id=213783 cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k3.12.9x40-122l. (custom kernel) arecord --version arecord: version 1.0.27.1 by Jaroslav Kysela <perex(a)perex.cz> I now trying with sox (rec test.wav) to check if sox is ok. Thank you for reading this!

3 2

[alsa-devel] [Query]: Play only selected channel data
by Prabhakar Lad 05 May '14

05 May '14

Hi, I am working on application which captures data from a microphone, my requirement is to play only left channel data is it possible to get to know the recieved data is for left channel or right ? Any hints clues would be helpful Regards, --Prabhakar

1 0

[alsa-devel] Regression Intel Corporation Haswell-ULT HD Audio Controller
by Delcypher 05 May '14

05 May '14

Hi, I was told to report my bug here because the ALSA bug tracker is no longer active. I'm having an issue with " Intel Corporation Haswell-ULT HD Audio Controller" card which is inside the Dell XPS13 9333 developer edition [1]. I've tried two different distributions * Arch Linux (kernel 3.14) Output of alsa-utlils-alsa-info.sh [2] * Ubuntu 14.04 LTS (kernel 3.13) Output of alsa-utlils-alsa-info.sh [3] The card behaves correctly under Ubuntu but under Arch Linux I've observed the following issues. - The microphone does not work correctly. In alsamixer I select "Internal Mic" as my capture device but when I do $ arecord -d 5 test.wav $ aplay test.wav test.wav contains silence. I've also tried enabling/disabling "CAPTURE" on the Capture channel at the same time - I don't really understand why this channel is here (seems a little bizarre) but enabling/disabling "CAPTURE" on this channel doesn't seem to make any difference anyway. Under Ubuntu the "arecord... aplay" command works correctly. - There probably isn't a way to reproduce this but when I first installed Arch Linux the sound card was put in a state where the headphone jack would not work at all. Sound would continue playing out of the speakers when the headphone jack was inserted and no sound would come out of the headphones. I did notice that if the headphone jack was only partially inserted I would get some sound sent to one of the headphones. My intuition is that the card is probably being incorrectly initialised because if you take a look at the output of alsa-utils-alsa-info.sh and diff them. Two very big differences I noticed were that - The card name (from lscpi) is "Intel Corporation Haswell-ULT HD Audio Controller" under Ubuntu but under Arch Linux it is "Intel Corporation Device 0a0c" - The card codec is "Realtek ALC668" under Ubuntu but is "Realtek ALC3661" under Arch Linux. One interesting I did notice is that under Arch Linux it is possible to use the microphone from the Audacity program by selecting "HDA Intel PCH: ALC3661Analog (hw0:0) Internal mic:0" as the input (this is not the default). Unfortunately I know of no way to make other applications use this. Any thoughts on how to fix this? [1] http://www.dell.com/uk/business/p/xps-13-linux/pd [2] https://gist.github.com/delcypher/11367224 [3] https://gist.github.com/delcypher/11367302 Thanks, Dan.

3 6

[alsa-devel] [PATCH 0/6] ASoC: Intel: rework the mfld platform driver
by Vinod Koul 05 May '14

05 May '14

This patch series reworks and reformats the existing platform driver we use. This is stepping support for seconds series of patches which will support Merrifield platform with DPCM based implementation Not much logical changes in code but modularize the existing code Vinod Koul (6): ASoC: Intel: remove FSF snail mail address ASoc: Intel: rename sst-mfld-platform.c ASoC: Intel: mark sst_set_stream_status as non static ASoC: Intel: split the pcm and compress to different files ASoC: Intel: remove unused sst-mfld platform dais ASoC: Intel: rename pcm dias to media dai sound/soc/intel/Makefile | 2 +- sound/soc/intel/sst-mfld-dsp.h | 4 - sound/soc/intel/sst-mfld-platform-compress.c | 228 ++++++++++++++++ ...sst-mfld-platform.c => sst-mfld-platform-pcm.c} | 278 ++------------------ sound/soc/intel/sst-mfld-platform.h | 9 +- 5 files changed, 254 insertions(+), 267 deletions(-) create mode 100644 sound/soc/intel/sst-mfld-platform-compress.c rename sound/soc/intel/{sst-mfld-platform.c => sst-mfld-platform-pcm.c} (62%)

1 6

[alsa-devel] ALC1150 can't playback SND_PCM_FORMAT_FLOAT without dmix
by Madura Anushanga 04 May '14

04 May '14

Hi, MSI z87-GD65 GAMING motherboard's ALC1150 plays SND_PCM_FORMAT_FLOAT with noise[1] other non float formats work well. As soon as I changed /etc/asound.conf from pcm.!default{ type hw card 1 } to pcm.dmixed { type asym playback.pcm { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio slave { channels 2 # make 6 or 5.1 channel pcm { format S16_LE # S32_LE rate 48000 # can also be 44100 type hw card 1 # your card device 0 # your device subdevice 0 #important? } period_size 1024 buffer_size 8192 } bindings { 0 0 1 1 # Uncomment below if using 6 channel # 2 2 # 3 3 # 4 4 # 5 5 } } capture.pcm "hw:0" } pcm.!default { type plug slave.pcm "dmixed" } SND_PCM_FORMAT_FLOAT plays without a problem. 1. https://github.com/manushanga/vrok/blob/master/outs/alsa.cpp - Madura A.

2 1

Re: [alsa-devel] [PATCH] ASoC: mc13783: Add devicetree support
by Lee Jones 03 May '14

03 May '14

> > > > This patch adds devicetree support for mc13783-codec. > > > > > > Applied, thanks. > > > > No Mark, no! > > What a problem, Lee? There's no problem with the patch, don't worry. Neither Mark, nor I can apply this patch independently, we need to create a shared branch somewhere, or we'll risk conflicts at merge time. -- Lee Jones Linaro STMicroelectronics Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

2 5

[alsa-devel] [PATCH 13/15] ASoC: SND_S3C_DMA_LEGACY needs S3C24XX_DMA
by Xia Kaixu 02 May '14

02 May '14

From: Arnd Bergmann <arnd(a)arndb.de> SND_S3C_DMA_LEGACY can only be set on S3C24xx, which does not (yet) support the dmaengine framework, so samsung_dma_get_ops() fails to link if S3C24XX_DMA is disabled: sound/built-in.o: In function `dma_hw_params': :(.text+0x7f310): undefined reference to `s3c_dma_get_ops' sound/built-in.o: In function `s3c_ac97_trigger': :(.text+0x7f7f0): undefined reference to `s3c_dma_get_ops' sound/built-in.o: In function `s3c_ac97_mic_trigger': :(.text+0x7f884): undefined reference to `s3c_dma_get_ops' sound/built-in.o: In function `s3c2412_i2s_trigger': :(.text+0x80944): undefined reference to `s3c2410_dma_ctrl' This makes sure S3C24XX_DMA is always enabled when we need it, just like we do it for the same dependency in SND_S3C24XX_I2S, which has the same problem. Selecting "S3C2410_DMA" as we did before does not actually have the intended effect, since that one is only used on the s3c2410 and s3c2442 SoCs of the s3c24xx family, but not the others, so we can remove this from Kconfig. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- sound/soc/samsung/Kconfig | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 99cc196..7b610a8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,7 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX select SND_S3C_DMA if !ARCH_S3C24XX select SND_S3C_DMA_LEGACY if ARCH_S3C24XX @@ -15,11 +14,11 @@ config SND_S3C_DMA tristate config SND_S3C_DMA_LEGACY + select S3C24XX_DMA tristate config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +26,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate @@ -83,7 +81,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8994 config SND_SOC_SAMSUNG_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_SOC_SAMSUNG && MACH_SMDK2443 - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -94,7 +91,6 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710 config SND_SOC_SAMSUNG_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_SOC_SAMSUNG && ARCH_S3C24XX - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 -- 1.7.9.5

4 6

[alsa-devel] [PATCH] ASoC: add SND_SOC_BYTES_EXT
by Vinod Koul 02 May '14

02 May '14

we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data for DSP modules Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc.h | 13 +++++++++++++ sound/soc/soc-core.c | 13 +++++++++++++ 2 files changed, 26 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..63202cf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -266,6 +266,13 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } +#define SND_SOC_BYTES_EXT(xname, xcount, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info_ext, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_bytes_ext) \ + {.max = xcount} } + #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -540,6 +547,8 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol); int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, @@ -1090,6 +1099,10 @@ struct soc_bytes { u32 mask; }; +struct soc_bytes_ext { + int max; +}; + /* multi register control */ struct soc_mreg_control { long min, max; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 051c006..48e4bba 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3297,6 +3297,19 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_bytes_put); +int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; + ucontrol->count = params->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); + /** * snd_soc_info_xr_sx - signed multi register info callback * @kcontrol: mreg control -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix block allocation so we only allocate blocks once.
by Liam Girdwood 02 May '14

02 May '14

Make sure we dont alloc blocks twice with requests spanning more than one block. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-firmware.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index f768710..c4e7126 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -376,10 +376,6 @@ static int block_alloc_fixed(struct sst_module *module, if (err < 0) return -ENOMEM; - /* add block */ - block->data_type = data->data_type; - list_move(&block->list, &dsp->used_block_list); - list_add(&block->module_list, &module->block_list); return 0; } -- 1.9.1

2 13

[alsa-devel] [PATCH] ASoC: Remove IS_ENABLED for INPUT in CS42L52 and WM8962
by Brian Austin 02 May '14

02 May '14

Now that INPUT is required for the CS42L52 and WM8962 we can remove the IS_ENABLED(INPUT) check in the drivers. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 12 ------------ sound/soc/codecs/wm8962.c | 12 ------------ 2 files changed, 24 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 8a7bde4..c0581c5 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -50,11 +50,9 @@ struct cs42l52_private { u8 mclksel; u32 mclk; u8 flags; -#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; -#endif }; static const struct reg_default cs42l52_reg_defaults[] = { @@ -962,7 +960,6 @@ static int cs42l52_resume(struct snd_soc_codec *codec) return 0; } -#if IS_ENABLED(CONFIG_INPUT) static int beep_rates[] = { 261, 522, 585, 667, 706, 774, 889, 1000, 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 @@ -1096,15 +1093,6 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL, CS42L52_BEEP_EN_MASK, 0); } -#else -static void cs42l52_init_beep(struct snd_soc_codec *codec) -{ -} - -static void cs42l52_free_beep(struct snd_soc_codec *codec) -{ -} -#endif static int cs42l52_probe(struct snd_soc_codec *codec) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 37986c8..b77e4c3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -74,11 +74,9 @@ struct wm8962_priv { struct regulator_bulk_data supplies[WM8962_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8962_NUM_SUPPLIES]; -#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; -#endif #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; @@ -3145,7 +3143,6 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(wm8962_mic_detect); -#if IS_ENABLED(CONFIG_INPUT) static int beep_rates[] = { 500, 1000, 2000, 4000, }; @@ -3277,15 +3274,6 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_BEEP_GENERATOR_1, WM8962_BEEP_ENA,0); } -#else -static void wm8962_init_beep(struct snd_soc_codec *codec) -{ -} - -static void wm8962_free_beep(struct snd_soc_codec *codec) -{ -} -#endif static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio) { -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback data
by Daniel Mack 02 May '14

02 May '14

From: Clemens Ladisch <clemens(a)ladisch.de> The TEAC UD-H01 firmware sends wrong feedback frequency values, thus causing the PC to send the samples at a wrong rate, which results in clicks and crackles in the output. Add a workaround to detect and fix the corruption. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> [mick37(a)gmx.de: use sender->udh01_fb_quirk rather than ep->udh01_fb_quirk in snd_usb_handle_sync_urb()] Reported-and-tested-by: Mick <mick37(a)gmx.de> Reported-and-tested-by: Andrea Messa <andr.messa(a)tiscali.it> Cc: <stable(a)vger.kernel.org> --- Clemens, I hope you're ok that I resend this patch. With Mick's minor amendment (use sender->udh01_fb_quirk rather than ep->udh01_fb_quirk), Andrea Messa also reported success with the same hardware model. Thanks, Daniel sound/usb/card.h | 1 + sound/usb/endpoint.c | 15 ++++++++++++++- 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 9867ab8..97acb90 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -92,6 +92,7 @@ struct snd_usb_endpoint { unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int syncmaxsize; /* sync endpoint packet size */ unsigned int fill_max:1; /* fill max packet size always */ + unsigned int udh01_fb_quirk:1; /* corrupted feedback data */ unsigned int datainterval; /* log_2 of data packet interval */ unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ unsigned char silence_value; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e70a87e..289f582 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -471,6 +471,10 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, ep->syncinterval = 3; ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); + + if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && + ep->syncmaxsize == 4) + ep->udh01_fb_quirk = 1; } list_add_tail(&ep->list, &chip->ep_list); @@ -1105,7 +1109,16 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, if (f == 0) return; - if (unlikely(ep->freqshift == INT_MIN)) { + if (unlikely(sender->udh01_fb_quirk)) { + /* + * The TEAC UD-H01 firmware sometimes changes the feedback value + * by +/- 0x1.0000. + */ + if (f < ep->freqn - 0x8000) + f += 0x10000; + else if (f > ep->freqn + 0x8000) + f -= 0x10000; + } else if (unlikely(ep->freqshift == INT_MIN)) { /* * The first time we see a feedback value, determine its format * by shifting it left or right until it matches the nominal -- 1.9.0

2 1

Re: [alsa-devel] [PATCH] ASoC: fsl: select SND_SOC_IMX_PCM_DMA where needed
by Arnd Bergmann 01 May '14

01 May '14

On Tuesday 29 April 2014 07:07:33 Alexander Shiyan wrote: > Mon, 28 Apr 2014 23:12:14 +0200 от Arnd Bergmann <arnd(a)arndb.de>: > > On Tuesday 29 April 2014 00:35:41 Alexander Shiyan wrote: > > > > > > So, I don't understand why this error happen, as well as I can not > > > reproduce this... > > > > It's probably CONFIG_SND_SOC_IMX_PCM_DMA=m and CONFIG_SND_SOC_FSL_SSI=y then. > > What is the intended behavior in this case? Should CONFIG_SND_SOC_FSL_SSI > > be forced to be a module as well? > > Hmm, yes... > I thought that I had already solved a similar problem for the earlier > version of the patch ... How about this? Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index fb26345..af0bb92 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -9,6 +9,7 @@ config SND_SOC_FSL_SAI config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" + depends on m || SND_SOC_IMX_PCM_DMA != m help Say Y if you want to add Synchronous Serial Interface (SSI) support for the Freescale CPUs. @@ -18,6 +19,7 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate "Sony/Philips Digital Interface module support" select REGMAP_MMIO + depends on m || SND_SOC_IMX_PCM_DMA != m help Say Y if you want to add Sony/Philips Digital Interface (SPDIF) support for the Freescale CPUs.

3 7

[alsa-devel] [PATCH 14/15] ASoC: Amstrad E3 needs TTY support for codec
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> The cx20442 codec driver used here requires the TTY layer to be enabled, or we get a link error: sound/built-in.o: In function `cx20442_codec_remove': cx20442.c:398: undefined reference to `tty_hangup' sound/built-in.o: In function `ams_delta_remove': ams-delta.c:613: undefined reference to `tty_unregister_ldisc' sound/built-in.o: In function `ams_delta_cx20442_init': ams-delta.c:559: undefined reference to `tty_register_ldisc' This adds the missing dependency in the E3 configuration, there was already one for the codec. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: linux-omap(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- sound/soc/omap/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 2796af9..d44463a 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -37,7 +37,7 @@ config SND_OMAP_SOC_RX51 config SND_OMAP_SOC_AMS_DELTA tristate "SoC Audio support for Amstrad E3 (Delta) videophone" - depends on SND_OMAP_SOC && MACH_AMS_DELTA + depends on SND_OMAP_SOC && MACH_AMS_DELTA && TTY select SND_OMAP_SOC_MCBSP select SND_SOC_CX20442 help -- 1.7.9.5

2 1

[alsa-devel] [PATCH 09/15] ASoC: Migo-R sound needs I2C
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> The WM8978 driver needs I2C to be enabled, so the SND_SIU_MIGOR option also requires this. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-kernel(a)lists.infradead.org --- sound/soc/sh/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index ff60e11..b43fdf0 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -56,7 +56,7 @@ config SND_SH7760_AC97 config SND_SIU_MIGOR tristate "SIU sound support on Migo-R" - depends on SH_MIGOR + depends on SH_MIGOR && I2C select SND_SOC_SH4_SIU select SND_SOC_WM8978 help -- 1.7.9.5

2 1

[alsa-devel] [PATCH 10/15] ASoC: TLV320AIC23 and Simtec Hermes audio
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> This codec requires I2C to be enabled, so any other option that selects it should also depend on I2C. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b09b5a4..9fd6f62 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -116,14 +116,14 @@ config SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23 tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" - depends on SND_SOC_SAMSUNG && ARCH_S3C24XX + depends on SND_SOC_SAMSUNG && ARCH_S3C24XX && I2C select SND_S3C24XX_I2S select SND_SOC_TLV320AIC23_I2C select SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_SIMTEC_HERMES tristate "SoC I2S Audio support for Simtec Hermes board" - depends on SND_SOC_SAMSUNG && ARCH_S3C24XX + depends on SND_SOC_SAMSUNG && ARCH_S3C24XX && I2C select SND_S3C24XX_I2S select SND_SOC_TLV320AIC3X select SND_SOC_SAMSUNG_SIMTEC -- 1.7.9.5

2 1

[alsa-devel] [PATCH 12/15] ASoC: Atmel WM8904 codec support needs I2C
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> The WM8904 codec driver needs I2C to be enabled, so the SND_ATMEL_SOC_WM8904 option also requires this. Found using randconfig build testing. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/atmel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 4789619..27e3fc4 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -35,7 +35,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_ATMEL_SOC_WM8904 tristate "Atmel ASoC driver for boards using WM8904 codec" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && I2C select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8904 -- 1.7.9.5

2 1

[alsa-devel] [PATCH 11/15] ASoC: WM0010 needs SPI
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> The missing dependency can lead to build errors, so make it explicit in Kconfig. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 9fd6f62..99cc196 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -201,7 +201,7 @@ config SND_SOC_SPEYSIDE select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 - select SND_SOC_WM0010 + select SND_SOC_WM0010 if SPI select SND_SOC_WM1250_EV1 config SND_SOC_TOBERMORY @@ -217,7 +217,7 @@ config SND_SOC_BELLS select SND_SOC_WM5102 select SND_SOC_WM5110 select SND_SOC_WM9081 - select SND_SOC_WM0010 + select SND_SOC_WM0010 if SPI select SND_SOC_WM1250_EV1 config SND_SOC_LOWLAND -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH v2 1/2] ASoC: Add driver for CLPS711X DAI interface
by Mark Brown 01 May '14

01 May '14

On Thu, May 01, 2014 at 10:54:32PM +0400, Alexander Shiyan wrote: > Thu, 1 May 2014 11:29:35 -0700 от Mark Brown <broonie(a)kernel.org>: > > This machinery seems pretty much identical to the FIQ code for i.MX - > > I know you don't think anything can be shared but can you please be more > > explicit about why? The contents of the FIQ are different but the > > start and stop mechanics don't look so different (and given that the > > general idea of what they do is very similar this isn't surprising to > > me). > That is, you really want to make some common code for this? Well, I'd like to understand why we're not doing common code in more detail - it's pretty fiddly and detailed stuff to be cut'n'pasting. On the other hand if it looks horrible to do that then perhaps that's actually the best way to go. Either factor it out or give a bit more detail on why that's not the way forwards. > OK, I will review this again, but where do you think it should be located? A file in sound/soc (soc-pcm-fiq.c or something) perhaps?

1 0

[alsa-devel] [PATCH v2 2/2] ASoC: cirrus: clps711x: Add bindings documentation for the CLPS711X DAI
by Alexander Shiyan 01 May '14

01 May '14

This patch adds the devicetree documentation for the Cirrus Logic CLPS711X Digital Audio Interface (DAI). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- .../bindings/sound/cirrus,clps711x-dai.txt | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt b/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt new file mode 100644 index 0000000..fd37774 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,clps711x-dai.txt @@ -0,0 +1,20 @@ +* Cirrus Logic CLPS711X Digital Audio Interface (DAI) + +Required properties: +- compatible: Shall contain "cirrus,clps711x-dai". +- reg: Should contain registers location and length. +- clocks: phandle + clock specifier pair of the CPU PLL reference + clock and (optionally) external MCLK reference clock. +- clock-names: The clock should be named "pll" for CPU PLL and "mclk" + for external MCLK. +- interrupts: Should contain DAI interrupt number. + +Example: + dai: dai@80002000 { + #sound-dai-cells = <0>; + compatible = "cirrus,ep7312-dai", "cirrus,clps711x-dai"; + reg = <0x80002000 0x604>; + clocks = <&clks 3>, <&clk12288>; + clock-names = "pll", "mclk"; + interrupts = <32>; + }; -- 1.8.3.2

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: Add driver for CLPS711X DAI interface
by Alexander Shiyan 01 May '14

01 May '14

This patch adds a driver for the DAI interface found on Cirrus Logic CLPS711X family CPUs. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/cirrus/Kconfig | 8 + sound/soc/cirrus/Makefile | 5 + sound/soc/cirrus/clps711x-dai.c | 506 ++++++++++++++++++++++++++++++++++++++++ sound/soc/cirrus/clps711x-dai.h | 61 +++++ sound/soc/cirrus/clps711x-fiq.S | 56 +++++ 5 files changed, 636 insertions(+) create mode 100644 sound/soc/cirrus/clps711x-dai.c create mode 100644 sound/soc/cirrus/clps711x-dai.h create mode 100644 sound/soc/cirrus/clps711x-fiq.S diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 5477c54..3c8d040 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,3 +1,11 @@ +config SND_CLPS711X_SOC + tristate "SoC Audio support for the Cirrus Logic CLPS711X CPUs" + depends on ARCH_CLPS711X || (ARM && COMPILE_TEST) + select FIQ + help + Say Y or M if you want to add support for codecs attached to + the CLPS711X DAI interface. + config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" depends on ARCH_EP93XX || COMPILE_TEST diff --git a/sound/soc/cirrus/Makefile b/sound/soc/cirrus/Makefile index 5514146..f5d72fc 100644 --- a/sound/soc/cirrus/Makefile +++ b/sound/soc/cirrus/Makefile @@ -1,3 +1,8 @@ +# CLPS711X Platform Support +snd-soc-clps711x-objs := clps711x-dai.o clps711x-fiq.o + +obj-$(CONFIG_SND_CLPS711X_SOC) += snd-soc-clps711x.o + # EP93xx Platform Support snd-soc-ep93xx-objs := ep93xx-pcm.o snd-soc-ep93xx-i2s-objs := ep93xx-i2s.o diff --git a/sound/soc/cirrus/clps711x-dai.c b/sound/soc/cirrus/clps711x-dai.c new file mode 100644 index 0000000..ed677c8 --- /dev/null +++ b/sound/soc/cirrus/clps711x-dai.c @@ -0,0 +1,506 @@ +/* + * Currus Logic CLPS711X DAI driver + * + * Copyright (C) 2014 Alexander Shiyan <shc_work(a)mail.ru> + * + * 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/clk.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> + +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/clps711x.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/fiq.h> + +#include "clps711x-dai.h" + +#define CLPS711X_FMTS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE) +#define CLPS711X_RATES (SNDRV_PCM_RATE_8000_48000) + +struct clps711x_dai { + u32 head; + u32 last_ptr; + + struct hrtimer hrt; + unsigned long reload; + + unsigned long pll_hz; + unsigned long ext_hz; + + void __iomem *base; + int irq; + + struct pt_regs regs; + struct fiq_handler fiq; + atomic_t running; + + struct snd_pcm_substream *substream; +}; + +static int clps711x_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_LEFT_J) + return -EINVAL; + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + return 0; +} + +static int clps711x_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct clps711x_dai *s = dev_get_drvdata(dai->dev); + + s->ext_hz = freq; + + return 0; +} + +static int clps711x_dai_update_best_err(unsigned int clk, unsigned int div, + unsigned int rate, int *besterr) +{ + int err = abs(DIV_ROUND_CLOSEST(clk, div) - rate); + + if ((*besterr < 0) || (*besterr > err)) { + *besterr = err; + return 0; + } + + return 1; +} + +static int clps711x_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct clps711x_dai *s = dev_get_drvdata(dai->dev); + int i, besterr = -1; + u32 dai64 = 0; + + if ((params_rate(params) < 8000) || (params_rate(params) > 48000)) + return -EINVAL; + + if (!s->pll_hz && !s->ext_hz) + return -EINVAL; + + /* Find best settings for desired samplerate */ + for (i = 0x01; (i < 0x80) && besterr && s->pll_hz; i++) + if (!clps711x_dai_update_best_err(s->pll_hz, 128 * i, + params_rate(params), + &besterr)) + dai64 = DAI64FS_AUDIV(i); + for (i = 0x01; (i < 0x80) && besterr && s->ext_hz; i++) + if (!clps711x_dai_update_best_err(s->ext_hz, 128 * i, + params_rate(params), + &besterr)) + dai64 = DAI64FS_AUDIV(i) | DAI64FS_AUDIOCLKSRC; + + writel(DAI64FS_I2SF64 | DAI64FS_AUDIOCLKEN | DAI64FS_MCLK256EN | dai64, + s->base + DAI64FS); + + return 0; +} + +static const struct snd_soc_dai_ops clps711x_dai_ops = { + .hw_params = clps711x_dai_hw_params, + .set_fmt = clps711x_dai_set_fmt, + .set_sysclk = clps711x_dai_set_sysclk, +}; + +static int clps711x_dai_probe(struct snd_soc_dai *dai) +{ + struct clps711x_dai *s = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, s); + + return 0; +} + +static struct snd_soc_dai_driver clps711x_dai_driver = { + .ops = &clps711x_dai_ops, + .probe = clps711x_dai_probe, + .playback = { + .stream_name = "Playback", + .formats = CLPS711X_FMTS, + .rates = CLPS711X_RATES, + .channels_min = 2, + .channels_max = 2, + }, +}; + +static const struct snd_soc_component_driver clps711x_i2s_component = { + .name = "clps711x-i2s", +}; + +static int clps711x_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct clps711x_dai *dai = snd_soc_platform_get_drvdata(rtd->platform); + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + dai->substream = substream; + dai->reload = (1000000000 / params_rate(params)) * + params_period_size(params) / 2; + + local_fiq_disable(); + get_fiq_regs(&dai->regs); + dai->regs.ARM_r8 = (u32)substream->runtime->dma_area; + set_fiq_regs(&dai->regs); + local_fiq_enable(); + + return 0; +} + +static int clps711x_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct clps711x_dai *dai = snd_soc_platform_get_drvdata(rtd->platform); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + atomic_set(&dai->running, 1); + hrtimer_start(&dai->hrt, ns_to_ktime(dai->reload), + HRTIMER_MODE_REL); + break; + case SNDRV_PCM_TRIGGER_STOP: + atomic_set(&dai->running, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t clps711x_pcm_ptr(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct clps711x_dai *dai = snd_soc_platform_get_drvdata(rtd->platform); + + return bytes_to_frames(substream->runtime, dai->last_ptr); +} + +static int clps711x_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void __user *buf, + snd_pcm_uframes_t count) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct clps711x_dai *dai = snd_soc_platform_get_drvdata(rtd->platform); + int sz = frames_to_bytes(substream->runtime, count); + + if (copy_from_user(substream->runtime->dma_area + dai->head, buf, sz)) + return -EFAULT; + + local_fiq_disable(); + dai->head += sz; + dai->head %= CLPS711X_SNDBUF_SIZE; + local_fiq_enable(); + + return 0; +} + +static int clps711x_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) +{ + /* Silence is provided in the FIQ interrupt routine */ + /* This empty function is necessary */ + return 0; +} + +static enum hrtimer_restart clps711x_pcm_timer(struct hrtimer *hrt) +{ + struct clps711x_dai *dai = container_of(hrt, struct clps711x_dai, hrt); + u32 delta; + + if (!atomic_read(&dai->running)) { + dai->head = 0; + dai->last_ptr = 0; + + local_fiq_disable(); + get_fiq_regs(&dai->regs); + dai->regs.ARM_r10 = 0; + set_fiq_regs(&dai->regs); + local_fiq_enable(); + + return HRTIMER_NORESTART; + } + + get_fiq_regs(&dai->regs); + + delta = CLPS711X_SNDBUF_SIZE + dai->regs.ARM_r10 - dai->last_ptr; + delta %= CLPS711X_SNDBUF_SIZE; + if ((delta >= dai->substream->runtime->period_size) || + (dai->regs.ARM_r10 == dai->head)) { + dai->last_ptr = dai->regs.ARM_r10; + snd_pcm_period_elapsed(dai->substream); + } + + hrtimer_forward_now(hrt, ns_to_ktime(dai->reload)); + + return HRTIMER_RESTART; +} + +static const struct snd_pcm_hardware clps711x_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = CLPS711X_FMTS, + .buffer_bytes_max = CLPS711X_SNDBUF_SIZE, + .period_bytes_min = 32, + .period_bytes_max = CLPS711X_SNDBUF_SIZE / 8, + .periods_min = 4, + .periods_max = CLPS711X_SNDBUF_SIZE / 64 - 1, +}; + +static int clps711x_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct clps711x_dai *dai = snd_soc_platform_get_drvdata(rtd->platform); + int ret; + + atomic_set(&dai->running, 0); + hrtimer_init(&dai->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + dai->hrt.function = clps711x_pcm_timer; + + dai->head = 0; + dai->last_ptr = 0; + + get_fiq_regs(&dai->regs); + dai->regs.ARM_r9 = (u32)&dai->head; + dai->regs.ARM_r10 = 0; + dai->regs.ARM_fp = (u32)dai->base; + set_fiq_regs(&dai->regs); + + enable_irq(dai->irq); + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + return snd_soc_set_runtime_hwparams(substream, &clps711x_pcm_hardware); +} + +static int clps711x_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct clps711x_dai *dai = snd_soc_platform_get_drvdata(rtd->platform); + + disable_irq(dai->irq); + hrtimer_cancel(&dai->hrt); + + return 0; +} + +static const struct snd_pcm_ops clps711x_pcm_ops = { + .open = clps711x_pcm_open, + .close = clps711x_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = clps711x_pcm_hw_params, + .hw_free = snd_pcm_lib_free_pages, + .trigger = clps711x_pcm_trigger, + .pointer = clps711x_pcm_ptr, + .copy = clps711x_pcm_copy, + .silence = clps711x_pcm_silence, +}; + +static int clps711x_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct clps711x_dai *dai = snd_soc_platform_get_drvdata(rtd->platform); + const char *devname = dev_name(rtd->platform->dev); + int ret; + + /* Request FIQ */ + dai->fiq.name = devname; + ret = claim_fiq(&dai->fiq); + if (ret) + return ret; + + /* Request FIQ interrupt */ + ret = request_irq(dai->irq, no_action, 0, devname, NULL); + if (ret) + return ret; + + /* Install FIQ handler */ + set_fiq_handler(&daifiq_start, &daifiq_end - &daifiq_start); + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), + CLPS711X_SNDBUF_SIZE, CLPS711X_SNDBUF_SIZE); +} + +static void clps711x_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct clps711x_dai *dai = snd_soc_platform_get_drvdata(rtd->platform); + + free_irq(dai->irq, NULL); + + release_fiq(&dai->fiq); + + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static struct snd_soc_platform_driver clps711x_soc_platform_drv = { + .ops = &clps711x_pcm_ops, + .pcm_new = clps711x_pcm_new, + .pcm_free = clps711x_pcm_free, +}; + +static int clps711x_enable_fifo(void __iomem *base, u32 channel) +{ + unsigned long timeout; + + writel(DAIDR2_FIFOEN | channel, base + DAIDR2); + timeout = jiffies + msecs_to_jiffies(5); + while (!(readl(base + DAISR) & DAISR_FIFO)) + if (time_is_before_jiffies(timeout)) + return -ETIMEDOUT; + + return 0; +} + +static int clps711x_dai_platform_probe(struct platform_device *pdev) +{ + u32 dai64 = DAI64FS_AUDIOCLKEN | DAI64FS_MCLK256EN | DAI64FS_I2SF64; + struct device *dev = &pdev->dev; + struct clps711x_dai *dai; + struct regmap *syscon; + struct resource *res; + struct clk *clk; + int ret; + + /* Check for proper buffer size */ + BUG_ON(!is_power_of_2(CLPS711X_SNDBUF_SIZE)); + + dai = devm_kzalloc(dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + dai->irq = platform_get_irq(pdev, 0); + if (dai->irq < 0) + return dai->irq; + + syscon = syscon_regmap_lookup_by_compatible("cirrus,clps711x-syscon3"); + if (IS_ERR(syscon)) + return PTR_ERR(syscon); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + dai->base = devm_ioremap_resource(dev, res); + if (IS_ERR(dai->base)) + return PTR_ERR(dai->base); + + clk = devm_clk_get(dev, "pll"); + if (IS_ERR(clk)) + return PTR_ERR(clk); + + dai->pll_hz = clk_get_rate(clk) / 2; + if (!dai->pll_hz) { + dai64 |= DAI64FS_AUDIOCLKSRC; + dev_notice(dev, "External MCLK will be used only\n"); + } + + clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(clk)) { + if (PTR_ERR(clk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else + dai->ext_hz = clk_get_rate(clk); + + platform_set_drvdata(pdev, dai); + + ret = devm_snd_soc_register_component(dev, &clps711x_i2s_component, + &clps711x_dai_driver, 1); + if (ret) + return ret; + + /* Enable DAI interface */ + regmap_update_bits(syscon, SYSCON_OFFSET, + SYSCON3_DAISEL | SYSCON3_128FS, SYSCON3_DAISEL); + + /* Clear interrupt flags */ + writel(~0, dai->base + DAISR); + + /* Enable DAI */ + writel(DAIR_RESERVED | DAIR_DAIEN | DAIR_ECS | DAIR_LCTM | DAIR_RCTM, + dai->base + DAIR); + + /* Set initial value to DAI register */ + writel(dai64 | DAI64FS_AUDIV(10), dai->base + DAI64FS); + + /* Enable FIFOs */ + ret = clps711x_enable_fifo(dai->base, DAIDR2_FIFOLEFT); + if (ret) + goto out_err; + ret = clps711x_enable_fifo(dai->base, DAIDR2_FIFORIGHT); + if (ret) + goto out_err; + + ret = snd_soc_register_platform(dev, &clps711x_soc_platform_drv); + if (!ret) + return 0; + +out_err: + /* Disable DAI */ + writel(DAIR_RESERVED, dai->base + DAIR); + + return ret; +} + +static int clps711x_dai_platform_remove(struct platform_device *pdev) +{ + struct clps711x_dai *dai = platform_get_drvdata(pdev); + + /* Disable DAI */ + writel(DAIR_RESERVED, dai->base + DAIR); + + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static const struct of_device_id clps711x_dai_dt_ids[] = { + { .compatible = "cirrus,clps711x-dai", }, + { } +}; +MODULE_DEVICE_TABLE(of, clps711x_dai_dt_ids); + +static struct platform_driver clps711x_dai_platform_driver = { + .driver = { + .name = "clps711x-dai", + .owner = THIS_MODULE, + .of_match_table = clps711x_dai_dt_ids, + }, + .probe = clps711x_dai_platform_probe, + .remove = clps711x_dai_platform_remove, +}; +module_platform_driver(clps711x_dai_platform_driver); + +MODULE_AUTHOR("Alexander Shiyan <shc_work(a)mail.ru>"); +MODULE_DESCRIPTION("CLPS711X DAI driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/cirrus/clps711x-dai.h b/sound/soc/cirrus/clps711x-dai.h new file mode 100644 index 0000000..c79d26a --- /dev/null +++ b/sound/soc/cirrus/clps711x-dai.h @@ -0,0 +1,61 @@ +/* + * Currus Logic CLPS711X DAI definitions + * + * Copyright (C) 2014 Alexander Shiyan <shc_work(a)mail.ru> + * + * 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 __CLPS711X_DAI_H +#define __CLPS711X_DAI_H + +#include <linux/sizes.h> + +#define CLPS711X_SNDBUF_SIZE SZ_128K + +#define DAIR (0x0000) +# define DAIR_RESERVED (0x0404) +# define DAIR_DAIEN (1 << 16) +# define DAIR_ECS (1 << 17) +# define DAIR_LCTM (1 << 19) +# define DAIR_LCRM (1 << 20) +# define DAIR_RCTM (1 << 21) +# define DAIR_RCRM (1 << 22) +# define DAIR_LBM (1 << 23) +#define DAIDR0 (0x0040) +#define DAIDR1 (0x0080) +#define DAIDR2 (0x00c0) +# define DAIDR2_FIFOEN (1 << 15) +# define DAIDR2_FIFOLEFT (0x0d << 16) +# define DAIDR2_FIFORIGHT (0x11 << 16) +#define DAISR (0x0100) +# define DAISR_RCTS (1 << 0) +# define DAISR_RCRS (1 << 1) +# define DAISR_LCTS (1 << 2) +# define DAISR_LCRS (1 << 3) +# define DAISR_RCTU (1 << 4) +# define DAISR_RCRO (1 << 5) +# define DAISR_LCTU (1 << 6) +# define DAISR_LCRO (1 << 7) +# define DAISR_RCNF (1 << 8) +# define DAISR_RCNE (1 << 9) +# define DAISR_LCNF (1 << 10) +# define DAISR_LCNE (1 << 11) +# define DAISR_FIFO (1 << 12) +#define DAI64FS (0x0600) +# define DAI64FS_I2SF64 (1 << 0) +# define DAI64FS_AUDIOCLKEN (1 << 1) +# define DAI64FS_AUDIOCLKSRC (1 << 2) +# define DAI64FS_MCLK256EN (1 << 3) +# define DAI64FS_LOOPBACK (1 << 5) +# define DAI64FS_AUDIV_MASK (0x7f00) +# define DAI64FS_AUDIV(x) (((x) << 8) & DAI64FS_AUDIV_MASK) + +#ifndef __ASSEMBLY__ +extern unsigned char daifiq_start, daifiq_end; +#endif + +#endif diff --git a/sound/soc/cirrus/clps711x-fiq.S b/sound/soc/cirrus/clps711x-fiq.S new file mode 100644 index 0000000..aa2dfb9 --- /dev/null +++ b/sound/soc/cirrus/clps711x-fiq.S @@ -0,0 +1,56 @@ +/* + * Currus Logic CLPS711X DAI FIQ Handler + * + * Copyright (C) 2014 Alexander Shiyan <shc_work(a)mail.ru> + * + * 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 <asm/assembler.h> +#include <linux/linkage.h> + +#include "clps711x-dai.h" + + .text + .arm + + .global daifiq_end + +ENTRY(daifiq_start) + @ r8 - Buffer address + @ r9 - Head Pointer + @ r10 - Tail Pointer + @ r11 - Registers base virtual address + # r12 - Temp + +daifiq_loop: + ldr r12, [r9] @ Get Head + subs r12, r12, r10 @ Compare Head & Tail + beq daifiq_play @ Send zeroes if Head = Tail + + ldr r12, [r8, r10] @ Get buffer value + add r10, r10, #4 @ Increment pointer + bic r10, r10, #CLPS711X_SNDBUF_SIZE @ Mask with buffer size + +daifiq_play: + @ Put data to FIFOs + strh r12, [r11, #DAIDR1] @ Left channel + mov r12, r12, lsr #16 + strh r12, [r11, #DAIDR0] @ Right channel + + @ Check DAI Flags (FIFOs not full) + ldr r12, [r11, #DAISR] + and r12, r12, #DAISR_RCNF | DAISR_LCNF + cmp r12, #DAISR_RCNF | DAISR_LCNF + beq daifiq_loop + + @ Clear DAI Interrupt Flags + mvn r12, #0 + str r12, [r11, #DAISR] + + @ Return from FIQ + subs pc, lr, #4 +daifiq_end: -- 1.8.3.2

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[alsa-devel] [PATCH 1/2] ASoC: fsl: Allow to select SAI device individually
by Xiubo Li 01 May '14

01 May '14

This will be useful for out-of-tree drivers since in-tree drivers could select it automatically. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/Kconfig | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7a7eaf3..fb73e35 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -3,9 +3,14 @@ menu "SoC Audio for Freescale CPUs" comment "Common SoC Audio options for Freescale CPUs:" config SND_SOC_FSL_SAI - tristate + tristate "Synchronous Audio Interface (SAI) module support" select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y if you want to add Synchronous Audio Interface (SAI) + support for the Freescale CPUs. + This option is only useful for out-of-tree drivers since + in-tree drivers select it automatically. config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" -- 1.8.4

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[alsa-devel] [PATCH 08/15] ASoC: TTC DKB audio needs I2C
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> The missing dependency can lead to build errors, so make it explicit in Kconfig. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Philipp Zabel <philipp.zabel(a)gmail.com> Cc: Paul Parsons <lost.distance(a)yahoo.com> Cc: Russell King <linux(a)arm.linux.org.uk> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-kernel(a)lists.infradead.org --- sound/soc/pxa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 6473052..6acb225 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -140,7 +140,7 @@ config SND_PXA910_SOC config SND_SOC_TTC_DKB bool "SoC Audio support for TTC DKB" - depends on SND_PXA910_SOC && MACH_TTC_DKB + depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y select PXA_SSP select SND_PXA_SOC_SSP select SND_MMP_SOC -- 1.7.9.5

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[alsa-devel] [PATCH 07/15] ASoC: UDA1380 needs I2C
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> The UDA1380 driver needs I2C to be enabled, so SND_SOC_SAMSUNG_H1940_UDA1380 and SND_SOC_SAMSUNG_RX1950_UDA1380 also require this. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 0c5e9e2..b09b5a4 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -130,7 +130,7 @@ config SND_SOC_SAMSUNG_SIMTEC_HERMES config SND_SOC_SAMSUNG_H1940_UDA1380 tristate "Audio support for the HP iPAQ H1940" - depends on SND_SOC_SAMSUNG && ARCH_H1940 + depends on SND_SOC_SAMSUNG && ARCH_H1940 && I2C select SND_S3C24XX_I2S select SND_SOC_UDA1380 help @@ -138,7 +138,7 @@ config SND_SOC_SAMSUNG_H1940_UDA1380 config SND_SOC_SAMSUNG_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" - depends on SND_SOC_SAMSUNG && MACH_RX1950 + depends on SND_SOC_SAMSUNG && MACH_RX1950 && I2C select SND_S3C24XX_I2S select SND_SOC_UDA1380 help -- 1.7.9.5

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[alsa-devel] [PATCH 06/15] ASoC: RX-51 audio needs I2C
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> The codec requires I2C to be enabled, so any other option that selects it should also depend on I2C. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-omap(a)vger.kernel.org --- sound/soc/omap/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e006593..2796af9 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) + depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 -- 1.7.9.5

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[alsa-devel] [PATCH 01/15] ASoC: CS42L51 and WM8962 codecs depend on INPUT
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> Building ARM randconfig got into a situation where CONFIG_INPUT is turned off and SND_SOC_ALL_CODECS is turned on, which failed for two codecs trying to use the input subsystem. Some other drivers also select one of these codecs and consequently need an explicit dependency added. Appending to the dependency list seems the easiest way out, since this is not a practical limitation. If anyone really needs to build these codecs for a kernel with no input support, a more sophisticated solution can be implemented. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Timur Tabi <timur(a)tabi.org> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 8 ++++---- sound/soc/fsl/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..d4260d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C - select SND_SOC_CS42L52 if I2C + select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI @@ -127,7 +127,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8955 if I2C select SND_SOC_WM8960 if I2C select SND_SOC_WM8961 if I2C - select SND_SOC_WM8962 if I2C + select SND_SOC_WM8962 if I2C && INPUT select SND_SOC_WM8971 if I2C select SND_SOC_WM8974 if I2C select SND_SOC_WM8978 if I2C @@ -282,7 +282,7 @@ config SND_SOC_CS42L51 config SND_SOC_CS42L52 tristate "Cirrus Logic CS42L52 CODEC" - depends on I2C + depends on I2C && INPUT config SND_SOC_CS42L73 tristate "Cirrus Logic CS42L73 CODEC" @@ -598,7 +598,7 @@ config SND_SOC_WM8961 config SND_SOC_WM8962 tristate "Wolfson Microelectronics WM8962 CODEC" - depends on I2C + depends on I2C && INPUT config SND_SOC_WM8971 tristate diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 338a916..f4069d0 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -187,7 +187,7 @@ config SND_SOC_EUKREA_TLV320 config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" - depends on OF && I2C + depends on OF && I2C && INPUT select SND_SOC_WM8962 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f2e2891..14568be 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -204,7 +204,7 @@ config SND_SOC_SPEYSIDE config SND_SOC_TOBERMORY tristate "Audio support for Wolfson Tobermory" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT select SND_SAMSUNG_I2S select SND_SOC_WM8962 -- 1.7.9.5

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[alsa-devel] TASCAM US-1800
by amand 01 May '14

01 May '14

Hello! I've just bought a TASCAM US-1800 soundcard. Unfortunately, there is no driver ready for linux. I started to gather some data from the windows USB driver in order to add support in linux. I am ready to invest time in this driver, but this is my first ever attempt. Presently, I try to reverse-engineering the data I have collect with wireshark. Somebody has ever worked on a US-1800 driver for linux ? Perhaps we can get some stuff from the us122l driver... By the way, any help will be appreciated ! -- amand.

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[alsa-devel] [PATCH 2/2] ASoC: mc13783: Add devicetree support
by Alexander Shiyan 01 May '14

01 May '14

This patch adds devicetree support for mc13783-codec. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- Documentation/devicetree/bindings/mfd/mc13xxx.txt | 3 +++ sound/soc/codecs/mc13783.c | 14 +++++++++++++- 2 files changed, 16 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/mfd/mc13xxx.txt b/Documentation/devicetree/bindings/mfd/mc13xxx.txt index 1413f39..8aba488 100644 --- a/Documentation/devicetree/bindings/mfd/mc13xxx.txt +++ b/Documentation/devicetree/bindings/mfd/mc13xxx.txt @@ -10,6 +10,9 @@ Optional properties: - fsl,mc13xxx-uses-touch : Indicate the touchscreen controller is being used Sub-nodes: +- codec: Contain the Audio Codec node. + - adc-port: Contain PMIC SSI port number used for ADC. + - dac-port: Contain PMIC SSI port number used for DAC. - leds : Contain the led nodes and initial register values in property "led-control". Number of register depends of used IC, for MC13783 is 6, for MC13892 is 4, for MC34708 is 1. See datasheet for bits definitions of diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 0b0a4ca..9965277 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -22,6 +22,7 @@ */ #include <linux/module.h> #include <linux/device.h> +#include <linux/of.h> #include <linux/mfd/mc13xxx.h> #include <linux/slab.h> #include <sound/core.h> @@ -748,6 +749,7 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) { struct mc13783_priv *priv; struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data; + struct device_node *np; int ret; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -758,7 +760,17 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) priv->adc_ssi_port = pdata->adc_ssi_port; priv->dac_ssi_port = pdata->dac_ssi_port; } else { - return -ENOSYS; + np = of_get_child_by_name(pdev->dev.parent->of_node, "codec"); + if (!np) + return -ENOSYS; + + ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); + if (ret) + return ret; + + ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); + if (ret) + return ret; } dev_set_drvdata(&pdev->dev, priv); -- 1.8.3.2

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[alsa-devel] [PATCH 1/2] ASoC: wm8804: Use devm_regulator_bulk_get()
by Fabio Estevam 01 May '14

01 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using devm_regulator_bulk_get() can make the code cleaner and smaller as we do not need to call regulator_bulk_free() in the error and remove paths. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8804.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 589455c..bbcad9f 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -535,7 +535,6 @@ static int wm8804_remove(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) regulator_unregister_notifier(wm8804->supplies[i].consumer, &wm8804->disable_nb[i]); - regulator_bulk_free(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); return 0; } @@ -549,7 +548,7 @@ static int wm8804_probe(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8804->supplies), + ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8804->supplies), wm8804->supplies); if (ret) { dev_err(codec->dev, "Failed to request supplies: %d\n", ret); @@ -574,7 +573,7 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804->supplies); if (ret) { dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_reg_get; + return ret; } id1 = snd_soc_read(codec, WM8804_RST_DEVID1); @@ -619,8 +618,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) err_reg_enable: regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); -err_reg_get: - regulator_bulk_free(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); return ret; } -- 1.8.3.2

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[alsa-devel] [PATCH 05/15] ASoC: nuc900: export nuc900_ac97_data
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> The symbol "nuc900_ac97_data" is used by the nuc900_pcm driver, which may be a loadable module, so we should export it. If one tries to build SND_SOC_NUC900 without SND_SOC_NUC900_AC97, the kernel fails to link because of the reference to nuc900_ac97_data. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Wan ZongShun <mcuos.com(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-kernel(a)lists.infradead.org --- sound/soc/nuc900/Kconfig | 1 + sound/soc/nuc900/nuc900-ac97.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/nuc900/Kconfig b/sound/soc/nuc900/Kconfig index a0ed1c6..7f0c954 100644 --- a/sound/soc/nuc900/Kconfig +++ b/sound/soc/nuc900/Kconfig @@ -4,6 +4,7 @@ config SND_SOC_NUC900 tristate "SoC Audio for NUC900 series" depends on ARCH_W90X900 + select SND_SOC_NUC900_AC97 help This option enables support for AC97 mode on the NUC900 SoC. diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 8987bf9..f2f6794 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -28,6 +28,7 @@ static DEFINE_MUTEX(ac97_mutex); struct nuc900_audio *nuc900_ac97_data; +EXPORT_SYMBOL_GPL(nuc900_ac97_data); static int nuc900_checkready(void) { -- 1.7.9.5

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[alsa-devel] [PATCH 04/15] ASoC: samsung-idma: avoid 64-bit division
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> dma_addr_t may be 64 bit wide, which causes a build failure when doing a division on it. Here it is safe to cast to an u32 type, which avoids the problem. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- sound/soc/samsung/idma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 3d5cf15..e9891b4 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -274,7 +274,7 @@ static irqreturn_t iis_irq(int irqno, void *dev_id) addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; addr += prtd->periodsz; - addr %= (prtd->end - prtd->start); + addr %= (u32)(prtd->end - prtd->start); addr += idma.lp_tx_addr; writel(addr, idma.regs + I2SLVL0ADDR); -- 1.7.9.5

3 2

[alsa-devel] [PATCH 03/15] ASoC: SMDK_WM8580_PCM needs REGMAP_I2C
by Xia Kaixu 01 May '14

01 May '14

From: Arnd Bergmann <arnd(a)arndb.de> This adds a missing dependency for SND_SOC_SMDK_WM8580_PCM to require REGMAP_I2C to be enabled, avoiding possible build erorrs. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- sound/soc/samsung/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 14568be..0c5e9e2 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -64,6 +64,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDK6440 || MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110) + depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S help @@ -178,6 +179,7 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF config SND_SOC_SMDK_WM8580_PCM tristate "SoC PCM Audio support for WM8580 on SMDK" depends on SND_SOC_SAMSUNG && (MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110) + depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_PCM help -- 1.7.9.5

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