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[alsa-devel] [PATCH v2 1/1] sst: select DW_DMAC_CORE since it's mandatory
by Andy Shevchenko 22 Oct '16

22 Oct '16

Select DW_DMAC_CORE like the rest of glue drivers do, e.g. drivers/dma/dw/Kconfig. While here group selectors under SND_SOC_INTEL_HASWELL and SND_SOC_INTEL_BAYTRAIL. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/Kconfig | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index fd5d1e0..44d1cd4 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -47,16 +47,19 @@ config SND_SOC_INTEL_SST_MATCH config SND_SOC_INTEL_HASWELL tristate + select DW_DMAC_CORE + select SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_FIRMWARE config SND_SOC_INTEL_BAYTRAIL tristate + select DW_DMAC_CORE + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -99,9 +102,7 @@ config SND_SOC_INTEL_BXT_RT298_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n) - select SND_SOC_INTEL_SST - select SND_SOC_INTEL_SST_FIRMWARE + depends on SND_SST_IPC_ACPI = n select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help @@ -112,9 +113,7 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n) - select SND_SOC_INTEL_SST - select SND_SOC_INTEL_SST_FIRMWARE + depends on SND_SST_IPC_ACPI = n select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help @@ -134,10 +133,7 @@ config SND_SOC_INTEL_BDW_RT5677_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on X86_INTEL_LPSS && I2C && DW_DMAC && \ - I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST + depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help -- 2.9.3

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Re: [alsa-devel] M-audio Audiophile Firewire
by Takashi Sakamoto 22 Oct '16

22 Oct '16

Hi, (CCed to alsa-devel, ffado-devel) On Oct 22 2016 07:37, Peti Pap wrote: > Do you have any advice about this soundcard? I own the unit and work with it on Linux system. > .... I tried use on Linux Mint (Sarah) 64 bit, and I installed ffado packs, > and jack too... but still only the blue led lighting, and no any sound. (On > the Jack "panel" I tried all of the possibble settings) Please show your environment; at least, kernel version and libffado2. Regards Takashi Sakamoto

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[alsa-devel] [PATCH 0/7] ALSA: Add rewinds disabled, delays, max_inflight_bytes
by Subhransu S. Prusty 21 Oct '16

21 Oct '16

Set of patches to fix issues with delays, hw_ptr fuzziness [1] and increased buffering w/ DSPs 1. Rewinds can be disabled when data written in ring buffer will never be validated. This allow for new HDaudio SPIB DMA functionality(allow fetch up to the application pointer, no rewind supported) 2. Report max in-flight bytes to avoid problems with stale data (like late wake-ups, rewinds) 3. add new estimate for USB startup delay. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093646.html Pierre-Louis Bossart (6): ALSA: core: let low-level driver or userspace disable rewinds ALSA: core: add .update_appl_ptr callback for pcm ops ALSA: core: add report of max inflight bytes ALSA: hda: add default value for max_inflight_bytes ALSA: usb: no_period_wake and max_inflight_bytes report ALSA: usb: take startup delay into account Ramesh Babu (1): ALSA: pcm: avoid mmap of control data if .update_appl_ptr is implemented include/sound/pcm.h | 4 +++ include/uapi/sound/asound.h | 6 ++-- sound/core/pcm_lib.c | 26 ++++++++++++++++++ sound/core/pcm_native.c | 51 +++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.c | 1 + sound/usb/card.h | 1 + sound/usb/pcm.c | 62 +++++++++++++++++++++++++++++++++++++----- 7 files changed, 141 insertions(+), 10 deletions(-) -- 1.9.1

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[alsa-devel] how to disable dynamic control of mic bias?
by Tim Harvey 21 Oct '16

21 Oct '16

I have a board design with a TLV320AIC3x codec + TS3A227E headset detect chip and am using the tlv320aic3x and ts3a227e drivers in mainline Linux. The TS3A227E serves a couple of purposes including support for multiple headset pinout standards and sensing buttons on the mic (ie see Android wired headset spec: https://source.android.com/devices/accessories/headset/specification.html). In order for the button detect to work mic bias from the codec needs to be enabled (at init time and regardless of playing or recording audio). What is the correct way to do this in either device-tree if using something like simple-audio-card or in a machine driver? I have not had success by setting the codec drivers 'idle_bias_off = false' - while this does keep mic bias on at init and after audio playback it kills it after record due to the 'Mic Bias' dapm widget in the case of the tlv320aic3x.c driver. I'm thinking I need to instead remove the routing or disable the 'Mic Bias' but I'm not clear how to do this? Regards, Tim

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[alsa-devel] [PATCH] ASoC: samsung: i2s: Fixup last IRQ unsafe spin lock call
by Charles Keepax 21 Oct '16

21 Oct '16

Unfortunately, I seem to have missed a case where an IRQ safe spinlock was required, in samsung_i2s_dai_remove, when I fixed up the other calls in this patch: 316fa9e09ad7 ("ASoC: samsung: Use IRQ safe spin lock calls") This causes a lockdep warning when unbinding and rebinding the audio card: [ 104.357664] CPU0 CPU1 [ 104.362174] ---- ---- [ 104.366692] lock(&(&pri_dai->spinlock)->rlock); [ 104.371372] local_irq_disable(); [ 104.377283] lock(&(&substream->self_group.lock)->rlock); [ 104.385259] lock(&(&pri_dai->spinlock)->rlock); [ 104.392469] <Interrupt> [ 104.395072] lock(&(&substream->self_group.lock)->rlock); [ 104.400710] [ 104.400710] *** DEADLOCK *** Fixes: ce8bcdbb61d9 ("ASoC: samsung: i2s: Protect more registers with a spinlock") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 7e32cf4..588aa01 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1029,12 +1029,13 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) { struct i2s_dai *i2s = snd_soc_dai_get_drvdata(dai); + unsigned long flags; if (!is_secondary(i2s)) { if (i2s->quirks & QUIRK_NEED_RSTCLR) { - spin_lock(i2s->lock); + spin_lock_irqsave(i2s->lock, flags); writel(0, i2s->addr + I2SCON); - spin_unlock(i2s->lock); + spin_unlock_irqrestore(i2s->lock, flags); } } -- 2.1.4

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[alsa-devel] [PATCH RESEND 0/2] Arizona Clocking Improvements
by Charles Keepax 21 Oct '16

21 Oct '16

Add support for requesting the input clocks for the CODEC through the clock framework. This is some what of an intermediate step until all the clocking itself in the CODEC can be brought under the clock framework but as there are machine drivers that want to be merged now with source clocks that require gating and difficult to resolve issues between the SPI and clock framework an intermediate solution seems warranted. The first patch in the original sending of this series is now in mainline through the MFD tree so these just need to go through ASoC now. Thanks, Charles Charles Keepax (2): ASoC: arizona: Add gating for clock when used for direct MCLK ASoC: arizona: Add gating for source clocks of the FLLs sound/soc/codecs/arizona.c | 100 ++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/cs47l24.c | 6 ++- sound/soc/codecs/wm5102.c | 9 +++- sound/soc/codecs/wm5110.c | 10 +++-- sound/soc/codecs/wm8997.c | 9 +++- sound/soc/codecs/wm8998.c | 6 ++- 7 files changed, 125 insertions(+), 17 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 1/2] ASoC: wm9705: Convert to regmap
by Lars-Peter Clausen 21 Oct '16

21 Oct '16

Currently the wm9712 driver still uses custom snd_soc_codec_driver IO callbacks. This has been deprecated for a while, so convert the wm9705 driver to use regmap for its IO. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Untested, but same changes as for all the other AC97 drivers, so hopefully it will just work. --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9705.c | 123 ++++++++++++++++++++++++---------------------- 2 files changed, 64 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 091c0f0..a92ea7c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1059,6 +1059,7 @@ config SND_SOC_WM9090 config SND_SOC_WM9705 tristate + select REGMAP_AC97 config SND_SOC_WM9712 tristate diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index dcdd055..70ab607 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/kernel.h> #include <linux/device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -25,26 +26,48 @@ #define WM9705_VENDOR_ID 0x574d4c05 #define WM9705_VENDOR_ID_MASK 0xffffffff -/* - * WM9705 register cache - */ -static const u16 wm9705_reg[] = { - 0x6150, 0x8000, 0x8000, 0x8000, /* 0x0 */ - 0x0000, 0x8000, 0x8008, 0x8008, /* 0x8 */ - 0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */ - 0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */ - 0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */ - 0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */ - 0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */ - 0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */ - 0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */ - 0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */ +static const struct reg_default wm9705_reg_defaults[] = { + { 0x02, 0x8000 }, + { 0x04, 0x8000 }, + { 0x06, 0x8000 }, + { 0x0a, 0x8000 }, + { 0x0c, 0x8008 }, + { 0x0e, 0x8008 }, + { 0x10, 0x8808 }, + { 0x12, 0x8808 }, + { 0x14, 0x8808 }, + { 0x16, 0x8808 }, + { 0x18, 0x8808 }, + { 0x1a, 0x0000 }, + { 0x1c, 0x8000 }, + { 0x20, 0x0000 }, + { 0x22, 0x0000 }, + { 0x26, 0x000f }, + { 0x28, 0x0605 }, + { 0x2a, 0x0000 }, + { 0x2c, 0xbb80 }, + { 0x32, 0xbb80 }, + { 0x34, 0x2000 }, + { 0x5a, 0x0000 }, + { 0x5c, 0x0000 }, + { 0x72, 0x0808 }, + { 0x74, 0x0000 }, + { 0x76, 0x0006 }, + { 0x78, 0x0000 }, + { 0x7a, 0x0000 }, +}; + +static const struct regmap_config wm9705_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x7e, + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = regmap_ac97_default_volatile, + + .reg_defaults = wm9705_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm9705_reg_defaults), }; static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = { @@ -203,39 +226,15 @@ static const struct snd_soc_dapm_route wm9705_audio_map[] = { {"Right ADC", NULL, "ADC PGA"}, }; -/* We use a register cache to enhance read performance. */ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - switch (reg) { - case AC97_RESET: - case AC97_VENDOR_ID1: - case AC97_VENDOR_ID2: - return soc_ac97_ops->read(ac97, reg); - default: - reg = reg >> 1; - - if (reg >= (ARRAY_SIZE(wm9705_reg))) - return -EIO; - - return cache[reg]; - } + return snd_soc_read(codec, reg); } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - soc_ac97_ops->write(ac97, reg, val); - reg = reg >> 1; - if (reg < (ARRAY_SIZE(wm9705_reg))) - cache[reg] = val; - - return 0; + return snd_soc_write(codec, reg, val); } static int ac97_prepare(struct snd_pcm_substream *substream, @@ -299,9 +298,9 @@ static struct snd_soc_dai_driver wm9705_dai[] = { #ifdef CONFIG_PM static int wm9705_soc_suspend(struct snd_soc_codec *codec) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - - soc_ac97_ops->write(ac97, AC97_POWERDOWN, 0xffff); + regcache_cache_bypass(codec->component.regmap, true); + ac97_write(codec, AC97_POWERDOWN, 0xffff); + regcache_cache_bypass(codec->component.regmap, false); return 0; } @@ -309,17 +308,14 @@ static int wm9705_soc_suspend(struct snd_soc_codec *codec) static int wm9705_soc_resume(struct snd_soc_codec *codec) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - int i, ret; - u16 *cache = codec->reg_cache; + int ret; ret = snd_ac97_reset(ac97, true, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); if (ret < 0) return ret; - for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { - soc_ac97_ops->write(ac97, i, cache[i>>1]); - } + regcache_sync(codec->component.regmap); return 0; } @@ -331,6 +327,8 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) static int wm9705_soc_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; + struct regmap *regmap; + int ret; ac97 = snd_soc_new_ac97_codec(codec, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); @@ -339,15 +337,26 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) return PTR_ERR(ac97); } + regmap = regmap_init_ac97(ac97, &wm9705_regmap_config); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + goto err_free_ac97_codec; + } + snd_soc_codec_set_drvdata(codec, ac97); + snd_soc_codec_init_regmap(codec, regmap); return 0; +err_free_ac97_codec: + snd_soc_free_ac97_codec(ac97); + return ret; } static int wm9705_soc_remove(struct snd_soc_codec *codec) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(ac97); return 0; } @@ -357,12 +366,6 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm9705 = { .remove = wm9705_soc_remove, .suspend = wm9705_soc_suspend, .resume = wm9705_soc_resume, - .read = ac97_read, - .write = ac97_write, - .reg_cache_size = ARRAY_SIZE(wm9705_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = wm9705_reg, .component_driver = { .controls = wm9705_snd_ac97_controls, -- 2.1.4

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[alsa-devel] [PATCH] ALSA: hda - Fix headset mic detection problem for two Dell laptops
by Hui Wang 21 Oct '16

21 Oct '16

They uses the codec ALC255, and have the different pin cfg definition from the ones in the existing pin quirk table. Now adding them into the table to fix the problem. Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- 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 8483fc2..b582d57 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5857,11 +5857,19 @@ enum { {0x1b, 0x02011020}, {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x14, 0x90170110}, + {0x1b, 0x01011020}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x14, 0x90170130}, {0x1b, 0x01014020}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x14, 0x90170130}, + {0x1b, 0x01011020}, + {0x21, 0x0221103f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x14, 0x90170130}, {0x1b, 0x02011020}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, -- 1.9.1

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[alsa-devel] Para-virtualized audio drivers
by Ughreja, Rakesh A 20 Oct '16

20 Oct '16

Hi Takashi and all, I have some need to develop para virtualized audio drivers which can work with XEN hypervisor. I see some attempt was made in the past. https://lists.xen.org/archives/html/xen-devel/2015-01/msg02125.html I see one more material but with no code. http://rd.springer.com/chapter/10.1007%2F978-94-007-2105-0_46 I am thinking to develop pv-audio-frontend (running on Dom1) and pv-audio-backend (running on Dom0) drivers all at kernel level. Backend driver owns the hardware. Frontend driver communicates with backend driver using XENbus framework just like storage and network PV drivers. Before I start, I wanted to check if there is any other work that has been done which I can take as base ? Regards, Rakesh

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Re: [alsa-devel] [PATCH] ASoC: Add jd function for rt5663.
by Mark Brown 20 Oct '16

20 Oct '16

On Thu, Oct 20, 2016 at 11:20:00AM +0000, Jack Yu wrote: > > Add initial setting for rt5663 jd to irq. > The patch will conflict with [PATCH] ASoC: rt5663: rename rt5668 as rt5663 v2, > please ignore this patch, we'll send another patch. It's already been applied - please send an incremental patch instead.

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[alsa-devel] [patch] ASoC: rt5663: fix a debug statement
by Dan Carpenter 20 Oct '16

20 Oct '16

We increment "i" before printing the debug statement. That makes it the wrong sleep_time[] information and Smatch complains that the last increment could be beyond the end of the array. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 01a18d8..00ff278 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1547,11 +1547,11 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) msleep(sleep_time[i]); val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) & 0x0003; + dev_dbg(codec->dev, "%s: MX-00e7 val=%x sleep %d\n", + __func__, val, sleep_time[i]); i++; if (val == 0x1 || val == 0x2 || val == 0x3) break; - dev_dbg(codec->dev, "%s: MX-00e7 val=%x sleep %d\n", - __func__, val, sleep_time[i]); } dev_dbg(codec->dev, "%s val = %d\n", __func__, val); switch (val) {

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[alsa-devel] [PATCH 0/2] topology: Support widgets' stream name and file inclusion for text conf file
by mengdong.lin＠linux.intel.com 20 Oct '16

20 Oct '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series doesn't require or cause any ABI updates. It doesn't need the kernel patches that trying to udpate ABI from v4 or v5. It just fix a missing feature and enhance text conf file to support including files to control the increasing size of text conf files. Mengdong Lin (2): topology: Fix missing stream name of widgets in text conf file topology: Add support for included files include/topology.h | 38 +++++ src/topology/dapm.c | 10 ++ src/topology/parser.c | 374 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/tplg_local.h | 17 +++ 4 files changed, 434 insertions(+), 5 deletions(-) -- 2.5.0

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Re: [alsa-devel] [PATCH] pcm_plug: Clear plugins on all error conditions
by Takashi Iwai 20 Oct '16

20 Oct '16

On Wed, 19 Oct 2016 08:10:28 +0200, Michael Forney wrote: > > Otherwise, they will linger after the error is returned (but pcm->setup == 0). > Then, if the caller tries to clean up and call snd_pcm_close(), the assertion > plug->gen.slave == plug->req_slave will fail. The changes look OK, but could you give your sign-off? thanks, Takashi > --- > src/pcm/pcm_plug.c | 4 +++- > 1 file changed, 3 insertions(+), 1 deletion(-) > > diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c > index 5639b9e..266707b 100644 > --- a/src/pcm/pcm_plug.c > +++ b/src/pcm/pcm_plug.c > @@ -652,8 +652,10 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, > (plug->ttable && !plug->ttable_ok)) { > snd_pcm_t *new; > int err; > - if (k >= sizeof(funcs)/sizeof(*funcs)) > + if (k >= sizeof(funcs)/sizeof(*funcs)) { > + snd_pcm_plug_clear(pcm); > return -EINVAL; > + } > err = funcs[k](pcm, &new, client, &p); > if (err < 0) { > snd_pcm_plug_clear(pcm); > -- > 2.10.1 >

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[alsa-devel] [PATCH] ASoC: rt5660: enable MCLK detection
by Bard Liao 20 Oct '16

20 Oct '16

There is a power saving mechanism in rt5660. It will turn off some unused power when MCLK is not present. We call that "MCLK detection" and it should be enabled by default. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5660.c | 4 ++++ sound/soc/codecs/rt5660.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 9f0933c..76cf76a 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1311,6 +1311,10 @@ static int rt5660_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + regmap_update_bits(rt5660->regmap, RT5660_GEN_CTRL1, + RT5660_AUTO_DIS_AMP | RT5660_MCLK_DET | RT5660_POW_CLKDET, + RT5660_AUTO_DIS_AMP | RT5660_MCLK_DET | RT5660_POW_CLKDET); + if (rt5660->pdata.dmic1_data_pin) { regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, RT5660_GP1_PIN_MASK, RT5660_GP1_PIN_DMIC1_SCL); diff --git a/sound/soc/codecs/rt5660.h b/sound/soc/codecs/rt5660.h index 6cdb926..bba18fb6 100644 --- a/sound/soc/codecs/rt5660.h +++ b/sound/soc/codecs/rt5660.h @@ -810,6 +810,9 @@ /* General Control 1 (0xfa) */ #define RT5660_PWR_VREF_HP (0x1 << 11) #define RT5660_PWR_VREF_HP_SFT 11 +#define RT5660_AUTO_DIS_AMP (0x1 << 6) +#define RT5660_MCLK_DET (0x1 << 5) +#define RT5660_POW_CLKDET (0x1 << 1) #define RT5660_DIG_GATE_CTRL (0x1) #define RT5660_DIG_GATE_CTRL_SFT 0 -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH v2 3/3] dt/bindings: Add milivolt, milliamps, and nanohenry units
by Paul.Handrigan＠cirrus.com 19 Oct '16

19 Oct '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Add millivolt, milliamp, and nanohenry units to the Electricity category. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- Documentation/devicetree/bindings/property-units.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/property-units.txt b/Documentation/devicetree/bindings/property-units.txt index 12278d7..92d6793 100644 --- a/Documentation/devicetree/bindings/property-units.txt +++ b/Documentation/devicetree/bindings/property-units.txt @@ -25,9 +25,12 @@ Distance Electricity ---------------------------------------- -microamp : micro amps +-milliamp : milli amps -ohms : Ohms -micro-ohms : micro Ohms -microvolt : micro volts +-millivolt : micro volts +-nanohenry : nano henries Temperature ---------------------------------------- -- 1.9.1

6 5

[alsa-devel] [ANNOUNCE] Sound Open Firmware
by Liam Girdwood 19 Oct '16

19 Oct '16

Sound Open Firmware is an open source, platform/architecture independent audio firmware and SDK intended for the Linux and open source communities. The intention of SOF is to provide an open audio firmware infrastructure and SDK with the following features :- o Optimised for the Linux kernel and userspace. o Provide features not found in proprietary firmwares. o Allow easier integration with audio hardware. o Foster innovation in userspace to support audio processing features. o Fully open source firmware and SDK. SOF currently supports the xtensa audio DSP architecture found in many modern devices. Platform support currently consists of the Intel Baytrail and Cherrytrail platforms (e.g. devices like the Minnowboard Max, UP board, Asus T100, etc). The firmware is modular and abstracts the architecture/platform layers like the Linux kernel, so can be ported to other DSP architectures or target platforms. The SDK provides a compiler, emulator and image builder tools. No other tools are needed to build and run the firmware. More information can be found here on the developer wiki :- http://alsa-project.org/main/index.php/Firmware Mailing list (subscribers only) is here :- http://alsa-project.org/mailman/listinfo/sound-open-firmware Thanks Liam

1 0

[alsa-devel] [SAMPLE-PATCH 0/x] ASoC: replace snd_soc_read/write
by Kuninori Morimoto 19 Oct '16

19 Oct '16

Hi Mark, Lars-Peter These are part of "replace snd_soc_read/write to snd_soc_component_read/write" patch-set. Full-patch-set will be over 100 patches. [0/x] - [x-1/x] : ASoC: use snd_soc_component_read/write on xxxx [x/x] : ASoC: remove snd_soc_component_read/write [0/x] - [x-1/x] are almost same patches. so, I pickuped few of them for reviewing to avoid patch flood on ML. If these review were OK, I will post full-patch-set, or send git-pull-request. Main purpose of these patches are replace current snd_soc_read/write to snd_soc_component_read/write to remove codec related function. Best regards --- Kuninori Morimoto

3 8

[alsa-devel] [PATCH 0/6] ASoC: rsnd: cleanup for IOMMU
by Kuninori Morimoto 19 Oct '16

19 Oct '16

Hi Mark I will post IOMMU support patches for rsnd driver. These are prepare/cleanup for it. Kuninori Morimoto (6): ASoC: rsnd: remove duplicate define of rsnd_dvc_of_node() ASoC: rsnd: amend .probe/.remove call for DPCM ASoC: rsnd: add rsnd_mod_next() for for_each_rsnd_mod_xxx() ASoC: rsnd: use for_each_rsnd_mod_xxx() on rsnd_dai_call() ASoC: rsnd: use for_each_rsnd_mod_xxx() on rsnd_rdai_continuance_probe() ASoC: rsnd: add rsnd_parse_of_node() and integrate rsnd_xxx_of_node sound/soc/sh/rcar/core.c | 115 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------------------------------ sound/soc/sh/rcar/dma.c | 11 +++++++++++ sound/soc/sh/rcar/dvc.c | 2 -- sound/soc/sh/rcar/rsnd.h | 56 +++++++++++++++++++++++++++++++++++++------------------- sound/soc/sh/rcar/ssi.c | 5 ++++- 5 files changed, 125 insertions(+), 64 deletions(-) Best regards --- Kuninori Morimoto

1 6

[alsa-devel] configuring machine specific DAPM for codec + headset jack detect
by Tim Harvey 19 Oct '16

19 Oct '16

Greetings, I have a TI TLV320AIC3105 codec on a board connected to a TS3A227E headset detect chip (which handles switching pins for OMTP vs std jacks as well as jack detect and headset buttons). The AIC3105 doesn't use several of the inputs and outputs yet those inputs and outputs still have mux widgets volume widgets that are not 'off'. Is there a way to easily disable them so they a) are not muxed into mixers, b) have their gains set to 0, and c) are pruned from the list of widgets so users don't have to muddle through them in mixers? I am also fighting both a pop/click issue and am wondering if there is any easy way to disable the DAPM power up/down sequence upon playback start/complete (my application is not power sensitive). I'm still trying to understand the source of the pop/click. Additionally the TS3A227E requires MIC bias to be kept enabled in order to detect headset button presses. I see that setting idle_bias_off to false will keep the MIC bias enabled but I don't see a way to specify that via device-tree. Regards, Tim

2 4

[alsa-devel] [PATCH] ASoC: PXA: Brownstone needs I2C
by Arnd Bergmann 18 Oct '16

18 Oct '16

I rand into a new build error with SND_MMP_SOC_BROWNSTONE: warning: (SND_MMP_SOC_BROWNSTONE && SND_SOC_SAMSUNG_SMDK_WM8994 && SND_SOC_SMDK_WM8994_PCM && SND_SOC_LITTLEMILL) selects MFD_WM8994 which has unmet direct dependencies (HAS_IOMEM && I2C) drivers/mfd/wm8994-core.c:688:1: error: data definition has no type or storage class [-Werror] drivers/mfd/wm8994-core.c:688:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] I don't see why this never showed up before, as the dependency seems to have been missing since the symbol was first introduced several years ago. This adds a dependency like the other drivers have. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- 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 f2bf8661dd21..823b5a236d8d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -208,7 +208,7 @@ config SND_PXA2XX_SOC_IMOTE2 config SND_MMP_SOC_BROWNSTONE tristate "SoC Audio support for Marvell Brownstone" - depends on SND_MMP_SOC && MACH_BROWNSTONE + depends on SND_MMP_SOC && MACH_BROWNSTONE && I2C select SND_MMP_SOC_SSPA select MFD_WM8994 select SND_SOC_WM8994 -- 2.9.0

1 0

[alsa-devel] [PATCH] ALSA: hda - allow 40 bit DMA mask for NVidia devices
by Ard Biesheuvel 18 Oct '16

18 Oct '16

Commit 49d9e77e72cf ("ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllers") simply disabled any DMA exceeding 32 bits for NVidia devices, even though they are capable of performing DMA up to 40 bits. On some architectures (such as arm64), system memory is not guaranteed to be 32-bit addressable by PCI devices, and so this change prevents NVidia devices from working on platforms such as AMD Seattle. Since the original commit already mentioned that up to 40 bits of DMA is supported, and given that the code has been updated in the meantime to support a 40 bit DMA mask on other devices, revert commit 49d9e77e72cf and explicitly set the DMA mask to 40 bits for NVidia devices. Signed-off-by: Ard Biesheuvel <ard.biesheuvel(a)linaro.org> --- sound/pci/hda/hda_intel.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c3469f756ec2..c64d986009a9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -341,8 +341,7 @@ enum { /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ - (AZX_DCAPS_NO_MSI | /*AZX_DCAPS_ALIGN_BUFSIZE |*/ \ - AZX_DCAPS_NO_64BIT | AZX_DCAPS_CORBRP_SELF_CLEAR |\ + (AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\ AZX_DCAPS_SNOOP_TYPE(NVIDIA)) #define AZX_DCAPS_PRESET_CTHDA \ @@ -1716,6 +1715,10 @@ static int azx_first_init(struct azx *chip) } } + /* NVidia hardware normally only supports up to 40 bits of DMA */ + if (chip->pci->vendor == PCI_VENDOR_ID_NVIDIA) + dma_bits = 40; + /* disable 64bit DMA address on some devices */ if (chip->driver_caps & AZX_DCAPS_NO_64BIT) { dev_dbg(card->dev, "Disabling 64bit DMA\n"); -- 2.7.4

3 7

[alsa-devel] [PATCH] ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializer for array
by Colin King 18 Oct '16

18 Oct '16

From: Colin Ian King <colin.king(a)canonical.com> Smatch reports below warnings: bxt_da7219_max98357a.c:352:9: warning: obsolete array initializer, use C99 syntax An earlier commit cleaned up similar warnings, however, a recent commit 43c02ede76a60de ("ASoC: Intel: Add DMIC channel constraint for bxt machine") re-introduced the older initializer style. So fix this warning to make the code consistent. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 865a21e..bff80b4 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -348,7 +348,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .dynamic = 1, .ops = &broxton_refcap_ops, }, - [BXT_DPCM_AUDIO_DMIC_CP] + [BXT_DPCM_AUDIO_DMIC_CP] = { .name = "Bxt Audio DMIC cap", .stream_name = "dmiccap", -- 2.9.3

3 2

[alsa-devel] [PATCH] sound/pci/asihpi: fix kernel memory disclosure
by Vlad Tsyrklevich 18 Oct '16

18 Oct '16

Some elements in hr are not cleared before being copied to user space, leaking kernel heap memory to user space. For example, this happens in the error handling code for the HPI_ADAPTER_DELETE case. Zero the memory before it's copied. Signed-off-by: Vlad Tsyrklevich <vlad(a)tsyrklevich.net> --- sound/pci/asihpi/hpioctl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index d17937b..7e3aa50 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -111,7 +111,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) return -EINVAL; hm = kmalloc(sizeof(*hm), GFP_KERNEL); - hr = kmalloc(sizeof(*hr), GFP_KERNEL); + hr = kzalloc(sizeof(*hr), GFP_KERNEL); if (!hm || !hr) { err = -ENOMEM; goto out; -- 2.7.0

2 1

[alsa-devel] [PATCH] ALSA: hda - Adding a new group of pin cfg into ALC295 pin quirk table
by Hui Wang 18 Oct '16

18 Oct '16

We have a new Dell laptop model which uses ALC295, the pin definition is different from the existing ones in the pin quirk table, to fix the headset mic detection and mic mute led's problem, we need to add the new pin defintion into the pin quirk table. Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b58e8c7..8483fc2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5811,8 +5811,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { #define ALC295_STANDARD_PINS \ {0x12, 0xb7a60130}, \ {0x14, 0x90170110}, \ - {0x17, 0x21014020}, \ - {0x18, 0x21a19030}, \ {0x21, 0x04211020} #define ALC298_STANDARD_PINS \ @@ -6039,7 +6037,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC292_STANDARD_PINS, {0x13, 0x90a60140}), SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC295_STANDARD_PINS), + ALC295_STANDARD_PINS, + {0x17, 0x21014020}, + {0x18, 0x21a19030}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC295_STANDARD_PINS, + {0x17, 0x21014040}, + {0x18, 0x21a19050}), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_STANDARD_PINS, {0x17, 0x90170110}), -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoc: wm8580: Add the wm8581 codec to the driver
by Matt Flax 18 Oct '16

18 Oct '16

This patch adds support for the wm8581 codec to the wm8580 driver. The wm8581 codec hardware adds a fourth DAC and otherwise is compatible with the wm8580 codec. of_device_id data is used to allow the driver to select the suitable software variables from the device tree codec selection. The wm8580_driver_data struct is used to store the number of DACs as well as the required stream names for both codecs. The snd_soc_dai_driver no longer lists the stream names and for playback the channels_max. These variables are set during i2c probe from the of_device_id supplied wm8580_driver_data struct. With knowledge of the number of DACs in use, the DAC4 controls, widgets and routes are added as required for DAC4. The device tree documentation for the wm8580 is altered to list the wm8581 codec support, as is the Kconfig file. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- Documentation/devicetree/bindings/sound/wm8580.txt | 4 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8580.c | 96 +++++++++++++++++++--- sound/soc/codecs/wm8580.h | 6 ++ 4 files changed, 94 insertions(+), 14 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8580.txt b/Documentation/devicetree/bindings/sound/wm8580.txt index 7d9821f..78fce9b 100644 --- a/Documentation/devicetree/bindings/sound/wm8580.txt +++ b/Documentation/devicetree/bindings/sound/wm8580.txt @@ -1,10 +1,10 @@ -WM8580 audio CODEC +WM8580 and WM8581 audio CODEC This device supports I2C only. Required properties: - - compatible : "wlf,wm8580" + - compatible : "wlf,wm8580", "wlf,wm8581" - reg : the I2C address of the device. diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c67667b..199bec8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -914,7 +914,7 @@ config SND_SOC_WM8523 depends on I2C config SND_SOC_WM8580 - tristate "Wolfson Microelectronics WM8523 CODEC" + tristate "Wolfson Microelectronics WM8580 and WM8581 CODECs" depends on I2C config SND_SOC_WM8711 diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index faa7287..a348549 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,5 +1,5 @@ /* - * wm8580.c -- WM8580 ALSA Soc Audio driver + * wm8580.c -- WM8580 and WM8581 ALSA Soc Audio driver * * Copyright 2008-12 Wolfson Microelectronics PLC. * @@ -12,6 +12,9 @@ * The WM8580 is a multichannel codec with S/PDIF support, featuring six * DAC channels and two ADC channels. * + * The WM8581 is a multichannel codec with S/PDIF support, featuring eight + * DAC channels and two ADC channels. + * * Currently only the primary audio interface is supported - S/PDIF and * the secondary audio interfaces are not. */ @@ -65,6 +68,8 @@ #define WM8580_DIGITAL_ATTENUATION_DACR2 0x17 #define WM8580_DIGITAL_ATTENUATION_DACL3 0x18 #define WM8580_DIGITAL_ATTENUATION_DACR3 0x19 +#define WM8580_DIGITAL_ATTENUATION_DACL4 0x1A +#define WM8580_DIGITAL_ATTENUATION_DACR4 0x1B #define WM8580_MASTER_DIGITAL_ATTENUATION 0x1C #define WM8580_ADC_CONTROL1 0x1D #define WM8580_SPDTXCHAN0 0x1E @@ -242,6 +247,7 @@ struct wm8580_priv { struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES]; struct pll_state a; struct pll_state b; + const struct wm8580_driver_data *drvdata; int sysclk[2]; }; @@ -306,6 +312,19 @@ SOC_DOUBLE("Capture Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 1), SOC_SINGLE("Capture High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0), }; +static const struct snd_kcontrol_new wm8580_dac4_snd_controls[] = { +SOC_DOUBLE_R_EXT_TLV("DAC4 Playback Volume", + WM8580_DIGITAL_ATTENUATION_DACL4, + WM8580_DIGITAL_ATTENUATION_DACR4, + 0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv), + +SOC_SINGLE("DAC4 Deemphasis Switch", WM8580_DAC_CONTROL3, 3, 1, 0), + +SOC_DOUBLE("DAC4 Invert Switch", WM8580_DAC_CONTROL4, 8, 7, 1, 0), + +SOC_SINGLE("DAC4 Switch", WM8580_DAC_CONTROL5, 3, 1, 1), +}; + static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1), SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1), @@ -324,6 +343,13 @@ SND_SOC_DAPM_INPUT("AINL"), SND_SOC_DAPM_INPUT("AINR"), }; +static const struct snd_soc_dapm_widget wm8580_dac4_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC4", "Playback", WM8580_PWRDN1, 5, 1), + +SND_SOC_DAPM_OUTPUT("VOUT4L"), +SND_SOC_DAPM_OUTPUT("VOUT4R"), +}; + static const struct snd_soc_dapm_route wm8580_dapm_routes[] = { { "VOUT1L", NULL, "DAC1" }, { "VOUT1R", NULL, "DAC1" }, @@ -338,6 +364,11 @@ static const struct snd_soc_dapm_route wm8580_dapm_routes[] = { { "ADC", NULL, "AINR" }, }; +static const struct snd_soc_dapm_route wm8580_dac4_dapm_routes[] = { + { "VOUT4L", NULL, "DAC4" }, + { "VOUT4R", NULL, "DAC4" }, +}; + /* PLL divisors */ struct _pll_div { u32 prescale:1; @@ -837,19 +868,16 @@ static const struct snd_soc_dai_ops wm8580_dai_ops_capture = { static struct snd_soc_dai_driver wm8580_dai[] = { { - .name = "wm8580-hifi-playback", .id = WM8580_DAI_PAIFRX, .playback = { .stream_name = "Playback", .channels_min = 1, - .channels_max = 6, .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, .ops = &wm8580_dai_ops_playback, }, { - .name = "wm8580-hifi-capture", .id = WM8580_DAI_PAIFTX, .capture = { .stream_name = "Capture", @@ -865,8 +893,22 @@ static struct snd_soc_dai_driver wm8580_dai[] = { static int wm8580_probe(struct snd_soc_codec *codec) { struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); int ret = 0; + switch (wm8580->drvdata->num_dacs) { + case 4: + snd_soc_add_codec_controls(codec, wm8580_dac4_snd_controls, + ARRAY_SIZE(wm8580_dac4_snd_controls)); + snd_soc_dapm_new_controls(dapm, wm8580_dac4_dapm_widgets, + ARRAY_SIZE(wm8580_dac4_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, wm8580_dac4_dapm_routes, + ARRAY_SIZE(wm8580_dac4_dapm_routes)); + break; + default: + break; + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); if (ret != 0) { @@ -914,12 +956,6 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8580 = { }, }; -static const struct of_device_id wm8580_of_match[] = { - { .compatible = "wlf,wm8580" }, - { }, -}; -MODULE_DEVICE_TABLE(of, wm8580_of_match); - static const struct regmap_config wm8580_regmap = { .reg_bits = 7, .val_bits = 9, @@ -932,10 +968,32 @@ static const struct regmap_config wm8580_regmap = { .volatile_reg = wm8580_volatile, }; +const struct wm8580_driver_data wm8580_data = { + .name_playback = "wm8580-hifi-playback", + .name_capture = "wm8580-hifi-capture", + .num_dacs = 3, +}; +EXPORT_SYMBOL_GPL(wm8580_data); + +const struct wm8580_driver_data wm8581_data = { + .name_playback = "wm8581-hifi-playback", + .name_capture = "wm8581-hifi-capture", + .num_dacs = 4, +}; +EXPORT_SYMBOL_GPL(wm8581_data); + +static const struct of_device_id wm8580_of_match[] = { + { .compatible = "wlf,wm8580", .data = &wm8580_data }, + { .compatible = "wlf,wm8581", .data = &wm8581_data }, + { }, +}; +MODULE_DEVICE_TABLE(of, wm8580_of_match); + #if IS_ENABLED(CONFIG_I2C) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + const struct of_device_id *of_id; struct wm8580_priv *wm8580; int ret, i; @@ -960,6 +1018,20 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8580); + of_id = of_match_device(wm8580_of_match, &i2c->dev); + if (of_id) + wm8580->drvdata = of_id->data; + + if (!wm8580->drvdata) { + dev_err(&i2c->dev, "failed to find driver data\n"); + return -EINVAL; + } + + /* Each dac supports stereo input */ + wm8580_dai[0].playback.channels_max = wm8580->drvdata->num_dacs * 2; + wm8580_dai[0].name = wm8580->drvdata->name_playback; + wm8580_dai[1].name = wm8580->drvdata->name_capture; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8580, wm8580_dai, ARRAY_SIZE(wm8580_dai)); @@ -973,7 +1045,8 @@ static int wm8580_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8580_i2c_id[] = { - { "wm8580", 0 }, + { "wm8580", (kernel_ulong_t)&wm8580_data }, + { "wm8581", (kernel_ulong_t)&wm8581_data }, { } }; MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); @@ -1014,4 +1087,5 @@ module_exit(wm8580_exit); MODULE_DESCRIPTION("ASoC WM8580 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.org>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 1d34656..868e400 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -15,6 +15,12 @@ #ifndef _WM8580_H #define _WM8580_H +struct wm8580_driver_data { + char name_playback[24]; + char name_capture[24]; + int num_dacs; +}; + #define WM8580_PLLA 1 #define WM8580_PLLB 2 -- 2.7.4

2 6

[alsa-devel] [PATCH v2] ASoC: rt5514: Add the DMIC initial delay to wait it ready.
by Oder Chiou 18 Oct '16

18 Oct '16

Due to the DMIC that needs time to initial after the MCLK is provided, the field of delay time is implemented by the platform data. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5514.h | 20 ++++++++++++++++++++ sound/soc/codecs/rt5514.c | 17 +++++++++++++++++ sound/soc/codecs/rt5514.h | 2 ++ 3 files changed, 39 insertions(+) create mode 100644 include/sound/rt5514.h diff --git a/include/sound/rt5514.h b/include/sound/rt5514.h new file mode 100644 index 0000000..ef18494 --- /dev/null +++ b/include/sound/rt5514.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt5514.h -- Platform data for RT5514 + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(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 __LINUX_SND_RT5514_H +#define __LINUX_SND_RT5514_H + +struct rt5514_platform_data { + unsigned int dmic_init_delay; +}; + +#endif + diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f24b7cf..0d7bbdf 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -452,6 +452,9 @@ static int rt5514_set_dmic_clk(struct snd_soc_dapm_widget *w, RT5514_CLK_DMIC_OUT_SEL_MASK, idx << RT5514_CLK_DMIC_OUT_SEL_SFT); + if (rt5514->pdata.dmic_init_delay) + msleep(rt5514->pdata.dmic_init_delay); + return idx; } @@ -1073,9 +1076,18 @@ static const struct of_device_id rt5514_of_match[] = { MODULE_DEVICE_TABLE(of, rt5514_of_match); #endif +static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) +{ + device_property_read_u32(dev, "realtek,dmic-init-delay", + &rt5514->pdata.dmic_init_delay); + + return 0; +} + static int rt5514_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt5514_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5514_priv *rt5514; int ret; unsigned int val; @@ -1087,6 +1099,11 @@ static int rt5514_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5514); + if (pdata) + rt5514->pdata = *pdata; + else if (i2c->dev.of_node) + rt5514_parse_dt(rt5514, &i2c->dev); + rt5514->i2c_regmap = devm_regmap_init_i2c(i2c, &rt5514_i2c_regmap); if (IS_ERR(rt5514->i2c_regmap)) { ret = PTR_ERR(rt5514->i2c_regmap); diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 229de0e..5d343fb 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -13,6 +13,7 @@ #define __RT5514_H__ #include <linux/clk.h> +#include <sound/rt5514.h> #define RT5514_DEVICE_ID 0x10ec5514 @@ -243,6 +244,7 @@ enum { }; struct rt5514_priv { + struct rt5514_platform_data pdata; struct snd_soc_codec *codec; struct regmap *i2c_regmap, *regmap; struct clk *mclk; -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH] ASoC: rt5514: Add the DMIC initial delay to wait it ready.
by Oder Chiou 18 Oct '16

18 Oct '16

Due to the DMIC that needs time to initial after the MCLK is provided, the field of delay time is implemented by the platform data. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5514.txt | 3 +++ sound/soc/codecs/rt5514.c | 17 +++++++++++++++++ sound/soc/codecs/rt5514.h | 2 ++ 3 files changed, 22 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 9cabfc1..a8074e7 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -13,6 +13,9 @@ Optional properties: - clocks: The phandle of the master clock to the CODEC - clock-names: Should be "mclk" +- realtek,dmic-init-delay + Set the DMIC initial delay to wait it ready. + Pins on the device (for linking into audio routes) for RT5514: * DMIC1L diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f24b7cf..0d7bbdf 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -452,6 +452,9 @@ static int rt5514_set_dmic_clk(struct snd_soc_dapm_widget *w, RT5514_CLK_DMIC_OUT_SEL_MASK, idx << RT5514_CLK_DMIC_OUT_SEL_SFT); + if (rt5514->pdata.dmic_init_delay) + msleep(rt5514->pdata.dmic_init_delay); + return idx; } @@ -1073,9 +1076,18 @@ static const struct of_device_id rt5514_of_match[] = { MODULE_DEVICE_TABLE(of, rt5514_of_match); #endif +static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) +{ + device_property_read_u32(dev, "realtek,dmic-init-delay", + &rt5514->pdata.dmic_init_delay); + + return 0; +} + static int rt5514_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt5514_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5514_priv *rt5514; int ret; unsigned int val; @@ -1087,6 +1099,11 @@ static int rt5514_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5514); + if (pdata) + rt5514->pdata = *pdata; + else if (i2c->dev.of_node) + rt5514_parse_dt(rt5514, &i2c->dev); + rt5514->i2c_regmap = devm_regmap_init_i2c(i2c, &rt5514_i2c_regmap); if (IS_ERR(rt5514->i2c_regmap)) { ret = PTR_ERR(rt5514->i2c_regmap); diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 229de0e..5d343fb 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -13,6 +13,7 @@ #define __RT5514_H__ #include <linux/clk.h> +#include <sound/rt5514.h> #define RT5514_DEVICE_ID 0x10ec5514 @@ -243,6 +244,7 @@ enum { }; struct rt5514_priv { + struct rt5514_platform_data pdata; struct snd_soc_codec *codec; struct regmap *i2c_regmap, *regmap; struct clk *mclk; -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH -next] ASoC: rk3399_gru_sound: Fix non static symbol warning
by Wei Yongjun 17 Oct '16

17 Oct '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Fixes the following sparse warning: sound/soc/rockchip/rk3399_gru_sound.c:41:14: warning: symbol 'rt5514_dmic_delay' was not declared. Should it be static? Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 9ed735a..0cbd235 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -38,7 +38,7 @@ #define SOUND_FS 256 -unsigned int rt5514_dmic_delay; +static unsigned int rt5514_dmic_delay; static struct snd_soc_jack rockchip_sound_jack;

1 0

[alsa-devel] Yamaha Montage class-compliant USB MIDI
by Chris Webb 17 Oct '16

17 Oct '16

I have a Yamaha Montage synthesiser sitting on my desk, which supports MIDI and multi-channel audio over USB. I'd like to get it talking to Linux ALSA. To my surprise, it offers two USB configurations, the normal Yamaha vendor-specific one (presumably very similar to the MOX defined in quirks-table.h:362) but also a class-compliant one. The class-complaint configuration nearly works out of the box. Audio and MIDI in (to the computer) work fine, but MIDI out (i.e. to the synth) is broken, at least in kernel 4.9-rc1 and 4.8. Here's the lsusb output: # lsusb -v -s 001:002 Bus 001 Device 002: ID 0499:1707 Yamaha Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x0499 Yamaha Corp. idProduct 0x1707 bcdDevice 1.02 iManufacturer 1 Yamaha Corporation iProduct 2 MONTAGE iSerial 0 bNumConfigurations 2 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 349 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 4 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 75 bmControl 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 1 bmAttributes 0x03 Internal programmable Clock bmControls 0x03 Clock Frequency Control (read/write) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 1 bNrChannels 6 bmChannelConfig 0x00000000 bmControls 0x0000 iChannelNames 16 Main L iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bSourceID 2 bCSourceID 1 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bCSourceID 1 bNrChannels 32 bmChannelConfig 0x00000000 bmControls 0x0000 iChannelNames 48 USB Main L iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 4 bCSourceID 1 bmControls 0x0000 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 6 bmChannelConfig 0x00000000 iChannelNames 16 Main L AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x07 EP 7 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0075 1x 117 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 32 bmChannelConfig 0x00000000 iChannelNames 48 USB Main L AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0271 1x 625 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 129 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 3 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 33 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 34 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 35 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 65 bNrInputPins 1 baSourceID( 0) 33 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 66 bNrInputPins 1 baSourceID( 0) 34 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 67 bNrInputPins 1 baSourceID( 0) 35 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 97 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 98 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 99 bNrInputPins 1 baSourceID( 0) 3 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 3 baAssocJackID( 0) 33 baAssocJackID( 1) 34 baAssocJackID( 2) 35 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 3 baAssocJackID( 0) 97 baAssocJackID( 1) 98 baAssocJackID( 2) 99 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 290 bNumInterfaces 5 bConfigurationValue 2 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 0a 24 01 00 01 34 00 02 01 02 ** UNRECOGNIZED: 0c 24 02 01 01 01 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 02 01 03 00 01 00 ** UNRECOGNIZED: 0c 24 02 03 03 06 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 04 01 01 00 03 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 01 01 00 ** UNRECOGNIZED: 0b 24 02 01 06 03 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x07 EP 7 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0075 1x 117 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 04 01 01 00 ** UNRECOGNIZED: 0b 24 02 01 20 03 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0271 1x 625 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 42 00 ** UNRECOGNIZED: 06 24 02 02 01 00 ** UNRECOGNIZED: 06 24 02 02 02 00 ** UNRECOGNIZED: 06 24 02 02 03 00 ** UNRECOGNIZED: 09 24 03 02 01 01 01 01 00 ** UNRECOGNIZED: 09 24 03 02 02 01 01 01 00 ** UNRECOGNIZED: 09 24 03 02 03 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 255 Vendor Specific Protocol iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 24 00 ** UNRECOGNIZED: 06 24 02 02 01 00 ** UNRECOGNIZED: 09 24 03 02 01 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 2 Device Status: 0x0001 Self Powered # Doing an aplaymidi -p 20:0 test.mid hangs without receiving the test notes at the synth end, and the kernel eventually reports: sound midiC1D0: rawmidi drain error (avail = 3700, buffer_size = 4096) Monitoring the USB traffic while this happens, # cat /sys/kernel/debug/usb/usbmon/1u ffff995b59da2780 93721891 S Bo:1:002:1 -115 16 = 0990404c 0991404c 0992404c 0993404c ffff995b59da2780 93721978 C Bo:1:002:1 0 16 > ffff995b59da2b40 93915884 S Bo:1:002:1 -115 12 = 09904000 09914000 09924000 ffff995b59da2b40 93915967 C Bo:1:002:1 0 12 > ffff995b59e563c0 93917882 S Bo:1:002:1 -115 4 = 09934000 ffff995b59e56540 94057875 S Bo:1:002:1 -115 4 = 09903e4a ffff995b59e56600 94059872 S Bo:1:002:1 -115 12 = 09913e4a 09923e4a 09933e4a ffff995b59e56780 94238881 S Bo:1:002:1 -115 16 = 09903e00 09913e00 09923e00 09933e00 ffff995b59e56840 94418886 S Bo:1:002:1 -115 16 = 09903c46 09913c46 09923c46 09933c46 ffff995b59da2780 94632882 S Bo:1:002:1 -115 4 = 0dd00800 ffff995b59da2b40 94633887 S Bo:1:002:1 -115 12 = 0dd10800 0dd20800 0dd30800 [hangs] so it looks like we stop getting callbacks for URBs after the first couple? This class-compliant interface apparently works okay on macOS/iOS, although macOS uses the vendor-specific interface when the proprietary drivers are installed. But I don't know how to get a dump of the USB traffic on either of those platforms for comparison. Looking at where the callbacks stop, I (naively) wonder if macOS serialises USB MIDI URBs or something like that, and I'm seeing a Yamaha bug with multiple URBs in flight? Is there anything more I can usefully do to further diagnose what's going on? Would it be better to attempt to support the (presumably very similar to existing MOX and more heavily tested) vendor-specific USB configuration? Is it possible to write a quirk that prefers this configuration, assuming similar structure to MOX and earlier Yamaha audio and MIDI devices? Best wishes, Chris.

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[alsa-devel] [PATCH] ASoC: rt5670: Enable MCLK detection
by Bard Liao 17 Oct '16

17 Oct '16

There is a power saving mechanism in rt5670. It will turn off some unused power when MCLK is not present. We call that "MCLK detection" and it should be enabled by default. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 4 +++- sound/soc/codecs/rt5670.h | 1 + 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 49caf13..6caac92 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2889,6 +2889,9 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + regmap_update_bits(rt5670->regmap, RT5670_DIG_MISC, + RT5670_MCLK_DET, RT5670_MCLK_DET); + if (rt5670->pdata.in2_diff) regmap_update_bits(rt5670->regmap, RT5670_IN2, RT5670_IN_DF2, RT5670_IN_DF2); @@ -2903,7 +2906,6 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); - regmap_update_bits(rt5670->regmap, RT5670_DIG_MISC, 0x8, 0x8); } if (rt5670->pdata.jd_mode) { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 3f1b0f1..5ba485c 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1914,6 +1914,7 @@ enum { #define RT5670_IF1_ADC1_IN2_SFT 11 #define RT5670_IF1_ADC2_IN1_SEL (0x1 << 10) #define RT5670_IF1_ADC2_IN1_SFT 10 +#define RT5670_MCLK_DET (0x1 << 3) /* General Control2 (0xfb) */ #define RT5670_RXDC_SRC_MASK (0x1 << 7) -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2] ASoC: rt5640: enable MCLK detection
by Bard Liao 17 Oct '16

17 Oct '16

There is a power saving mechanism in rt5640. It will turn off some unused power when MCLK is not present. We call that "MCLK detection" and it should be enabled by default. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Improve commit message. --- sound/soc/codecs/rt5640.c | 3 +++ sound/soc/codecs/rt5640.h | 2 ++ 2 files changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 3cc1135..0a7378f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2407,6 +2407,9 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + regmap_update_bits(rt5640->regmap, RT5640_DUMMY1, + RT5640_MCLK_DET, RT5640_MCLK_DET); + if (rt5640->pdata.in1_diff) regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, RT5640_IN_DF1, RT5640_IN_DF1); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 90c8871..22c017c 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -1970,6 +1970,8 @@ #define RT5640_ZCD_HP_DIS (0x0 << 15) #define RT5640_ZCD_HP_EN (0x1 << 15) +/* General Control 1 (0xfa) */ +#define RT5640_MCLK_DET (0x1 << 11) /* Codec Private Register definition */ /* 3D Speaker Control (0x63) */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: rt5640: enable MCLK detection
by Bard Liao 17 Oct '16

17 Oct '16

Enable MCLK detection. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 3 +++ sound/soc/codecs/rt5640.h | 2 ++ 2 files changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 3cc1135..0a7378f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2407,6 +2407,9 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + regmap_update_bits(rt5640->regmap, RT5640_DUMMY1, + RT5640_MCLK_DET, RT5640_MCLK_DET); + if (rt5640->pdata.in1_diff) regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, RT5640_IN_DF1, RT5640_IN_DF1); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 90c8871..22c017c 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -1970,6 +1970,8 @@ #define RT5640_ZCD_HP_DIS (0x0 << 15) #define RT5640_ZCD_HP_EN (0x1 << 15) +/* General Control 1 (0xfa) */ +#define RT5640_MCLK_DET (0x1 << 11) /* Codec Private Register definition */ /* 3D Speaker Control (0x63) */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/2 v2][resend] DesignWare HDMI I2S suport
by Kuninori Morimoto 17 Oct '16

17 Oct '16

Hi Archit, Mark Cc Thierry, Russell These are resend of v2 of DesignWare HDMI I2S support patches. (I added drm-bridge maintainer Archit on To:) It will use ALSA SoC hdmi-codec driver, but we can't use it as-is at this point. 1) patch tidyup hdmi-codec driver to enable dw-hdmi I2S support. 2) patch is based on it. Difficult is that these 2 patches are under different maintainer. 1) Mark ? 2) Archit ? Kuninori Morimoto (2): 1) ASoC: hdmi-codec: enable multi probe for same device 2) drm: bridge: add DesignWare HDMI I2S audio support drivers/gpu/drm/bridge/Kconfig | 8 ++++++++ drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 +++++++ drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 130 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 ++++++++++++++++++-- drivers/gpu/drm/bridge/dw-hdmi.h | 21 +++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++--- 7 files changed, 251 insertions(+), 5 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/7] constify snd_soc_ops structures
by Julia Lawall 17 Oct '16

17 Oct '16

Constify snd_soc_ops structures. --- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/imx-wm8962.c | 2 +- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-scu-card.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 ++-- sound/soc/intel/boards/bxt_rt298.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 4 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 ++-- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 6 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 6 +++--- sound/soc/intel/boards/skl_rt286.c | 4 ++-- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 6 +++--- sound/soc/rockchip/rockchip_max98090.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- 32 files changed, 46 insertions(+), 46 deletions(-)

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[alsa-devel] PCM params and Sub unit direction:Invalid argument
by vbs 17 Oct '16

17 Oct '16

Hello gentlemen, I have a problem using libasound (1.1.0-0ubuntu1) in a program (Reicast emulator) with my system. When trying to initialize the ALSA device in the code by invoking "snd_pcm_hw_params" I get "Invalid argument". I stripped the problem down to a simple demo program: http://pastebin.com/DMQrcYft When I run it I get: Unable to set hw parameters: Invalid argument So the call to "snd_pcm_hw_params" fails on my system. I am using Lubuntu x64 16.04 on a NUC D54250. This is my .asoundrc: pcm.softvol { type softvol slave { pcm "hw:0,3" } control { name "Master" card 0 } } pcm.!default { type plug slave.pcm "softvol" } I created a softvol device to have a master volume mixer (some programs rely on it). This is the output of "aplay -L": http://pastebin.com/k0qkfNnd Two points I have noticed: 1. The problem disappears when not using the softvol device. When I replace my .asoundrc with this, then it works: pcm.!default { type hw card 0 device 3 } So the problems might be related to the defined softvol device. 2. In the sample program there is this code: int dir; ... rc=snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); The variable dir is not initialized. But AFAIK that parameter in the function "snd_pcm_hw_params_set_rate_near" is also used as an input value. But the uninitialized variable will just have an undefined state. So this looks fishy to me. When I change the definition to "int dir=0" then it actually works! With "int dir=1" the problem happens again. I would think this is just a bug in the code but the strange thing is that the official example does it in the same way (calling "snd_pcm_hw_params_set_buffer_time_near" with an uninitialized dir variable): http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html <http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html> How can that unintialized variable be correct if it really is used as an input value? Maybe someone has thoughts on this? Is the code correct? Especially regarding the unintialized variable? Are there maybe problems with my .asoundrc softvol device? Big thanks in advance guys! Cheers, Robin

1 0

[alsa-devel] [PATCH 1/3] ASoC: ak4641: Remove unused DAI ID defines
by Lars-Peter Clausen 15 Oct '16

15 Oct '16

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. This driver was merged after the change of DAIs are matched was made and the IDs were never used and presumably never will be, so remove them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4641.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/ak4641.h b/sound/soc/codecs/ak4641.h index 4a26324..38b9e93 100644 --- a/sound/soc/codecs/ak4641.h +++ b/sound/soc/codecs/ak4641.h @@ -38,10 +38,4 @@ #define AK4641_CACHEREGNUM 0x14 - - -#define AK4641_DAI_HIFI 0 -#define AK4641_DAI_VOICE 1 - - #endif -- 2.1.4

1 2

[alsa-devel] [PATCH] ASoC: es8328: Remove some unused defines
by Lars-Peter Clausen 15 Oct '16

15 Oct '16

The es8328 drivers seems to have used the wm8753 driver as a template. Its header file contains a few defines that are copied verbatim from the wm8753 header that do not really apply to this driver and are unused. So remove those defines. This seems safe as they deal with PLL and clock divider settings and the es8328 driver neither implements a set_pll() nor a set_clkdiv() callback. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/es8328.h | 37 ------------------------------------- 1 file changed, 37 deletions(-) diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 1a736e7..8930322 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -278,43 +278,6 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_REG_MAX 0x35 -#define ES8328_PLL1 0 -#define ES8328_PLL2 1 - -/* clock inputs */ -#define ES8328_MCLK 0 -#define ES8328_PCMCLK 1 - -/* clock divider id's */ -#define ES8328_PCMDIV 0 -#define ES8328_BCLKDIV 1 -#define ES8328_VXCLKDIV 2 - -/* PCM clock dividers */ -#define ES8328_PCM_DIV_1 (0 << 6) -#define ES8328_PCM_DIV_3 (2 << 6) -#define ES8328_PCM_DIV_5_5 (3 << 6) -#define ES8328_PCM_DIV_2 (4 << 6) -#define ES8328_PCM_DIV_4 (5 << 6) -#define ES8328_PCM_DIV_6 (6 << 6) -#define ES8328_PCM_DIV_8 (7 << 6) - -/* BCLK clock dividers */ -#define ES8328_BCLK_DIV_1 (0 << 7) -#define ES8328_BCLK_DIV_2 (1 << 7) -#define ES8328_BCLK_DIV_4 (2 << 7) -#define ES8328_BCLK_DIV_8 (3 << 7) - -/* VXCLK clock dividers */ -#define ES8328_VXCLK_DIV_1 (0 << 6) -#define ES8328_VXCLK_DIV_2 (1 << 6) -#define ES8328_VXCLK_DIV_4 (2 << 6) -#define ES8328_VXCLK_DIV_8 (3 << 6) -#define ES8328_VXCLK_DIV_16 (4 << 6) - -#define ES8328_DAI_HIFI 0 -#define ES8328_DAI_VOICE 1 - #define ES8328_1536FS 1536 #define ES8328_1024FS 1024 #define ES8328_768FS 768 -- 2.1.4

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Re: [alsa-devel] [FFADO-devel] Clicking issues with the output of an Onyx Blackbird
by Jordi Torres 15 Oct '16

15 Oct '16

> (CCed to alsa-devel) I sent this issue to alsa mailist 6 days ago but I supouse that hit the spam filter. Even I published an alsa-info http://www.alsa-project.org/db/?f=cb6745b6d0acdb5d9375096ec8dc008f6ff49cd7 When I read the wiki about reporting bugs, I understood that kernel related bugs goes to bugzilla.kernel.org and user-land bugs goes to the devel mail-list. So I supouse that I should sent this issue to the bugzilla as it's considered a kernel bug > Well, could I request you to gather information of packet sequence? Here is the trace of packets: https://drive.google.com/file/d/0Bz44Dr5G2FeJSXBHMzc5WTFmOWs/view?usp=shari… The trace includes the opening of alsa device and playing a sine wave for 30 seconds My linux box has two firewire ports I don't know the details about the host controller but could be feasible sniff the connection between the other computer with official drivers and the DAW? Could that be useful? PD: Resent due to using the wrong sender address

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[alsa-devel] struct snd_soc_codec_driver nested and duplicated fields
by Christian N 15 Oct '16

15 Oct '16

Hi I'm tryng to understand serveral links between asoc structures to avoid bad codec/machine driver implementation (TEF663x for iMX6). I saw that several structures have pointer to structures which one could have pointer to the first one, and serveral fields are duplicated. Some examples snd_soc_codec_driver->probe snd_soc_codec_driver->snd_soc_dapm_widget but snd_soc_codec_driver->snd_soc_component_driver->probe (duplicated?) snd_soc_codec_driver->snd_soc_component_driver->snd_soc_dapm_widget (duplicated?) and snd_soc_codec_driver->snd_soc_component_driver->snd_soc_dapm_widget->snd_soc_dapm_context-> snd_soc_component (hey is recursive???!!!!!) I would like figure out why several fields are duplicated in structures that could be nested. Is a compatibility issue? thanks in advance

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[alsa-devel] [TDM help please] multiplex out a stereo I2S line
by Matt Flax 14 Oct '16

14 Oct '16

Hi there, Problem is that I need multiplex all 6 channels out a stereo I2S line. I have a working system ... the CPU thinks it is stereo I2S, but the codec thinks it is TDM multichannel. When I aplay a pre-interleaved wav file, it works ! Check the video here : https://www.kickstarter.com/projects/1250664710/1299781340?token=e465b540 Problem is that I need to multiplex all 6 channels down the stereo I2S line. I have tried to do this with the asoundrc by attempting to declare the same device 3 times and interleaving there, however have failed. I am happy to alter my machine driver to implement this ... Is there a way to present 3 stereo subdevices to the user, where each of these subdevcies is interleaved to form the TDM before sending to the codec ? Is there perhaps a better or more elegant way to go about this ? thanks Matt

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Re: [alsa-devel] inconsistent reference counting for snd-soc-skl
by Subhransu S. Prusty 13 Oct '16

13 Oct '16

On Thu, Oct 13, 2016 at 10:19:37PM +0900, Takashi Sakamoto wrote: > Hi, > > I find an issue of snd-soc-skl due to inconsistent reference counting. > See this call graph. > > skl_probe() > ->skl_first_init() > ->skl_i915_init() > ->snd_hdac_display_power() (CONFIG_SND_SOC_HDAC_HDMI is enabled) > ->bus->i915_power_refcount++ > ->all error pathes > ->skl_free() > ->snd_hda_i915_exit() > ->WARN_ON(!bus->i915_power_refcount) > > This might be also hit by rmmod(8). I see this bug in kernel 4.8 and > pre-4.9. Thanks Takashi for reporting. We will fix this. Regards, Subhransu > > > Regards > > Takashi Sakamoto --

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Re: [alsa-devel] [FFADO-devel] Clicking issues with the output of an Onyx Blackbird
by Jordi Torres 13 Oct '16

13 Oct '16

> (CCed to alsa-devel) I sent this issue to alsa mailist 6 days ago but I supouse that hit the spam filter. Even I published an alsa-info http://www.alsa-project.org/db/?f=cb6745b6d0acdb5d9375096ec8dc008f6ff49cd7 When I read the wiki about reporting bugs, I understood that kernel related bugs goes to bugzilla.kernel.org and user-land bugs goes to the devel mail-list. So I supouse that I should sent this issue to the bugzilla as it's considered a kernel bug > Well, could I request you to gather information of packet sequence? Here is the trace of packets: https://drive.google.com/file/d/0Bz44Dr5G2FeJSXBHMzc5WTFmOWs/view?usp=shari… The trace includes the opening of alsa device and playing a sine wave for 30 seconds My linux box has two firewire ports I don't know the details about the host controller but could be feasible sniff the connection between the other computer with official drivers and the DAW? Could that be useful?

1 0

[alsa-devel] Applied "ASoC: cs4270: fix DAPM stream name mismatch" to the asoc tree
by Mark Brown 13 Oct '16

13 Oct '16

The patch ASoC: cs4270: fix DAPM stream name mismatch has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From aa5f920993bda2095952177eea79bc8e58ae6065 Mon Sep 17 00:00:00 2001 From: murray foster <mrafoster(a)gmail.com> Date: Sun, 9 Oct 2016 13:28:45 -0700 Subject: [PATCH] ASoC: cs4270: fix DAPM stream name mismatch Mismatching stream names in DAPM route and widget definitions are causing compilation errors. Fixing these names allows the cs4270 driver to compile and function. [Errors must be at probe time not compile time -- broonie] Signed-off-by: Murray Foster <mrafoster(a)gmail.com> Acked-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/cs4270.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 18baea2f7d65..84f86745c30e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -148,11 +148,11 @@ SND_SOC_DAPM_OUTPUT("AOUTR"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { - { "Capture", NULL, "AINA" }, - { "Capture", NULL, "AINB" }, + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, - { "AOUTA", NULL, "Playback" }, - { "AOUTB", NULL, "Playback" }, + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, }; /** -- 2.9.3

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[alsa-devel] [PATCH v2 2/3] ASoC: cs35l34: Add device tree bindings file for cs35l34
by Paul.Handrigan＠cirrus.com 13 Oct '16

13 Oct '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Add device tree bindings file for the cs35l34 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l34.txt | 64 ++++++++++++++++++++++ 1 file changed, 64 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l34.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l34.txt b/Documentation/devicetree/bindings/sound/cs35l34.txt new file mode 100644 index 0000000..b218ead --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l34.txt @@ -0,0 +1,64 @@ +CS35L34 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l34" + + - reg : the I2C address of the device for I2C. + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-vtge-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from VP to 8V with + increments of 100mV. + + - cirrus,boost-nanohenry: Inductor value for boost converter. The value is + in nH and they can be values of 1000nH, 1100nH, 1200nH, 1500nH, and 2200nH. + +Optional properties: + + - reset-gpios: GPIO used to reset the amplifier. + + - interrupt-parent : Specifies the phandle of the interrupt controller to + which the IRQs from CS35L34 are delivered to. + - interrupts : IRQ line info CS35L34. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-peak-milliamp : Boost converter peak current limit in mA. The + range starts at 1200mA and goes to a maximum of 3840mA with increments of + 80mA. The default value is 2480mA. + + - cirrus,i2s-sdinloc : ADSP SDIN I2S channel location. Indicates whether the + received mono data is in the left or right portion of the I2S frame + according to the AD0 pin or directly via this configuration. + 0x0 (Default) = Selected by AD0 input (if AD0 = LOW, use left channel), + 0x2 = Left, + 0x1 = Selected by the inversion of the AD0 input (if AD0 = LOW, use right + channel), + 0x3 = Right. + + - cirrus,gain-zc-disable: Boolean property. If set, the gain change will take + effect without waiting for a zero cross. + + - cirrus,tdm-rising-edge: Boolean property. If set, data is on the rising edge of + SCLK. Otherwise, data is on the falling edge of SCLK. + + +Example: + +cs35l34: cs35l34@40 { + compatible = "cirrus,cs35l34"; + reg = <0x40>; + + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + reset-gpios = <&gpio 10 0>; + + cirrus,boost-vtge-milltvolt = <8000>; /* 8V */ + cirrus,boost-ind-nanohenry = <1000>; /* 1uH */ + cirrus,boost-peak-milliamp = <3000>; /* 3A */ +}; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: cs35l34: Initial commit of the cs35l34 CODEC driver.
by Paul.Handrigan＠cirrus.com 13 Oct '16

13 Oct '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l34 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l34.h | 35 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l34.c | 1252 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l34.h | 269 ++++++++++ 5 files changed, 1563 insertions(+) create mode 100644 include/sound/cs35l34.h create mode 100644 sound/soc/codecs/cs35l34.c create mode 100644 sound/soc/codecs/cs35l34.h diff --git a/include/sound/cs35l34.h b/include/sound/cs35l34.h new file mode 100644 index 0000000..9c927cf --- /dev/null +++ b/include/sound/cs35l34.h @@ -0,0 +1,35 @@ +/* + * linux/sound/cs35l34.h -- Platform data for CS35l34 + * + * Copyright (c) 2016 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 __CS35L34_H +#define __CS35L34_H + +struct cs35l34_platform_data { + /* Set AIF to half drive strength */ + bool aif_half_drv; + /* Digital Soft Ramp Disable */ + bool digsft_disable; + /* Amplifier Invert */ + bool amp_inv; + /* Peak current (mA) */ + unsigned int boost_peak; + /* Boost inductor value (nH) */ + unsigned int boost_ind; + /* Boost Controller Voltage Setting (mV) */ + unsigned int boost_vtge; + /* Gain Change Zero Cross */ + bool gain_zc_disable; + /* SDIN Left/Right Selection */ + unsigned int i2s_sdinloc; + /* TDM Rising Edge */ + bool tdm_rising_edge; +}; + +#endif /* __CS35L34_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c67667b..bf2f26d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -48,6 +48,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C + select SND_SOC_CS35L34 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -399,6 +400,10 @@ config SND_SOC_CS35L33 tristate "Cirrus Logic CS35L33 CODEC" depends on I2C +config SND_SOC_CS35L34 + tristate "Cirrus Logic CS35L34 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 958cd49..3f0cd90 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -38,6 +38,7 @@ snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o +snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -263,6 +264,7 @@ obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o +obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(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 diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c new file mode 100644 index 0000000..4045383 --- /dev/null +++ b/sound/soc/codecs/cs35l34.c @@ -0,0 +1,1252 @@ +/* + * cs35l34.c -- CS35l34 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(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/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/pm_runtime.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l34.h> + +#include "cs35l34.h" + +#define PDN_DONE_ATTEMPTS 10 +#define CS35L34_START_DELAY 50 + +struct cs35l34_private { + struct snd_soc_codec *codec; + struct cs35l34_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + int mclk_int; + bool tdm_mode; + struct gpio_desc *reset_gpio; /* Active-low reset GPIO */ +}; + +static const struct reg_default cs35l34_reg[] = { + {CS35L34_PWRCTL1, 0x01}, + {CS35L34_PWRCTL2, 0x19}, + {CS35L34_PWRCTL3, 0x01}, + {CS35L34_ADSP_CLK_CTL, 0x08}, + {CS35L34_MCLK_CTL, 0x11}, + {CS35L34_AMP_INP_DRV_CTL, 0x01}, + {CS35L34_AMP_DIG_VOL_CTL, 0x12}, + {CS35L34_AMP_DIG_VOL, 0x00}, + {CS35L34_AMP_ANLG_GAIN_CTL, 0x0F}, + {CS35L34_PROTECT_CTL, 0x06}, + {CS35L34_AMP_KEEP_ALIVE_CTL, 0x04}, + {CS35L34_BST_CVTR_V_CTL, 0x00}, + {CS35L34_BST_PEAK_I, 0x10}, + {CS35L34_BST_RAMP_CTL, 0x87}, + {CS35L34_BST_CONV_COEF_1, 0x24}, + {CS35L34_BST_CONV_COEF_2, 0x24}, + {CS35L34_BST_CONV_SLOPE_COMP, 0x4E}, + {CS35L34_BST_CONV_SW_FREQ, 0x08}, + {CS35L34_CLASS_H_CTL, 0x0D}, + {CS35L34_CLASS_H_HEADRM_CTL, 0x0D}, + {CS35L34_CLASS_H_RELEASE_RATE, 0x08}, + {CS35L34_CLASS_H_FET_DRIVE_CTL, 0x41}, + {CS35L34_CLASS_H_STATUS, 0x05}, + {CS35L34_VPBR_CTL, 0x0A}, + {CS35L34_VPBR_VOL_CTL, 0x90}, + {CS35L34_VPBR_TIMING_CTL, 0x6A}, + {CS35L34_PRED_MAX_ATTEN_SPK_LOAD, 0x95}, + {CS35L34_PRED_BROWNOUT_THRESH, 0x1C}, + {CS35L34_PRED_BROWNOUT_VOL_CTL, 0x00}, + {CS35L34_PRED_BROWNOUT_RATE_CTL, 0x10}, + {CS35L34_PRED_WAIT_CTL, 0x10}, + {CS35L34_PRED_ZVP_INIT_IMP_CTL, 0x08}, + {CS35L34_PRED_MAN_SAFE_VPI_CTL, 0x80}, + {CS35L34_VPBR_ATTEN_STATUS, 0x00}, + {CS35L34_PRED_BRWNOUT_ATT_STATUS, 0x00}, + {CS35L34_SPKR_MON_CTL, 0xC6}, + {CS35L34_ADSP_I2S_CTL, 0x00}, + {CS35L34_ADSP_TDM_CTL, 0x00}, + {CS35L34_TDM_TX_CTL_1_VMON, 0x00}, + {CS35L34_TDM_TX_CTL_2_IMON, 0x04}, + {CS35L34_TDM_TX_CTL_3_VPMON, 0x03}, + {CS35L34_TDM_TX_CTL_4_VBSTMON, 0x07}, + {CS35L34_TDM_TX_CTL_5_FLAG1, 0x08}, + {CS35L34_TDM_TX_CTL_6_FLAG2, 0x09}, + {CS35L34_TDM_TX_SLOT_EN_1, 0x00}, + {CS35L34_TDM_TX_SLOT_EN_2, 0x00}, + {CS35L34_TDM_TX_SLOT_EN_3, 0x00}, + {CS35L34_TDM_TX_SLOT_EN_4, 0x00}, + {CS35L34_TDM_RX_CTL_1_AUDIN, 0x40}, + {CS35L34_TDM_RX_CTL_3_ALIVE, 0x04}, + {CS35L34_MULT_DEV_SYNCH1, 0x00}, + {CS35L34_MULT_DEV_SYNCH2, 0x80}, + {CS35L34_PROT_RELEASE_CTL, 0x00}, + {CS35L34_DIAG_MODE_REG_LOCK, 0x00}, + {CS35L34_DIAG_MODE_CTL_1, 0x00}, + {CS35L34_DIAG_MODE_CTL_2, 0x00}, + {CS35L34_INT_MASK_1, 0xFF}, + {CS35L34_INT_MASK_2, 0xFF}, + {CS35L34_INT_MASK_3, 0xFF}, + {CS35L34_INT_MASK_4, 0xFF}, + {CS35L34_INT_STATUS_1, 0x30}, + {CS35L34_INT_STATUS_2, 0x05}, + {CS35L34_INT_STATUS_3, 0x00}, + {CS35L34_INT_STATUS_4, 0x00}, + {CS35L34_OTP_TRIM_STATUS, 0x00}, +}; + +static bool cs35l34_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L34_DEVID_AB: + case CS35L34_DEVID_CD: + case CS35L34_DEVID_E: + case CS35L34_FAB_ID: + case CS35L34_REV_ID: + case CS35L34_INT_STATUS_1: + case CS35L34_INT_STATUS_2: + case CS35L34_INT_STATUS_3: + case CS35L34_INT_STATUS_4: + case CS35L34_CLASS_H_STATUS: + case CS35L34_VPBR_ATTEN_STATUS: + case CS35L34_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l34_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L34_CHIP_ID: + case CS35L34_DEVID_AB: + case CS35L34_DEVID_CD: + case CS35L34_DEVID_E: + case CS35L34_FAB_ID: + case CS35L34_REV_ID: + case CS35L34_PWRCTL1: + case CS35L34_PWRCTL2: + case CS35L34_PWRCTL3: + case CS35L34_ADSP_CLK_CTL: + case CS35L34_MCLK_CTL: + case CS35L34_AMP_INP_DRV_CTL: + case CS35L34_AMP_DIG_VOL_CTL: + case CS35L34_AMP_DIG_VOL: + case CS35L34_AMP_ANLG_GAIN_CTL: + case CS35L34_PROTECT_CTL: + case CS35L34_AMP_KEEP_ALIVE_CTL: + case CS35L34_BST_CVTR_V_CTL: + case CS35L34_BST_PEAK_I: + case CS35L34_BST_RAMP_CTL: + case CS35L34_BST_CONV_COEF_1: + case CS35L34_BST_CONV_COEF_2: + case CS35L34_BST_CONV_SLOPE_COMP: + case CS35L34_BST_CONV_SW_FREQ: + case CS35L34_CLASS_H_CTL: + case CS35L34_CLASS_H_HEADRM_CTL: + case CS35L34_CLASS_H_RELEASE_RATE: + case CS35L34_CLASS_H_FET_DRIVE_CTL: + case CS35L34_CLASS_H_STATUS: + case CS35L34_VPBR_CTL: + case CS35L34_VPBR_VOL_CTL: + case CS35L34_VPBR_TIMING_CTL: + case CS35L34_PRED_MAX_ATTEN_SPK_LOAD: + case CS35L34_PRED_BROWNOUT_THRESH: + case CS35L34_PRED_BROWNOUT_VOL_CTL: + case CS35L34_PRED_BROWNOUT_RATE_CTL: + case CS35L34_PRED_WAIT_CTL: + case CS35L34_PRED_ZVP_INIT_IMP_CTL: + case CS35L34_PRED_MAN_SAFE_VPI_CTL: + case CS35L34_VPBR_ATTEN_STATUS: + case CS35L34_PRED_BRWNOUT_ATT_STATUS: + case CS35L34_SPKR_MON_CTL: + case CS35L34_ADSP_I2S_CTL: + case CS35L34_ADSP_TDM_CTL: + case CS35L34_TDM_TX_CTL_1_VMON: + case CS35L34_TDM_TX_CTL_2_IMON: + case CS35L34_TDM_TX_CTL_3_VPMON: + case CS35L34_TDM_TX_CTL_4_VBSTMON: + case CS35L34_TDM_TX_CTL_5_FLAG1: + case CS35L34_TDM_TX_CTL_6_FLAG2: + case CS35L34_TDM_TX_SLOT_EN_1: + case CS35L34_TDM_TX_SLOT_EN_2: + case CS35L34_TDM_TX_SLOT_EN_3: + case CS35L34_TDM_TX_SLOT_EN_4: + case CS35L34_TDM_RX_CTL_1_AUDIN: + case CS35L34_TDM_RX_CTL_3_ALIVE: + case CS35L34_MULT_DEV_SYNCH1: + case CS35L34_MULT_DEV_SYNCH2: + case CS35L34_PROT_RELEASE_CTL: + case CS35L34_DIAG_MODE_REG_LOCK: + case CS35L34_DIAG_MODE_CTL_1: + case CS35L34_DIAG_MODE_CTL_2: + case CS35L34_INT_MASK_1: + case CS35L34_INT_MASK_2: + case CS35L34_INT_MASK_3: + case CS35L34_INT_MASK_4: + case CS35L34_INT_STATUS_1: + case CS35L34_INT_STATUS_2: + case CS35L34_INT_STATUS_3: + case CS35L34_INT_STATUS_4: + case CS35L34_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l34_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L34_INT_STATUS_1: + case CS35L34_INT_STATUS_2: + case CS35L34_INT_STATUS_3: + case CS35L34_INT_STATUS_4: + return true; + default: + return false; + } +} + +static int cs35l34_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->tdm_mode) + regmap_update_bits(priv->regmap, CS35L34_PWRCTL3, + CS35L34_PDN_TDM, 0x00); + + ret = regmap_update_bits(priv->regmap, CS35L34_PWRCTL1, + CS35L34_PDN_ALL, 0); + if (ret < 0) { + dev_err(codec->dev, "Cannot set Power bits %d\n", ret); + return ret; + } + usleep_range(5000, 5100); + break; + case SND_SOC_DAPM_POST_PMD: + if (priv->tdm_mode) { + regmap_update_bits(priv->regmap, CS35L34_PWRCTL3, + CS35L34_PDN_TDM, CS35L34_PDN_TDM); + } + ret = regmap_update_bits(priv->regmap, CS35L34_PWRCTL1, + CS35L34_PDN_ALL, CS35L34_PDN_ALL); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + +static int cs35l34_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; + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + priv->tdm_mode = true; + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) + snd_soc_update_bits(codec, CS35L34_TDM_RX_CTL_1_AUDIN, + CS35L34_X_LOC, slot); + + /* scan tx_mask: vmon(2 slots); imon (2 slots); vpmon (1 slot) + * vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_3_VPMON, + CS35L34_X_STATE | CS35L34_X_LOC, + CS35L34_X_STATE | CS35L34_X_LOC); + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_4_VBSTMON, + CS35L34_X_STATE | CS35L34_X_LOC, + CS35L34_X_STATE | CS35L34_X_LOC); + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_1_VMON, + CS35L34_X_STATE | CS35L34_X_LOC, slot); + + /* configure IMON_TX_LOC */ + if (slot_num == 4) { + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_2_IMON, + CS35L34_X_STATE | CS35L34_X_LOC, slot); + } + /* configure VPMON_TX_LOC */ + if (slot_num == 3) { + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_3_VPMON, + CS35L34_X_STATE | CS35L34_X_LOC, slot); + } + /* configure VBSTMON_TX_LOC */ + if (slot_num == 7) { + snd_soc_update_bits(codec, + CS35L34_TDM_TX_CTL_4_VBSTMON, + CS35L34_X_STATE | CS35L34_X_LOC, slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L34_TDM_TX_SLOT_EN_4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + snd_soc_update_bits(codec, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l34_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(priv->regmap, CS35L34_BST_CVTR_V_CTL, + CS35L34_BST_CVTL_MASK, priv->pdata.boost_vtge); + usleep_range(5000, 5100); + regmap_update_bits(priv->regmap, CS35L34_PROTECT_CTL, + CS35L34_MUTE, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L34_BST_CVTR_V_CTL, + CS35L34_BST_CVTL_MASK, 0); + regmap_update_bits(priv->regmap, CS35L34_PROTECT_CTL, + CS35L34_MUTE, CS35L34_MUTE); + usleep_range(5000, 5100); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 300, 100, 0); + + +static const struct snd_kcontrol_new cs35l34_snd_controls[] = { + SOC_SINGLE_SX_TLV("Digital Volume", CS35L34_AMP_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("Amp Gain Volume", CS35L34_AMP_ANLG_GAIN_CTL, + 0, 0xF, 0, amp_gain_tlv), +}; + + +static int cs35l34_mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + int ret, i; + unsigned int reg; + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + ret = regmap_read(priv->regmap, CS35L34_AMP_DIG_VOL_CTL, + &reg); + if (ret != 0) { + pr_err("%s regmap read failure %d\n", __func__, ret); + return ret; + } + if (reg & CS35L34_AMP_DIGSFT) + msleep(40); + else + usleep_range(2000, 2100); + + for (i = 0; i < PDN_DONE_ATTEMPTS; i++) { + ret = regmap_read(priv->regmap, CS35L34_INT_STATUS_2, + &reg); + if (ret != 0) { + pr_err("%s regmap read failure %d\n", + __func__, ret); + return ret; + } + if (reg & CS35L34_PDN_DONE) + break; + + usleep_range(5000, 5100); + } + if (i == PDN_DONE_ATTEMPTS) + pr_err("%s Device did not power down properly\n", + __func__); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget cs35l34_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L34_PWRCTL3, + 1, 1, cs35l34_sdin_event, + SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L34_PWRCTL3, 2, 1), + + SND_SOC_DAPM_SUPPLY("EXTCLK", CS35L34_PWRCTL3, 7, 1, + cs35l34_mclk_event, SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VPST"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L34_PWRCTL2, 7, 1), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L34_PWRCTL2, 6, 1), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L34_PWRCTL3, 3, 1), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L34_PWRCTL3, 4, 1), + SND_SOC_DAPM_ADC("CLASS H", NULL, CS35L34_PWRCTL2, 5, 1), + SND_SOC_DAPM_ADC("BOOST", NULL, CS35L34_PWRCTL2, 2, 1), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L34_PWRCTL2, 0, 1, NULL, 0, + cs35l34_main_amp_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cs35l34_audio_map[] = { + {"SDIN", NULL, "AMP Playback"}, + {"BOOST", NULL, "SDIN"}, + {"CLASS H", NULL, "BOOST"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, + + {"VPMON ADC", NULL, "CLASS H"}, + {"VBSTMON ADC", NULL, "CLASS H"}, + {"SPK", NULL, "VPMON ADC"}, + {"SPK", NULL, "VBSTMON ADC"}, + + {"IMON ADC", NULL, "ISENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + {"SDOUT", NULL, "IMON ADC"}, + {"SDOUT", NULL, "VMON ADC"}, + {"AMP Capture", NULL, "SDOUT"}, + + {"SDIN", NULL, "EXTCLK"}, + {"SDOUT", NULL, "EXTCLK"}, +}; + +struct cs35l34_mclk_div { + int mclk; + int srate; + u8 adsp_rate; +}; + +static struct cs35l34_mclk_div cs35l34_mclk_coeffs[] = { + + /* MCLK, Sample Rate, adsp_rate */ + + {5644800, 11025, 0x1}, + {5644800, 22050, 0x4}, + {5644800, 44100, 0x7}, + + {6000000, 8000, 0x0}, + {6000000, 11025, 0x1}, + {6000000, 12000, 0x2}, + {6000000, 16000, 0x3}, + {6000000, 22050, 0x4}, + {6000000, 24000, 0x5}, + {6000000, 32000, 0x6}, + {6000000, 44100, 0x7}, + {6000000, 48000, 0x8}, + + {6144000, 8000, 0x0}, + {6144000, 11025, 0x1}, + {6144000, 12000, 0x2}, + {6144000, 16000, 0x3}, + {6144000, 22050, 0x4}, + {6144000, 24000, 0x5}, + {6144000, 32000, 0x6}, + {6144000, 44100, 0x7}, + {6144000, 48000, 0x8}, +}; + +static int cs35l34_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l34_mclk_coeffs); i++) { + if (cs35l34_mclk_coeffs[i].mclk == mclk && + cs35l34_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l34_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L34_ADSP_CLK_CTL, + 0x80, 0x80); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L34_ADSP_CLK_CTL, + 0x80, 0x00); + break; + default: + return -EINVAL; + } + return 0; +} + +static int cs35l34_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 cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + int srate = params_rate(params); + int ret; + + int coeff = cs35l34_get_mclk_coeff(priv->mclk_int, srate); + + if (coeff < 0) { + dev_err(codec->dev, "ERROR: Invalid mclk %d and/or srate %d\n", + priv->mclk_int, srate); + return coeff; + } + + ret = regmap_update_bits(priv->regmap, CS35L34_ADSP_CLK_CTL, + CS35L34_ADSP_RATE, cs35l34_mclk_coeffs[coeff].adsp_rate); + if (ret != 0) + dev_err(codec->dev, "Failed to set clock state %d\n", ret); + + return ret; +} + +static unsigned int cs35l34_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 +}; + + +static struct snd_pcm_hw_constraint_list cs35l34_constraints = { + .count = ARRAY_SIZE(cs35l34_src_rates), + .list = cs35l34_src_rates, +}; + +static int cs35l34_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l34_constraints); + return 0; +} + + +static int cs35l34_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + + struct snd_soc_codec *codec = dai->codec; + + if (tristate) + snd_soc_update_bits(codec, CS35L34_PWRCTL3, + CS35L34_PDN_SDOUT, CS35L34_PDN_SDOUT); + else + snd_soc_update_bits(codec, CS35L34_PWRCTL3, + CS35L34_PDN_SDOUT, 0); + return 0; +} + +static int cs35l34_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l34_private *cs35l34 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (freq) { + case CS35L34_MCLK_5644: + value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_5P6448; + cs35l34->mclk_int = freq; + break; + case CS35L34_MCLK_6: + value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P0000; + cs35l34->mclk_int = freq; + break; + case CS35L34_MCLK_6144: + value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P1440; + cs35l34->mclk_int = freq; + break; + case CS35L34_MCLK_11289: + value = CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_5P6448; + cs35l34->mclk_int = freq / 2; + break; + case CS35L34_MCLK_12: + value = CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_6P0000; + cs35l34->mclk_int = freq / 2; + break; + case CS35L34_MCLK_12288: + value = CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_6P1440; + cs35l34->mclk_int = freq / 2; + break; + default: + dev_err(codec->dev, "ERROR: Invalid Frequency %d\n", freq); + cs35l34->mclk_int = 0; + return -EINVAL; + } + regmap_update_bits(cs35l34->regmap, CS35L34_MCLK_CTL, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK, value); + return 0; +} + +static const struct snd_soc_dai_ops cs35l34_ops = { + .startup = cs35l34_pcm_startup, + .set_tristate = cs35l34_set_tristate, + .set_fmt = cs35l34_set_dai_fmt, + .hw_params = cs35l34_pcm_hw_params, + .set_sysclk = cs35l34_dai_set_sysclk, + .set_tdm_slot = cs35l34_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l34_dai = { + .name = "cs35l34", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = CS35L34_RATES, + .formats = CS35L34_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = CS35L34_RATES, + .formats = CS35L34_FORMATS, + }, + .ops = &cs35l34_ops, + .symmetric_rates = 1, +}; + +static int cs35l34_boost_inductor(struct cs35l34_private *cs35l34, + unsigned int inductor) +{ + struct snd_soc_codec *codec = cs35l34->codec; + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_1, 0x24); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_2, 0x24); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SLOPE_COMP, + 0x4E); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SW_FREQ, 0); + break; + case 1200: /* 1.2 uH */ + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_1, 0x20); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_2, 0x20); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SLOPE_COMP, + 0x47); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SW_FREQ, 1); + break; + case 1500: /* 1.5uH */ + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_1, 0x20); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_2, 0x20); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SLOPE_COMP, + 0x3C); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SW_FREQ, 2); + break; + case 2200: /* 2.2uH */ + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_1, 0x19); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_2, 0x25); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SLOPE_COMP, + 0x23); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SW_FREQ, 3); + break; + default: + dev_err(codec->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + return 0; +} + +static int cs35l34_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct cs35l34_private *cs35l34 = snd_soc_codec_get_drvdata(codec); + + pm_runtime_get_sync(codec->dev); + + /* Set over temperature warning attenuation to 6 dB */ + regmap_update_bits(cs35l34->regmap, CS35L34_PROTECT_CTL, + CS35L34_OTW_ATTN_MASK, 0x8); + + /* Set Power control registers 2 and 3 to have everything + * powered down at initialization + */ + regmap_write(cs35l34->regmap, CS35L34_PWRCTL2, 0xFD); + regmap_write(cs35l34->regmap, CS35L34_PWRCTL3, 0x1F); + + /* Set mute bit at startup */ + regmap_update_bits(cs35l34->regmap, CS35L34_PROTECT_CTL, + CS35L34_MUTE, CS35L34_MUTE); + + /* Set Platform Data */ + if (cs35l34->pdata.boost_peak) + regmap_update_bits(cs35l34->regmap, CS35L34_BST_PEAK_I, + CS35L34_BST_PEAK_MASK, + cs35l34->pdata.boost_peak); + + if (cs35l34->pdata.gain_zc_disable) + regmap_update_bits(cs35l34->regmap, CS35L34_PROTECT_CTL, + CS35L34_GAIN_ZC_MASK, 0); + else + regmap_update_bits(cs35l34->regmap, CS35L34_PROTECT_CTL, + CS35L34_GAIN_ZC_MASK, CS35L34_GAIN_ZC_MASK); + + if (cs35l34->pdata.aif_half_drv) + regmap_update_bits(cs35l34->regmap, CS35L34_ADSP_CLK_CTL, + CS35L34_ADSP_DRIVE, 0); + + if (cs35l34->pdata.digsft_disable) + regmap_update_bits(cs35l34->regmap, CS35L34_AMP_DIG_VOL_CTL, + CS35L34_AMP_DIGSFT, 0); + + if (cs35l34->pdata.amp_inv) + regmap_update_bits(cs35l34->regmap, CS35L34_AMP_DIG_VOL_CTL, + CS35L34_INV, CS35L34_INV); + + if (cs35l34->pdata.boost_ind) + ret = cs35l34_boost_inductor(cs35l34, cs35l34->pdata.boost_ind); + + if (cs35l34->pdata.i2s_sdinloc) + regmap_update_bits(cs35l34->regmap, CS35L34_ADSP_I2S_CTL, + CS35L34_I2S_LOC_MASK, + cs35l34->pdata.i2s_sdinloc << CS35L34_I2S_LOC_SHIFT); + + if (cs35l34->pdata.tdm_rising_edge) + regmap_update_bits(cs35l34->regmap, CS35L34_ADSP_TDM_CTL, + 1, 1); + + pm_runtime_put_sync(codec->dev); + + return ret; +} + + +static struct snd_soc_codec_driver soc_codec_dev_cs35l34 = { + .probe = cs35l34_probe, + + .component_driver = { + .dapm_widgets = cs35l34_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l34_dapm_widgets), + .dapm_routes = cs35l34_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l34_audio_map), + .controls = cs35l34_snd_controls, + .num_controls = ARRAY_SIZE(cs35l34_snd_controls), + }, +}; + +static struct regmap_config cs35l34_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L34_MAX_REGISTER, + .reg_defaults = cs35l34_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l34_reg), + .volatile_reg = cs35l34_volatile_register, + .readable_reg = cs35l34_readable_register, + .precious_reg = cs35l34_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs35l34_handle_of_data(struct i2c_client *i2c_client, + struct cs35l34_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + + if (of_property_read_u32(np, "cirrus,boost-vtge-millivolt", + &val) >= 0) { + /* Boost Voltage has a maximum of 8V */ + if (val > 8000 || (val < 3300 && val > 0)) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + if (val == 0) + pdata->boost_vtge = 0; /* Use VP */ + else + pdata->boost_vtge = ((val - 3300)/100) + 1; + } else { + dev_warn(&i2c_client->dev, + "Boost Voltage not specified. Using VP\n"); + } + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val) >= 0) { + if (val > 3840 || val < 1200) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %d mA\n", val); + return -EINVAL; + } + pdata->boost_peak = ((val - 1200)/80) + 1; + } + + pdata->aif_half_drv = of_property_read_bool(np, + "cirrus,aif-half-drv"); + pdata->digsft_disable = of_property_read_bool(np, + "cirrus,digsft-disable"); + + pdata->gain_zc_disable = of_property_read_bool(np, + "cirrus,gain-zc-disable"); + pdata->amp_inv = of_property_read_bool(np, "cirrus,amp-inv"); + + if (of_property_read_u32(np, "cirrus,i2s-sdinloc", &val) >= 0) + pdata->i2s_sdinloc = val; + if (of_property_read_u32(np, "cirrus,tdm-rising-edge", &val) >= 0) + pdata->tdm_rising_edge = val; + + return 0; +} + +static irqreturn_t cs35l34_irq_thread(int irq, void *data) +{ + struct cs35l34_private *cs35l34 = data; + struct snd_soc_codec *codec = cs35l34->codec; + unsigned int sticky1, sticky2, sticky3, sticky4; + unsigned int mask1, mask2, mask3, mask4, current1; + + + /* ack the irq by reading all status registers */ + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_4, &sticky4); + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_3, &sticky3); + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_2, &sticky2); + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_1, &sticky1); + + regmap_read(cs35l34->regmap, CS35L34_INT_MASK_4, &mask4); + regmap_read(cs35l34->regmap, CS35L34_INT_MASK_3, &mask3); + regmap_read(cs35l34->regmap, CS35L34_INT_MASK_2, &mask2); + regmap_read(cs35l34->regmap, CS35L34_INT_MASK_1, &mask1); + + if (!(sticky1 & ~mask1) && !(sticky2 & ~mask2) && !(sticky3 & ~mask3) + && !(sticky4 & ~mask4)) + return IRQ_NONE; + + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_1, &current1); + + if (sticky1 & CS35L34_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L34_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_CAL_ERR_RLS, 0); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_CAL_ERR_RLS, + CS35L34_CAL_ERR_RLS); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_CAL_ERR_RLS, 0); + /* note: amp will re-calibrate on next resume */ + } + } + + if (sticky1 & CS35L34_ALIVE_ERR) + dev_err(codec->dev, "Alive error\n"); + + if (sticky1 & CS35L34_AMP_SHORT) { + dev_crit(codec->dev, "Amp short error\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L34_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_SHORT_RLS, 0); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_SHORT_RLS, + CS35L34_SHORT_RLS); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_SHORT_RLS, 0); + } + } + + if (sticky1 & CS35L34_OTW) { + dev_crit(codec->dev, "Over temperature warning\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L34_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTW_RLS, 0); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTW_RLS, + CS35L34_OTW_RLS); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTW_RLS, 0); + } + } + + if (sticky1 & CS35L34_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L34_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTE_RLS, 0); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTE_RLS, + CS35L34_OTE_RLS); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTE_RLS, 0); + } + } + + if (sticky3 & CS35L34_BST_HIGH) { + dev_crit(codec->dev, "VBST too high error; powering off!\n"); + regmap_update_bits(cs35l34->regmap, CS35L34_PWRCTL2, + CS35L34_PDN_AMP, CS35L34_PDN_AMP); + regmap_update_bits(cs35l34->regmap, CS35L34_PWRCTL1, + CS35L34_PDN_ALL, CS35L34_PDN_ALL); + } + + if (sticky3 & CS35L34_LBST_SHORT) { + dev_crit(codec->dev, "LBST short error; powering off!\n"); + regmap_update_bits(cs35l34->regmap, CS35L34_PWRCTL2, + CS35L34_PDN_AMP, CS35L34_PDN_AMP); + regmap_update_bits(cs35l34->regmap, CS35L34_PWRCTL1, + CS35L34_PDN_ALL, CS35L34_PDN_ALL); + } + + return IRQ_HANDLED; +} + +static const char * const cs35l34_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l34_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l34_private *cs35l34; + struct cs35l34_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int i; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs35l34 = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l34_private), + GFP_KERNEL); + if (!cs35l34) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs35l34); + cs35l34->regmap = devm_regmap_init_i2c(i2c_client, &cs35l34_regmap); + if (IS_ERR(cs35l34->regmap)) { + ret = PTR_ERR(cs35l34->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + cs35l34->num_core_supplies = ARRAY_SIZE(cs35l34_core_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l34_core_supplies); i++) + cs35l34->core_supplies[i].supply = cs35l34_core_supplies[i]; + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l34->num_core_supplies, + cs35l34->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l34->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l34_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 = cs35l34_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + cs35l34->pdata = *pdata; + } + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l34_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l34", cs35l34); + if (ret != 0) + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + cs35l34->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset-gpios", GPIOD_OUT_LOW); + if (IS_ERR(cs35l34->reset_gpio)) + return PTR_ERR(cs35l34->reset_gpio); + + gpiod_set_value_cansleep(cs35l34->reset_gpio, 1); + + msleep(CS35L34_START_DELAY); + + ret = regmap_read(cs35l34->regmap, CS35L34_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l34->regmap, CS35L34_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l34->regmap, CS35L34_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L34_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35l34 Device ID (%X). Expected ID %X\n", + devid, CS35L34_CHIP_ID); + ret = -ENODEV; + goto err_regulator; + } + + ret = regmap_read(cs35l34->regmap, CS35L34_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_regulator; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35l34 (%x), Revision: %02X\n", devid, + reg & 0xFF); + + /* Unmask critical interrupts */ + regmap_update_bits(cs35l34->regmap, CS35L34_INT_MASK_1, + CS35L34_M_CAL_ERR | CS35L34_M_ALIVE_ERR | + CS35L34_M_AMP_SHORT | CS35L34_M_OTW | + CS35L34_M_OTE, 0); + regmap_update_bits(cs35l34->regmap, CS35L34_INT_MASK_3, + CS35L34_M_BST_HIGH | CS35L34_M_LBST_SHORT, 0); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l34, &cs35l34_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, + "%s: Register codec failed\n", __func__); + goto err_regulator; + } + + return 0; + +err_regulator: + regulator_bulk_disable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + return ret; +} + +static int cs35l34_i2c_remove(struct i2c_client *client) +{ + struct cs35l34_private *cs35l34 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l34->reset_gpio) + gpiod_set_value_cansleep(cs35l34->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + return 0; +} + +static int __maybe_unused cs35l34_runtime_resume(struct device *dev) +{ + struct cs35l34_private *cs35l34 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", + ret); + return ret; + } + + regcache_cache_only(cs35l34->regmap, false); + + gpiod_set_value_cansleep(cs35l34->reset_gpio, 1); + msleep(CS35L34_START_DELAY); + + ret = regcache_sync(cs35l34->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + return 0; +err: + regcache_cache_only(cs35l34->regmap, true); + regulator_bulk_disable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + return ret; +} + +static int __maybe_unused cs35l34_runtime_suspend(struct device *dev) +{ + struct cs35l34_private *cs35l34 = dev_get_drvdata(dev); + + regcache_cache_only(cs35l34->regmap, true); + regcache_mark_dirty(cs35l34->regmap); + + gpiod_set_value_cansleep(cs35l34->reset_gpio, 0); + + regulator_bulk_disable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l34_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l34_runtime_suspend, + cs35l34_runtime_resume, + NULL) +}; + +static const struct of_device_id cs35l34_of_match[] = { + {.compatible = "cirrus,cs35l34"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l34_of_match); + +static const struct i2c_device_id cs35l34_id[] = { + {"cs35l34", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs35l34_id); + +static struct i2c_driver cs35l34_i2c_driver = { + .driver = { + .name = "cs35l34", + .pm = &cs35l34_pm_ops, + .of_match_table = cs35l34_of_match, + + }, + .id_table = cs35l34_id, + .probe = cs35l34_i2c_probe, + .remove = cs35l34_i2c_remove, + +}; + +static int __init cs35l34_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&cs35l34_i2c_driver); + if (ret != 0) { + pr_err("Failed to register CS35l34 I2C driver: %d\n", ret); + return ret; + } + return 0; +} +module_init(cs35l34_modinit); + +static void __exit cs35l34_exit(void) +{ + i2c_del_driver(&cs35l34_i2c_driver); +} +module_exit(cs35l34_exit); + +MODULE_DESCRIPTION("ASoC CS35l34 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l34.h b/sound/soc/codecs/cs35l34.h new file mode 100644 index 0000000..bcd54f1 --- /dev/null +++ b/sound/soc/codecs/cs35l34.h @@ -0,0 +1,269 @@ +/* + * cs35l34.h -- CS35L34 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(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 __CS35L34_H__ +#define __CS35L34_H__ + +#define CS35L34_CHIP_ID 0x00035A34 +#define CS35L34_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L34_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L34_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L34_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L34_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L34_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L34_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L34_PWRCTL3 0x08 /* Power Ctl 3 */ +#define CS35L34_ADSP_CLK_CTL 0x0A /* (ADSP) Clock Ctl */ +#define CS35L34_MCLK_CTL 0x0B /* Master Clocking Ctl */ +#define CS35L34_AMP_INP_DRV_CTL 0x14 /* Amp Input Drive Ctl */ +#define CS35L34_AMP_DIG_VOL_CTL 0x15 /* Amplifier Dig Volume Ctl */ +#define CS35L34_AMP_DIG_VOL 0x16 /* Amplifier Dig Volume */ +#define CS35L34_AMP_ANLG_GAIN_CTL 0x17 /* Amplifier Analog Gain Ctl */ +#define CS35L34_PROTECT_CTL 0x18 /* Amp Gain - Prot Ctl Param */ +#define CS35L34_AMP_KEEP_ALIVE_CTL 0x1A /* Amplifier Keep Alive Ctl */ +#define CS35L34_BST_CVTR_V_CTL 0x1D /* Boost Conv Voltage Ctl */ +#define CS35L34_BST_PEAK_I 0x1E /* Boost Conv Peak Current */ +#define CS35L34_BST_RAMP_CTL 0x20 /* Boost Conv Soft Ramp Ctl */ +#define CS35L34_BST_CONV_COEF_1 0x21 /* Boost Conv Coefficients 1 */ +#define CS35L34_BST_CONV_COEF_2 0x22 /* Boost Conv Coefficients 2 */ +#define CS35L34_BST_CONV_SLOPE_COMP 0x23 /* Boost Conv Slope Comp */ +#define CS35L34_BST_CONV_SW_FREQ 0x24 /* Boost Conv L BST SW Freq */ +#define CS35L34_CLASS_H_CTL 0x30 /* CLS H Control */ +#define CS35L34_CLASS_H_HEADRM_CTL 0x31 /* CLS H Headroom Ctl */ +#define CS35L34_CLASS_H_RELEASE_RATE 0x32 /* CLS H Release Rate */ +#define CS35L34_CLASS_H_FET_DRIVE_CTL 0x33 /* CLS H Weak FET Drive Ctl */ +#define CS35L34_CLASS_H_STATUS 0x38 /* CLS H Status */ +#define CS35L34_VPBR_CTL 0x3A /* VPBR Ctl */ +#define CS35L34_VPBR_VOL_CTL 0x3B /* VPBR Volume Ctl */ +#define CS35L34_VPBR_TIMING_CTL 0x3C /* VPBR Timing Ctl */ +#define CS35L34_PRED_MAX_ATTEN_SPK_LOAD 0x40 /* PRD Max Atten / Spkr Load */ +#define CS35L34_PRED_BROWNOUT_THRESH 0x41 /* PRD Brownout Threshold */ +#define CS35L34_PRED_BROWNOUT_VOL_CTL 0x42 /* PRD Brownout Volume Ctl */ +#define CS35L34_PRED_BROWNOUT_RATE_CTL 0x43 /* PRD Brownout Rate Ctl */ +#define CS35L34_PRED_WAIT_CTL 0x44 /* PRD Wait Ctl */ +#define CS35L34_PRED_ZVP_INIT_IMP_CTL 0x46 /* PRD ZVP Initial Imp Ctl */ +#define CS35L34_PRED_MAN_SAFE_VPI_CTL 0x47 /* PRD Manual Safe VPI Ctl */ +#define CS35L34_VPBR_ATTEN_STATUS 0x4B /* VPBR Attenuation Status */ +#define CS35L34_PRED_BRWNOUT_ATT_STATUS 0x4C /* PRD Brownout Atten Status */ +#define CS35L34_SPKR_MON_CTL 0x4E /* Speaker Monitoring Ctl */ +#define CS35L34_ADSP_I2S_CTL 0x50 /* ADSP I2S Ctl */ +#define CS35L34_ADSP_TDM_CTL 0x51 /* ADSP TDM Ctl */ +#define CS35L34_TDM_TX_CTL_1_VMON 0x52 /* TDM TX Ctl 1 (VMON) */ +#define CS35L34_TDM_TX_CTL_2_IMON 0x53 /* TDM TX Ctl 2 (IMON) */ +#define CS35L34_TDM_TX_CTL_3_VPMON 0x54 /* TDM TX Ctl 3 (VPMON) */ +#define CS35L34_TDM_TX_CTL_4_VBSTMON 0x55 /* TDM TX Ctl 4 (VBSTMON) */ +#define CS35L34_TDM_TX_CTL_5_FLAG1 0x56 /* TDM TX Ctl 5 (FLAG1) */ +#define CS35L34_TDM_TX_CTL_6_FLAG2 0x57 /* TDM TX Ctl 6 (FLAG2) */ +#define CS35L34_TDM_TX_SLOT_EN_1 0x5A /* TDM TX Slot Enable */ +#define CS35L34_TDM_TX_SLOT_EN_2 0x5B /* TDM TX Slot Enable */ +#define CS35L34_TDM_TX_SLOT_EN_3 0x5C /* TDM TX Slot Enable */ +#define CS35L34_TDM_TX_SLOT_EN_4 0x5D /* TDM TX Slot Enable */ +#define CS35L34_TDM_RX_CTL_1_AUDIN 0x5E /* TDM RX Ctl 1 */ +#define CS35L34_TDM_RX_CTL_3_ALIVE 0x60 /* TDM RX Ctl 3 (ALIVE) */ +#define CS35L34_MULT_DEV_SYNCH1 0x62 /* Multidevice Synch */ +#define CS35L34_MULT_DEV_SYNCH2 0x63 /* Multidevice Synch 2 */ +#define CS35L34_PROT_RELEASE_CTL 0x64 /* Protection Release Ctl */ +#define CS35L34_DIAG_MODE_REG_LOCK 0x68 /* Diagnostic Mode Reg Lock */ +#define CS35L34_DIAG_MODE_CTL_1 0x69 /* Diagnostic Mode Ctl 1 */ +#define CS35L34_DIAG_MODE_CTL_2 0x6A /* Diagnostic Mode Ctl 2 */ +#define CS35L34_INT_MASK_1 0x70 /* Interrupt Mask 1 */ +#define CS35L34_INT_MASK_2 0x71 /* Interrupt Mask 2 */ +#define CS35L34_INT_MASK_3 0x72 /* Interrupt Mask 3 */ +#define CS35L34_INT_MASK_4 0x73 /* Interrupt Mask 4 */ +#define CS35L34_INT_STATUS_1 0x74 /* Interrupt Status 1 */ +#define CS35L34_INT_STATUS_2 0x75 /* Interrupt Status 2 */ +#define CS35L34_INT_STATUS_3 0x76 /* Interrupt Status 3 */ +#define CS35L34_INT_STATUS_4 0x77 /* Interrupt Status 4 */ +#define CS35L34_OTP_TRIM_STATUS 0x7E /* OTP Trim Status */ + +#define CS35L34_MAX_REGISTER 0x7F +#define CS35L34_REGISTER_COUNT 0x4E + +#define CS35L34_MCLK_5644 5644800 +#define CS35L34_MCLK_6144 6144000 +#define CS35L34_MCLK_6 6000000 +#define CS35L34_MCLK_11289 11289600 +#define CS35L34_MCLK_12 12000000 +#define CS35L34_MCLK_12288 12288000 + +/* CS35L34_PWRCTL1 */ +#define CS35L34_SFT_RST (1 << 7) +#define CS35L34_DISCHG_FLT (1 << 1) +#define CS35L34_PDN_ALL 1 + +/* CS35L34_PWRCTL2 */ +#define CS35L34_PDN_VMON (1 << 7) +#define CS35L34_PDN_IMON (1 << 6) +#define CS35L34_PDN_CLASSH (1 << 5) +#define CS35L34_PDN_VPBR (1 << 4) +#define CS35L34_PDN_PRED (1 << 3) +#define CS35L34_PDN_BST (1 << 2) +#define CS35L34_PDN_AMP 1 + +/* CS35L34_PWRCTL3 */ +#define CS35L34_MCLK_DIS (1 << 7) +#define CS35L34_PDN_VBSTMON_OUT (1 << 4) +#define CS35L34_PDN_VMON_OUT (1 << 3) +/* Tristate the ADSP SDOUT when in I2C mode */ +#define CS35L34_PDN_SDOUT (1 << 2) +#define CS35L34_PDN_SDIN (1 << 1) +#define CS35L34_PDN_TDM 1 + +/* CS35L34_ADSP_CLK_CTL */ +#define CS35L34_ADSP_RATE 0xF +#define CS35L34_ADSP_DRIVE (1 << 4) +#define CS35L34_ADSP_M_S (1 << 7) + +/* CS35L34_MCLK_CTL */ +#define CS35L34_MCLK_DIV (1 << 4) +#define CS35L34_MCLK_RATE_MASK 0x7 +#define CS35L34_MCLK_RATE_6P1440 0x2 +#define CS35L34_MCLK_RATE_6P0000 0x1 +#define CS35L34_MCLK_RATE_5P6448 0x0 +#define CS35L34_MCLKDIS (1 << 7) +#define CS35L34_MCLKDIV2 (1 << 6) +#define CS35L34_SDOUT_3ST_TDM (1 << 5) +#define CS35L34_INT_FS_RATE (1 << 4) +#define CS35L34_ADSP_FS 0xF + +/* CS35L34_AMP_INP_DRV_CTL */ +#define CS35L34_DRV_STR_SRC (1 << 1) +#define CS35L34_DRV_STR 1 + +/* CS35L34_AMP_DIG_VOL_CTL */ +#define CS35L34_AMP_DSR_RATE_MASK 0xF0 +#define CS35L34_AMP_DSR_RATE_SHIFT (1 << 4) +#define CS35L34_NOTCH_DIS (1 << 3) +#define CS35L34_AMP_DIGSFT (1 << 1) +#define CS35L34_INV 1 + +/* CS35L34_PROTECT_CTL */ +#define CS35L34_OTW_ATTN_MASK 0xC +#define CS35L34_OTW_THRD_MASK 0x3 +#define CS35L34_MUTE (1 << 5) +#define CS35L34_GAIN_ZC (1 << 4) +#define CS35L34_GAIN_ZC_MASK 0x10 +#define CS35L34_GAIN_ZC_SHIFT 4 + +/* CS35L34_AMP_KEEP_ALIVE_CTL */ +#define CS35L34_ALIVE_WD_DIS (1 << 2) + +/* CS35L34_BST_CVTR_V_CTL */ +#define CS35L34_BST_CVTL_MASK 0x3F + +/* CS35L34_BST_PEAK_I */ +#define CS35L34_BST_PEAK_MASK 0x3F + +/* CS35L34_ADSP_I2S_CTL */ +#define CS35L34_I2S_LOC_MASK 0xC +#define CS35L34_I2S_LOC_SHIFT 2 + +/* CS35L34_MULT_DEV_SYNCH2 */ +#define CS35L34_SYNC2_MASK 0xF + +/* CS35L34_PROT_RELEASE_CTL */ +#define CS35L34_CAL_ERR_RLS (1 << 7) +#define CS35L34_SHORT_RLS (1 << 2) +#define CS35L34_OTW_RLS (1 << 1) +#define CS35L34_OTE_RLS 1 + +/* CS35L34_INT_MASK_1 */ +#define CS35L34_M_CAL_ERR_SHIFT 7 +#define CS35L34_M_CAL_ERR (1 << CS35L34_M_CAL_ERR_SHIFT) +#define CS35L34_M_ALIVE_ERR_SHIFT 5 +#define CS35L34_M_ALIVE_ERR (1 << CS35L34_M_ALIVE_ERR_SHIFT) +#define CS35L34_M_ADSP_CLK_SHIFT 4 +#define CS35L34_M_ADSP_CLK_ERR (1 << CS35L34_M_ADSP_CLK_SHIFT) +#define CS35L34_M_MCLK_SHIFT 3 +#define CS35L34_M_MCLK_ERR (1 << CS35L34_M_MCLK_SHIFT) +#define CS35L34_M_AMP_SHORT_SHIFT 2 +#define CS35L34_M_AMP_SHORT (1 << CS35L34_M_AMP_SHORT_SHIFT) +#define CS35L34_M_OTW_SHIFT 1 +#define CS35L34_M_OTW (1 << CS35L34_M_OTW_SHIFT) +#define CS35L34_M_OTE_SHIFT 0 +#define CS35L34_M_OTE (1 << CS35L34_M_OTE_SHIFT) + +/* CS35L34_INT_MASK_2 */ +#define CS35L34_M_PDN_DONE_SHIFT 4 +#define CS35L34_M_PDN_DONE (1 << CS35L34_M_PDN_DONE_SHIFT) +#define CS35L34_M_PRED_SHIFT 3 +#define CS35L34_M_PRED_ERR (1 << CS35L34_M_PRED_SHIFT) +#define CS35L34_M_PRED_CLR_SHIFT 2 +#define CS35L34_M_PRED_CLR (1 << CS35L34_M_PRED_CLR_SHIFT) +#define CS35L34_M_VPBR_SHIFT 1 +#define CS35L34_M_VPBR_ERR (1 << CS35L34_M_VPBR_SHIFT) +#define CS35L34_M_VPBR_CLR_SHIFT 0 +#define CS35L34_M_VPBR_CLR (1 << CS35L34_M_VPBR_CLR_SHIFT) + +/* CS35L34_INT_MASK_3 */ +#define CS35L34_M_BST_HIGH_SHIFT 4 +#define CS35L34_M_BST_HIGH (1 << CS35L34_M_BST_HIGH_SHIFT) +#define CS35L34_M_BST_HIGH_FLAG_SHIFT 3 +#define CS35L34_M_BST_HIGH_FLAG (1 << CS35L34_M_BST_HIGH_FLAG_SHIFT) +#define CS35L34_M_BST_IPK_FLAG_SHIFT 2 +#define CS35L34_M_BST_IPK_FLAG (1 << CS35L34_M_BST_IPK_FLAG_SHIFT) +#define CS35L34_M_LBST_SHORT_SHIFT 0 +#define CS35L34_M_LBST_SHORT (1 << CS35L34_M_LBST_SHORT_SHIFT) + +/* CS35L34_INT_MASK_4 */ +#define CS35L34_M_VMON_OVFL_SHIFT 3 +#define CS35L34_M_VMON_OVFL (1 << CS35L34_M_VMON_OVFL_SHIFT) +#define CS35L34_M_IMON_OVFL_SHIFT 2 +#define CS35L34_M_IMON_OVFL (1 << CS35L34_M_IMON_OVFL_SHIFT) +#define CS35L34_M_VPMON_OVFL_SHIFT 1 +#define CS35L34_M_VPMON_OVFL (1 << CS35L34_M_VPMON_OVFL_SHIFT) +#define CS35L34_M_VBSTMON_OVFL_SHIFT 1 +#define CS35L34_M_VBSTMON_OVFL (1 << CS35L34_M_VBSTMON_OVFL_SHIFT) + +/* CS35L34_INT_1 */ +#define CS35L34_CAL_ERR (1 << CS35L34_M_CAL_ERR_SHIFT) +#define CS35L34_ALIVE_ERR (1 << CS35L34_M_ALIVE_ERR_SHIFT) +#define CS35L34_M_ADSP_CLK_ERR (1 << CS35L34_M_ADSP_CLK_SHIFT) +#define CS35L34_MCLK_ERR (1 << CS35L34_M_MCLK_SHIFT) +#define CS35L34_AMP_SHORT (1 << CS35L34_M_AMP_SHORT_SHIFT) +#define CS35L34_OTW (1 << CS35L34_M_OTW_SHIFT) +#define CS35L34_OTE (1 << CS35L34_M_OTE_SHIFT) + +/* CS35L34_INT_2 */ +#define CS35L34_PDN_DONE (1 << CS35L34_M_PDN_DONE_SHIFT) +#define CS35L34_PRED_ERR (1 << CS35L34_M_PRED_SHIFT) +#define CS35L34_PRED_CLR (1 << CS35L34_M_PRED_CLR_SHIFT) +#define CS35L34_VPBR_ERR (1 << CS35L34_M_VPBR_SHIFT) +#define CS35L34_VPBR_CLR (1 << CS35L34_M_VPBR_CLR_SHIFT) + +/* CS35L34_INT_3 */ +#define CS35L34_BST_HIGH (1 << CS35L34_M_BST_HIGH_SHIFT) +#define CS35L34_BST_HIGH_FLAG (1 << CS35L34_M_BST_HIGH_FLAG_SHIFT) +#define CS35L34_BST_IPK_FLAG (1 << CS35L34_M_BST_IPK_FLAG_SHIFT) +#define CS35L34_LBST_SHORT (1 << CS35L34_M_LBST_SHORT_SHIFT) + +/* CS35L34_INT_4 */ +#define CS35L34_VMON_OVFL (1 << CS35L34_M_VMON_OVFL_SHIFT) +#define CS35L34_IMON_OVFL (1 << CS35L34_M_IMON_OVFL_SHIFT) +#define CS35L34_VPMON_OVFL (1 << CS35L34_M_VPMON_OVFL_SHIFT) +#define CS35L34_VBSTMON_OVFL (1 << CS35L34_M_VBSTMON_OVFL_SHIFT) + +/* CS35L34_{RX,TX}_X */ +#define CS35L34_X_STATE_SHIFT 7 +#define CS35L34_X_STATE (1 << CS35L34_X_STATE_SHIFT) +#define CS35L34_X_LOC_SHIFT 0 +#define CS35L34_X_LOC (0x1F << CS35L34_X_LOC_SHIFT) + +#define CS35L34_RATES (SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_32000) +#define CS35L34_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#endif -- 1.9.1

2 1

Re: [alsa-devel] Clicking issues with the output of an Onyx Blackbird
by Takashi Sakamoto 13 Oct '16

13 Oct '16

Hi, (CCed to alsa-devel) I'm an ALSA developer currently commits to drivers for Audio and Music units on IEEE 1394 bus. On Oct 13 2016 04:15, Jordi Torres wrote: > I have an Onyx Blackbird that uses a firewire DICE II chip. This > device works correctly using a Power Mac G5 with Mac OS X 10.5 but > using another computer with linux, all works but there is a clicking > issue in the output of the device. > > I tested the input and works correctly without the clicking. I tried > use the alsa stack without jackd and it has the same issue. I tried vary > a lot of configurations and I found that clicking sound varies when > the sample rate is different > > ffado-diag: http://pastebin.com/u7g93GqD > ffado-test Discover: http://pastebin.com/qBMQwWMq > > I provide samples of the clicking sound in the next file: > https://drive.google.com/file/d/0Bz44Dr5G2FeJOEdCT21sLUFFWlU/view?usp=shari… > These samples are Audacity projects where 1st track is the original > sound and 2nd track is the Blackbird ouput sound that was recorded in > a Focusrite Saffire PRO I/O 10 using input line > The spectrum shows regular patterns that could be useful to the issue > > If anyone needs more info, please don't hesitate to ask Current ALSA dice driver has an issue of packet timestamping, which causes periodical noises. This issue comes from quirks of packet sequence to/from Dice chipset. In detail, see: [alsa-devel] Dice packet sequence quirk and ALSA firewire stack in Linux 4.6 http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107715.html I know this issue is potentially in current FFADO implementation, too. Well, could I request you to gather information of packet sequence? In Linux 4.7, ALSA firewire stack get tracepoints. Via the tracepoints, we can gather enough information to diagnose the sequence from user space. To gather the information, please use trace-cmd (or perf) command, like: $ sudo trace-cmd list | grep fire snd_firewire_lib:in_packet snd_firewire_lib:out_packet $ sudo trace-cmd record -e snd_firewire_lib:in_packet -e snd_firewire_lib:out_packet Then execute ALSA applications. When terminate trace-cmd, you can get trace.dat in your current working directory. Please send it to me via some better ways (it may be too big to attach e-mail). When read the content of data, run: $ trace-cmd report trace.dat In detail, read: http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/fi… Regards Takashi Sakamoto

1 0

[alsa-devel] [patch] ALSA: line6: fix a crash in line6_hwdep_write()
by Dan Carpenter 12 Oct '16

12 Oct '16

The error checking here is messed up so we could end up dereferencing -EFAULT. Fixes: a16039cbf1a1 ('ALSA: line6: Add hwdep interface to access the POD control messages') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 14e587e..90009c0 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -604,8 +604,8 @@ line6_hwdep_write(struct snd_hwdep *hwdep, const char __user *data, long count, } data_copy = memdup_user(data, count); - if (IS_ERR(ERR_PTR)) - return -ENOMEM; + if (IS_ERR(data_copy)) + return PTR_ERR(data_copy); rv = line6_send_raw_message(line6, data_copy, count);

3 2

[alsa-devel] [PATCH] ALSA: seq: fix passing wrong pointer in function call of compatibility layer
by Takashi Sakamoto 12 Oct '16

12 Oct '16

This commit is a fix for Linux 4.9-rc1. In former commit, a function call of compatibility layer for ALSA sequencer core was obsoleted by an alternative. Although, the alternative gets a pointer to kernel stack due to mis-programming. As a result, ALSA sequencer core unexpectedly refers over kernel stack. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 8ce8eb601c71 ("ALSA: seq: add an alternative way to handle ioctl requests") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/seq/seq_compat.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_compat.c b/sound/core/seq/seq_compat.c index fce5697..8c35072 100644 --- a/sound/core/seq/seq_compat.c +++ b/sound/core/seq/seq_compat.c @@ -58,7 +58,7 @@ static int snd_seq_call_port_info_ioctl(struct snd_seq_client *client, unsigned goto error; data->kernel = NULL; - err = snd_seq_kernel_client_ctl(client->number, cmd, &data); + err = snd_seq_kernel_client_ctl(client->number, cmd, data); if (err < 0) goto error; -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix a failure of micmute led when having multi adcs
by Hui Wang 12 Oct '16

12 Oct '16

On a Dell laptop, there is no global adcs for all input devices, so the input devices use the different adc, as a result, dyn_adc_switch is set to true. In this situation, it is safe to control the micmute led according to user's choice of muting/unmuting the current input device, since only current input device path is active, while other input device pathes are inactive and powered down. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- You can find the alsa-info.txt for this machine from: http://paste.ubuntu.com/23306165/ sound/pci/hda/dell_wmi_helper.c | 2 +- sound/pci/hda/thinkpad_helper.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 9c22f95..19d41da 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -49,7 +49,7 @@ static void alc_fixup_dell_wmi(struct hda_codec *codec, removefunc = true; if (dell_led_set_func(DELL_LED_MICMUTE, false) >= 0) { dell_led_value = 0; - if (spec->gen.num_adc_nids > 1) + if (spec->gen.num_adc_nids > 1 && !spec->gen.dyn_adc_switch) codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); else { dell_old_cap_hook = spec->gen.cap_sync_hook; diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index f0955fd..6a23302 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -62,7 +62,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = false; } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->num_adc_nids > 1) + if (spec->num_adc_nids > 1 && !spec->dyn_adc_switch) codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); else { -- 1.9.1

2 1

[alsa-devel] More than 32-channels on arecord?
by Rob Nertney 12 Oct '16

12 Oct '16

The manpages says 32 channels max, but I was curious as to where this limitation will begin to affect things if I go higher than 32. Thanks, Rob

1 0

[alsa-devel] [bug report] ALSA: seq: obsolete change of address limit
by Dan Carpenter 12 Oct '16

12 Oct '16

Hello Takashi Sakamoto, The patch e12ec251e4db: "ALSA: seq: obsolete change of address limit" from Aug 13, 2016, leads to the following static checker warning: sound/core/seq/seq_compat.c:61 snd_seq_call_port_info_ioctl() warn: did you mean to pass the address of 'data' sound/core/seq/seq_compat.c 45 static int snd_seq_call_port_info_ioctl(struct snd_seq_client *client, unsigned int cmd, 46 struct snd_seq_port_info32 __user *data32) 47 { 48 int err = -EFAULT; 49 struct snd_seq_port_info *data; 50 51 data = kmalloc(sizeof(*data), GFP_KERNEL); 52 if (!data) 53 return -ENOMEM; 54 55 if (copy_from_user(data, data32, sizeof(*data32)) || 56 get_user(data->flags, &data32->flags) || 57 get_user(data->time_queue, &data32->time_queue)) 58 goto error; 59 data->kernel = NULL; 60 61 err = snd_seq_kernel_client_ctl(client->number, cmd, &data); This should almost certainly be "cmd, data);" without the &. Have you tested this? It eventually gets passed to functions like snd_seq_ioctl_create_port(). 62 if (err < 0) 63 goto error; 64 65 if (copy_to_user(data32, data, sizeof(*data32)) || 66 put_user(data->flags, &data32->flags) || 67 put_user(data->time_queue, &data32->time_queue)) 68 err = -EFAULT; 69 70 error: 71 kfree(data); 72 return err; 73 } regards, dan carpenter

2 1

[alsa-devel] [PATCH] ASoC: core: allow control index different from 0
by Arnaud Pouliquen 11 Oct '16

11 Oct '16

To differentiate control with same name only device field can be used. But some applications like iecset use control index to differentiate controls linked to several PCM devices or DAIs. This patch suppress index overwriting to allow to use control index field. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4afa8db..39bc1a9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2175,7 +2175,6 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, char *name = NULL; memcpy(&template, _template, sizeof(template)); - template.index = 0; if (!long_name) long_name = template.name; -- 1.9.1

3 10

[alsa-devel] [PATCH v2] ASoC: cs4270: fix DAPM stream name mismatch
by murray foster 11 Oct '16

11 Oct '16

Mismatching stream names in DAPM route and widget definitions are causing compilation errors. Fixing these names allows the cs4270 driver to compile and function. Signed-off-by: Murray Foster <mrafoster(a)gmail.com> Acked-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- sound/soc/codecs/cs4270.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e07807d..3670086 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -148,11 +148,11 @@ SND_SOC_DAPM_OUTPUT("AOUTR"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { - { "Capture", NULL, "AINA" }, - { "Capture", NULL, "AINB" }, + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, - { "AOUTA", NULL, "Playback" }, - { "AOUTB", NULL, "Playback" }, + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, }; /** -- 2.4.9

2 1

[alsa-devel] [PATCH] ASoC: topology: Reenable use from userspace
by Mark Brown 11 Oct '16

11 Oct '16

We had inserted a #error into the topology UAPI code to ensure that the ABI was not adopted by userspace while final review and testing was ongoing. Since we are now comfortable that the ABI is suitable for production remove the #error, future changes will include provision for backwards compatibility. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Are we OK with this? Looking for some acks and reviews... include/uapi/sound/asoc.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 247c50b..51b8066 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -18,12 +18,6 @@ #include <linux/types.h> #include <sound/asound.h> -#ifndef __KERNEL__ -#error This API is an early revision and not enabled in the current -#error kernel release, it will be enabled in a future kernel version -#error with incompatible changes to what is here. -#endif - /* * Maximum number of channels topology kcontrol can represent. */ -- 2.6.1

4 6

[alsa-devel] [PATCH] ASoC: Intel: haswell depends on sst-firmware
by Arnd Bergmann 11 Oct '16

11 Oct '16

The Intel Haswell audio support fails to link if CONFIG_SND_SOC_INTEL_SST_FIRMWARE is disabled: sst-haswell-dsp.c: undefined reference to `sst_mem_block_register' sst-haswell-dsp.c: undefined reference to `sst_mem_block_unregister_all' sst-haswell-dsp.c: undefined reference to `sst_module_alloc_blocks' sst-haswell-dsp.c: undefined reference to `sst_module_free' sst-haswell-dsp.c: undefined reference to `sst_module_new' sst-haswell-pcm.c: undefined reference to `sst_module_get_from_id' sst-haswell-pcm.c: undefined reference to `sst_module_runtime_restore' sst-haswell-pcm.c: undefined reference to `sst_module_runtime_save' ERROR: "sst_block_alloc_scratch" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_block_free_scratch" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_dsp_dma_copyfrom" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_dsp_dma_copyto" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_dsp_dma_get_channel" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_dsp_dma_put_channel" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_dsp_free" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_dsp_get_offset" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_dsp_new" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_fw_free_all" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_fw_new" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_fw_reload" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_fw_unload" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_module_runtime_alloc_blocks" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_module_runtime_get_from_id" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! ERROR: "sst_module_runtime_new" [sound/soc/intel/haswell/snd-soc-sst-haswell-pcm.ko] undefined! This moves the 'select' statement from two of the three haswell based users into the line that is used by all of them, so make it harder to get wrong and to fix the existing randconfig regressions. Fixes: 2d995e5dc283 ("ASoC: Intel: boards: Add bdw-rt5677 machine driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 26eb5a0a5575..fd5d1e091038 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_INTEL_SST_MATCH config SND_SOC_INTEL_HASWELL tristate + select SND_SOC_INTEL_SST_FIRMWARE config SND_SOC_INTEL_BAYTRAIL tristate @@ -56,7 +57,6 @@ config SND_SOC_INTEL_HASWELL_MACH depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM depends on DW_DMAC_CORE select SND_SOC_INTEL_SST - select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -138,7 +138,6 @@ config SND_SOC_INTEL_BROADWELL_MACH I2C_DESIGNWARE_PLATFORM depends on DW_DMAC_CORE select SND_SOC_INTEL_SST - select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help -- 2.9.0

2 1

[alsa-devel] [PATCH 0/5] topology: Upgrade ABI to match kernel v4.8
by mengdong.lin＠linux.intel.com 11 Oct '16

11 Oct '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Now the topology ABI version in kernel is 5, added types for BE (Backend) DAIs and sig_bits in stream caps. But the ABI version in alsa-lib is still 4, without above ABI update. Since the kernel merge window is open and the remaining topology ABI update (v3 kernel patch series) cannot be accepted atm. So we use this series to fix the ABI mismatch between kernel v4.8 and user space, by adding types for BE DAIs and sig_bits to stream caps and pumping topolpgy ABI version to 5 as well. This patch series also completes the support for BE DAIs. Guneshwor Singh (3): topology: Parse BE DAIs in text conf file topology: Support configuring BE DAIs by C API topology: Export BE DAIs to the binary for kernel Mengdong Lin (2): topology: ABI - Add the types for BE DAI topology: ABI - Add sig_bits to stream caps include/sound/asoc.h | 35 ++++++- include/topology.h | 45 +++++++++ src/topology/builder.c | 11 +++ src/topology/data.c | 4 + src/topology/elem.c | 4 + src/topology/parser.c | 16 ++++ src/topology/pcm.c | 238 ++++++++++++++++++++++++++++++++++++++++++++++ src/topology/tplg_local.h | 7 ++ 8 files changed, 358 insertions(+), 2 deletions(-) -- 2.5.0

3 8

Re: [alsa-devel] Audio on mx6q udoo
by Fabio Estevam 10 Oct '16

10 Oct '16

Hi Maciej, On Thu, Oct 6, 2016 at 8:17 PM, Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> wrote: > If I remember correctly I had some problem if SDMA driver was built-in > into kernel (it didn't load firmware correctly or so). > Not sure that the problem was with sound but maybe this is worth checking. I am not loading any external SDMA firmware. Haven't had a chance to try your config yet, but will investigate on the SDMA module versus built-in and also about the SDMA firmware. Thanks

1 0

[alsa-devel] Preferred way to keep mixer mute in sync with digital_mute callback
by Pascal Huerst 10 Oct '16

10 Oct '16

Hey everyone, I'm currently having an issue with cs4271. The codec plays back, even if mute is activated in alsa mixer. This is caused by the digital_mute callback, that does not care about the current mute status in alsa mixer. As a user, I would expect, that if mute is set in alsa mixer, I don't get any output if I play something back. (But probably I'm wrong?). On the other hand, wm8350 codec uses a flag in the driver struct, that stores the current state of mute in alsa mixer, which is taken into consideration in the digital_mute callback. But to make this possible, one can not just use: SOC_*() for example, but has to create a put-function using: SOC_*_EXT() only to get and store that flag. Is this really the way to go, or did I miss something? Thanks, Pascal

1 0

Re: [alsa-devel] [PATCH v2] ASoC: sgtl5000: add headphone and LINEOUT mute controls
by Mark Brown 10 Oct '16

10 Oct '16

On Mon, Oct 10, 2016 at 02:52:38PM +0200, Richard Leitner wrote: > On 09/09/2016 12:17 PM, Mark Brown wrote: > > Please don't send content free pings and please allow a reasonable time > > for review. People get busy, go on holiday, attend conferences and so > > on so unless there is some reason for urgency (like critical bug fixes) > > please allow at least a couple of weeks for review. If there have been > > review comments then people may be waiting for those to be addressed. > > Sending content free pings just adds to the mail volume (if they are > > seen at all) and if something has gone wrong you'll have to resend the > > patches anyway. > So it's been a month now, is this a reasonable time to ping? To repeat please don't send content free pings at all for the reasons explained above.

1 0

[alsa-devel] [PATCH] ALSA: line6: Fix POD X3 Live audio input
by Andrej Krutak 10 Oct '16

10 Oct '16

The commit c039aaa77a7d1d9375665a8b59ec16dc7d23e259 was incomplete, missing part of the setup for Live. This makes also audio input work, in addition to audio output. Signed-off-by: Andrej Krutak <dev(a)andree.sk> Reported-by: Eddi De Pieri <eddi(a)depieri.net> Fixes: c039aaa77a7d1d9375665a8b59ec16dc7d23e259 --- sound/usb/line6/podhd.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 193eb29..ac40525 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -317,7 +317,8 @@ static int podhd_init(struct usb_line6 *line6, if (pod->line6.properties->capabilities & LINE6_CAP_PCM) { /* initialize PCM subsystem: */ err = line6_init_pcm(line6, - (id->driver_info == LINE6_PODX3) ? &podx3_pcm_properties : + (id->driver_info == LINE6_PODX3 || + id->driver_info == LINE6_PODX3LIVE) ? &podx3_pcm_properties : &podhd_pcm_properties); if (err < 0) return err; -- 2.7.4

2 1

[alsa-devel] [PATCH] conf/ucm: skylake: add skylake-rt286 conf files
by Vinod Koul 10 Oct '16

10 Oct '16

This add the UCM conf files for skylake enabling I2S audio, HDMI and DMIC ports. Signed-off-by: Karthik D M <karthik.d.m(a)intel.com> Signed-off-by: Nishit Sharma <nishitx.sharma(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/skylake-rt286/Hdmi1 | 20 ++++ src/conf/ucm/skylake-rt286/Hdmi2 | 20 ++++ src/conf/ucm/skylake-rt286/HiFi | 129 ++++++++++++++++++++++++++ src/conf/ucm/skylake-rt286/Makefile.am | 4 + src/conf/ucm/skylake-rt286/skylake-rt286.conf | 60 ++++++++++++ 7 files changed, 235 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/skylake-rt286/Hdmi1 create mode 100644 src/conf/ucm/skylake-rt286/Hdmi2 create mode 100644 src/conf/ucm/skylake-rt286/HiFi create mode 100644 src/conf/ucm/skylake-rt286/Makefile.am create mode 100644 src/conf/ucm/skylake-rt286/skylake-rt286.conf diff --git a/configure.ac b/configure.ac index a1de230e588c..0dfbe81c6595 100644 --- a/configure.ac +++ b/configure.ac @@ -673,6 +673,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/PAZ00/Makefile \ src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ + src/conf/ucm/skylake-rt286/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/DB410c/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 7bf432e70e26..a11d324e785b 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 DB410c +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c diff --git a/src/conf/ucm/skylake-rt286/Hdmi1 b/src/conf/ucm/skylake-rt286/Hdmi1 new file mode 100644 index 000000000000..0e00fc844301 --- /dev/null +++ b/src/conf/ucm/skylake-rt286/Hdmi1 @@ -0,0 +1,20 @@ +# Usecase for device HDMI1/Display Port stereo playback on Intel SKYLAKE/KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi1" { + Comment "HDMI/Display Port 1 Stereo" + + EnableSequence [ + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:skylakert286,4" + PlaybackChannels "2" + PlaybackPriority "3" + JackControl "HDMI/DP, pcm=4 Jack" + } +} + diff --git a/src/conf/ucm/skylake-rt286/Hdmi2 b/src/conf/ucm/skylake-rt286/Hdmi2 new file mode 100644 index 000000000000..89b31aa4b8f9 --- /dev/null +++ b/src/conf/ucm/skylake-rt286/Hdmi2 @@ -0,0 +1,20 @@ +# Usecase for device HDMI2/Display Port stereo playback on Intel SKYLAKE/KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi2" { + Comment "HDMI/Display Port 2 Stereo" + + EnableSequence [ + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:skylakert286,5" + PlaybackChannels "2" + PlaybackPriority "4" + JackControl "HDMI/DP, pcm=5 Jack" + } +} + diff --git a/src/conf/ucm/skylake-rt286/HiFi b/src/conf/ucm/skylake-rt286/HiFi new file mode 100644 index 000000000000..41638f4cb9c8 --- /dev/null +++ b/src/conf/ucm/skylake-rt286/HiFi @@ -0,0 +1,129 @@ +# Usecase for stereo playback Speaker and Headset, Recording on DMIC and Headset MIC. +# For Audio in I2S mode on Intel SKYLAKE/KABYLAKE platforms + +SectionVerb { + + EnableSequence [ + cdev "hw:skylakert286" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + ] + + DisableSequence [ + cdev "hw:skylakert286" + ] + + Value { + TQ "HiFi" + CapturePCM "hw:skylakert286,1" + PlaybackPCM "hw:skylakert286,0" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + ] + EnableSequence [ + cdev "hw:skylakert286" + cset "name='HPO L Switch' 1" + cset "name='HPO R Switch' 1" + cset "name='Headphone Jack Switch' 1" + cset "name='Speaker Playback Switch' 0,0" + ] + + DisableSequence [ + ] + Value { + PlaybackChannels "2" + PlaybackPriority "1" + JackControl "Headphone Jack" + JackHWMute "Speaker" + } +} + +SectionDevice."Speaker" { + Comment "Speaker" + + ConflictingDevice [ + "Headphones" + ] + EnableSequence [ + cdev "hw:skylakert286" + cset "name='SPO Switch' 1" + cset "name='Speaker Playback Switch' 1,1" + cset "name='Speaker Switch' 1" + cset "name='HPO L Switch' 0" + cset "name='HPO R Switch' 0" + ] + + DisableSequence [ + ] + Value { + PlaybackChannels "2" + PlaybackPriority "1" + JackHWMute "Headphones" + } +} + +SectionDevice."Headset" { + Comment "Headset Mic" + + ConflictingDevice [ + "dmiccap" + ] + EnableSequence [ + cdev "hw:skylakert286" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='ADC0 Capture Switch' 1,1" + cset "name='ADC 0 Mux' 0" + cset "name='Mic Jack Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 0" + ] + + DisableSequence [ + cdev "hw:skylakert286" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 0" + ] + Value { + CaptureChannels "2" + CapturePriority "2" + JackControl "Mic Jack" + } +} + + +SectionDevice."dmiccap" { + Comment "DMIC Stereo" + + ConflictingDevice [ + "Headpset" + ] + + EnableSequence [ + + cdev "hw:skylakert286" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 0" + cset "name='ADC0 Capture Switch' 0,0" + cset "name='Mic Jack Switch' 0" + cset "name='ADC 0 Mux' 0" + + ] + + DisableSequence [ + cdev "hw:skylakert286" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 0" + cset "name='ADC0 Capture Switch' 1,1" + cset "name='Mic Jack Switch' 1" + cset "name='ADC 0 Mux' 2" + ] + Value { + CaptureChannels "2" + CapturePriority "2" + } +} + diff --git a/src/conf/ucm/skylake-rt286/Makefile.am b/src/conf/ucm/skylake-rt286/Makefile.am new file mode 100644 index 000000000000..9d6a85a5bdcf --- /dev/null +++ b/src/conf/ucm/skylake-rt286/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/skylake-rt286 +ucm_DATA = skylake-rt286.conf HiFi Hdmi1 Hdmi2 +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/skylake-rt286/skylake-rt286.conf b/src/conf/ucm/skylake-rt286/skylake-rt286.conf new file mode 100644 index 000000000000..64e852c479e3 --- /dev/null +++ b/src/conf/ucm/skylake-rt286/skylake-rt286.conf @@ -0,0 +1,60 @@ +# UCM for Intel SKYLAKE/KABYLAKE platforms +# For Audio in I2S mode + +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play and record HiFi quality Music" +} + +SectionUseCase."Hdmi1" { + File "Hdmi1" + Comment "Play on Hdmi/DP 1" +} + +SectionUseCase."Hdmi2" { + File "Hdmi2" + Comment "Play on Hdmi/DP 2" +} + +ValueDefaults { + PlaybackCTL "hw:skylakert286" + CaptureCTL "hw:skylakert286" + CaptureChannels "2" + CapturePriority "2" +} + +SectionDefaults [ + cdev "hw:skylakert286" + cset "name='Headphone Jack Switch' 1" + cset "name='Speaker Switch' 1" + cset "name='Speaker Playback Switch' 0,0" + cset "name='Front DAC Switch' 1" + cset "name='Front RECMIX Switch' 0" + cset "name='Mic Jack Switch' 1" + cset "name='ADC 0 Mux' 2" + cset "name='ADC 1 Mux' 0" + cset "name='ADC0 Capture Volume' 105,105" + cset "name='ADC0 Capture Switch' 1,1" + cset "name='AMIC Volume' 3" + cset "name='DAC0 Playback Volume' 100,100" + cset "name='HPO L Switch' 0" + cset "name='HPO Mux' 0" + cset "name='HPO R Switch' 0" + cset "name='Pin 5 Mux' 1" + cset "name='Pin 6 Mux' 2" + cset "name='Pin 7 Mux' 3" + cset "name='RECMIX Beep Switch' 0" + cset "name='RECMIX Line1 Switch' 0" + cset "name='RECMIX Mic1 Switch' 0" + cset "name='SPK Mux' 0" + cset "name='SPO Switch' 1" + cset "name='codec0_out mo codec0_in mi Switch' 0" + cset "name='codec0_out mo dmic01_hifi_in mi Switch' 0" + cset "name='codec0_out mo media0_in mi Switch' 1" + cset "name='codec1_out mo codec0_in mi Switch' 0" + cset "name='codec1_out mo dmic01_hifi_in mi Switch' 0" + cset "name='codec1_out mo media0_in mi Switch' 0" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo media0_in mi Switch' 0" +] -- 1.9.1

2 1

[alsa-devel] 18i20 2nd Gen no ALSA mixer
by Happy 10 Oct '16

10 Oct '16

It was reported that Focusrite 18i20 Gen 2 seems not 100% compatible with the 18i20 Gen 1. It can work, but no ALSA Mixer support. What information does ALSA team need to get this fixed (USB Device ID) P.S: How to pull the ALSA driver code to to change it and test such fixon Ubuntu ? Any tutorial, step by step instruction ?

1 0

[alsa-devel] [PATCH] conf/ucm: skylake: add skylake-rt296 conf files
by Vinod Koul 07 Oct '16

07 Oct '16

This add the UCM conf files for skylake enabling I2S audio, HDMI and DMIC ports. Signed-off-by: Karthik D M <karthik.d.m(a)intel.com> Signed-off-by: Nishit Sharma <nishitx.sharma(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/skylake-rt286/Hdmi1 | 20 ++++ src/conf/ucm/skylake-rt286/Hdmi2 | 20 ++++ src/conf/ucm/skylake-rt286/HiFi | 129 ++++++++++++++++++++++++++ src/conf/ucm/skylake-rt286/Makefile.am | 4 + src/conf/ucm/skylake-rt286/skylake-rt286.conf | 60 ++++++++++++ 7 files changed, 235 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/skylake-rt286/Hdmi1 create mode 100644 src/conf/ucm/skylake-rt286/Hdmi2 create mode 100644 src/conf/ucm/skylake-rt286/HiFi create mode 100644 src/conf/ucm/skylake-rt286/Makefile.am create mode 100644 src/conf/ucm/skylake-rt286/skylake-rt286.conf diff --git a/configure.ac b/configure.ac index a1de230e588c..0dfbe81c6595 100644 --- a/configure.ac +++ b/configure.ac @@ -673,6 +673,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/PAZ00/Makefile \ src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ + src/conf/ucm/skylake-rt286/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/DB410c/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 7bf432e70e26..a11d324e785b 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 DB410c +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c diff --git a/src/conf/ucm/skylake-rt286/Hdmi1 b/src/conf/ucm/skylake-rt286/Hdmi1 new file mode 100644 index 000000000000..0e00fc844301 --- /dev/null +++ b/src/conf/ucm/skylake-rt286/Hdmi1 @@ -0,0 +1,20 @@ +# Usecase for device HDMI1/Display Port stereo playback on Intel SKYLAKE/KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi1" { + Comment "HDMI/Display Port 1 Stereo" + + EnableSequence [ + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:skylakert286,4" + PlaybackChannels "2" + PlaybackPriority "3" + JackControl "HDMI/DP, pcm=4 Jack" + } +} + diff --git a/src/conf/ucm/skylake-rt286/Hdmi2 b/src/conf/ucm/skylake-rt286/Hdmi2 new file mode 100644 index 000000000000..89b31aa4b8f9 --- /dev/null +++ b/src/conf/ucm/skylake-rt286/Hdmi2 @@ -0,0 +1,20 @@ +# Usecase for device HDMI2/Display Port stereo playback on Intel SKYLAKE/KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi2" { + Comment "HDMI/Display Port 2 Stereo" + + EnableSequence [ + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:skylakert286,5" + PlaybackChannels "2" + PlaybackPriority "4" + JackControl "HDMI/DP, pcm=5 Jack" + } +} + diff --git a/src/conf/ucm/skylake-rt286/HiFi b/src/conf/ucm/skylake-rt286/HiFi new file mode 100644 index 000000000000..41638f4cb9c8 --- /dev/null +++ b/src/conf/ucm/skylake-rt286/HiFi @@ -0,0 +1,129 @@ +# Usecase for stereo playback Speaker and Headset, Recording on DMIC and Headset MIC. +# For Audio in I2S mode on Intel SKYLAKE/KABYLAKE platforms + +SectionVerb { + + EnableSequence [ + cdev "hw:skylakert286" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + ] + + DisableSequence [ + cdev "hw:skylakert286" + ] + + Value { + TQ "HiFi" + CapturePCM "hw:skylakert286,1" + PlaybackPCM "hw:skylakert286,0" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + ] + EnableSequence [ + cdev "hw:skylakert286" + cset "name='HPO L Switch' 1" + cset "name='HPO R Switch' 1" + cset "name='Headphone Jack Switch' 1" + cset "name='Speaker Playback Switch' 0,0" + ] + + DisableSequence [ + ] + Value { + PlaybackChannels "2" + PlaybackPriority "1" + JackControl "Headphone Jack" + JackHWMute "Speaker" + } +} + +SectionDevice."Speaker" { + Comment "Speaker" + + ConflictingDevice [ + "Headphones" + ] + EnableSequence [ + cdev "hw:skylakert286" + cset "name='SPO Switch' 1" + cset "name='Speaker Playback Switch' 1,1" + cset "name='Speaker Switch' 1" + cset "name='HPO L Switch' 0" + cset "name='HPO R Switch' 0" + ] + + DisableSequence [ + ] + Value { + PlaybackChannels "2" + PlaybackPriority "1" + JackHWMute "Headphones" + } +} + +SectionDevice."Headset" { + Comment "Headset Mic" + + ConflictingDevice [ + "dmiccap" + ] + EnableSequence [ + cdev "hw:skylakert286" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='ADC0 Capture Switch' 1,1" + cset "name='ADC 0 Mux' 0" + cset "name='Mic Jack Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 0" + ] + + DisableSequence [ + cdev "hw:skylakert286" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 0" + ] + Value { + CaptureChannels "2" + CapturePriority "2" + JackControl "Mic Jack" + } +} + + +SectionDevice."dmiccap" { + Comment "DMIC Stereo" + + ConflictingDevice [ + "Headpset" + ] + + EnableSequence [ + + cdev "hw:skylakert286" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 0" + cset "name='ADC0 Capture Switch' 0,0" + cset "name='Mic Jack Switch' 0" + cset "name='ADC 0 Mux' 0" + + ] + + DisableSequence [ + cdev "hw:skylakert286" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 0" + cset "name='ADC0 Capture Switch' 1,1" + cset "name='Mic Jack Switch' 1" + cset "name='ADC 0 Mux' 2" + ] + Value { + CaptureChannels "2" + CapturePriority "2" + } +} + diff --git a/src/conf/ucm/skylake-rt286/Makefile.am b/src/conf/ucm/skylake-rt286/Makefile.am new file mode 100644 index 000000000000..9d6a85a5bdcf --- /dev/null +++ b/src/conf/ucm/skylake-rt286/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/skylake-rt286 +ucm_DATA = skylake-rt286.conf HiFi Hdmi1 Hdmi2 +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/skylake-rt286/skylake-rt286.conf b/src/conf/ucm/skylake-rt286/skylake-rt286.conf new file mode 100644 index 000000000000..64e852c479e3 --- /dev/null +++ b/src/conf/ucm/skylake-rt286/skylake-rt286.conf @@ -0,0 +1,60 @@ +# UCM for Intel SKYLAKE/KABYLAKE platforms +# For Audio in I2S mode + +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play and record HiFi quality Music" +} + +SectionUseCase."Hdmi1" { + File "Hdmi1" + Comment "Play on Hdmi/DP 1" +} + +SectionUseCase."Hdmi2" { + File "Hdmi2" + Comment "Play on Hdmi/DP 2" +} + +ValueDefaults { + PlaybackCTL "hw:skylakert286" + CaptureCTL "hw:skylakert286" + CaptureChannels "2" + CapturePriority "2" +} + +SectionDefaults [ + cdev "hw:skylakert286" + cset "name='Headphone Jack Switch' 1" + cset "name='Speaker Switch' 1" + cset "name='Speaker Playback Switch' 0,0" + cset "name='Front DAC Switch' 1" + cset "name='Front RECMIX Switch' 0" + cset "name='Mic Jack Switch' 1" + cset "name='ADC 0 Mux' 2" + cset "name='ADC 1 Mux' 0" + cset "name='ADC0 Capture Volume' 105,105" + cset "name='ADC0 Capture Switch' 1,1" + cset "name='AMIC Volume' 3" + cset "name='DAC0 Playback Volume' 100,100" + cset "name='HPO L Switch' 0" + cset "name='HPO Mux' 0" + cset "name='HPO R Switch' 0" + cset "name='Pin 5 Mux' 1" + cset "name='Pin 6 Mux' 2" + cset "name='Pin 7 Mux' 3" + cset "name='RECMIX Beep Switch' 0" + cset "name='RECMIX Line1 Switch' 0" + cset "name='RECMIX Mic1 Switch' 0" + cset "name='SPK Mux' 0" + cset "name='SPO Switch' 1" + cset "name='codec0_out mo codec0_in mi Switch' 0" + cset "name='codec0_out mo dmic01_hifi_in mi Switch' 0" + cset "name='codec0_out mo media0_in mi Switch' 1" + cset "name='codec1_out mo codec0_in mi Switch' 0" + cset "name='codec1_out mo dmic01_hifi_in mi Switch' 0" + cset "name='codec1_out mo media0_in mi Switch' 0" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='media0_out mo dmic01_hifi_in mi Switch' 1" + cset "name='media0_out mo media0_in mi Switch' 0" +] -- 1.9.1

1 1

[alsa-devel] [PATCH REPOST] ASoC: Intel: Atom: Fix message handling during drop stream
by Harsha Priya 07 Oct '16

07 Oct '16

If a stream is being dropped, the period_elapsed received then after need not be processed. Processing of the period_elapsed message causes a time out in drop stream response processing. This patch adds a condition to skip period_elapsed message processing if the stream is in INIT state. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 8afa6fe..bfc8899 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -267,6 +267,9 @@ static void process_fw_async_msg(struct intel_sst_drv *sst_drv_ctx, "Period elapsed rcvd for pipe id 0x%x\n", pipe_id); stream = &sst_drv_ctx->streams[str_id]; + /* If stream is dropped, skip processing this message*/ + if (stream->status == STREAM_INIT) + break; if (stream->period_elapsed) stream->period_elapsed(stream->pcm_substream); if (stream->compr_cb) -- 1.9.1

1 0

[alsa-devel] [PATCH] [PATCH REPOST] ASoC: Intel: Atom: Fix message handling during drop stream
by Harsha Priya 07 Oct '16

07 Oct '16

If a stream is being dropped, the period_elapsed received then after need not be processed. Processing of the period_elapsed message causes a time out in drop stream response processing. This patch adds a condition to skip period_elapsed message processing if the stream is in INIT state. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 8afa6fe..bfc8899 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -267,6 +267,9 @@ static void process_fw_async_msg(struct intel_sst_drv *sst_drv_ctx, "Period elapsed rcvd for pipe id 0x%x\n", pipe_id); stream = &sst_drv_ctx->streams[str_id]; + /* If stream is dropped, skip processing this message*/ + if (stream->status == STREAM_INIT) + break; if (stream->period_elapsed) stream->period_elapsed(stream->pcm_substream); if (stream->compr_cb) -- 1.9.1

1 0

Re: [alsa-devel] [PATCH] ASoC: rl6347a: Use dev_err_once for I2C communication error prints
by Mark Brown 07 Oct '16

07 Oct '16

On Wed, Oct 05, 2016 at 05:05:36PM +0300, Jarkko Nikula wrote: > It's difficult to guess from bunch of "ret=-121" errors what driver and > device actually caused them. Since struct i2c_client has the dev pointer > use that for dev_err_once() in order to see the device and not spam the > log needlessly. dev_err() is obviously an improvement (as would a better message be) but dev_err_once() seems like a big jump that we don't normally do for our I/O errors - transient errors do happen from time to time (especially when people get their power up sequences wrong) and it can be useful to know if something is just dead, failed gradually over time or just had a single transient error. It's also helpful since a lot of the errors that do occur result in long timeouts so it's not always clear that we're grinding through lots of I/Os if we don't say anything. Am I missing something about this case?

2 1

[alsa-devel] Audio on mx6q udoo
by Fabio Estevam 07 Oct '16

07 Oct '16

Hi Maciej, I am running today's linux-next on a imx6q-udoo board. Removed USB support to avoid the hang and also enabled CONFIG_SND_SOC_AC97_CODEC=y. When I start a playback it seems to be running: # aplay /root/clarinet.wav Playing WAVE '/root/clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz ,but I don't get any sound out of the CN7 green jack. I put the volumes at the maximum via alsamixer. Any ideas? Thanks, Fabio Estevam

2 4

[alsa-devel] [PATCH v2] ASoC: cs4270: fix DAPM stream name mismatch
by murray foster 07 Oct '16

07 Oct '16

Mismatching stream names in DAPM route and widget definitions are causing compilation errors. Fixing these names allows the cs4270 driver to compile and function. Signed-off-by: Murray Foster <mrafoster(a)gmail.com> --- Changes in v2: - Remove the 's' typo at the beginning of the codec driver. sound/soc/codecs/cs4270.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e07807d..3670086 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -148,11 +148,11 @@ SND_SOC_DAPM_OUTPUT("AOUTR"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { - { "Capture", NULL, "AINA" }, - { "Capture", NULL, "AINB" }, + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, - { "AOUTA", NULL, "Playback" }, - { "AOUTB", NULL, "Playback" }, + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, }; /** -- 2.4.9

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[alsa-devel] [PATCH v3] clk: x86: Add Atom PMC platform clocks
by Irina Tirdea 06 Oct '16

06 Oct '16

The BayTrail and CherryTrail platforms provide platform clocks through their Power Management Controller (PMC). The SoC supports up to 6 clocks (PMC_PLT_CLK[5:0]) with a frequency of either 19.2 MHz (PLL) or 25 MHz (XTAL) for BayTrail an a frequency of 19.2 MHz (XTAL) for CherryTrail. These clocks are available for general system use, where appropriate, and each have Control & Frequency register fields associated with them. For example, the usage for platform clocks suggested in the datasheet is the following: PLT_CLK[2:0] - Camera PLT_CLK[3] - Audio Codec PLT_CLK[4] - PLT_CLK[5] - COMMs Signed-off-by: Irina Tirdea <irina.tirdea(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Notes: Submitting this patch through the clock tree as requested by Mark Brown. This patch specifically enables the audio MCLK required by Baytrail CR devices (support already merged in Mark's tree) Changes from v2: - move clk platform data structures to a separate include file - store clk_hw pointer for the fixed rate clocks Changes from v1: - register the clk device as child of pmc device - pass iomem pointer from pmc driver to clk driver to avoid using pmc_atom_read()/write() and use readl/writel API instead - use devm_clk_hw_register/clkdev_hw_create instead of clk_register/clkdev_create arch/x86/Kconfig | 1 + arch/x86/include/asm/pmc_atom.h | 3 + arch/x86/platform/atom/pmc_atom.c | 78 +++++- drivers/clk/x86/Makefile | 1 + drivers/clk/x86/clk-byt-plt.c | 381 ++++++++++++++++++++++++++++++ include/linux/platform_data/clk-byt-plt.h | 31 +++ 6 files changed, 492 insertions(+), 3 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c create mode 100644 include/linux/platform_data/clk-byt-plt.h diff --git a/arch/x86/Kconfig b/arch/x86/Kconfig index 2a1f0ce..3e603a8 100644 --- a/arch/x86/Kconfig +++ b/arch/x86/Kconfig @@ -2744,6 +2744,7 @@ config X86_DMA_REMAP config PMC_ATOM def_bool y depends on PCI + select COMMON_CLK config VMD depends on PCI_MSI diff --git a/arch/x86/include/asm/pmc_atom.h b/arch/x86/include/asm/pmc_atom.h index aa8744c..2d310cf 100644 --- a/arch/x86/include/asm/pmc_atom.h +++ b/arch/x86/include/asm/pmc_atom.h @@ -50,6 +50,9 @@ BIT_ORED_DEDICATED_IRQ_GPSC | \ BIT_SHARED_IRQ_GPSS) +/* Platform clock control registers */ +#define PMC_CLK_CTL_0 0x60 + /* The timers acumulate time spent in sleep state */ #define PMC_S0IR_TMR 0x80 #define PMC_S0I1_TMR 0x84 diff --git a/arch/x86/platform/atom/pmc_atom.c b/arch/x86/platform/atom/pmc_atom.c index 964ff4f..b353893 100644 --- a/arch/x86/platform/atom/pmc_atom.c +++ b/arch/x86/platform/atom/pmc_atom.c @@ -21,6 +21,8 @@ #include <linux/debugfs.h> #include <linux/seq_file.h> #include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/platform_data/clk-byt-plt.h> #include <asm/pmc_atom.h> @@ -37,6 +39,11 @@ struct pmc_reg_map { const struct pmc_bit_map *pss; }; +struct pmc_data { + const struct pmc_reg_map *map; + const struct pmc_clk *clks; +}; + struct pmc_dev { u32 base_addr; void __iomem *regmap; @@ -50,6 +57,29 @@ struct pmc_dev { static struct pmc_dev pmc_device; static u32 acpi_base_addr; +static const struct pmc_clk byt_clks[] = { + { + .name = "xtal", + .freq = 25000000, + .parent_name = NULL, + }, + { + .name = "pll", + .freq = 19200000, + .parent_name = "xtal", + }, + {}, +}; + +static const struct pmc_clk cht_clks[] = { + { + .name = "xtal", + .freq = 19200000, + .parent_name = NULL, + }, + {}, +}; + static const struct pmc_bit_map d3_sts_0_map[] = { {"LPSS1_F0_DMA", BIT_LPSS1_F0_DMA}, {"LPSS1_F1_PWM1", BIT_LPSS1_F1_PWM1}, @@ -169,6 +199,16 @@ static const struct pmc_reg_map cht_reg_map = { .pss = cht_pss_map, }; +static const struct pmc_data byt_data = { + .map = &byt_reg_map, + .clks = byt_clks, +}; + +static const struct pmc_data cht_data = { + .map = &cht_reg_map, + .clks = cht_clks, +}; + static inline u32 pmc_reg_read(struct pmc_dev *pmc, int reg_offset) { return readl(pmc->regmap + reg_offset); @@ -382,10 +422,36 @@ static int pmc_dbgfs_register(struct pmc_dev *pmc) } #endif /* CONFIG_DEBUG_FS */ +static int pmc_setup_clks(struct pci_dev *pdev, void __iomem *pmc_regmap, + const struct pmc_data *pmc_data) +{ + struct platform_device *clkdev; + struct pmc_clk_data *clk_data; + + clk_data = kzalloc(sizeof(*clk_data), GFP_KERNEL); + if (!clk_data) + return -ENOMEM; + + clk_data->base = pmc_regmap + PMC_CLK_CTL_0; + clk_data->clks = pmc_data->clks; + + clkdev = platform_device_register_data(&pdev->dev, "clk-byt-plt", -1, + clk_data, sizeof(*clk_data)); + if (IS_ERR(clkdev)) { + kfree(clk_data); + return PTR_ERR(clkdev); + } + + kfree(clk_data); + + return 0; +} + static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) { struct pmc_dev *pmc = &pmc_device; - const struct pmc_reg_map *map = (struct pmc_reg_map *)ent->driver_data; + const struct pmc_data *data = (struct pmc_data *)ent->driver_data; + const struct pmc_reg_map *map = data->map; int ret; /* Obtain ACPI base address */ @@ -414,6 +480,12 @@ static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) if (ret) dev_warn(&pdev->dev, "debugfs register failed\n"); + /* Register platform clocks - PMC_PLT_CLK [5:0] */ + ret = pmc_setup_clks(pdev, pmc->regmap, data); + if (ret) + dev_warn(&pdev->dev, "platform clocks register failed: %d\n", + ret); + pmc->init = true; return ret; } @@ -424,8 +496,8 @@ static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) * used by pci_match_id() call below. */ static const struct pci_device_id pmc_pci_ids[] = { - { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_VLV_PMC), (kernel_ulong_t)&byt_reg_map }, - { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_CHT_PMC), (kernel_ulong_t)&cht_reg_map }, + { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_VLV_PMC), (kernel_ulong_t)&byt_data }, + { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_CHT_PMC), (kernel_ulong_t)&cht_data }, { 0, }, }; diff --git a/drivers/clk/x86/Makefile b/drivers/clk/x86/Makefile index 0478138..cbdc8cc 100644 --- a/drivers/clk/x86/Makefile +++ b/drivers/clk/x86/Makefile @@ -1,2 +1,3 @@ clk-x86-lpss-objs := clk-lpt.o obj-$(CONFIG_X86_INTEL_LPSS) += clk-x86-lpss.o +obj-$(CONFIG_PMC_ATOM) += clk-byt-plt.o diff --git a/drivers/clk/x86/clk-byt-plt.c b/drivers/clk/x86/clk-byt-plt.c new file mode 100644 index 0000000..9dbf4b0 --- /dev/null +++ b/drivers/clk/x86/clk-byt-plt.c @@ -0,0 +1,381 @@ +/* + * Intel Atom platform clocks driver for BayTrail and CherryTrail SoC. + * + * Copyright (C) 2016, Intel Corporation + * Author: Irina Tirdea <irina.tirdea(a)intel.com> + * + * 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/clk-provider.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/clkdev.h> +#include <linux/platform_data/clk-byt-plt.h> + +#define PLT_CLK_NAME_BASE "pmc_plt_clk_" +#define PLT_CLK_DRIVER_NAME "clk-byt-plt" + +#define PMC_CLK_CTL_SIZE 4 +#define PMC_CLK_NUM 6 +#define PMC_MASK_CLK_CTL GENMASK(1, 0) +#define PMC_MASK_CLK_FREQ BIT(2) +#define PMC_CLK_CTL_GATED_ON_D3 0x0 +#define PMC_CLK_CTL_FORCE_ON 0x1 +#define PMC_CLK_CTL_FORCE_OFF 0x2 +#define PMC_CLK_CTL_RESERVED 0x3 +#define PMC_CLK_FREQ_XTAL 0x0 /* 25 MHz */ +#define PMC_CLK_FREQ_PLL 0x4 /* 19.2 MHz */ + +struct clk_plt_fixed { + struct clk_hw *clk; + struct clk_lookup *lookup; +}; + +struct clk_plt { + struct clk_hw hw; + void __iomem *reg; + struct clk_lookup *lookup; + spinlock_t lock; +}; + +#define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) + +struct clk_plt_data { + struct clk_plt_fixed **parents; + u8 nparents; + struct clk_plt *clks[PMC_CLK_NUM]; +}; + +static inline int plt_reg_to_parent(int reg) +{ + switch (reg & PMC_MASK_CLK_FREQ) { + case PMC_CLK_FREQ_XTAL: + return 0; /* index 0 in parents[] */ + case PMC_CLK_FREQ_PLL: + return 1; /* index 1 in parents[] */ + } + + return 0; +} + +static inline int plt_parent_to_reg(int index) +{ + switch (index) { + case 0: /* index 0 in parents[] */ + return PMC_CLK_FREQ_XTAL; + case 1: /* index 0 in parents[] */ + return PMC_CLK_FREQ_PLL; + } + + return PMC_CLK_FREQ_XTAL; +} + +static inline int plt_reg_to_enabled(int reg) +{ + switch (reg & PMC_MASK_CLK_CTL) { + case PMC_CLK_CTL_GATED_ON_D3: + case PMC_CLK_CTL_FORCE_ON: + return 1; /* enabled */ + case PMC_CLK_CTL_FORCE_OFF: + case PMC_CLK_CTL_RESERVED: + default: + return 0; /* disabled */ + } +} + +static void plt_clk_reg_update(struct clk_plt *clk, u32 mask, u32 val) +{ + u32 orig, tmp; + unsigned long flags = 0; + + spin_lock_irqsave(&clk->lock, flags); + + orig = readl(clk->reg); + + tmp = orig & ~mask; + tmp |= val & mask; + + if (tmp != orig) + writel(tmp, clk->reg); + + spin_unlock_irqrestore(&clk->lock, flags); +} + +static int plt_clk_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_plt *clk = to_clk_plt(hw); + + plt_clk_reg_update(clk, PMC_MASK_CLK_FREQ, plt_parent_to_reg(index)); + + return 0; +} + +static u8 plt_clk_get_parent(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + u32 value; + + value = readl(clk->reg); + + return plt_reg_to_parent(value); +} + +static int plt_clk_enable(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + + plt_clk_reg_update(clk, PMC_MASK_CLK_CTL, PMC_CLK_CTL_FORCE_ON); + + return 0; +} + +static void plt_clk_disable(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + + plt_clk_reg_update(clk, PMC_MASK_CLK_CTL, PMC_CLK_CTL_FORCE_OFF); +} + +static int plt_clk_is_enabled(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + u32 value; + + value = readl(clk->reg); + + return plt_reg_to_enabled(value); +} + +static const struct clk_ops plt_clk_ops = { + .enable = plt_clk_enable, + .disable = plt_clk_disable, + .is_enabled = plt_clk_is_enabled, + .get_parent = plt_clk_get_parent, + .set_parent = plt_clk_set_parent, + .determine_rate = __clk_mux_determine_rate, +}; + +static struct clk_plt *plt_clk_register(struct platform_device *pdev, int id, + void __iomem *base, + const char **parent_names, + int num_parents) +{ + struct clk_plt *pclk; + struct clk_init_data init; + int ret; + + pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); + if (!pclk) + return ERR_PTR(-ENOMEM); + + init.name = kasprintf(GFP_KERNEL, "%s%d", PLT_CLK_NAME_BASE, id); + init.ops = &plt_clk_ops; + init.flags = 0; + init.parent_names = parent_names; + init.num_parents = num_parents; + + pclk->hw.init = &init; + pclk->reg = base + id * PMC_CLK_CTL_SIZE; + spin_lock_init(&pclk->lock); + + ret = devm_clk_hw_register(&pdev->dev, &pclk->hw); + if (ret) + goto err_free_init; + + pclk->lookup = clkdev_hw_create(&pclk->hw, init.name, NULL); + if (!pclk->lookup) { + ret = -ENOMEM; + goto err_free_init; + } + + kfree(init.name); + + return pclk; + +err_free_init: + kfree(init.name); + return ERR_PTR(ret); +} + +static void plt_clk_unregister(struct clk_plt *pclk) +{ + clkdev_drop(pclk->lookup); +} + +static struct clk_plt_fixed *plt_clk_register_fixed_rate(struct platform_device *pdev, + const char *name, + const char *parent_name, + unsigned long fixed_rate) +{ + struct clk_plt_fixed *pclk; + int ret = 0; + + pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); + if (!pclk) + return ERR_PTR(-ENOMEM); + + pclk->clk = clk_hw_register_fixed_rate(&pdev->dev, name, parent_name, + 0, fixed_rate); + if (IS_ERR(pclk->clk)) + return ERR_CAST(pclk->clk); + + pclk->lookup = clkdev_hw_create(pclk->clk, name, NULL); + if (!pclk->lookup) { + ret = -ENOMEM; + goto err_clk_unregister; + } + + return pclk; + +err_clk_unregister: + clk_hw_unregister_fixed_rate(pclk->clk); + return ERR_PTR(ret); +} + +static void plt_clk_unregister_fixed_rate(struct clk_plt_fixed *pclk) +{ + clkdev_drop(pclk->lookup); + clk_hw_unregister_fixed_rate(pclk->clk); +} + +static const char **plt_clk_register_parents(struct platform_device *pdev, + struct clk_plt_data *data, + const struct pmc_clk *clks) +{ + const char **parent_names; + int i, err; + + data->nparents = 0; + while (clks[data->nparents].name) + data->nparents++; + + data->parents = devm_kzalloc(&pdev->dev, + sizeof(*data->parents) * data->nparents, + GFP_KERNEL); + if (!data->parents) { + err = -ENOMEM; + goto err_out; + } + + parent_names = kcalloc(data->nparents, sizeof(*parent_names), + GFP_KERNEL); + if (!parent_names) { + err = -ENOMEM; + goto err_out; + } + + for (i = 0; i < data->nparents; i++) { + data->parents[i] = + plt_clk_register_fixed_rate(pdev, clks[i].name, + clks[i].parent_name, + clks[i].freq); + if (IS_ERR(data->parents[i])) { + err = PTR_ERR(data->parents[i]); + goto err_unreg; + } + parent_names[i] = kstrdup_const(clks[i].name, GFP_KERNEL); + } + + return parent_names; + +err_unreg: + for (i--; i >= 0; i--) { + plt_clk_unregister_fixed_rate(data->parents[i]); + kfree_const(parent_names[i]); + } + kfree(parent_names); +err_out: + data->nparents = 0; + return ERR_PTR(err); +} + +static void plt_clk_unregister_parents(struct clk_plt_data *data) +{ + int i; + + for (i = 0; i < data->nparents; i++) + plt_clk_unregister_fixed_rate(data->parents[i]); +} + +static int plt_clk_probe(struct platform_device *pdev) +{ + struct clk_plt_data *data; + int i, err; + const char **parent_names; + const struct pmc_clk_data *pmc_data; + + pmc_data = dev_get_platdata(&pdev->dev); + if (!pmc_data || !pmc_data->clks) + return -EINVAL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + parent_names = plt_clk_register_parents(pdev, data, pmc_data->clks); + if (IS_ERR(parent_names)) + return PTR_ERR(parent_names); + + for (i = 0; i < PMC_CLK_NUM; i++) { + data->clks[i] = plt_clk_register(pdev, i, pmc_data->base, + parent_names, data->nparents); + if (IS_ERR(data->clks[i])) { + err = PTR_ERR(data->clks[i]); + goto err_unreg_clk_plt; + } + } + + for (i = 0; i < data->nparents; i++) + kfree_const(parent_names[i]); + kfree(parent_names); + + dev_set_drvdata(&pdev->dev, data); + return 0; + +err_unreg_clk_plt: + for (i--; i >= 0; i--) + plt_clk_unregister(data->clks[i]); + plt_clk_unregister_parents(data); + for (i = 0; i < data->nparents; i++) + kfree_const(parent_names[i]); + kfree(parent_names); + return err; +} + +static int plt_clk_remove(struct platform_device *pdev) +{ + struct clk_plt_data *data; + int i; + + data = dev_get_drvdata(&pdev->dev); + if (!data) + return 0; + + for (i = 0; i < PMC_CLK_NUM; i++) + plt_clk_unregister(data->clks[i]); + plt_clk_unregister_parents(data); + return 0; +} + +static struct platform_driver plt_clk_driver = { + .driver = { + .name = PLT_CLK_DRIVER_NAME, + }, + .probe = plt_clk_probe, + .remove = plt_clk_remove, +}; +module_platform_driver(plt_clk_driver); + +MODULE_DESCRIPTION("Intel Atom platform clocks driver"); +MODULE_AUTHOR("Irina Tirdea <irina.tirdea(a)intel.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/include/linux/platform_data/clk-byt-plt.h b/include/linux/platform_data/clk-byt-plt.h new file mode 100644 index 0000000..e6bca9c --- /dev/null +++ b/include/linux/platform_data/clk-byt-plt.h @@ -0,0 +1,31 @@ +/* + * Intel Atom platform clocks for BayTrail and CherryTrail SoC. + * + * Copyright (C) 2016, Intel Corporation + * Author: Irina Tirdea <irina.tirdea(a)intel.com> + * + * 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. + */ + +#ifndef __CLK_BYT_PLT_H +#define __CLK_BYT_PLT_H + +struct pmc_clk { + const char *name; + unsigned long freq; + const char *parent_name; +}; + +struct pmc_clk_data { + void __iomem *base; + const struct pmc_clk *clks; +}; + +#endif /* __CLK_BYT_PLT_H */ -- 1.9.1

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[alsa-devel] [PATCH] ASoC: topology: Reenable use from userspace
by Mark Brown 06 Oct '16

06 Oct '16

We had inserted a #error into the topology UAPI code to ensure that the ABI was not adopted by userspace while final review and testing was ongoing. The idea was that some finishing touches would be made to the ABI before declaring it stable and suitable for use in production but this has not yet happened as more than a year later revisions to the ABI are still onging. The reality however is that people are shipping topology files in production and these ABI changes are causing practical issues for users and we can't break userspace. This makes this error pointless so we should remove it. Signed-off-by: Mark Brown <broonie(a)kernel.org> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/asoc.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 33d00a4ce656..819d895edfdc 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -18,12 +18,6 @@ #include <linux/types.h> #include <sound/asound.h> -#ifndef __KERNEL__ -#error This API is an early revision and not enabled in the current -#error kernel release, it will be enabled in a future kernel version -#error with incompatible changes to what is here. -#endif - /* * Maximum number of channels topology kcontrol can represent. */ -- 2.9.3

1 0

[alsa-devel] [PATCH v2 0/4] ASoC: tas571x: fix of several issues for TAS5717
by Petr Kulhavy 06 Oct '16

06 Oct '16

An extensive usage of the TAS571x driver with the TAS5717 chip revealed several issues which in the first place caused audio to fail to play. This set of one-line and trivial patches fixes them. The issues were: * improper handling of the power-down signal * some timings not obeyed * channel mixer volume controls in the wrong chip structure ;-) Petr Kulhavy (4): ASoC: tas571x: extend the t_i2c time to comply with TAS5721 ASoC: tas571x: remove improper PDN signal usage in set_bias_level ASoC: tas571x: wait 50ms after oscillator trim ASoC: tas571x: move mixer volume controls from TAS5711 to TAS5717 sound/soc/codecs/tas571x.c | 37 ++++++++++++------------------------- 1 file changed, 12 insertions(+), 25 deletions(-) -- 2.7.4

2 5

[alsa-devel] [PATCH 0/5] ASoC: tas571x: fix of several issues for TAS5717
by Petr Kulhavy 05 Oct '16

05 Oct '16

An extensive usage of the TAS571x driver with the TAS5717 chip revealed several issues which in the first place caused audio to fail to play. This set of one-line and trivial patches fixes them. The issues were: * improper handling of the power-down signal * some timings not obeyed * bias level not saved in the driver * channel mixer volume controls in the wrong chip structure ;-) Petr Kulhavy (5): ASoC: tas571x: fix missing bias save level ASoC: tas571x: extend the t_i2c time to comply with TAS5721 ASoC: tas571x: remove improper PDN signal usage in set_bias_level ASoC: tas571x: wait 50ms after oscillator trim ASoC: tas571x: move mixer volume controls from TAS5711 to TAS5717 sound/soc/codecs/tas571x.c | 38 +++++++++++++------------------------- 1 file changed, 13 insertions(+), 25 deletions(-) -- 2.7.4

3 14

[alsa-devel] [PATCH] ASoC: Intel: report JACK_LINEOUT event
by sathyanarayana.nujella＠intel.com 05 Oct '16

05 Oct '16

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> This patch updates Jack type bitmask to include SND_JACK_LINEOUT while creating a new jack, so that LINEOUT events are reported properly. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 6532b8f..865a21e 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -130,8 +130,8 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) */ ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &broxton_headset, - NULL, 0); + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &broxton_headset, NULL, 0); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- 2.8.4

2 1

[alsa-devel] Applied "ASoC: wm_adsp: Remove duplicate set of kcontrol->iface" to the asoc tree
by Mark Brown 05 Oct '16

05 Oct '16

The patch ASoC: wm_adsp: Remove duplicate set of kcontrol->iface has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e23cd27da59cd2245fc4450f183e4c1223021eca Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Date: Mon, 3 Oct 2016 11:40:38 +0100 Subject: [PATCH] ASoC: wm_adsp: Remove duplicate set of kcontrol->iface Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b943dde8dbe5..7320fcacd86a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -967,7 +967,6 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); if (!kcontrol) return -ENOMEM; - kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; kcontrol->name = ctl->name; kcontrol->info = wm_coeff_info; -- 2.9.3

1 0

[alsa-devel] [PATCH] ASoC: sun4i-i2s: fix odd_ptr_err.cocci warnings
by Julia Lawall 05 Oct '16

05 Oct '16

sound/soc/sunxi/sun4i-i2s.c:633:5-11: inconsistent IS_ERR and PTR_ERR on line 635. PTR_ERR should normally access the value just tested by IS_ERR Generated by: scripts/coccinelle/tests/odd_ptr_err.cocci Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- Please take the patch only if it's a positive warning. Thanks! sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -632,7 +632,7 @@ static int sun4i_i2s_probe(struct platfo reset_apb = devm_reset_control_get(&pdev->dev, "apb_reset"); if (IS_ERR(reset_apb)) { dev_err(&pdev->dev, "Can't get apb reset\n"); - return PTR_ERR(i2s->mod_clk); + return PTR_ERR(reset_apb); } ret = reset_control_deassert(reset_apb);

2 1

[alsa-devel] [PATCH 0/6] Arizona FLL fixes and Clocking Improvements
by Charles Keepax 04 Oct '16

04 Oct '16

Hi, The first 3 patches in this series fix some small issues with the FLL configuration in the driver. The second 3 patches add support for requesting the input clocks for the CODEC through the clock framework. This is some what of an intermediate step until all the clocking itself in the CODEC can be brought under the clock framework but as there are machine drivers that want to be merged now with source clocks that require gating and difficult to resolve issues between the SPI and clock framework an intermediate solution seems warranted. Whilst the two halfs of this series do fairly different things, I have sent them as once series as they touch a lot of the same code, so this seemed the simpler approach to pushing them both through. The series is based of Mark's ASoC for-next tree. Thanks, Charles Charles Keepax (5): ASoC: arizona: Correct handling of FLL theta in synchroniser mode ASoC: arizona: Allow specification of base for arizona_is_enabled_fll ASoC: arizona: Avoid changing SYNC_ENA whilst the FLL_ENA is set ASoC: arizona: Add gating for clock when used for direct MCLK ASoC: arizona: Add gating for source clocks of the FLLs Sylwester Nawrocki (1): mfd: arizona: Add gating of external MCLKn clocks drivers/mfd/arizona-core.c | 30 +++++++++- include/linux/mfd/arizona/core.h | 9 +++ sound/soc/codecs/arizona.c | 124 +++++++++++++++++++++++++++++++++++---- sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/cs47l24.c | 6 +- sound/soc/codecs/wm5102.c | 9 ++- sound/soc/codecs/wm5110.c | 10 +++- sound/soc/codecs/wm8997.c | 9 ++- sound/soc/codecs/wm8998.c | 6 +- 9 files changed, 179 insertions(+), 26 deletions(-) -- 2.1.4

4 17

[alsa-devel] [PATCH 1/2] ASoC: wm9712: Convert to regmap
by Lars-Peter Clausen 04 Oct '16

04 Oct '16

Currently the wm9712 driver still uses custom snd_soc_codec_driver IO callbacks. This has been deprecated for a while, so convert the wm9712 driver to use regmap for its IO. Tested-by: Marek Vasut <marex(a)denx.de> Acked-by: Marek Vasut <marex(a)denx.de> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9712.c | 143 +++++++++++++++++++++++++--------------------- 2 files changed, 79 insertions(+), 65 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c67667b..f472254 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1057,6 +1057,7 @@ config SND_SOC_WM9705 config SND_SOC_WM9712 tristate + select REGMAP_AC97 config SND_SOC_WM9713 tristate diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 557709e..d2d0d2b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/kernel.h> #include <linux/device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -32,31 +33,66 @@ struct wm9712_priv { struct mutex lock; }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg); -static int ac97_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val); +static const struct reg_default wm9712_reg_defaults[] = { + { 0x02, 0x8000 }, + { 0x04, 0x8000 }, + { 0x06, 0x8000 }, + { 0x08, 0x0f0f }, + { 0x0a, 0xaaa0 }, + { 0x0c, 0xc008 }, + { 0x0e, 0x6808 }, + { 0x10, 0xe808 }, + { 0x12, 0xaaa0 }, + { 0x14, 0xad00 }, + { 0x16, 0x8000 }, + { 0x18, 0xe808 }, + { 0x1a, 0x3000 }, + { 0x1c, 0x8000 }, + { 0x20, 0x0000 }, + { 0x22, 0x0000 }, + { 0x26, 0x000f }, + { 0x28, 0x0605 }, + { 0x2a, 0x0410 }, + { 0x2c, 0xbb80 }, + { 0x2e, 0xbb80 }, + { 0x32, 0xbb80 }, + { 0x34, 0x2000 }, + { 0x4c, 0xf83e }, + { 0x4e, 0xffff }, + { 0x50, 0x0000 }, + { 0x52, 0x0000 }, + { 0x56, 0xf83e }, + { 0x58, 0x0008 }, + { 0x5c, 0x0000 }, + { 0x60, 0xb032 }, + { 0x62, 0x3e00 }, + { 0x64, 0x0000 }, + { 0x76, 0x0006 }, + { 0x78, 0x0001 }, + { 0x7a, 0x0000 }, +}; -/* - * WM9712 register cache - */ -static const u16 wm9712_reg[] = { - 0x6174, 0x8000, 0x8000, 0x8000, /* 6 */ - 0x0f0f, 0xaaa0, 0xc008, 0x6808, /* e */ - 0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */ - 0xe808, 0x3000, 0x8000, 0x0000, /* 1e */ - 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ - 0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */ - 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ - 0x0000, 0x2000, 0x0000, 0x0000, /* 3e */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 46 */ - 0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */ - 0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */ - 0x0008, 0x0000, 0x0000, 0x0000, /* 5e */ - 0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ - 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ - 0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */ +static bool wm9712_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AC97_REC_GAIN: + return true; + default: + return regmap_ac97_default_volatile(dev, reg); + } +} + +static const struct regmap_config wm9712_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x7e, + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm9712_volatile_reg, + + .reg_defaults = wm9712_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm9712_reg_defaults), }; #define HPL_MIXER 0x0 @@ -488,35 +524,13 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || - reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || - reg == AC97_REC_GAIN) - return soc_ac97_ops->read(wm9712->ac97, reg); - else { - reg = reg >> 1; - - if (reg >= (ARRAY_SIZE(wm9712_reg))) - return -EIO; - - return cache[reg]; - } + return snd_soc_read(codec, reg); } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - soc_ac97_ops->write(wm9712->ac97, reg, val); - reg = reg >> 1; - if (reg < (ARRAY_SIZE(wm9712_reg))) - cache[reg] = val; - - return 0; + return snd_soc_write(codec, reg, val); } static int ac97_prepare(struct snd_pcm_substream *substream, @@ -619,8 +633,7 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, static int wm9712_soc_resume(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - int i, ret; - u16 *cache = codec->reg_cache; + int ret; ret = snd_ac97_reset(wm9712->ac97, true, WM9712_VENDOR_ID, WM9712_VENDOR_ID_MASK); @@ -629,15 +642,8 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret == 0) { - /* Sync reg_cache with the hardware after cold reset */ - for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) { - if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || - (i > 0x58 && i != 0x5c)) - continue; - soc_ac97_ops->write(wm9712->ac97, i, cache[i>>1]); - } - } + if (ret == 0) + regcache_sync(codec->component.regmap); return ret; } @@ -645,6 +651,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) static int wm9712_soc_probe(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap; int ret; wm9712->ac97 = snd_soc_new_ac97_codec(codec, WM9712_VENDOR_ID, @@ -655,16 +662,28 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) return ret; } + regmap = regmap_init_ac97(wm9712->ac97, &wm9712_regmap_config); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + goto err_free_ac97_codec; + } + + snd_soc_codec_init_regmap(codec, regmap); + /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); return 0; +err_free_ac97_codec: + snd_soc_free_ac97_codec(wm9712->ac97); + return ret; } static int wm9712_soc_remove(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9712->ac97); return 0; } @@ -673,14 +692,8 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm9712 = { .probe = wm9712_soc_probe, .remove = wm9712_soc_remove, .resume = wm9712_soc_resume, - .read = ac97_read, - .write = ac97_write, .set_bias_level = wm9712_set_bias_level, .suspend_bias_off = true, - .reg_cache_size = ARRAY_SIZE(wm9712_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = wm9712_reg, .component_driver = { .controls = wm9712_snd_ac97_controls, -- 2.1.4

3 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Remove duplicate set of kcontrol->iface
by Richard Fitzgerald 03 Oct '16

03 Oct '16

Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b943dde..7320fca 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -962,17 +962,16 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) int ret; if (!ctl || !ctl->name) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); if (!kcontrol) return -ENOMEM; - kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; kcontrol->name = ctl->name; kcontrol->info = wm_coeff_info; kcontrol->get = wm_coeff_get; kcontrol->put = wm_coeff_put; kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; kcontrol->tlv.c = snd_soc_bytes_tlv_callback; kcontrol->private_value = (unsigned long)&ctl->bytes_ext; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: da7219: Improve pop/click performance for sensitive HPs
by Adam Thomson 03 Oct '16

03 Oct '16

Currently on some headsets slight pops can be heard during DAPM power-up/down. This can also be witnessed during the HP detect procedure. This patch addresses the issue by adjusting DAPM power sequencing slightly, the introduction of delays and use of minimum HP gain to avoid such noise artefacts. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Changes are based on top of commit: ASoC: da7219: Disable AAD if codec is not a wake-up source sound/soc/codecs/da7219-aad.c | 18 ++++-- sound/soc/codecs/da7219.c | 139 ++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/da7219.h | 5 ++ 3 files changed, 140 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 2b8914d..6274d79 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -204,10 +204,19 @@ static void da7219_aad_hptest_work(struct work_struct *work) snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL, DA7219_MIXOUT_R_AMP_EN_MASK, DA7219_MIXOUT_R_AMP_EN_MASK); - snd_soc_write(codec, DA7219_HP_L_CTRL, - DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK); - snd_soc_write(codec, DA7219_HP_R_CTRL, - DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK, + DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK, + DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + msleep(DA7219_SETTLING_DELAY); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_MUTE_EN_MASK | + DA7219_HP_L_AMP_MIN_GAIN_EN_MASK, 0); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_MUTE_EN_MASK | + DA7219_HP_R_AMP_MIN_GAIN_EN_MASK, 0); /* * If we're running from the internal oscillator then give audio paths @@ -244,6 +253,7 @@ static void da7219_aad_hptest_work(struct work_struct *work) regcache_mark_dirty(da7219->regmap); regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL, DA7219_HP_R_CTRL); + msleep(DA7219_SETTLING_DELAY); regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL, DA7219_MIXOUT_R_CTRL); regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L, diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 65f7e98..7c26ee3 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -823,6 +823,85 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, } } +static int da7219_settling_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + msleep(DA7219_SETTLING_DELAY); + break; + default: + break; + } + + return 0; +} + +static int da7219_mixout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + u8 hp_ctrl, min_gain_mask; + + switch (w->reg) { + case DA7219_MIXOUT_L_CTRL: + hp_ctrl = DA7219_HP_L_CTRL; + min_gain_mask = DA7219_HP_L_AMP_MIN_GAIN_EN_MASK; + break; + case DA7219_MIXOUT_R_CTRL: + hp_ctrl = DA7219_HP_R_CTRL; + min_gain_mask = DA7219_HP_R_AMP_MIN_GAIN_EN_MASK; + break; + default: + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + /* Enable minimum gain on HP to avoid pops */ + snd_soc_update_bits(codec, hp_ctrl, min_gain_mask, + min_gain_mask); + + msleep(DA7219_MIN_GAIN_DELAY); + + break; + case SND_SOC_DAPM_POST_PMU: + /* Remove minimum gain on HP */ + snd_soc_update_bits(codec, hp_ctrl, min_gain_mask, 0); + + break; + } + + return 0; +} + +static int da7219_gain_ramp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_PRE_PMD: + /* Ensure nominal gain ramping for DAPM sequence */ + da7219->gain_ramp_ctrl = + snd_soc_read(codec, DA7219_GAIN_RAMP_CTRL); + snd_soc_write(codec, DA7219_GAIN_RAMP_CTRL, + DA7219_GAIN_RAMP_RATE_NOMINAL); + break; + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + /* Restore previous gain ramp settings */ + snd_soc_write(codec, DA7219_GAIN_RAMP_CTRL, + da7219->gain_ramp_ctrl); + break; + } + + return 0; +} + /* * DAPM Widgets @@ -906,30 +985,46 @@ static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { ARRAY_SIZE(da7219_st_out_filtr_mix_controls)), /* DACs */ - SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT, - DA7219_NO_INVERT), - SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT, - DA7219_NO_INVERT), + SND_SOC_DAPM_DAC_E("DACL", NULL, DA7219_DAC_L_CTRL, + DA7219_DAC_L_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("DACR", NULL, DA7219_DAC_R_CTRL, + DA7219_DAC_R_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Output PGAs */ - SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL, - DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL, - DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL, - DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), - SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL, - DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA_E("Mixout Left PGA", DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mixout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Mixout Right PGA", DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mixout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY_S("Headphone Left PGA", 1, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S("Headphone Right PGA", 1, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Output Supplies */ - SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT, - DA7219_NO_INVERT, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 0, DA7219_CP_CTRL, + DA7219_CP_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU), /* Outputs */ SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), + + /* Pre/Post Power */ + SND_SOC_DAPM_PRE("Pre Power Gain Ramp", da7219_gain_ramp_event), + SND_SOC_DAPM_POST("Post Power Gain Ramp", da7219_gain_ramp_event), }; @@ -1002,8 +1097,8 @@ static const struct snd_soc_dapm_route da7219_audio_map[] = { {"Mixout Left PGA", NULL, "DACL"}, {"Mixout Right PGA", NULL, "DACR"}, - {"Headphone Left PGA", NULL, "Mixout Left PGA"}, - {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + {"HPL", NULL, "Mixout Left PGA"}, + {"HPR", NULL, "Mixout Right PGA"}, {"HPL", NULL, "Headphone Left PGA"}, {"HPR", NULL, "Headphone Right PGA"}, @@ -1711,6 +1806,14 @@ static int da7219_probe(struct snd_soc_codec *codec) DA7219_HP_R_AMP_RAMP_EN_MASK, DA7219_HP_R_AMP_RAMP_EN_MASK); + /* Default minimum gain on HP to avoid pops during DAPM sequencing */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_MIN_GAIN_EN_MASK, + DA7219_HP_L_AMP_MIN_GAIN_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_MIN_GAIN_EN_MASK, + DA7219_HP_R_AMP_MIN_GAIN_EN_MASK); + /* Default infinite tone gen, start/stop by Kcontrol */ snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 66d3bad..6baba74 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -777,6 +777,10 @@ #define DA7219_SYS_STAT_CHECK_RETRIES 6 #define DA7219_SYS_STAT_CHECK_DELAY 50 +/* Power up/down Delays */ +#define DA7219_SETTLING_DELAY 40 +#define DA7219_MIN_GAIN_DELAY 30 + enum da7219_clk_src { DA7219_CLKSRC_MCLK = 0, DA7219_CLKSRC_MCLK_SQR, @@ -814,6 +818,7 @@ struct da7219_priv { bool master; bool alc_en; + u8 gain_ramp_ctrl; }; #endif /* __DA7219_H */ -- 1.9.3

1 0

[alsa-devel] [RFC PATCH v2 0/2] support DP MST audio
by libin.yang＠linux.intel.com 01 Oct '16

01 Oct '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patchset starts to support DP MST audio. This patchset is based on drm-intel-nightly on git://anongit.freedesktop.org/drm-intel Libin Yang (2): ALSA: hda - add DP mst verb support ALSA: hda - add DP mst audio support sound/pci/hda/hda_codec.c | 77 +++++++++++++++- sound/pci/hda/hda_codec.h | 3 + sound/pci/hda/patch_hdmi.c | 216 ++++++++++++++++++++++++++++++++++----------- 3 files changed, 241 insertions(+), 55 deletions(-) -- 1.9.1

3 5

[alsa-devel] [GIT PULL] ASoC updates for v4.9
by Mark Brown 30 Sep '16

30 Sep '16

The following changes since commit 08895a8b6b06ed2323cd97a36ee40a116b3db8ed: Linux 4.8-rc8 (2016-09-25 18:47:13 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.9 for you to fetch changes up to 513e43efafe329dad7b5794583b67ac898dcbdca: Merge remote-tracking branches 'asoc/topic/tpa6130a2', 'asoc/topic/ux500', 'asoc/topic/wm8960', 'asoc/topic/wm8962' and 'asoc/topic/wm8991' into asoc-next (2016-09-29 12:44:49 -0700) ---------------------------------------------------------------- ASoC: Updates for v4.9 Apart from the cleanups done by Morimoto-san this has very much been a driver focused release with very little generic change: - A big factoring out of the simple-card code to allow it to be shared more with the rcar generic card from Kuninori Morimoto. - Removal of some operations duplicated on the CODEC level, again by Kuninori Morimoto. - Lots more machine support for x86 systems. - New drivers for Nuvoton NAU88C10, Realtek RT5660 and RT5663. ---------------------------------------------------------------- Adam Thomson (9): ASoC: da7213: Improve driver efficiency with regards to MCLK usage ASoC: da7213: Refactor sysclk(), pll() functions to improve handling ASoC: da7213: Improve 32KHz mode PLL locking ASoC: da7219: Make more efficient use of MCLK within driver ASoC: da7218: Remove 32KHz PLL mode from driver ASoC: da7218: Improve driver efficiency with regards to MCLK usage ASoC: da7219: Support HP detect procedure when MCLK not present ASoC: da7219: Reset codec gracefully, if still active ASoC: da7219: Disable AAD if codec is not a wake-up source Arnaud Pouliquen (2): ASoC: sti: suppress inappropriate DT fields ASoC: sti: suppress inappropriate fields for sti sound card Arnd Bergmann (3): ASoC: rockchip: use SPI dependency for rt5514 ASoC: rt5663: refine error handling ASoC: cq93vc: duplicated callback function goes to component Axel Lin (4): ASoC: cs53l30: Constify cs53l30_mclk_coeffs and cs53l30_mclkx_coeffs tables ASoC: cs53l30: Constify cs53l30_mclk_coeffs and cs53l30_mclkx_coeffs tables ASoC: cs42l73: Constify cs42l73_mclk_coeffs and cs42l73_mclkx_coeffs tables ASoC: nau8810: Fix memory leak in nau8810_eq_put error path Baoyou Xie (3): ASoC: atmel-pdmic: mark symbols static where possible ASoC: mediatek: mark symbols static where possible ASoC: fix W=1 build warnings Bard Liao (1): ASoC: add rt5663 codec driver Bhaktipriya Shridhar (1): ASoC: tlv320dac33: Remove deprecated create_singlethread_workqueue Charles Keepax (13): ASoC: arizona: Connect ASRCs to both clock domains ASoC: arizona: Correct handling of FLL theta in synchroniser mode ASoC: arizona: Allow specification of base for arizona_is_enabled_fll ASoC: arizona: Avoid changing SYNC_ENA whilst the FLL_ENA is set ASoC: wm_adsp: Separate concept of booted and running ASoC: wm_adsp: Move control cache sync out of preloader ASoC: arizona: Attach SYSCLK to DSP preloaders ASoC: wm_adsp: Make DSP preloader a supply widget ASoC: wm_adsp: Use usleep_range for short delay ASoC: wm_adsp: Put DSP into low power state between loading and running ASoC: wm_adsp: Allow preloader to do the final shutdown of the DSP ASoC: arizona: Add debug prints for output power up/down times ASoC: arizona: Add output power up/down delays for speaker path Chen-Yu Tsai (3): ASoC: dapm: Fix possible uninitialized variable in snd_soc_dapm_get_volsw() ASoC: dapm: Fix value setting for _ENUM_DOUBLE MUX's second channel ASoC: dapm: Fix kcontrol creation for output driver widget Cheng-Yi Chiang (1): ASoC: rockchip: Support headset button function on rk3399 Christophe JAILLET (1): ASoC: omap-pcm: off-by-one in 'omap_pcm_limit_supported_formats' Colin Ian King (13): ASoC: Intel: sst: fix ix spelling mistake: "capablities" -> "capabilites" ASoC: tas5086: fix typo: "Inavlid" -> "Invalid" ASoC: wm8962: fix spelling mistake "mesurement" -> "measurement" ASoC: hdac_hdmi: fix spelling mistake "montior" -> "monitor" ASoC: fsl_esai: fix spelling mistake "Transmition" -> "Transmission" ASoC: ux500: fix spelling mistake "Unsopported" -> "Unsupported" ASoC: Intel: sst: fix to spelling mistake: "susupend" -> "suspend" ASoC: Intel: Skylake: add missing \n to end of dev_* messages ASoC: max98926: add missing \n to end of dev_err/dev_dbg messages ASoC: samsung: add missing \n to end of pr_err messages ASoC: Intel: atom: add missing \n to end of dev_err/dev_dbg messages ASoC: Intel: remove status, it is shadowing status of a higher scope ASoC: Intel: Skylake: fix memory leak of module on error exit path Danny Milosavljevic (3): ASoC: sun4i-codec: Rename some sun7i-only registers ASoC: sun4i-codec: Add custom regmap configs ASoC: sun4i-codec: Rename sun4i_codec_widgets for consistency Dharageswari R (6): ASoC: Intel: Skylake: modify snd_skl_get_module_info args ASoC: Intel: Skylake: Unload all the loadable modules ASoC: Intel: Skylake: Add module instance id generation APIs ASoC: Intel: Skylake: Use private instance id of modules in IPC ASoC: Intel: Skylake: Update to use instance ids generated ASoC: Intel: Skylake: Add table for module id for quick ref Fabio Estevam (4): ASoC: fsl_sai: Use 'np' variable ASoC: fsl_asrc: Propagate the real error code ASoC: fsl_asrc: Remove unneeded driver registration message ASoC: fsl_asrc: Use np variable Guneshwor Singh (2): ALSA - hda: Add support for parsing new HDA capabilities ALSA - hda: Add support for link audio time reporting Harsha Priya (1): ASoC: Intel: Atom: Fix message handling during drop stream Hsin-Yu Chao (1): ASoC: da7219: software reset codec at probe Irina Tirdea (2): ASoC: Intel: bytcr_rt5640: add MCLK support AsoC: Intel: Add quirks for MinnowBoard MAX Jaswinder Jassal (1): ASoC: core: fix shift used for second item in snd_soc_get_enum_double Jeeja KP (2): ASoC: Intel: Skylake: Fix DMA control config size ASoC: dpcm: Don't apply symmetry for BE with hw param fixup John Hsu (3): ASoC: nau8810: Add driver for Nuvoton codec chip NAU88C10 ASoC: nau8810: fix compile warning in loopback switch control ASoC: nau8825: fix bug in FLL parameter John Keeping (2): ASoC: rt5677: Add ACPI support ASoC: Intel: boards: Add bdw-rt5677 machine driver Julia Lawall (5): ASoC: use of_property_read_bool ASoC: constify snd_soc_codec_driver structures ASoC: constify snd_pcm_ops structures ASoC: ac97: constify gpio_chip structures ASoC: constify gpio_chip structures Kranthi G (1): ASoC: Intel: Skylake: add support for tplg manifest load Kuninori Morimoto (203): ASoC: simple-card: use asoc_simple_card_parse_clk() ASoC: rsrc-card: use asoc_simple_card_parse_clk() ASoC: simple-card-utils: add asoc_simple_card_parse_clk() ASoC: simple-card-utils: add asoc_simple_card_parse_dai() ASoC: simple-card: use asoc_simple_card_parse_dai() ASoC: rsrc-card: use asoc_simple_card_parse_dai() ASoC: simple-card: remove asoc_simple_card_sub_parse_of() ASoC: hdmi-codec: enable multi probe for same device ASoC: codec duplicated callback function goes to component on sun4i-codec ASoC: codec duplicated callback function goes to component on pistachio-internal-dac ASoC: remove codec duplicated callback function ASoC: codec duplicated callback function goes to component on wm8985 ASoC: codec duplicated callback function goes to component on wm8978 ASoC: codec duplicated callback function goes to component on wm8974 ASoC: codec duplicated callback function goes to component on wm8903 ASoC: codec duplicated callback function goes to component on wm8804 ASoC: codec duplicated callback function goes to component on wm8776 ASoC: codec duplicated callback function goes to component on wm8770 ASoC: codec duplicated callback function goes to component on wm8753 ASoC: codec duplicated callback function goes to component on wm8750 ASoC: codec duplicated callback function goes to component on wm8741 ASoC: codec duplicated callback function goes to component on wm8737 ASoC: codec duplicated callback function goes to component on wm8731 ASoC: codec duplicated callback function goes to component on wm8728 ASoC: codec duplicated callback function goes to component on wm8711 ASoC: codec duplicated callback function goes to component on wm8580 ASoC: codec duplicated callback function goes to component on wm8523 ASoC: codec duplicated callback function goes to component on wm8510 ASoC: codec duplicated callback function goes to component on tlv320aic3x ASoC: codec duplicated callback function goes to component on tlv320aic31xx ASoC: codec duplicated callback function goes to component on tlv320aic23 ASoC: codec duplicated callback function goes to component on tfa9879 ASoC: codec duplicated callback function goes to component on tas5720 ASoC: codec duplicated callback function goes to component on tas571x ASoC: codec duplicated callback function goes to component on tas5086 ASoC: codec duplicated callback function goes to component on tas2552 ASoC: codec duplicated callback function goes to component on sti-sas ASoC: codec duplicated callback function goes to component on sta350 ASoC: codec duplicated callback function goes to component on sta32x ASoC: codec duplicated callback function goes to component on ssm4567 ASoC: codec duplicated callback function goes to component on ssm2602 ASoC: codec duplicated callback function goes to component on spdif_transmitter ASoC: codec duplicated callback function goes to component on spdif_receiver ASoC: codec duplicated callback function goes to component on sgtl5000 ASoC: codec duplicated callback function goes to component on rt5631 ASoC: codec duplicated callback function goes to component on rt5616 ASoC: codec duplicated callback function goes to component on pcm512x ASoC: codec duplicated callback function goes to component on pcm3168a ASoC: codec duplicated callback function goes to component on pcm179x ASoC: codec duplicated callback function goes to component on pcm1681 ASoC: codec duplicated callback function goes to component on max9860 ASoC: codec duplicated callback function goes to component on inno_rk3036 ASoC: codec duplicated callback function goes to component on gtm601 ASoC: codec duplicated callback function goes to component on es8328 ASoC: codec duplicated callback function goes to component on cs53l30 ASoC: codec duplicated callback function goes to component on cs4349 ASoC: codec duplicated callback function goes to component on cs42xx8 ASoC: codec duplicated callback function goes to component on cs42l73 ASoC: codec duplicated callback function goes to component on cs42l56 ASoC: codec duplicated callback function goes to component on cs42l52 ASoC: codec duplicated callback function goes to component on cs42l51 ASoC: codec duplicated callback function goes to component on cs4271 ASoC: codec duplicated callback function goes to component on cs4270 ASoC: codec duplicated callback function goes to component on cs4265 ASoC: codec duplicated callback function goes to component on cs35l33 ASoC: codec duplicated callback function goes to component on cs35l32 ASoC: codec duplicated callback function goes to component on bt-sco ASoC: codec duplicated callback function goes to component on ak5386 ASoC: codec duplicated callback function goes to component on ak4642 ASoC: codec duplicated callback function goes to component on ak4613 ASoC: codec duplicated callback function goes to component on ak4554 ASoC: codec duplicated callback function goes to component on ak4104 ASoC: codec duplicated callback function goes to component on adau7002 ASoC: codec duplicated callback function goes to component on adau1701 ASoC: codec duplicated callback function goes to component on ac97 ASoC: codec duplicated callback function goes to component on ad1980 ASoC: codec duplicated callback function goes to component on ad73311 ASoC: codec duplicated callback function goes to component on adau1373 ASoC: codec duplicated callback function goes to component on ad1836 ASoC: codec duplicated callback function goes to component on ad193x ASoC: codec duplicated callback function goes to component on 88pm860x-codec ASoC: codec duplicated callback function goes to component on atmel-classd ASoC: codec duplicated callback function goes to component on atmel-pdmic ASoC: codec duplicated callback function goes to component on ab8500-codec ASoC: codec duplicated callback function goes to component on adau1761 ASoC: codec duplicated callback function goes to component on adau1781 ASoC: codec duplicated callback function goes to component on adau1977 ASoC: codec duplicated callback function goes to component on adav80x ASoC: codec duplicated callback function goes to component on ads117x ASoC: codec duplicated callback function goes to component on ak4535 ASoC: codec duplicated callback function goes to component on ak4641 ASoC: codec duplicated callback function goes to component on ak4671 ASoC: codec duplicated callback function goes to component on alc5632 ASoC: codec duplicated callback function goes to component on cs47l24 ASoC: codec duplicated callback function goes to component on cx20442 ASoC: codec duplicated callback function goes to component on da7210 ASoC: codec duplicated callback function goes to component on da7213 ASoC: codec duplicated callback function goes to component on da7218 ASoC: codec duplicated callback function goes to component on da7219 ASoC: codec duplicated callback function goes to component on da732x ASoC: codec duplicated callback function goes to component on da9055 ASoC: codec duplicated callback function goes to component on dmic ASoC: codec duplicated callback function goes to component on hdmi-codec ASoC: codec duplicated callback function goes to component on isabelle ASoC: codec duplicated callback function goes to component on jz4740 ASoC: codec duplicated callback function goes to component on lm49453 ASoC: codec duplicated callback function goes to component on wm98088 ASoC: codec duplicated callback function goes to component on wm98095 ASoC: codec duplicated callback function goes to component on wm98357a ASoC: codec duplicated callback function goes to component on wm9850 ASoC: codec duplicated callback function goes to component on wm9867 ASoC: codec duplicated callback function goes to component on max98925 ASoC: codec duplicated callback function goes to component on max98926 ASoC: codec duplicated callback function goes to component on mc13783 ASoC: codec duplicated callback function goes to component on ml26124 ASoC: codec duplicated callback function goes to component on nau8825 ASoC: codec duplicated callback function goes to component on pcm3008 ASoC: codec duplicated callback function goes to component on rt286 ASoC: codec duplicated callback function goes to component on rt298 ASoC: codec duplicated callback function goes to component on rt5514 ASoC: codec duplicated callback function goes to component on rt5640 ASoC: codec duplicated callback function goes to component on rt5645 ASoC: codec duplicated callback function goes to component on rt5651 ASoC: codec duplicated callback function goes to component on rt5659 ASoC: codec duplicated callback function goes to component on rt5670 ASoC: codec duplicated callback function goes to component on rt5677 ASoC: codec duplicated callback function goes to component on si476x ASoC: codec duplicated callback function goes to component on sn95031 ASoC: codec duplicated callback function goes to component on ssm2518 ASoC: codec duplicated callback function goes to component on sta529 ASoC: codec duplicated callback function goes to component on stac9766 ASoC: codec duplicated callback function goes to component on tlv320aic26 ASoC: codec duplicated callback function goes to component on tlv320aic32x4 ASoC: codec duplicated callback function goes to component on tlv320dac33 ASoC: codec duplicated callback function goes to component on twl4030 ASoC: codec duplicated callback function goes to component on twl6040 ASoC: codec duplicated callback function goes to component on uda134x ASoC: codec duplicated callback function goes to component on uda1380 ASoC: codec duplicated callback function goes to component on wl1273 ASoC: codec duplicated callback function goes to component on wm0010 ASoC: codec duplicated callback function goes to component on wm1250-ev1 ASoC: codec duplicated callback function goes to component on wm2000 ASoC: codec duplicated callback function goes to component on wm2200 ASoC: codec duplicated callback function goes to component on wm5100 ASoC: codec duplicated callback function goes to component on wm5102 ASoC: codec duplicated callback function goes to component on wm5110 ASoC: codec duplicated callback function goes to component on wm8350 ASoC: codec duplicated callback function goes to component on wm8400 ASoC: codec duplicated callback function goes to component on wm8727 ASoC: codec duplicated callback function goes to component on wm8782 ASoC: codec duplicated callback function goes to component on wm8900 ASoC: codec duplicated callback function goes to component on wm8940 ASoC: codec duplicated callback function goes to component on wm8955 ASoC: codec duplicated callback function goes to component on wm8961 ASoC: codec duplicated callback function goes to component on wm8971 ASoC: codec duplicated callback function goes to component on wm8983 ASoC: codec duplicated callback function goes to component on wm8988 ASoC: codec duplicated callback function goes to component on wm8990 ASoC: codec duplicated callback function goes to component on wm8991 ASoC: codec duplicated callback function goes to component on wm8995 ASoC: codec duplicated callback function goes to component on wm8996 ASoC: codec duplicated callback function goes to component on wm8997 ASoC: codec duplicated callback function goes to component on wm8998 ASoC: codec duplicated callback function goes to component on wm9081 ASoC: codec duplicated callback function goes to component on wm9705 ASoC: codec duplicated callback function goes to component on wm9712 ASoC: codec duplicated callback function goes to component on wm9713 ASoC: lpass-platform: don't use snd_soc_pcm_set_drvdata() ASoC: remove snd_soc_pcm_set/get_drvdata() ASoC: simple-card-utils: add asoc_simple_card_init_dai() ASoC: simple-card: use asoc_simple_card_init_dai() ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() ASoC: rsrc-card: use asoc_simple_card_init_dai() ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() ASoC: simple-card-utils: add asoc_simple_card_canonicalize_cpu() ASoC: simple-card: use asoc_simple_card_canonicalize_cpu() ASoC: rsrc-card: use asoc_simple_card_canonicalize_cpu() ASoC: simple-card-utils: add asoc_simple_card_clean_reference() ASoC: simple-card: use asoc_simple_card_clean_reference() ASoC: rsrc-card: use asoc_simple_card_clean_reference() ASoC: rsrc-card: remove board specific compatible ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase1 ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase2 ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase3 ASoC: simple-scu-card: remove unused definitions ASoC: simple-scu-card: is GPL v2 ASoC: simple-scu-card: tidyup codec daifmt handling method ASoC: simple-scu-card: tidyup asoc_simple_card_parse_links() method ASoC: simple-scu-card: depends on CONFIG_OF ASoC: simple-scu-card: add 2 CPU 1 Codec example ASoC: simple-scu-card: code sync: move asoc_simple_card_of_match ASoC: simple-scu-card: code sync: tidyup TDM setting position ASoC: simple-card: call of_node_put() for dai_link ASoC: simple-card: tidyup mclk-fs of_property_read_u32() usage ASoC: simple-card: code sync: tidyup simple_priv_to_xxx() macro ASoC: simple-card: code sync: use simple_priv_to_props() macro ASoC: simple-card: is GPL v2 ASoC: simple-card: code sync: tidyup white line ASoC: simple-card: code sync: use tag ASoC: simple-card: code sync: rename num_link to num ASoC: simple-card: use kzalloc() for dai_props / dai_link ASoC: rsnd: add SNDRV_PCM_TRIGGER_SUSPEND/RESUME Marcel Ziswiler (1): ASoC: tegra: add tegra sgtl5000 machine driver Marcus Cooper (2): ASoC: sunxi: Add bindings for sun6i to SPDIF ASoC: sunxi: compatibility for sun6i to SPDIF Marek Vasut (6): ASoC: fsl_asrc: use flat regmap cache ASoC: fsl_esai: use flat regmap cache ASoC: fsl_spdif: use flat regmap cache ASoC: fsl_ssi: use flat regmap cache ASoC: fsl_ssi: Remove .num_reg_defaults_raw from regmap_config ASoC: trivial: system spelling fix Mark Brown (30): Merge branch 'fix/simple' of git://git.kernel.org/.../broonie/sound into asoc-simple Merge branch 'topic/hda-link-time' of git://git.kernel.org/.../tiwai/sound into asoc-intel Merge branch 'topic/rt5640' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'fix/omap' of git://git.kernel.org/.../broonie/sound into asoc-omap Merge branch 'topic/simple' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge tag 'asoc-fix-v4.8-rc4' into asoc-arizona Merge branch 'topic/rt5663' of git://git.kernel.org/.../broonie/sound into asoc-rt5660 ASoC: hdac_hdmi: Drop use of audio component framework to read ELD Merge branch 'topic/codec-component' of git://git.kernel.org/.../broonie/sound into asoc-sunxi Merge branch 'fix/da7219' of git://git.kernel.org/.../broonie/sound into asoc-da7219 ASoC: fsl: Fix lockups with recent cache changes Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/da7219', 'asoc/fix/nau8825', 'asoc/fix/rt5514' and 'asoc/fix/shift' into asoc-linus Merge remote-tracking branch 'asoc/fix/tpa6130a2' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/ak4104', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/codec-component' into asoc-next Merge remote-tracking branches 'asoc/topic/cs35l30', 'asoc/topic/cs42l73', 'asoc/topic/cs53l30' and 'asoc/topic/da7213' into asoc-next Merge remote-tracking branches 'asoc/topic/da7218', 'asoc/topic/da7219' and 'asoc/topic/dpcm' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/fsl', 'asoc/topic/gpio-chip' and 'asoc/topic/hdmi' into asoc-next Merge remote-tracking branches 'asoc/topic/max98371', 'asoc/topic/max9867', 'asoc/topic/max98926' and 'asoc/topic/mtk' into asoc-next Merge remote-tracking branches 'asoc/topic/nau8810', 'asoc/topic/of-bool', 'asoc/topic/omap' and 'asoc/topic/platform-drvdata' into asoc-next Merge remote-tracking branches 'asoc/topic/qcom', 'asoc/topic/rockchip' and 'asoc/topic/rt5616' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5659', 'asoc/topic/rt5660', 'asoc/topic/rt5677' and 'asoc/topic/samsung' into asoc-next Merge remote-tracking branches 'asoc/topic/sgtl5000', 'asoc/topic/simple', 'asoc/topic/stac9766', 'asoc/topic/sti' and 'asoc/topic/sunxi' into asoc-next Merge remote-tracking branches 'asoc/topic/tas5086', 'asoc/topic/tegra', 'asoc/topic/tlv320aic31xx', 'asoc/topic/tlv320dac33' and 'asoc/topic/topology' into asoc-next Merge remote-tracking branches 'asoc/topic/tpa6130a2', 'asoc/topic/ux500', 'asoc/topic/wm8960', 'asoc/topic/wm8962' and 'asoc/topic/wm8991' into asoc-next Markus Elfring (1): ASoC: max98371: Delete unnecessary assignment for the field "owner" Mengdong Lin (3): ASoC: topology: ABI - Add the types for BE DAI ASoC: topology: Add support for configuring existing BE DAIs ASoC: topology: ABI - Add sig_bits to stream caps Nariman Poushin (1): ASoC: arizona: Don't change the FLLn_GAIN before entering FREERUN Nicolas Iooss (2): ASoC: simple-card-utils: add __printf attribute ASoC: Intel: Atom: add a missing star in a memcpy call Nicolin Chen (5): ASoC: cs53l30: Do not ignore errors if mclk is specified ASoC: rt5659: Add mclk controls ASoC: tegra_rt5640: Correct a copy and paste typo in the comments ASoC: rt5659: Enable IRQ output for GPIO1 pin ASoC: rt5659: Fix incorrect register addresses Nikita Yushchenko (5): ASoC: tpa6130a2: fix volume setting when no stream is running ASoC: simple-card: add support for aux devices ASoC: tpa6130a2: unmerge power enable error path from power disable path ASoC: tlv320aic31xx: add explicit support for tlv320dac31xx ASoC: tlv320aic31xx: do not declare support for mono DAI Oder Chiou (2): ASoC: rt5514: make the volume TLV to match the units 0.01dB ASoC: rt5660: add rt5660 codec driver Peter Meerwald-Stadler (2): ASoC: dapm: Fix typos in comment ASoC: atmel-pdmic: Fix typos in comments and error messages Peter Robinson (1): ASoC: sunxi: depend on ARCH_SUNXI Peter Ujfalusi (2): ASoC: omap-mcbsp: Enable TX/RX under and overflow interrupts ASoC: davinci-mcasp: off-by-one in davinci_mcasp_hw_rule_format() Pierre-Louis Bossart (17): ASoC: rt5640: add internal clock source support ASoC: Intel: bytcr-rt5640: add Asus T100TAF quirks ASoC: Intel: bytcr_rt5640: quirk for mono speaker ASoC: Intel: bytcr_rt5640: enable differential mic quirk ASoC: Intel: Atom: auto-detection of Baytrail-CR ASoC: Intel: Atom: add definitions for modem/SSP0 interface ASoC: Intel: atom: enable configuration of SSP0 ASoC: Intel: bytcr_rt5640: add SSP2_AIF2 routing ASoC: Intel: bytcr_rt56040: additional routing quirks ASoC: Intel: bytcr_rt5640: fix dai/clock setup for SSP0 routing ASoC: Intel: bytcr_rt5640: default routing and quirks on Baytrail-CR ASoC: Intel: bytcr_rt5640: add IN3 map ASoC: Intel: bytcr_rt5640: Add quirk for Teclast X98 Air 3G tablet ASoC: Intel: bytcr_rt5640: log quirks ASoC: Intel: atom: fix 0-day warnings ASoC: Intel: bytcr_rt5640: quirk for Acer Aspire SWS-012 ASoC: Intel: Atom: fix IOSF_MBI dependency Ramesh Babu (2): ASoC: Intel: Skylake: Add library loading IPCs ASoC: Intel: Skylake: Add library loading support Randy Li (1): ASoC: exynos: organize the asoc audio into a menu Richard Fitzgerald (1): ASoC: arizona: Wait for resume before enabling FLL Richard Leitner (1): ASoC: sgtl5000: add headphone and LINEOUT mute controls Samaga Krishna (1): ASoC: Intel: Skylake: Add 32bit support Sandeep Tayal (1): ASoC: hdac_hdmi: use audio component framework to read ELD Senthilnathan Veppur (2): ASoC: Intel: Skylake: add additional args to module parsing ASoC: Intel: Skylake: Add module processing domain support Shreyas NC (4): ASoC: uapi: Intel: Skylake: Define vendor specific tokens ASoC: Intel: Skylake: Parse vendor tokens to build module data ASoC: Intel: Skylake: Remove dfw config and associated structures ASoC: Intel: Skylake: Parse manifest data Srinivas Kandagatla (2): ASoC: qcom: apq8016-sbc: add board specific dapm mic widgets ASoC: qcom: apq8016-sbc: add support to routing via DT Sugar Zhang (2): ASoC: rockchip: i2s: restore register during runtime_suspend/resume cycle ASoC: rockchip: spdif: restore register during runtime_suspend/resume cycle Sylwester Nawrocki (15): ASoC: dwc: Drop DMA channel names assignment ASoC: samsung: Drop usage of struct s3c_dma_params from i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from s3c24xx-i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from s3c2412-i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from ac97.c ASoC: samsung: Drop usage of struct s3c_dma_params from spdif.c ASoC: samsung: Drop usage of of struct s3c_dma_params from pcm.c ASoC: samsung: Remove unused now unused struct s3c_dma_params ASoC: L3 bus: Add default gpio ops ARM: S3C24XX: Specify audio codec platform_data for mini2440 board ASoC: uda134x: Optionally initialize L3 ops to default GPIO ops ASoC: s3c24xx_uda134x: Remove unused power() callback ASoC: s3c24xx_uda134x: Drop initialization of codec's platform data ASoC: samsung: Convert s3c24xx_uda134x to use devm_snd_soc_register_card() ASoC: samsung: s3c24xx_uda134x: debug/error trace cleanup Takashi Sakamoto (11): ASoC: rt5616: add static qualifier for file local symbols ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array ASoC: sst-bxt-rt298: fix obsoleted initializers for array ASOC: tpa6130a2: add static qualifier for file local symbols ASoC: wm8991: fix wrong usage of DECLARE_TLV_DB_LINEAR() ASoC: wm8991: remove unused variable ASoC: stac9766: fix wrong usage of DECLARE_TLV_DB_LINEAR() ASoC: da7219: fix inappropriate condition statement ASoC: max9867: remove usage of obsoleted TLV-related macro ASoC: rt5616: remove usage of obsoleted TLV-related macro ASoC: wm8960: remove usage of obsoleted TLV-related macro Vinod Koul (13): ASoC: Intel: Skylake: Move modules query to runtime ASoC: Intel: Skylake: modify skl_get_dsp_ops() ASoC: Intel: Skylake: split fw and dsp initialization ASoC: Intel: Skylake: Parse UUIDs once ASoC: Intel: Bxt: Parse UUIDs once ASoC: Intel: Skylake: Fix a comment style ALSA: hda - move bus_parse_capabilities to core ALSA: convert users to core bus_parse_capabilities ALSA - Ext hda: remove bus_parse_capabilities ASoC: Intel: Skylake: Populate modules after loading ASoC: Intel: Skylake: remove module id query at runtime ASoC: Intel: Skylake: check manifest size ASoC: Intel: Skylake: Fix the inverted logic check Vishal Thanki (1): ASoC: ak4104: Limit the supported sample rates Wei Yongjun (6): ASoC: atmel-pdmic: add missing clk_disable_unprepare() on error in atmel_pdmic_cpu_dai_startup() ASoC: topology: Fix error return code in soc_tplg_dapm_widget_create() ASoC: omap-abe-twl6040: Use devm_snd_soc_register_card() ASoC: samsung: i2s: Add missing clk_disable_unprepare() on error in samsung_i2s_probe() ASoC: sti: fix missing clk_disable_unprepare() on error in uni_player_start() ASoC: rt5663: fix sparse warnings Wonjoon Lee (1): ASoC: rk3399_gru_sound: fix recording pop at first attempt Xing Zheng (2): ASoC: rockchip: Add machine driver for RK3399 GRU Boards ASoC: rockchip: Add support rt5514 dsp summy dailink Yong Zhi (1): ASoC: Intel: Add DMIC channel constraint for bxt machine kbuild test robot (2): ASoC: Intel: Skylake: fix noderef.cocci warnings ASoC: rt5663: fix platform_no_drv_owner.cocci warnings 彭东林 (1): ASoC: dpcm: print dai_link name of BE other than FE. .../devicetree/bindings/sound/nau8810.txt | 16 + .../bindings/sound/nvidia,tegra-audio-sgtl5000.txt | 42 + .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 30 + .../bindings/sound/renesas,rsrc-card.txt | 75 - .../bindings/sound/rockchip,rk3399-gru-sound.txt | 22 + Documentation/devicetree/bindings/sound/rt5659.txt | 3 + Documentation/devicetree/bindings/sound/rt5660.txt | 47 + Documentation/devicetree/bindings/sound/rt5663.txt | 30 + .../devicetree/bindings/sound/simple-card.txt | 37 + .../devicetree/bindings/sound/simple-scu-card.txt | 110 + .../devicetree/bindings/sound/st,sti-asoc-card.txt | 52 +- .../bindings/sound/sunxi,sun4i-spdif.txt | 3 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 9 +- arch/arm/mach-s3c24xx/mach-mini2440.c | 19 + include/sound/da7219.h | 2 + include/sound/hda_register.h | 36 + include/sound/hdaudio.h | 13 + include/sound/hdaudio_ext.h | 12 - include/sound/l3.h | 15 +- include/sound/rt5660.h | 31 + include/sound/s3c24xx_uda134x.h | 1 - include/sound/simple_card_utils.h | 35 + include/sound/soc.h | 19 - include/uapi/sound/Kbuild | 1 + include/uapi/sound/asoc.h | 35 +- include/uapi/sound/snd_sst_tokens.h | 214 ++ sound/hda/ext/hdac_ext_controller.c | 91 +- sound/hda/ext/hdac_ext_stream.c | 46 +- sound/hda/hdac_controller.c | 75 + sound/pci/hda/hda_controller.c | 203 +- sound/pci/hda/hda_controller.h | 3 + sound/pci/hda/hda_intel.c | 17 + sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/atmel/atmel-classd.c | 6 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel-pdmic.c | 20 +- sound/soc/codecs/88pm860x-codec.c | 14 +- sound/soc/codecs/Kconfig | 18 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ab8500-codec.c | 24 +- sound/soc/codecs/ac97.c | 10 +- sound/soc/codecs/ad1836.c | 14 +- sound/soc/codecs/ad193x.c | 14 +- sound/soc/codecs/ad1980.c | 14 +- sound/soc/codecs/ad73311.c | 10 +- sound/soc/codecs/adau1373.c | 14 +- sound/soc/codecs/adau1701.c | 15 +- sound/soc/codecs/adau1761.c | 14 +- sound/soc/codecs/adau1781.c | 14 +- sound/soc/codecs/adau1977.c | 14 +- sound/soc/codecs/adau7002.c | 10 +- sound/soc/codecs/adav80x.c | 14 +- sound/soc/codecs/ads117x.c | 10 +- sound/soc/codecs/ak4104.c | 15 +- sound/soc/codecs/ak4535.c | 14 +- sound/soc/codecs/ak4554.c | 10 +- sound/soc/codecs/ak4613.c | 14 +- sound/soc/codecs/ak4641.c | 14 +- sound/soc/codecs/ak4642.c | 14 +- sound/soc/codecs/ak4671.c | 14 +- sound/soc/codecs/ak5386.c | 10 +- sound/soc/codecs/alc5632.c | 14 +- sound/soc/codecs/arizona.c | 78 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/bt-sco.c | 10 +- sound/soc/codecs/cq93vc.c | 6 +- sound/soc/codecs/cs35l32.c | 15 +- sound/soc/codecs/cs35l33.c | 15 +- sound/soc/codecs/cs4265.c | 15 +- sound/soc/codecs/cs4270.c | 14 +- sound/soc/codecs/cs4271.c | 14 +- sound/soc/codecs/cs42l51.c | 14 +- sound/soc/codecs/cs42l52.c | 15 +- sound/soc/codecs/cs42l56.c | 15 +- sound/soc/codecs/cs42l73.c | 19 +- sound/soc/codecs/cs42xx8.c | 14 +- sound/soc/codecs/cs4349.c | 15 +- sound/soc/codecs/cs47l24.c | 26 +- sound/soc/codecs/cs53l30.c | 23 +- sound/soc/codecs/cx20442.c | 10 +- sound/soc/codecs/da7210.c | 15 +- sound/soc/codecs/da7213.c | 145 +- sound/soc/codecs/da7213.h | 12 +- sound/soc/codecs/da7218.c | 49 +- sound/soc/codecs/da7218.h | 2 - sound/soc/codecs/da7219-aad.c | 108 +- sound/soc/codecs/da7219-aad.h | 11 +- sound/soc/codecs/da7219.c | 95 +- sound/soc/codecs/da7219.h | 8 + sound/soc/codecs/da732x.c | 14 +- sound/soc/codecs/da9055.c | 15 +- sound/soc/codecs/dmic.c | 10 +- sound/soc/codecs/es8328.c | 14 +- sound/soc/codecs/gtm601.c | 10 +- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/codecs/hdmi-codec.c | 79 +- sound/soc/codecs/inno_rk3036.c | 14 +- sound/soc/codecs/isabelle.c | 14 +- sound/soc/codecs/jz4740.c | 14 +- sound/soc/codecs/l3.c | 71 +- sound/soc/codecs/lm49453.c | 14 +- sound/soc/codecs/max98088.c | 14 +- sound/soc/codecs/max98095.c | 14 +- sound/soc/codecs/max98357a.c | 10 +- sound/soc/codecs/max98371.c | 1 - sound/soc/codecs/max9850.c | 14 +- sound/soc/codecs/max9860.c | 14 +- sound/soc/codecs/max9867.c | 19 +- sound/soc/codecs/max98925.c | 14 +- sound/soc/codecs/max98926.c | 20 +- sound/soc/codecs/mc13783.c | 14 +- sound/soc/codecs/ml26124.c | 14 +- sound/soc/codecs/nau8810.c | 884 ++++++ sound/soc/codecs/nau8810.h | 281 ++ sound/soc/codecs/nau8825.c | 16 +- sound/soc/codecs/pcm1681.c | 14 +- sound/soc/codecs/pcm179x.c | 14 +- sound/soc/codecs/pcm3008.c | 10 +- sound/soc/codecs/pcm3168a.c | 14 +- sound/soc/codecs/pcm512x.c | 14 +- sound/soc/codecs/rt286.c | 14 +- sound/soc/codecs/rt298.c | 14 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/rt5514.c | 20 +- sound/soc/codecs/rt5514.h | 4 +- sound/soc/codecs/rt5616.c | 23 +- sound/soc/codecs/rt5631.c | 14 +- sound/soc/codecs/rt5640.c | 17 +- sound/soc/codecs/rt5640.h | 1 + sound/soc/codecs/rt5645.c | 14 +- sound/soc/codecs/rt5651.c | 14 +- sound/soc/codecs/rt5659.c | 41 +- sound/soc/codecs/rt5659.h | 7 +- sound/soc/codecs/rt5660.c | 1353 ++++++++ sound/soc/codecs/rt5660.h | 847 ++++++ sound/soc/codecs/rt5663.c | 3218 ++++++++++++++++++++ sound/soc/codecs/rt5663.h | 1121 +++++++ sound/soc/codecs/rt5670.c | 14 +- sound/soc/codecs/rt5677.c | 69 +- sound/soc/codecs/sgtl5000.c | 17 +- sound/soc/codecs/si476x.c | 10 +- sound/soc/codecs/sn95031.c | 14 +- sound/soc/codecs/spdif_receiver.c | 10 +- sound/soc/codecs/spdif_transmitter.c | 10 +- sound/soc/codecs/ssm2518.c | 14 +- sound/soc/codecs/ssm2602.c | 14 +- sound/soc/codecs/ssm4567.c | 14 +- sound/soc/codecs/sta32x.c | 14 +- sound/soc/codecs/sta350.c | 14 +- sound/soc/codecs/sta529.c | 6 +- sound/soc/codecs/stac9766.c | 14 +- sound/soc/codecs/sti-sas.c | 8 +- sound/soc/codecs/tas2552.c | 14 +- sound/soc/codecs/tas5086.c | 16 +- sound/soc/codecs/tas571x.c | 14 +- sound/soc/codecs/tas5720.c | 14 +- sound/soc/codecs/tfa9879.c | 15 +- sound/soc/codecs/tlv320aic23.c | 14 +- sound/soc/codecs/tlv320aic26.c | 14 +- sound/soc/codecs/tlv320aic31xx.c | 218 +- sound/soc/codecs/tlv320aic31xx.h | 2 + sound/soc/codecs/tlv320aic32x4.c | 14 +- sound/soc/codecs/tlv320aic3x.c | 14 +- sound/soc/codecs/tlv320dac33.c | 31 +- sound/soc/codecs/tpa6130a2.c | 49 +- sound/soc/codecs/twl4030.c | 14 +- sound/soc/codecs/twl6040.c | 14 +- sound/soc/codecs/uda134x.c | 17 +- sound/soc/codecs/uda1380.c | 14 +- sound/soc/codecs/wl1273.c | 14 +- sound/soc/codecs/wm0010.c | 12 +- sound/soc/codecs/wm1250-ev1.c | 13 +- sound/soc/codecs/wm2000.c | 16 +- sound/soc/codecs/wm2200.c | 16 +- sound/soc/codecs/wm5100.c | 18 +- sound/soc/codecs/wm5102.c | 27 +- sound/soc/codecs/wm5110.c | 28 +- sound/soc/codecs/wm8350.c | 16 +- sound/soc/codecs/wm8400.c | 16 +- sound/soc/codecs/wm8510.c | 16 +- sound/soc/codecs/wm8523.c | 16 +- sound/soc/codecs/wm8580.c | 16 +- sound/soc/codecs/wm8711.c | 16 +- sound/soc/codecs/wm8727.c | 12 +- sound/soc/codecs/wm8728.c | 16 +- sound/soc/codecs/wm8731.c | 16 +- sound/soc/codecs/wm8737.c | 16 +- sound/soc/codecs/wm8741.c | 12 +- sound/soc/codecs/wm8750.c | 16 +- sound/soc/codecs/wm8753.c | 16 +- sound/soc/codecs/wm8770.c | 16 +- sound/soc/codecs/wm8776.c | 16 +- sound/soc/codecs/wm8782.c | 12 +- sound/soc/codecs/wm8804.c | 10 +- sound/soc/codecs/wm8900.c | 16 +- sound/soc/codecs/wm8903.c | 18 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 16 +- sound/soc/codecs/wm8955.c | 16 +- sound/soc/codecs/wm8960.c | 7 +- sound/soc/codecs/wm8961.c | 16 +- sound/soc/codecs/wm8962.c | 6 +- sound/soc/codecs/wm8971.c | 16 +- sound/soc/codecs/wm8974.c | 16 +- sound/soc/codecs/wm8978.c | 16 +- sound/soc/codecs/wm8983.c | 16 +- sound/soc/codecs/wm8985.c | 16 +- sound/soc/codecs/wm8988.c | 14 +- sound/soc/codecs/wm8990.c | 16 +- sound/soc/codecs/wm8991.c | 44 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8995.c | 14 +- sound/soc/codecs/wm8996.c | 18 +- sound/soc/codecs/wm8997.c | 16 +- sound/soc/codecs/wm8998.c | 26 +- sound/soc/codecs/wm9081.c | 16 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9705.c | 16 +- sound/soc/codecs/wm9712.c | 16 +- sound/soc/codecs/wm9713.c | 16 +- sound/soc/codecs/wm_adsp.c | 88 +- sound/soc/codecs/wm_adsp.h | 7 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/dwc/designware_i2s.c | 2 - sound/soc/fsl/fsl_asrc.c | 12 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_esai.c | 8 +- sound/soc/fsl/fsl_sai.c | 6 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/simple-card-utils.c | 141 + sound/soc/generic/simple-card.c | 305 +- sound/soc/generic/simple-scu-card.c | 345 +++ sound/soc/img/pistachio-internal-dac.c | 14 +- sound/soc/intel/Kconfig | 12 + sound/soc/intel/atom/sst-atom-controls.c | 32 +- sound/soc/intel/atom/sst-atom-controls.h | 6 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst.c | 5 +- sound/soc/intel/atom/sst/sst_acpi.c | 97 +- sound/soc/intel/atom/sst/sst_ipc.c | 3 + sound/soc/intel/atom/sst/sst_pvt.c | 14 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5677.c | 347 +++ sound/soc/intel/boards/bxt_da7219_max98357a.c | 75 +- sound/soc/intel/boards/bxt_rt298.c | 14 +- sound/soc/intel/boards/bytcr_rt5640.c | 507 ++- sound/soc/intel/common/sst-acpi.c | 1 + sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- sound/soc/intel/skylake/bxt-sst.c | 118 +- sound/soc/intel/skylake/skl-messages.c | 67 +- sound/soc/intel/skylake/skl-pcm.c | 60 +- sound/soc/intel/skylake/skl-sst-cldma.c | 4 +- sound/soc/intel/skylake/skl-sst-dsp.h | 18 +- sound/soc/intel/skylake/skl-sst-ipc.c | 41 +- sound/soc/intel/skylake/skl-sst-ipc.h | 12 +- sound/soc/intel/skylake/skl-sst-utils.c | 176 +- sound/soc/intel/skylake/skl-sst.c | 49 +- sound/soc/intel/skylake/skl-topology.c | 918 +++++- sound/soc/intel/skylake/skl-topology.h | 15 +- sound/soc/intel/skylake/skl-tplg-interface.h | 95 +- sound/soc/intel/skylake/skl.c | 6 +- sound/soc/intel/skylake/skl.h | 2 + sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 5 +- sound/soc/omap/mcbsp.c | 3 +- sound/soc/omap/omap-abe-twl6040.c | 14 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 20 + sound/soc/qcom/lpass-platform.c | 16 +- sound/soc/qcom/lpass.h | 2 + sound/soc/rockchip/Kconfig | 12 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 397 +++ sound/soc/rockchip/rockchip_i2s.c | 10 +- sound/soc/rockchip/rockchip_spdif.c | 12 +- sound/soc/samsung/Kconfig | 57 +- sound/soc/samsung/ac97.c | 34 +- sound/soc/samsung/dma.h | 20 +- sound/soc/samsung/dmaengine.c | 31 - sound/soc/samsung/i2s.c | 56 +- sound/soc/samsung/idma.c | 1 - sound/soc/samsung/pcm.c | 26 +- sound/soc/samsung/s3c-i2s-v2.c | 3 +- sound/soc/samsung/s3c-i2s-v2.h | 4 +- sound/soc/samsung/s3c2412-i2s.c | 24 +- sound/soc/samsung/s3c24xx-i2s.c | 24 +- sound/soc/samsung/s3c24xx_uda134x.c | 134 +- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - sound/soc/samsung/spdif.c | 12 +- sound/soc/sh/Kconfig | 6 - sound/soc/sh/rcar/Makefile | 3 - sound/soc/sh/rcar/core.c | 4 + sound/soc/sh/rcar/rsrc-card.c | 486 --- sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/soc-ac97.c | 2 +- sound/soc/soc-core.c | 15 +- sound/soc/soc-dapm.c | 23 +- sound/soc/soc-ops.c | 2 +- sound/soc/soc-pcm.c | 13 +- sound/soc/soc-topology.c | 115 +- sound/soc/soc-utils.c | 2 +- sound/soc/sti/sti_uniperif.c | 130 +- sound/soc/sti/uniperif.h | 28 +- sound/soc/sti/uniperif_player.c | 83 +- sound/soc/sti/uniperif_reader.c | 46 +- sound/soc/sunxi/Kconfig | 1 + sound/soc/sunxi/sun4i-codec.c | 60 +- sound/soc/sunxi/sun4i-spdif.c | 17 +- sound/soc/tegra/Kconfig | 11 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 212 ++ sound/soc/ux500/ux500_msp_dai.c | 6 +- 318 files changed, 15145 insertions(+), 3112 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nau8810.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt delete mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5660.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5663.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-scu-card.txt create mode 100644 include/sound/rt5660.h create mode 100644 include/uapi/sound/snd_sst_tokens.h create mode 100644 sound/soc/codecs/nau8810.c create mode 100644 sound/soc/codecs/nau8810.h create mode 100644 sound/soc/codecs/rt5660.c create mode 100644 sound/soc/codecs/rt5660.h create mode 100644 sound/soc/codecs/rt5663.c create mode 100644 sound/soc/codecs/rt5663.h create mode 100644 sound/soc/generic/simple-scu-card.c create mode 100644 sound/soc/intel/boards/bdw-rt5677.c create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c delete mode 100644 sound/soc/sh/rcar/rsrc-card.c create mode 100644 sound/soc/tegra/tegra_sgtl5000.c

2 1

[alsa-devel] [PATCH v2 0/5][alsa-lib] ctl: sync UAPI header for TLV-related macros and apply misc changes
by Takashi Sakamoto 30 Sep '16

30 Sep '16

Hi, This patchset updates my former one: [alsa-devel] [PATCH 0/4][alsa-lib] ctl: sync UAPI header for TLV-related macros and apply misc changes http://mailman.alsa-project.org/pipermail/alsa-devel/2016-September/112853.… In this month, UAPI header of 4.9-pre kernel newly include existent macros related to TLV operation for user land. This is to allow applications to register TLV information to element set which they added. This patchset is for related changes in ALSA user land library. The macros are put into 'include/sound/tlv.h'. However corresponding macros are not put into 'include/control.h' yet because currently I have no better idea for library APIs to construct tlv information. This patchset also improves some corrections for documentation which committed with my misunderstandings. Takashi Sakamoto (5): Update include/sound/tlv.h from 4.9-pre kernel uapi test: use actual information for TLV operation ctl: improve API documentation for TLV operation ctl: improve documentation about TLV-related APIs ctl: correct documentation about TLV feature include/sound/tlv.h | 77 ++++++++++++++++ src/control/control.c | 47 +++++++--- test/user-ctl-element-set.c | 214 ++++++++++++++++++++++++++++++++++++-------- 3 files changed, 289 insertions(+), 49 deletions(-) -- 2.7.4

2 6

[alsa-devel] Alsa-lib strings?
by David Henningsson 30 Sep '16

30 Sep '16

Hi, I'm making some rust bindings for alsa-lib [1] and Rust is quite picky about strings; i e, requires them to be UTF-8. 1a) can I count on alsa-lib's strings (everything from card name and mixer identifiers to various descriptions) to be UTF-8 or perhaps even 7-bit? If not, what encoding is to be expected? 1b) In what way can buggy drivers violate the answers to 1a? 2) Can I count on the strings to be zero terminated? E g, I looked at snd_ctl_elem_id_set_name <http://www.alsa-project.org/alsa-doc/alsa-lib/group___control.html#ga813d02…> that sets all 44 characters without ensuring a zero at the last character, but snd_ctl_elem_id_get_name <http://www.alsa-project.org/alsa-doc/alsa-lib/group___control.html#gaa6cfea…> does not say anything about a max length, it just gives out a *const char. The question applies to all other strings as well. // David [1] https://docs.rs/alsa/

3 2

[alsa-devel] [PATCH] ASoC: fsl: Fix lockups with recent cache changes
by Mark Brown 29 Sep '16

29 Sep '16

The recent series of changes to the caching in the SSI driver have caused a number of problems to appear in some test systems. These are still not fully understood but we're coming up to the merge window so for now let's revert commit 7de2763d9b3 (ASoC: fsl_ssi: Remove .num_reg_defaults_raw from regmap_config) as backing that out seems to resolve the problem on affected systems. Reported-by: Maciej S. Szmigiero" <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: Marek Vasut <marex(a)denx.de> --- sound/soc/fsl/fsl_ssi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b73c102f1cf2..50349437d961 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -177,6 +177,7 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, + .num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1, .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg, @@ -1500,6 +1501,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) * don't have SACC{ST,EN,DIS} regs. */ regconfig.max_register = CCSR_SSI_SRMSK; + regconfig.num_reg_defaults_raw = + CCSR_SSI_SRMSK / sizeof(uint32_t) + 1; } ret = of_property_match_string(np, "clock-names", "ipg"); -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix memory leak of module on error exit path
by Colin King 29 Sep '16

29 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Currently there is a memory leak of module on a ENOMEM return path. Fix this by kfree'ing module before returning. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/skl-sst-utils.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index d3120ba..8dc0303 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -354,8 +354,10 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, module->max_instance = mod_entry->instance_max_count; size = sizeof(int) * mod_entry->instance_max_count; module->instance_id = devm_kzalloc(ctx->dev, size, GFP_KERNEL); - if (!module->instance_id) + if (!module->instance_id) { + kfree(module); return -ENOMEM; + } list_add_tail(&module->list, &skl->uuid_list); -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: rsnd: add SNDRV_PCM_TRIGGER_SUSPEND/RESUME
by Kuninori Morimoto 29 Sep '16

29 Sep '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch adds SNDRV_PCM_TRIGGER_SUSPEND/RESUME. Otherwise, it breaks rsnd driver internal start/stop counter when suspend/resume. This issue was reported/tested by Hiep Tested-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f718a20..f181410 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -576,6 +576,7 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: rsnd_dai_stream_init(io, substream); ret = rsnd_dai_call(init, io, priv); @@ -592,6 +593,7 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, break; case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: ret = rsnd_dai_call(irq, io, priv, 0); ret |= rsnd_dai_call(stop, io, priv); -- 1.9.1

2 1

[alsa-devel] [alsa-lib][PATCHv2] ucm: docs: typeset lists of identifiers explicitly
by Antonio Ospite 29 Sep '16

29 Sep '16

Doxygen doesn't preserve formatting that relies only on indentation, typeset lists of identifiers explicitly. This makes the HTML docs a lot more readable. This change comes along the lines of commit 72aa0f8332fb ("ucm: reformat snd_use_case_get() doc"). Some TABs has been added too in order to preserve the aligned look when reading the source code. Signed-off-by: Antonio Ospite <ao2(a)ao2.it> --- Changes since v1: * fix a typo where a '=' (equal) was used instead of a '-' (minus) at the start of one line. Thanks, Antonio include/use-case.h | 57 +++++++++++++++++++++++++++--------------------------- 1 file changed, 29 insertions(+), 28 deletions(-) diff --git a/include/use-case.h b/include/use-case.h index 9aac6e2..8911645 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -192,20 +192,21 @@ int snd_use_case_free_list(const char *list[], int items); * \return Number of list entries if success, otherwise a negative error code * * Defined identifiers: - * NULL - get card list - * (in pair cardname+comment) - * _verbs - get verb list - * (in pair verb+comment) - * _devices[/{verb}] - get list of supported devices - * (in pair device+comment) - * _modifiers[/{verb}]- get list of supported modifiers - * (in pair modifier+comment) - * TQ[/{verb}] - get list of TQ identifiers - * _enadevs - get list of enabled devices - * _enamods - get list of enabled modifiers + * - NULL - get card list + * (in pair cardname+comment) + * - _verbs - get verb list + * (in pair verb+comment) + * - _devices[/{verb}] - get list of supported devices + * (in pair device+comment) + * - _modifiers[/{verb}] - get list of supported modifiers + * (in pair modifier+comment) + * - TQ[/{verb}] - get list of TQ identifiers + * - _enadevs - get list of enabled devices + * - _enamods - get list of enabled modifiers + * + * - _supporteddevs/{modifier}|{device}[/{verb}] - list of supported devices + * - _conflictingdevs/{modifier}|{device}[/{verb}] - list of conflicting devices * - * _supporteddevs/{modifier}|{device}[/{verb}] - list of supported devices - * _conflictingdevs/{modifier}|{device}[/{verb}] - list of conflicting devices * Note that at most one of the supported/conflicting devs lists has * any entries, and when neither is present, all devices are supported. * @@ -331,8 +332,8 @@ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, * \return Zero if success, otherwise a negative error code * * Known identifiers: - * _devstatus/{device} - return status for given device - * _modstatus/{modifier} - return status for given modifier + * - _devstatus/{device} - return status for given device + * - _modstatus/{modifier} - return status for given modifier */ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, const char *identifier, @@ -346,19 +347,19 @@ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, * \return Zero if success, otherwise a negative error code * * Known identifiers: - * _verb - set current verb = value - * _enadev - enable given device = value - * _disdev - disable given device = value - * _swdev/{old_device} - new_device = value - * - disable old_device and then enable new_device - * - if old_device is not enabled just return - * - check transmit sequence firstly - * _enamod - enable given modifier = value - * _dismod - disable given modifier = value - * _swmod/{old_modifier} - new_modifier = value - * - disable old_modifier and then enable new_modifier - * - if old_modifier is not enabled just return - * - check transmit sequence firstly + * - _verb - set current verb = value + * - _enadev - enable given device = value + * - _disdev - disable given device = value + * - _swdev/{old_device} - new_device = value + * - disable old_device and then enable new_device + * - if old_device is not enabled just return + * - check transmit sequence firstly + * - _enamod - enable given modifier = value + * - _dismod - disable given modifier = value + * - _swmod/{old_modifier} - new_modifier = value + * - disable old_modifier and then enable new_modifier + * - if old_modifier is not enabled just return + * - check transmit sequence firstly */ int snd_use_case_set(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 2.9.3

2 1

[alsa-devel] [alsa-lib][PATCH] ucm: fix crash when calling snd_use_case_geti() with no device or modifier
by Antonio Ospite 29 Sep '16

29 Sep '16

When calling snd_use_case_geti(uc_mgr, "_devstatus", &lvalue) the code ends up calling device_status(uc_mgr, NULL), which result in a crash in strcmp(dev->name, NULL), when there are enabled devices. This happens because snd_use_case_geti() allows a "_devstatus" identifier even if it's only supposed to allow the form "_devstatus/{device}". So check that the device name is not null. The same issue occurs with "_modstatus", this change fixes that as well. Signed-off-by: Antonio Ospite <ao2(a)ao2.it> --- Hi, the bug can be reproduced with a command like this: # alsaucm -n -b - <<EOM open bytcr-rt5640 reset set _verb HiFi set _device Speaker geti _devstatus EOM Segmentation fault I decided to add the check once per command instead of doing this: @@ -1525,7 +1525,8 @@ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, goto __end; } } else { - str = NULL; + err = -EINVAL; + goto __end; } because the function body seems to be prepared to accept other forms of identifiers, possibly even without a "prefix/suffix" structure. Ciao ciao, Antonio P.S. the source code in src/ucm has a mixed indentation style of TABs and spaces, would you accept a patch to uniform the style? src/ucm/main.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/src/ucm/main.c b/src/ucm/main.c index 24d9510..d5e418e 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1528,12 +1528,20 @@ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, str = NULL; } if (check_identifier(identifier, "_devstatus")) { + if(!str) { + err = -EINVAL; + goto __end; + } err = device_status(uc_mgr, str); if (err >= 0) { *value = err; err = 0; } } else if (check_identifier(identifier, "_modstatus")) { + if(!str) { + err = -EINVAL; + goto __end; + } err = modifier_status(uc_mgr, str); if (err >= 0) { *value = err; -- 2.9.3

2 1

[alsa-devel] [PATCH v2 0/5] ASOC: fix sparse warnings
by Takashi Sakamoto 29 Sep '16

29 Sep '16

Hi, This patchset updates my previous one, and for Linux 4.8-rc8. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-July/110466.html This is to fix some sparse warnings in ALSA SoC part. Below items are going to be fixed: * fix a condition statement with a logical operation * purge old way to initialize array * complement missing 'static' and 'const' qualifier for file local symbols Takashi Sakamoto (5): ASOC: da7219: fix inappropriate condition statement ASOC: sst-bxt-rt298: fix obsoleted initializers for array ASOC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array ASOC: rt5616: add static qualifier for file local symbols ASOC: tpa6130a2: add static qualifier for file local symbols sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/rt5616.c | 4 ++-- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 ++++++------ sound/soc/intel/boards/bxt_rt298.c | 14 +++++++------- 5 files changed, 17 insertions(+), 17 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: cs4341: Add driver for CS4341 CODEC
by Alexander Shiyan 29 Sep '16

29 Sep '16

Initial commit of the Cirrus Logic CS4341. This is a very simple, playback only, stereo DAC. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/Kconfig | 7 ++ sound/soc/codecs/Makefile | 1 + sound/soc/codecs/cs4341.c | 281 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 289 insertions(+) create mode 100644 sound/soc/codecs/cs4341.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1cd6ab3..9853e79 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -458,6 +458,13 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +config SND_SOC_CS4341 + tristate "Cirrus Logic CS4341 CODEC" + depends on SPI_MASTER + select REGMAP_SPI + help + Enable support for Cirrus Logic (Crystal) CS4341 CODEC. + # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 58036af..f699105 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -50,6 +50,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4341-objs := cs4341.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o snd-soc-cs53l30-objs := cs53l30.o diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c new file mode 100644 index 0000000..74907ad --- /dev/null +++ b/sound/soc/codecs/cs4341.c @@ -0,0 +1,281 @@ +/* + * Cirrus Logic CS4341A ALSA SoC Codec Driver + * + * Author: Alexander Shiyan <shc_work(a)mail.ru>, 2016 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +/* + * CS4341 registers: + * High byte - SPI chip address (0x20) + * Low byte - Memory Address Pointer + */ +#define CS4341_REG_MODE1 0x00 +#define CS4341_REG_MODE2 0x01 +#define CS4341_REG_MIX 0x02 +#define CS4341_REG_VOLA 0x03 +#define CS4341_REG_VOLB 0x04 + +#define CS4341_MODE2_DIF (7 << 4) +#define CS4341_MODE2_DIF_I2S_24 (0 << 4) +#define CS4341_MODE2_DIF_I2S_16 (1 << 4) +#define CS4341_MODE2_DIF_LJ_24 (2 << 4) +#define CS4341_MODE2_DIF_RJ_24 (3 << 4) +#define CS4341_MODE2_DIF_RJ_16 (5 << 4) + +#define CS4341_VOLX_MUTE (1 << 7) + +struct cs4341_priv { + unsigned int fmt; + struct regmap *regmap; + struct regmap_config regcfg; +}; + +static const struct reg_default cs4341_reg_defaults[] = { + { CS4341_REG_MODE1, 0x00 }, + { CS4341_REG_MODE2, 0x82 }, + { CS4341_REG_MIX, 0x49 }, + { CS4341_REG_VOLA, 0x80 }, + { CS4341_REG_VOLB, 0x80 }, +}; + +static bool cs4341_reg_readable(struct device *dev, unsigned int reg) +{ + return false; +} + +static int cs4341_set_fmt(struct snd_soc_dai *dai, unsigned int format) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs4341_priv *cs4341 = snd_soc_codec_get_drvdata(codec); + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + goto fmt_err; + } + + switch (format & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + goto fmt_err; + } + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4341->fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + return 0; + default: + break; + } + +fmt_err: + dev_err(codec->dev, "Invalid DAI format 0x%08x\n", format); + + return -EINVAL; +} + +static int cs4341_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 cs4341_priv *cs4341 = snd_soc_codec_get_drvdata(codec); + int bits24 = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + bits24 = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + dev_err(codec->dev, "Unsupported PCM format\n"); + return -EINVAL; + } + + switch (cs4341->fmt) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, CS4341_REG_MODE2, CS4341_MODE2_DIF, + bits24 ? CS4341_MODE2_DIF_I2S_24 : + CS4341_MODE2_DIF_I2S_16); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, CS4341_REG_MODE2, CS4341_MODE2_DIF, + CS4341_MODE2_DIF_LJ_24); + break; + case SND_SOC_DAIFMT_RIGHT_J: + snd_soc_update_bits(codec, CS4341_REG_MODE2, CS4341_MODE2_DIF, + bits24 ? CS4341_MODE2_DIF_RJ_24 : + CS4341_MODE2_DIF_RJ_16); + break; + default: + dev_err(codec->dev, "Unsupported DAI format\n"); + return -EINVAL; + } + + return 0; +} + +static int cs4341_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int ret; + + ret = snd_soc_update_bits(codec, CS4341_REG_VOLA, CS4341_VOLX_MUTE, + mute ? CS4341_VOLX_MUTE : 0); + if (ret < 0) + return ret; + + return snd_soc_update_bits(codec, CS4341_REG_VOLB, CS4341_VOLX_MUTE, + mute ? CS4341_VOLX_MUTE : 0); +} + +static DECLARE_TLV_DB_SCALE(out_tlv, -9000, 100, 0); + +static const char *deemph[] = { + "Disabled", "44.1 kHz", "48 kHz", "32 kHz", +}; + +static const struct soc_enum deemph_enum = + SOC_ENUM_SINGLE(CS4341_REG_MODE2, 2, 4, deemph); + +static const char *srzc[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static const struct soc_enum srzc_enum = + SOC_ENUM_SINGLE(CS4341_REG_MIX, 5, 4, srzc); + + +static const struct snd_soc_dapm_widget cs4341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OutA"), + SND_SOC_DAPM_OUTPUT("OutB"), +}; + +static const struct snd_soc_dapm_route cs4341_routes[] = { + {"OutA", NULL, "HiFi DAC"}, + {"OutB", NULL, "HiFi DAC"}, + {"DAC Playback", NULL, "OutA"}, + {"DAC Playback", NULL, "OutB"}, +}; + +static const struct snd_kcontrol_new cs4341_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4341_REG_VOLA, CS4341_REG_VOLB, 0, 90, 1, out_tlv), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", srzc_enum), + SOC_SINGLE("Auto-Mute Switch", CS4341_REG_MODE2, 7, 1, 0), + SOC_SINGLE("Popguard Transient Control", CS4341_REG_MODE2, 1, 1, 0), +}; + +static const struct snd_soc_dai_ops cs4341_dai_ops = { + .set_fmt = cs4341_set_fmt, + .hw_params = cs4341_hw_params, + .digital_mute = cs4341_digital_mute, +}; + +static struct snd_soc_dai_driver cs4341_dai = { + .name = "cs4341a-hifi", + .symmetric_rates = 1, + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &cs4341_dai_ops, +}; + +static struct snd_soc_codec_driver soc_codec_cs4341 = { + .dapm_widgets = cs4341_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4341_dapm_widgets), + .dapm_routes = cs4341_routes, + .num_dapm_routes = ARRAY_SIZE(cs4341_routes), + .controls = cs4341_controls, + .num_controls = ARRAY_SIZE(cs4341_controls), +}; + +static int cs4341_spi_probe(struct spi_device *spi) +{ + struct cs4341_priv *cs4341; + int i, ret; + + cs4341 = devm_kzalloc(&spi->dev, sizeof(*cs4341), GFP_KERNEL); + if (!cs4341) + return -ENOMEM; + + spi->bits_per_word = spi->bits_per_word ? : 8; + spi->max_speed_hz = spi->max_speed_hz ? : 6000000; + ret = spi_setup(spi); + if (ret) + return ret; + + cs4341->regcfg.reg_bits = 16; + cs4341->regcfg.val_bits = 8; + cs4341->regcfg.write_flag_mask = 0x20; + cs4341->regcfg.max_register = CS4341_REG_VOLB; + cs4341->regcfg.cache_type = REGCACHE_FLAT; + cs4341->regcfg.readable_reg = cs4341_reg_readable; + cs4341->regcfg.reg_defaults = cs4341_reg_defaults; + cs4341->regcfg.num_reg_defaults = ARRAY_SIZE(cs4341_reg_defaults); + cs4341->regmap = devm_regmap_init_spi(spi, &cs4341->regcfg); + if (IS_ERR(cs4341->regmap)) + return PTR_ERR(cs4341->regmap); + + spi_set_drvdata(spi, cs4341); + + for (i = 0; i < ARRAY_SIZE(cs4341_reg_defaults); i++) + regmap_write(cs4341->regmap, cs4341_reg_defaults[i].reg, + cs4341_reg_defaults[i].def); + + return snd_soc_register_codec(&spi->dev, &soc_codec_cs4341, &cs4341_dai, 1); +} + +static int cs4341_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + + return 0; +} + +static const struct of_device_id __maybe_unused cs4341_dt_ids[] = { + { .compatible = "cirrus,cs4341a", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4341_dt_ids); + +static struct spi_driver cs4341_spi_driver = { + .driver = { + .name = "cs4341", + .of_match_table = of_match_ptr(cs4341_dt_ids), + }, + .probe = cs4341_spi_probe, + .remove = cs4341_spi_remove, +}; +module_spi_driver(cs4341_spi_driver); + +MODULE_LICENSE("GPL"); +MODULE_AUTHOR("Alexander Shiyan <shc_work(a)mail.ru>"); +MODULE_DESCRIPTION("Cirrus Logic CS4341 ALSA SoC Codec Driver"); -- 2.7.3

3 5

[alsa-devel] [PATCH 1/5] ASoC: fsl_asrc: use flat regmap cache
by Marek Vasut 28 Sep '16

28 Sep '16

Same as commit ce492b3b8f99cf9d2f807ec22d8805c996a09503 Subject: drm/fsl-dcu: use flat regmap cache Using flat regmap cache instead of RB-tree to avoid the following lockdep warning on driver load: WARNING: CPU: 0 PID: 1 at kernel/locking/lockdep.c:2871 lockdep_trace_alloc+0x104/0x128 DEBUG_LOCKS_WARN_ON(irqs_disabled_flags(flags)) The RB-tree regmap cache needs to allocate new space on first writes. However, allocations in an atomic context (e.g. when a spinlock is held) are not allowed. The function regmap_write calls map->lock, which acquires a spinlock in the fast_io case. Since the driver uses MMIO, the regmap bus of type regmap_mmio is being used which has fast_io set to true. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Timur Tabi <timur(a)tabi.org> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> --- sound/soc/fsl/fsl_asrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index b7157ce..1d82f68 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -726,7 +726,7 @@ static const struct regmap_config fsl_asrc_regmap_config = { .readable_reg = fsl_asrc_readable_reg, .volatile_reg = fsl_asrc_volatile_reg, .writeable_reg = fsl_asrc_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; /** -- 2.9.3

5 30

[alsa-devel] [PATCH 0/5] ALSA: soc: fix sparse warnings
by Takashi Sakamoto 28 Sep '16

28 Sep '16

Hi, This patchset is to fix some sparse warnings in ALSA SoC part. Below items are going to be fixed: * fix a condition statement with a logical operation * purge old way to initialize array * missing 'static' qualifier for file local symbols Although I got many warnings from Intel SST drivers due to confusions of address spaces (__iomem and __kernel), I still left them as they are, because they require more codes to fix. I hope Intel developers will do additional work for them... Takashi Sakamoto (5): ALSA: soc-da7219: fix inappropriate condition statement ALSA: soc-sst-bxt-rt298: fix obsoleted initializers for array ALSA: soc-sst-bxt-da7219_max98357a: fix obsoleted initializers for array ALSA: soc-rt5616: add static qualifier for file local symbols ALSA: soc-tpa6130a2: add static qualifier for file local symbols sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/rt5616.c | 4 ++-- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 ++++++------ sound/soc/intel/boards/bxt_rt298.c | 14 +++++++------- 5 files changed, 17 insertions(+), 17 deletions(-) -- 2.7.4

4 13

[alsa-devel] [PATCH] ASoC: trivial: system spelling fix
by Marek Vasut 28 Sep '16

28 Sep '16

Do s/ststem/system/ . Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b95ae49..d023959 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -13,4 +13,4 @@ config SND_SIMPLE_SCU_CARD select SND_SIMPLE_CARD_UTILS help This option enables generic simple SCU sound card support. - It supports DPCM of multi CPU single Codec ststem. + It supports DPCM of multi CPU single Codec system. -- 2.9.3

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: max9867/rt5616/wm8960: remove usage of obsoleted TLV-related macro
by Takashi Sakamoto 28 Sep '16

28 Sep '16

This patchset update my previous one: [alsa-devel] [PATCH] ASOC: max9867/rt5616/wm8960: remove usage of obsoleted TLV-related macro Changes v2: - Split to several patches for each modules Takashi Sakamoto (3): ASoC: max9867: remove usage of obsoleted TLV-related macro ASoC: rt5616: remove usage of obsoleted TLV-related macro ASoC: wm8960: remove usage of obsoleted TLV-related macro sound/soc/codecs/max9867.c | 5 ++--- sound/soc/codecs/rt5616.c | 5 ++--- sound/soc/codecs/wm8960.c | 5 ++--- 3 files changed, 6 insertions(+), 9 deletions(-) -- 2.7.4

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[alsa-devel] [RFC PATCH 0/3] support DP MST audio
by libin.yang＠linux.intel.com 28 Sep '16

28 Sep '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patchset starts to support DP MST audio. This patchset is based on drm-intel-nightly on git://anongit.freedesktop.org/drm-intel Libin Yang (3): ALSA: hda - codec add DP MST support for connection list ALSA: hda - add DP mst verb support ALSA: hda - add DP mst audio support sound/pci/hda/hda_codec.c | 129 ++++++++++++++++++----- sound/pci/hda/hda_codec.h | 18 ++-- sound/pci/hda/hda_generic.c | 18 ++-- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/patch_analog.c | 18 ++-- sound/pci/hda/patch_hdmi.c | 228 +++++++++++++++++++++++++++++++---------- sound/pci/hda/patch_realtek.c | 22 ++-- sound/pci/hda/patch_sigmatel.c | 4 +- sound/pci/hda/patch_via.c | 10 +- 9 files changed, 328 insertions(+), 121 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: do not declare support for mono DAI
by Nikita Yushchenko 27 Sep '16

27 Sep '16

This hardware supports only 2-channel DAI, even mono ADC digital output has two channels with the same data. Having min_channels=1 results in broken playback of mono files in setups where CPU DAI supports mono. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- sound/soc/codecs/tlv320aic31xx.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 725173b12725..be1a64bfd320 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1213,7 +1213,7 @@ static struct snd_soc_dai_driver dac31xx_dai_driver[] = { .name = "tlv32dac31xx-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC31XX_RATES, .formats = AIC31XX_FORMATS, @@ -1228,14 +1228,14 @@ static struct snd_soc_dai_driver aic31xx_dai_driver[] = { .name = "tlv320aic31xx-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC31XX_RATES, .formats = AIC31XX_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC31XX_RATES, .formats = AIC31XX_FORMATS, -- 2.1.4

2 1

[alsa-devel] [PATCH 0/3] ASOC: wm8991/stac9766: fix wrong usage of TLV-related macro
by Takashi Sakamoto 27 Sep '16

27 Sep '16

Hi, This patchset is for Linux 4.8-rc8. In current implementation of drivers for wm899a and stac9766, DECLARE_TLV_DB_LINEAR() is used to represent information about attenuation and gain of volumes. According to their datasheet, the attenuation and volumes are first-degree equation with dB coefficiency. It's not linear and usage of the macro is wrong[1]. This patchset fixes the bug. As of 2016, these CODECs seem to EOL. So this patchset is just for maintenance purpose to remove misuse of TLV-related macros from tree. [1] Re: [RFC] TLV entry for linear volume http://www.spinics.net/lists/alsa-devel/msg03485.html Takashi Sakamoto (3): ASOC: wm8991: fix wrong usage of DECLARE_TLV_DB_LINEAR() ASOC: wm8991: remove unused variable ASOC: stac9766: fix wrong usage of DECLARE_TLV_DB_LINEAR() sound/soc/codecs/stac9766.c | 8 ++++---- sound/soc/codecs/wm8991.c | 28 +++++++++++++++++++--------- 2 files changed, 23 insertions(+), 13 deletions(-) -- 2.7.4

3 8

[alsa-devel] [PATCH 1/2] ASoC: arizona: Add debug prints for output power up/down times
by Charles Keepax 27 Sep '16

27 Sep '16

When debugging it is useful to check the total power up/down delay that is executed as part of the coalesced output delay. This patch adds some debug prints for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 780cf4f..eaf1dbb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -891,6 +891,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT3R_ENA_SHIFT: priv->out_up_pending--; if (!priv->out_up_pending) { + dev_dbg(codec->dev, "Power up delay: %d\n", + priv->out_up_delay); msleep(priv->out_up_delay); priv->out_up_delay = 0; } @@ -925,6 +927,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT3R_ENA_SHIFT: priv->out_down_pending--; if (!priv->out_down_pending) { + dev_dbg(codec->dev, "Power down delay: %d\n", + priv->out_down_delay); msleep(priv->out_down_delay); priv->out_down_delay = 0; } -- 2.1.4

2 4

[alsa-devel] Applied "ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array" to the asoc tree
by Mark Brown 27 Sep '16

27 Sep '16

The patch ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 43d443dc257c00ef3a3f940b6abfb7537c4fcbe8 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Date: Tue, 27 Sep 2016 07:45:09 +0900 Subject: [PATCH] ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array Sparse reports below warnings. bxt_da7219_max98357a.c:250:9: warning: obsolete array initializer, use C99 syntax bxt_da7219_max98357a.c:275:9: warning: obsolete array initializer, use C99 syntax bxt_da7219_max98357a.c:290:9: warning: obsolete array initializer, use C99 syntax bxt_da7219_max98357a.c:304:9: warning: obsolete array initializer, use C99 syntax bxt_da7219_max98357a.c:317:9: warning: obsolete array initializer, use C99 syntax There's no need to use obsoleted way. This commit fixes it. Fixes: 723bad3fef8b (ASoC: Intel: Add Broxton-P Dialog Maxim machine driver) Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index bb4533c305c7..6532b8f0ab2f 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -302,7 +302,7 @@ static struct snd_soc_ops broxton_refcap_ops = { /* broxton digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link broxton_dais[] = { /* Front End DAI links */ - [BXT_DPCM_AUDIO_PB] + [BXT_DPCM_AUDIO_PB] = { .name = "Bxt Audio Port", .stream_name = "Audio", @@ -318,7 +318,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .dpcm_playback = 1, .ops = &broxton_da7219_fe_ops, }, - [BXT_DPCM_AUDIO_CP] + [BXT_DPCM_AUDIO_CP] = { .name = "Bxt Audio Capture Port", .stream_name = "Audio Record", @@ -333,7 +333,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .dpcm_capture = 1, .ops = &broxton_da7219_fe_ops, }, - [BXT_DPCM_AUDIO_REF_CP] + [BXT_DPCM_AUDIO_REF_CP] = { .name = "Bxt Audio Reference cap", .stream_name = "Refcap", @@ -362,7 +362,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .dynamic = 1, .ops = &broxton_dmic_ops, }, - [BXT_DPCM_AUDIO_HDMI1_PB] + [BXT_DPCM_AUDIO_HDMI1_PB] = { .name = "Bxt HDMI Port1", .stream_name = "Hdmi1", @@ -375,7 +375,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .nonatomic = 1, .dynamic = 1, }, - [BXT_DPCM_AUDIO_HDMI2_PB] + [BXT_DPCM_AUDIO_HDMI2_PB] = { .name = "Bxt HDMI Port2", .stream_name = "Hdmi2", @@ -388,7 +388,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .nonatomic = 1, .dynamic = 1, }, - [BXT_DPCM_AUDIO_HDMI3_PB] + [BXT_DPCM_AUDIO_HDMI3_PB] = { .name = "Bxt HDMI Port3", .stream_name = "Hdmi3", -- 2.9.3

1 0

[alsa-devel] [PATCH] ASOC: max9867/rt5616/wm8960: remove usage of obsoleted TLV-related macro
by Takashi Sakamoto 27 Sep '16

27 Sep '16

TLV_DB_RANGE_HEAD macro was obsoleted by commit bf1d1c9b6179 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()"). This commit removes usage of the macro, with the obsoleting macro renamed to SNDRV_CTL_TLVD_DECLARE_DB_RANGE(). Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: patches(a)opensource.wolfsonmicro.com Cc: anish kumar <yesanishhere(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/soc/codecs/max9867.c | 5 ++--- sound/soc/codecs/rt5616.c | 5 ++--- sound/soc/codecs/wm8960.c | 5 ++--- 3 files changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 2a22fdd..1aa838f 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -38,11 +38,10 @@ static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const unsigned int max98088_micboost_tlv[] = { - TLV_DB_RANGE_HEAD(2), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), -}; +); static const struct snd_kcontrol_new max9867_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f527b5b..851704f 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -294,8 +294,7 @@ 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), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(bst_tlv, 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), @@ -303,7 +302,7 @@ static unsigned int bst_tlv[] = { 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 struct snd_kcontrol_new rt5616_snd_controls[] = { /* Headphone Output Volume */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index d7f444f..3d8c756 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -226,11 +226,10 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1); -static const unsigned int micboost_tlv[] = { - TLV_DB_RANGE_HEAD(2), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(micboost_tlv, 0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0), 2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0), -}; +); static const struct snd_kcontrol_new wm8960_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, -- 2.7.4

3 2

[alsa-devel] [PATCH] ALSA: hda - Add the top speaker pin config for HP Spectre x360
by Takashi Iwai 27 Sep '16

27 Sep '16

HP Spectre x360 with CX20724 codec has two speaker outputs while the BIOS sets up only the bottom one (NID 0x17) and disables the top one (NID 0x1d). This patch adds a fixup simply defining the proper pincfg for NID 0x1d so that the top speaker works as is. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=169071 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56fefbd85782..ed62748a6d55 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -261,6 +261,7 @@ enum { CXT_FIXUP_HP_530, CXT_FIXUP_CAP_MIX_AMP_5047, CXT_FIXUP_MUTE_LED_EAPD, + CXT_FIXUP_HP_SPECTRE, }; /* for hda_fixup_thinkpad_acpi() */ @@ -765,6 +766,14 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_mute_led_eapd, }, + [CXT_FIXUP_HP_SPECTRE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* enable NID 0x1d for the speaker on top */ + { 0x1d, 0x91170111 }, + { } + } + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -814,6 +823,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), + SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.10.0

1 0

[alsa-devel] Applied "ASoC: Intel: Add DMIC channel constraint for bxt machine" to the asoc tree
by Mark Brown 27 Sep '16

27 Sep '16

The patch ASoC: Intel: Add DMIC channel constraint for bxt machine has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 43c02ede76a60de38504f4f4fd6042f677be093a Mon Sep 17 00:00:00 2001 From: Yong Zhi <yong.zhi(a)intel.com> Date: Mon, 26 Sep 2016 13:02:29 -0700 Subject: [PATCH] ASoC: Intel: Add DMIC channel constraint for bxt machine Add channel and rate constraints for Refcap and dmiccap devices respectively. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 63 +++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3774b117d365..bb4533c305c7 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -37,6 +37,7 @@ enum { BXT_DPCM_AUDIO_PB = 0, BXT_DPCM_AUDIO_CP, BXT_DPCM_AUDIO_REF_CP, + BXT_DPCM_AUDIO_DMIC_CP, BXT_DPCM_AUDIO_HDMI1_PB, BXT_DPCM_AUDIO_HDMI2_PB, BXT_DPCM_AUDIO_HDMI3_PB, @@ -252,6 +253,52 @@ static struct snd_soc_ops broxton_da7219_ops = { .hw_free = broxton_da7219_hw_free, }; +static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels->min = channels->max = DUAL_CHANNEL; + + return 0; +} + +static int broxton_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops broxton_dmic_ops = { + .startup = broxton_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int broxton_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static struct snd_soc_ops broxton_refcap_ops = { + .startup = broxton_refcap_startup, +}; + /* broxton digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link broxton_dais[] = { /* Front End DAI links */ @@ -299,6 +346,21 @@ static struct snd_soc_dai_link broxton_dais[] = { .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, + .ops = &broxton_refcap_ops, + }, + [BXT_DPCM_AUDIO_DMIC_CP] + { + .name = "Bxt Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &broxton_dmic_ops, }, [BXT_DPCM_AUDIO_HDMI1_PB] { @@ -382,6 +444,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .codec_dai_name = "dmic-hifi", .platform_name = "0000:00:0e.0", .ignore_suspend = 1, + .be_hw_params_fixup = broxton_dmic_fixup, .dpcm_capture = 1, .no_pcm = 1, }, -- 2.9.3

1 0

[alsa-devel] [PATCH] ASoC: Intel: Add DMIC channel constraint for bxt machine
by Yong Zhi 26 Sep '16

26 Sep '16

Add channel and rate constraints for Refcap and dmiccap devices respectively. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 63 +++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3774b11..bb4533c 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -37,6 +37,7 @@ enum { BXT_DPCM_AUDIO_PB = 0, BXT_DPCM_AUDIO_CP, BXT_DPCM_AUDIO_REF_CP, + BXT_DPCM_AUDIO_DMIC_CP, BXT_DPCM_AUDIO_HDMI1_PB, BXT_DPCM_AUDIO_HDMI2_PB, BXT_DPCM_AUDIO_HDMI3_PB, @@ -252,6 +253,52 @@ static struct snd_soc_ops broxton_da7219_ops = { .hw_free = broxton_da7219_hw_free, }; +static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels->min = channels->max = DUAL_CHANNEL; + + return 0; +} + +static int broxton_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops broxton_dmic_ops = { + .startup = broxton_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int broxton_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static struct snd_soc_ops broxton_refcap_ops = { + .startup = broxton_refcap_startup, +}; + /* broxton digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link broxton_dais[] = { /* Front End DAI links */ @@ -299,6 +346,21 @@ static struct snd_soc_dai_link broxton_dais[] = { .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, + .ops = &broxton_refcap_ops, + }, + [BXT_DPCM_AUDIO_DMIC_CP] + { + .name = "Bxt Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &broxton_dmic_ops, }, [BXT_DPCM_AUDIO_HDMI1_PB] { @@ -382,6 +444,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .codec_dai_name = "dmic-hifi", .platform_name = "0000:00:0e.0", .ignore_suspend = 1, + .be_hw_params_fixup = broxton_dmic_fixup, .dpcm_capture = 1, .no_pcm = 1, }, -- 1.9.1

1 0

[alsa-devel] [PATCH v3 0/3] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 26 Sep '16

26 Sep '16

Introduce mechanism to detect sun7i and provide a different regmap different compared to sun4i Allwinner A10. The controls will be extended in a forthcoming patch - it is necessary to distinguish between sun4i and sun7i controls because they have different registers. Renamed SUN4I_CODEC_AC_SYS_VERI to SUN7I_CODEC_AC_DAC_CAL and renamed SUN4I_CODEC_AC_MIC_PHONE_CAL to SUN7I_CODEC_AC_MIC_PHONE_CAL because these are actually not present on Allwinner A10. Handle quirks by regmap config and codec and select the correct quirks automatically. Danny Milosavljevic (3): ASoC: sun4i-codec: rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field name. ASoC: rename "SUN4I_CODEC_SYS_VERI" to "SUN7I_CODEC_AC_DAC_CAL"; rename "SUN4I_CODEC_AC_MIC_PHONE_CAL" to "SUN7I_CODEC_AC_MIC_PHONE_CAL". ASoC: sun4i-codec: Add custom regmap configs for the A10 and A20 variants. sound/soc/sunxi/sun4i-codec.c | 45 ++++++++++++++++++++++++++++++++++--------- 1 file changed, 36 insertions(+), 9 deletions(-)

3 8

[alsa-devel] [PATCH] ASoC: ak4104: Limit the supported sample rates
by Vishal Thanki 26 Sep '16

26 Sep '16

Currently the driver exposes range of sample rates between 8KHz to 192KHz as a part of dai_driver. This does not hold true as the limited number sample rates are allowed in hw_params DAI callback. This patch limits the sample rates exposed via dai_driver. Signed-off-by: Vishal Thanki <vishalthanki(a)gmail.com> --- sound/soc/codecs/ak4104.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index bdca879..1a9d233 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -163,7 +163,10 @@ static struct snd_soc_dai_driver ak4104_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE -- 2.4.11

2 1

[alsa-devel] [PATCH 0/2] ASoC: da7219: Codec soft reset and AAD improvements
by Adam Thomson 26 Sep '16

26 Sep '16

This patch set makes the following updates to the driver code: 1) Ensures codec is properly reset at startup, and if previously active then disable audio paths prior to reset. 2) Disables AAD in suspend, if device is not a wake-up source. Changes are based on top of latest code introduced under the following commits: ASoC: da7219: software reset codec at probe ASoC: da7219: Support HP detect procedure when MCLK not present Adam Thomson (2): ASoC: da7219: Reset codec gracefully, if still active ASoC: da7219: Disable AAD if codec is not a wake-up source include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219-aad.c | 56 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7219-aad.h | 5 ++++ sound/soc/codecs/da7219.c | 60 +++++++++++++++++++++++++++++++------------ sound/soc/codecs/da7219.h | 6 +++++ 5 files changed, 113 insertions(+), 16 deletions(-) -- 1.9.3

2 4

[alsa-devel] Applied "ASoC: dpcm: print dai_link name of BE other than FE." to the asoc tree
by Mark Brown 26 Sep '16

26 Sep '16

The patch ASoC: dpcm: print dai_link name of BE other than FE. has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 94d215cc6fc5f0a5d202ed2b12fa0b14392cc2e4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?=E5=BD=AD=E4=B8=9C=E6=9E=97?= <pengdonglin(a)smartisan.com> Date: Mon, 26 Sep 2016 08:29:31 +0000 Subject: [PATCH] ASoC: dpcm: print dai_link name of BE other than FE. When operating the BE, we should print out the dai_link name of BE other than FE. This is useful when analyzing the kernel log. Signed-off-by: Donglin Peng <pengdonglin(a)smartisan.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c1bfe9d8406d..d56a16a0f6fa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1793,7 +1793,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: close BE %s\n", - dpcm->fe->dai_link->name); + be->dai_link->name); soc_pcm_close(be_substream); be_substream->runtime = NULL; @@ -1859,7 +1859,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: hw_free BE %s\n", - dpcm->fe->dai_link->name); + be->dai_link->name); soc_pcm_hw_free(be_substream); @@ -1937,7 +1937,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: hw_params BE %s\n", - dpcm->fe->dai_link->name); + be->dai_link->name); ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params); if (ret < 0) { @@ -2017,7 +2017,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, int ret; dev_dbg(dpcm->be->dev, "ASoC: trigger BE %s cmd %d\n", - dpcm->fe->dai_link->name, cmd); + dpcm->be->dai_link->name, cmd); ret = soc_pcm_trigger(substream, cmd); if (ret < 0) @@ -2232,7 +2232,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", - dpcm->fe->dai_link->name); + be->dai_link->name); ret = soc_pcm_prepare(be_substream); if (ret < 0) { -- 2.9.3

1 0

[alsa-devel] [PATCH v6 0/3] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 26 Sep '16

26 Sep '16

Introduce mechanism to detect sun7i and provide a different regmap (different compared to sun4i Allwinner A10). The controls will be extended in a forthcoming patch - it is necessary to distinguish between sun4i and sun7i controls because they have different registers. Rename SUN4I_CODEC_AC_SYS_VERI to SUN7I_CODEC_AC_DAC_CAL and rename SUN4I_CODEC_AC_MIC_PHONE_CAL to SUN7I_CODEC_AC_MIC_PHONE_CAL because these are actually not present on Allwinner A10. Handle quirks by regmap config and codec and select the correct quirks automatically. Danny Milosavljevic (3): ASoC: sun4i-codec: Rename sun4i_codec_widgets for consistency ASoC: sun4i-codec: Rename some sun7i-only registers ASoC: sun4i-codec: Add custom regmap configs sound/soc/sunxi/sun4i-codec.c | 50 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 41 insertions(+), 9 deletions(-)

3 6

[alsa-devel] [PATCH] ALSA: control: cage TLV_DB_RANGE_HEAD in kernel land because it was obsoleted
by Takashi Sakamoto 26 Sep '16

26 Sep '16

In commit bf1d1c9b6179 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()"), the new macro was added so that "dB range information can be specified without having to count the items manually for TLV_DB_RANGE_HEAD()". In short, TLV_DB_RANGE_HEAD macro was obsoleted. In commit 46e860f76804 ("ALSA: rename TLV-related macros so that they're friendly to user applications"), TLV-related macros are exposed for applications in user land to get content of data structured by Type/Length/Value shape. The commit managed to expose TLV-related macros as many as possible, while obsoleted TLV_DB_RANGE_HEAD() was included to the list of exposed macros. This situation brings some confusions to application developers because they might think all exposed macros have their own purpose and useful for applications. For the reason, this commit moves TLV_DB_RANGE_HEAD macro from UAPI header to a header for kernel land, again. The above commit is done within the same development period for kernel 4.9, thus not published yet. This commit might certainly brings no confusions to user land. Reference: commit bf1d1c9b6179 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()") Reference: commit 46e860f76804 ("ALSA: rename TLV-related macros so that they're friendly to user applications") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/tlv.h | 9 ++++++++- include/uapi/sound/tlv.h | 3 --- 2 files changed, 8 insertions(+), 4 deletions(-) diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 6e2e773..3677ebb 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -46,8 +46,15 @@ #define TLV_DB_RANGE_ITEM SNDRV_CTL_TLVD_DB_RANGE_ITEM #define DECLARE_TLV_DB_RANGE SNDRV_CTL_TLVD_DECLARE_DB_RANGE -#define TLV_DB_RANGE_HEAD SNDRV_CTL_TLVD_DB_RANGE_HEAD #define TLV_DB_GAIN_MUTE SNDRV_CTL_TLVD_DB_GAIN_MUTE +/* + * The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR. + * This is an old fasion and obsoleted by commit bf1d1c9b6179("ALSA: tlv: add + * DECLARE_TLV_DB_RANGE()"). + */ +#define TLV_DB_RANGE_HEAD(num) \ + SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) + #endif /* __SOUND_TLV_H */ diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index f3c198f..b4df440 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -94,9 +94,6 @@ unsigned int name[] = { \ SNDRV_CTL_TLVD_DB_RANGE_ITEM(__VA_ARGS__) \ } -/* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */ -#define SNDRV_CTL_TLVD_DB_RANGE_HEAD(num) \ - SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int) #define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999 -- 2.7.4

3 3

[alsa-devel] [PATCH v2] ASoC: rk3399_gru_sound: fix recording pop at first attempt
by Xing Zheng 26 Sep '16

26 Sep '16

From: Wonjoon Lee <woojoo.lee(a)samsung.com> Pop happens when mclk applied but dmic's own boot-time Specify dmic delay times in dt to make sure clocks are ready earlier than dmic working Signed-off-by: Wonjoon Lee <woojoo.lee(a)samsung.com> Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v2: - rename dmic-delay to dmic-wakeup-delay-ms .../bindings/sound/rockchip,rk3399-gru-sound.txt | 7 +++++++ sound/soc/rockchip/rk3399_gru_sound.c | 14 ++++++++++++++ 2 files changed, 21 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt index f19b6c8..eac91db 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -6,10 +6,17 @@ Required properties: connected to the codecs - rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs +Optional properties: +- dmic-wakeup-delay-ms : specify delay time (ms) for DMIC ready. + If this option is specified, which means it's required dmic need + delay for DMIC to ready so that rt5514 can avoid recording before + DMIC send valid data + Example: sound { compatible = "rockchip,rk3399-gru-sound"; rockchip,cpu = <&i2s0>; rockchip,codec = <&max98357a &rt5514 &da7219>; + dmic-wakeup-delay-ms = <20>; }; diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index ee06489..9ed735a 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -38,6 +38,8 @@ #define SOUND_FS 256 +unsigned int rt5514_dmic_delay; + static struct snd_soc_jack rockchip_sound_jack; static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { @@ -123,6 +125,9 @@ static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Wait for DMIC stable */ + msleep(rt5514_dmic_delay); + return 0; } @@ -343,6 +348,15 @@ static int rockchip_sound_probe(struct platform_device *pdev) return -ENODEV; } + /* Set DMIC delay */ + ret = device_property_read_u32(&pdev->dev, "dmic-delay", + &rt5514_dmic_delay); + if (ret) { + rt5514_dmic_delay = 0; + dev_dbg(&pdev->dev, + "no optional property 'dmic-delay' found, default: no delay\n"); + } + rockchip_dailinks[DAILINK_RT5514_DSP].cpu_name = kstrdup_const(dev_name(dev), GFP_KERNEL); rockchip_dailinks[DAILINK_RT5514_DSP].cpu_dai_name = kstrdup_const(dev_name(dev), GFP_KERNEL); rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL); -- 1.9.1

3 3

[alsa-devel] Applied "ASoC: hdac_hdmi: Drop use of audio component framework to read ELD" to the asoc tree
by Mark Brown 26 Sep '16

26 Sep '16

The patch ASoC: hdac_hdmi: Drop use of audio component framework to read ELD has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 88b456b0f4a2a4c10c141a7d83113a8ce2164463 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Mon, 26 Sep 2016 08:59:50 -0700 Subject: [PATCH] ASoC: hdac_hdmi: Drop use of audio component framework to read ELD The audio component framework code has not yet landed in the i915 driver so drop the use of the API for the time being. Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: Jeeja KP <jeeja.kp(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 201 +++++++++++++++++++++++++++++++------------ 1 file changed, 145 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 537f61aa27fa..4e181b270d95 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -46,10 +46,6 @@ #define ELD_MAX_SIZE 256 #define ELD_FIXED_BYTES 20 -#define ELD_VER_CEA_861D 2 -#define ELD_VER_PARTIAL 31 -#define ELD_MAX_MNL 16 - struct hdac_hdmi_cvt_params { unsigned int channels_min; unsigned int channels_max; @@ -85,6 +81,8 @@ struct hdac_hdmi_pin { hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; struct hdac_hdmi_eld eld; struct hdac_ext_device *edev; + int repoll_count; + struct delayed_work work; struct mutex lock; bool chmap_set; unsigned char chmap[8]; /* ALSA API channel-map */ @@ -175,6 +173,80 @@ static int hdac_hdmi_eld_limit_formats(struct snd_pcm_runtime *runtime, } + /* HDMI ELD routines */ +static unsigned int hdac_hdmi_get_eld_data(struct hdac_device *codec, + hda_nid_t nid, int byte_index) +{ + unsigned int val; + + val = snd_hdac_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, + byte_index); + + dev_dbg(&codec->dev, "HDMI: ELD data byte %d: 0x%x\n", + byte_index, val); + + return val; +} + +static int hdac_hdmi_get_eld_size(struct hdac_device *codec, hda_nid_t nid) +{ + return snd_hdac_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, + AC_DIPSIZE_ELD_BUF); +} + +/* + * This function queries the ELD size and ELD data and fills in the buffer + * passed by user + */ +static int hdac_hdmi_get_eld(struct hdac_device *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size) +{ + int i, size, ret = 0; + + /* + * ELD size is initialized to zero in caller function. If no errors and + * ELD is valid, actual eld_size is assigned. + */ + + size = hdac_hdmi_get_eld_size(codec, nid); + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE) { + dev_err(&codec->dev, "HDMI: invalid ELD buf size %d\n", size); + return -ERANGE; + } + + /* set ELD buffer */ + for (i = 0; i < size; i++) { + unsigned int val = hdac_hdmi_get_eld_data(codec, nid, i); + /* + * Graphics driver might be writing to ELD buffer right now. + * Just abort. The caller will repoll after a while. + */ + if (!(val & AC_ELDD_ELD_VALID)) { + dev_err(&codec->dev, + "HDMI: invalid ELD data byte %d\n", i); + ret = -EINVAL; + goto error; + } + val &= AC_ELDD_ELD_DATA; + /* + * The first byte cannot be zero. This can happen on some DVI + * connections. Some Intel chips may also need some 250ms delay + * to return non-zero ELD data, even when the graphics driver + * correctly writes ELD content before setting ELD_valid bit. + */ + if (!val && !i) { + dev_err(&codec->dev, "HDMI: 0 ELD data\n"); + ret = -EINVAL; + goto error; + } + buf[i] = val; + } + + *eld_size = size; +error: + return ret; +} + static int hdac_hdmi_setup_stream(struct hdac_ext_device *hdac, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format) @@ -987,59 +1059,32 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) return hdac_hdmi_query_cvt_params(&edev->hdac, cvt); } -static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, +static void hdac_hdmi_parse_eld(struct hdac_ext_device *edev, struct hdac_hdmi_pin *pin) { - unsigned int ver, mnl; - - ver = (pin->eld.eld_buffer[DRM_ELD_VER] & DRM_ELD_VER_MASK) - >> DRM_ELD_VER_SHIFT; - - if (ver != ELD_VER_CEA_861D && ver != ELD_VER_PARTIAL) { - dev_dbg(&edev->hdac.dev, "HDMI: Unknown ELD version %d\n", ver); - return -EINVAL; - } - - mnl = (pin->eld.eld_buffer[DRM_ELD_CEA_EDID_VER_MNL] & - DRM_ELD_MNL_MASK) >> DRM_ELD_MNL_SHIFT; - - if (mnl > ELD_MAX_MNL) { - dev_dbg(&edev->hdac.dev, "HDMI: MNL Invalid %d\n", mnl); - return -EINVAL; - } - pin->eld.info.spk_alloc = pin->eld.eld_buffer[DRM_ELD_SPEAKER]; - - return 0; } -static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin) +static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) { struct hdac_ext_device *edev = pin->edev; struct hdac_hdmi_priv *hdmi = edev->private_data; struct hdac_hdmi_pcm *pcm; - int size; + int val; - mutex_lock(&hdmi->pin_mutex); - pin->eld.monitor_present = false; + pin->repoll_count = repoll; - size = snd_hdac_acomp_get_eld(&edev->hdac, pin->nid, -1, - &pin->eld.monitor_present, pin->eld.eld_buffer, - ELD_MAX_SIZE); + pm_runtime_get_sync(&edev->hdac.dev); + val = snd_hdac_codec_read(&edev->hdac, pin->nid, 0, + AC_VERB_GET_PIN_SENSE, 0); - if (size > 0) { - size = min(size, ELD_MAX_SIZE); - if (hdac_hdmi_parse_eld(edev, pin) < 0) - size = -EINVAL; - } + dev_dbg(&edev->hdac.dev, "Pin sense val %x for pin: %d\n", + val, pin->nid); - if (size > 0) { - pin->eld.eld_valid = true; - pin->eld.eld_size = size; - } else { - pin->eld.eld_valid = false; - pin->eld.eld_size = 0; - } + + mutex_lock(&hdmi->pin_mutex); + pin->eld.monitor_present = !!(val & AC_PINSENSE_PRESENCE); + pin->eld.eld_valid = !!(val & AC_PINSENSE_ELDV); pcm = hdac_hdmi_get_pcm(edev, pin); @@ -1061,23 +1106,66 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin) } mutex_unlock(&hdmi->pin_mutex); - return; + goto put_hdac_device; } if (pin->eld.monitor_present && pin->eld.eld_valid) { - if (pcm) { - dev_dbg(&edev->hdac.dev, - "jack report for pcm=%d\n", - pcm->pcm_id); + /* TODO: use i915 component for reading ELD later */ + if (hdac_hdmi_get_eld(&edev->hdac, pin->nid, + pin->eld.eld_buffer, + &pin->eld.eld_size) == 0) { - snd_jack_report(pcm->jack, SND_JACK_AVOUT); - } + if (pcm) { + dev_dbg(&edev->hdac.dev, + "jack report for pcm=%d\n", + pcm->pcm_id); + + snd_jack_report(pcm->jack, SND_JACK_AVOUT); + } + hdac_hdmi_parse_eld(edev, pin); + + print_hex_dump_debug("ELD: ", + DUMP_PREFIX_OFFSET, 16, 1, + pin->eld.eld_buffer, pin->eld.eld_size, + true); + } else { + pin->eld.monitor_present = false; + pin->eld.eld_valid = false; + + if (pcm) { + dev_dbg(&edev->hdac.dev, + "jack report for pcm=%d\n", + pcm->pcm_id); - print_hex_dump_debug("ELD: ", DUMP_PREFIX_OFFSET, 16, 1, - pin->eld.eld_buffer, pin->eld.eld_size, false); + snd_jack_report(pcm->jack, 0); + } + } } mutex_unlock(&hdmi->pin_mutex); + + /* + * Sometimes the pin_sense may present invalid monitor + * present and eld_valid. If ELD data is not valid, loop few + * more times to get correct pin sense and valid ELD. + */ + if ((!pin->eld.monitor_present || !pin->eld.eld_valid) && repoll) + schedule_delayed_work(&pin->work, msecs_to_jiffies(300)); + +put_hdac_device: + pm_runtime_put_sync(&edev->hdac.dev); +} + +static void hdac_hdmi_repoll_eld(struct work_struct *work) +{ + struct hdac_hdmi_pin *pin = + container_of(to_delayed_work(work), struct hdac_hdmi_pin, work); + + /* picked from legacy HDA driver */ + if (pin->repoll_count++ > 6) + pin->repoll_count = 0; + + hdac_hdmi_present_sense(pin, pin->repoll_count); } static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) @@ -1096,6 +1184,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) pin->edev = edev; mutex_init(&pin->lock); + INIT_DELAYED_WORK(&pin->work, hdac_hdmi_repoll_eld); return 0; } @@ -1306,7 +1395,7 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port) list_for_each_entry(pin, &hdmi->pin_list, head) { if (pin->nid == pin_nid) - hdac_hdmi_present_sense(pin); + hdac_hdmi_present_sense(pin, 1); } } @@ -1407,7 +1496,7 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec) } list_for_each_entry(pin, &hdmi->pin_list, head) - hdac_hdmi_present_sense(pin); + hdac_hdmi_present_sense(pin, 1); /* Imp: Store the card pointer in hda_codec */ edev->card = dapm->card->snd_card; @@ -1472,7 +1561,7 @@ static void hdmi_codec_complete(struct device *dev) * all pins here. */ list_for_each_entry(pin, &hdmi->pin_list, head) - hdac_hdmi_present_sense(pin); + hdac_hdmi_present_sense(pin, 1); pm_runtime_put_sync(&edev->hdac.dev); } -- 2.9.3

1 0

[alsa-devel] [PATCH v2 1/4] ASoC: wm_adsp: Use usleep_range for short delay
by Charles Keepax 26 Sep '16

26 Sep '16

Replace the 1ms msleep in wm_adsp2_ena with a usleep_range, as per normal guidance on delay functions. Also tighten up the delay a little as 1ms was quite generous. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- New since the first series of this patch chain. sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 24485ec..4188c37 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2237,7 +2237,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) if (val & ADSP2_RAM_RDY) break; - msleep(1); + usleep_range(250, 500); } if (!(val & ADSP2_RAM_RDY)) { -- 2.1.4

2 6

[alsa-devel] [PATCH v2] ASoC: simple-card: add support for aux devices
by Nikita Yushchenko 26 Sep '16

26 Sep '16

This patch makes it possible to use simple-card in setups where separate amplifier chip is connected to codec's output. Changes from v1: - moved example usage from commit message to Documentation/, as suggested by Kuninori Morimoto, - fixed typo in example usage. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- .../devicetree/bindings/sound/simple-card.txt | 37 ++++++++++++++++++++++ sound/soc/generic/simple-card.c | 34 ++++++++++++++++++++ 2 files changed, 71 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 59d8628..c7a9393 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -22,6 +22,8 @@ Optional properties: headphones are attached. - simple-audio-card,mic-det-gpio : Reference to GPIO that signals when a microphone is attached. +- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. Optional subnodes: @@ -162,3 +164,38 @@ sound { }; }; }; + +Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec +through TPA6130A2 amplifier to headphones: + +&i2c0 { + codec: tlv320dac3100@18 { + compatible = "ti,tlv320dac3100"; + ... + } + + amp: tpa6130a2@60 { + compatible = "ti,tpa6130a2"; + ... + } +} + +sound { + compatible = "simple-audio-card"; + ... + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = ... + }; +}; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 43295f0..f989f34 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -417,6 +417,36 @@ dai_link_of_err: return ret; } +static int asoc_simple_card_parse_aux_devs(struct device_node *node, + struct simple_card_data *priv) +{ + struct device *dev = simple_priv_to_dev(priv); + struct device_node *aux_node; + int i, n, len; + + if (!of_find_property(node, PREFIX "aux-devs", &len)) + return 0; /* Ok to have no aux-devs */ + + n = len / sizeof(__be32); + if (n <= 0) + return -EINVAL; + + priv->snd_card.aux_dev = devm_kzalloc(dev, + n * sizeof(*priv->snd_card.aux_dev), GFP_KERNEL); + if (!priv->snd_card.aux_dev) + return -ENOMEM; + + for (i = 0; i < n; i++) { + aux_node = of_parse_phandle(node, PREFIX "aux-devs", i); + if (!aux_node) + return -EINVAL; + priv->snd_card.aux_dev[i].codec_of_node = aux_node; + } + + priv->snd_card.num_aux_devs = n; + return 0; +} + static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv) { @@ -474,6 +504,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (ret) return ret; + ret = asoc_simple_card_parse_aux_devs(node, priv); + if (ret) + return ret; + return 0; } -- 2.1.4

3 7

[alsa-devel] [PATCH] ASoC: tpa6130a2: fix volume setting when no stream is running
by Nikita Yushchenko 26 Sep '16

26 Sep '16

After moving tpa6130a2 power management to DAPM, if chip can be physically powered off (either reset_gpio is defined, or regulator indeed removes power), then volume change no longer works unless chip is on due to a running stream. Fix that by entering regcache cache_only mode while chip is off. Move regcache calls to tpa6130a2_power() to get them at driver init time as well. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- sound/soc/codecs/tpa6130a2.c | 49 +++++++++++++++++++++++--------------------- 1 file changed, 26 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index f1ea052..3b6faed 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -52,7 +52,7 @@ struct tpa6130a2_data { static int tpa6130a2_power(struct tpa6130a2_data *data, bool enable) { - int ret; + int ret = 0, ret2; if (enable) { ret = regulator_enable(data->supply); @@ -64,20 +64,34 @@ static int tpa6130a2_power(struct tpa6130a2_data *data, bool enable) /* Power on */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); + + /* Sync registers */ + regcache_cache_only(data->regmap, false); + ret = regcache_sync(data->regmap); + if (ret != 0) { + dev_err(data->dev, + "Failed to sync registers: %d\n", ret); + goto regcache_sync_failed; + } } else { + /* Powered off device does not retain registers. While device + * is off, any register updates (i.e. volume changes) should + * happen in cache only. + */ + regcache_mark_dirty(data->regmap); +regcache_sync_failed: + regcache_cache_only(data->regmap, true); + /* Power off */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - ret = regulator_disable(data->supply); - if (ret != 0) { + ret2 = regulator_disable(data->supply); + if (ret2 != 0) { dev_err(data->dev, - "Failed to disable supply: %d\n", ret); - return ret; + "Failed to disable supply: %d\n", ret2); + return ret ? ret : ret2; } - - /* device regs does not match the cache state anymore */ - regcache_mark_dirty(data->regmap); } return ret; @@ -88,25 +102,14 @@ static int tpa6130a2_power_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); struct tpa6130a2_data *data = snd_soc_component_get_drvdata(c); - int ret; - /* before widget power up */ if (SND_SOC_DAPM_EVENT_ON(event)) { - /* Turn on the chip */ - tpa6130a2_power(data, true); - /* Sync the registers */ - ret = regcache_sync(data->regmap); - if (ret < 0) { - dev_err(c->dev, "Failed to initialize chip\n"); - tpa6130a2_power(data, false); - return ret; - } - /* after widget power down */ + /* Before widget power up: turn chip on, sync registers */ + return tpa6130a2_power(data, true); } else { - tpa6130a2_power(data, false); + /* After widget power down: turn chip off */ + return tpa6130a2_power(data, false); } - - return 0; } /* -- 2.1.4

2 5

[alsa-devel] [asoc:topic/intel 53/55] sound/soc/codecs/hdac_hdmi.c:1028:5: warning: passing argument 4 of 'snd_hdac_acomp_get_eld' from incompatible pointer type
by kbuild test robot 26 Sep '16

26 Sep '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 2d995e5dc283adbfbf9c1eb81bf35ca7af2d22a6 commit: fe27f4e0545d3fc1b0518fafb4fe0460d757651d [53/55] ASoC: hdac_hdmi: use audio component framework to read ELD config: x86_64-randconfig-n0-09251512 (attached as .config) compiler: gcc-4.8 (Debian 4.8.4-1) 4.8.4 reproduce: git checkout fe27f4e0545d3fc1b0518fafb4fe0460d757651d # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): sound/soc/codecs/hdac_hdmi.c: In function 'hdac_hdmi_present_sense': sound/soc/codecs/hdac_hdmi.c:1028:5: warning: passing argument 3 of 'snd_hdac_acomp_get_eld' makes pointer from integer without a cast [enabled by default] ELD_MAX_SIZE); ^ In file included from sound/soc/codecs/hdac_hdmi.c:30:0: include/sound/hda_i915.h:14:5: note: expected 'bool *' but argument is of type 'int' int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, ^ >> sound/soc/codecs/hdac_hdmi.c:1028:5: warning: passing argument 4 of 'snd_hdac_acomp_get_eld' from incompatible pointer type [enabled by default] ELD_MAX_SIZE); ^ In file included from sound/soc/codecs/hdac_hdmi.c:30:0: include/sound/hda_i915.h:14:5: note: expected 'char *' but argument is of type 'bool *' int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, ^ sound/soc/codecs/hdac_hdmi.c:1028:5: warning: passing argument 5 of 'snd_hdac_acomp_get_eld' makes integer from pointer without a cast [enabled by default] ELD_MAX_SIZE); ^ In file included from sound/soc/codecs/hdac_hdmi.c:30:0: include/sound/hda_i915.h:14:5: note: expected 'int' but argument is of type 'char *' int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, ^ sound/soc/codecs/hdac_hdmi.c:1028:5: error: too many arguments to function 'snd_hdac_acomp_get_eld' ELD_MAX_SIZE); ^ In file included from sound/soc/codecs/hdac_hdmi.c:30:0: include/sound/hda_i915.h:14:5: note: declared here int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, ^ vim +/snd_hdac_acomp_get_eld +1028 sound/soc/codecs/hdac_hdmi.c 1012 1013 return 0; 1014 } 1015 1016 static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin) 1017 { 1018 struct hdac_ext_device *edev = pin->edev; 1019 struct hdac_hdmi_priv *hdmi = edev->private_data; 1020 struct hdac_hdmi_pcm *pcm; 1021 int size; 1022 1023 mutex_lock(&hdmi->pin_mutex); 1024 pin->eld.monitor_present = false; 1025 1026 size = snd_hdac_acomp_get_eld(&edev->hdac, pin->nid, -1, 1027 &pin->eld.monitor_present, pin->eld.eld_buffer, > 1028 ELD_MAX_SIZE); 1029 1030 if (size > 0) { 1031 size = min(size, ELD_MAX_SIZE); 1032 if (hdac_hdmi_parse_eld(edev, pin) < 0) 1033 size = -EINVAL; 1034 } 1035 1036 if (size > 0) { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] DAPM: codec to codec link documentation[RFC]
by anish kumar 26 Sep '16

26 Sep '16

Hello, I had a tough time figuring out how to get codec to codec link to work. So thought of making it easier for other people who wants to use the same. I will appreciate any inputs for below documentation. Excuse my diagrams if it doesn't render on your browser. I am still learning as how to best draw txt diagrams. Mostly the flow of audio is always from CPU to codec so your system will look as below: ---------- ---------- | | dai | | CPU -------> codec | | | | ---------- ---------- In case your system looks as below: ---------- | | codec-2 | | ---------- | dai-2 | ---------- ---------- | | dai-1 | | CPU -------> codec-1 | | | | ---------- ---------- | dai-3 | ---------- | | codec-3 | | ---------- Suppose codec-2 is a bluetooth chip and codec-3 is connected to a speaker and you have a below scenario: codec-2 will receive the audio data and the user wants to play that audio through codec-3 without involving the CPU.This aforementioned case is the ideal case when codec to codec connection should be used. Your dai_link should appear as below in your machine file: static const struct snd_soc_pcm_stream dummy_params = { .formats = SNDRV_PCM_FMTBIT_S24_LE, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, }; { .name = "your_name", .stream_name = "your_stream_name", .cpu_dai_name = "snd-soc-dummy-dai", .codec_name = "codec-2, .codec_dai_name = "codec-2-dai_name", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, .params = &dummy_params, }, { .name = "your_name", .stream_name = "your_stream_name", .cpu_dai_name = "snd-soc-dummy-dai", .codec_name = "codec-3, .codec_dai_name = "codec-3-dai_name", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, .params = &dummy_params, }, Note the "params" callback which lets the dapm know that this dai_link is a codec to codec connection. Also, in above code cpu_dai should be replaced with your actual cpu dai but in case you don't have a actual cpu dai then dummy will do. You can browse the speyside.c for an actual example code in mainline. In dapm core a route is created between cpu_dai playback widget and codec_dai capture widget for playback path and vice-versa is true for capture path. In order for this aforementioned route to get triggered, DAPM needs to find a valid endpoint which could be either a sink or source widget corresponding to playback and capture path respectively. Below is what you can use it to trigger the widgets provided you have stream name ending with "Playback" and "Capture" for cpu and codec dai's. static const struct snd_soc_dapm_widget aif_dapm_widgets[] = { SND_SOC_DAPM_SPK("dummyspk", NULL), SND_SOC_DAPM_MIC("dummymic", NULL), }; static const struct snd_soc_dapm_route audio_i2s_map[] = { {"dummyspk", NULL, "Playback"}, {"Capture", NULL, "dummymic"}, }; Thanks,

2 3

[alsa-devel] [PATCH] ALSA: hda - Adding one more ALC255 pin definition for headset problem
by Hui Wang 26 Sep '16

26 Sep '16

We have two new Dell laptop models, they have the same ALC255 pin definition, but not in the pin quirk table yet, as a result, the headset microphone can't work. After adding the definition in the table, the headset microphone works well. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1fe8750..b58e8c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5855,6 +5855,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170120}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x14, 0x90170110}, + {0x1b, 0x02011020}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x14, 0x90170130}, {0x1b, 0x01014020}, {0x21, 0x0221103f}), -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: usb-line6: use the same declaration as definition in header for MIDI manufacturer ID
by Takashi Sakamoto 25 Sep '16

25 Sep '16

Currently, usb-line6 module exports an array of MIDI manufacturer ID and usb-pod module uses it. However, the declaration is not the definition in common header. The difference is explicit length of array. Although compiler calculates it and everything goes well, it's better to use the same representation between definition and declaration. This commit fills the length of array for usb-line6 module. As a small good sub-effect, this commit suppress below warnings from static analysis by sparse v0.5.0. sound/usb/line6/driver.c:274:43: error: cannot size expression sound/usb/line6/driver.c:275:16: error: cannot size expression sound/usb/line6/driver.c:276:16: error: cannot size expression sound/usb/line6/driver.c:277:16: error: cannot size expression Fixes: 705ececd1c60 ("Staging: add line6 usb driver") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index f9224ec..ad7efc0 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -30,7 +30,7 @@ /* This is Line 6's MIDI manufacturer ID. */ -const unsigned char line6_midi_id[] = { +const unsigned char line6_midi_id[3] = { 0x00, 0x01, 0x0c }; EXPORT_SYMBOL_GPL(line6_midi_id); -- 2.7.4

2 1

[alsa-devel] [PATCH v5 0/3] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 25 Sep '16

25 Sep '16

ASoC: sun4i-codec: Introduce mechanism to detect sun7i and provide a different regmap different compared to sun4i Allwinner A10. The controls will be extended in a forthcoming patch - it is necessary to distinguish between sun4i and sun7i controls because they have different registers. Renamed SUN4I_CODEC_AC_SYS_VERI to SUN7I_CODEC_AC_DAC_CAL and renamed SUN4I_CODEC_AC_MIC_PHONE_CAL to SUN7I_CODEC_AC_MIC_PHONE_CAL because these are actually not present on Allwinner A10. Handle quirks by regmap config and codec and select the correct quirks automatically. Danny Milosavljevic (3): ASoC: sun4i-codec: rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field name. ASoC: rename "SUN4I_CODEC_SYS_VERI" to "SUN7I_CODEC_AC_DAC_CAL"; rename "SUN4I_CODEC_AC_MIC_PHONE_CAL" to "SUN7I_CODEC_AC_MIC_PHONE_CAL". ASoC: sun4i-codec: Add custom regmap configs for the A10 and A20 variants. sound/soc/sunxi/sun4i-codec.c | 50 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 41 insertions(+), 9 deletions(-)

3 8

[alsa-devel] [PATCH v2 0/4] ASoC: Intel: Skylake: Add support for generating unique module ids
by Vinod Koul 25 Sep '16

25 Sep '16

Driver needs to send unique module instance id to firmware while creating the module and uses this id to communicate with DSP for setting parameters while audio use case is ongoing. But, we have upper bound of instance ID. The current IDs are coming from topology but it doesn't know the upper bound and can't assign unique id's subject to upper bounds as we can create a big graph but not all parts running at same time. So this series first adds APIs to allocate/free a 128bit unique id which is built on top of ffz(). It also adds a table for finding modules of neighbouring modules. Then driver patches are updated to use this. Dharageswari R (4): ASoC: Intel: Skylake: Add module instance id generation APIs ASoC: Intel: Skylake: Use private instance id of modules in IPC ASoC: Intel: Skylake: Add table for module id for quick ref ASoC: Intel: Skylake: Update to use instance ids generated sound/soc/intel/skylake/skl-messages.c | 24 +++-- sound/soc/intel/skylake/skl-sst-dsp.h | 6 ++ sound/soc/intel/skylake/skl-sst-utils.c | 150 ++++++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.c | 32 +++++- sound/soc/intel/skylake/skl-topology.h | 11 ++ sound/soc/intel/skylake/skl-tplg-interface.h | 3 +- 6 files changed, 211 insertions(+), 15 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH 0/9] ASoC: Intel: Skylake: Driver updates
by Vinod Koul 25 Sep '16

25 Sep '16

So we had some driver updates collated, time to send them upstream :) - manifest data size was not checked so that is fixed - previous conversion to use core parsing for HDA capabilities was wrong at one place, that is fixed. - DMA config size was wrong, that is also fixed - Add 32bit format - The module instance IDs are manged in driver now. Dharageswari R (5): ASoC: Intel: Skylake: Unload all the loadable modules ASoC: Intel: Skylake: Create dynamic instance ids for DSP modules ASoC: Intel: Skylake: Use private instance id of modules in IPC ASoC: Intel: Skylake: Table for module instance id and private id ASoC: Intel: Skylake: Override the actual instance id's to pvt_id's Jeeja KP (1): ASoC: Intel: Skylake: Fix DMA control config size Samaga Krishna (1): ASoC: Intel: Skylake: Add 32bit support Vinod Koul (2): ASoC: Intel: Skylake: check manifest size ASoC: Intel: Skylake: Fix the inverted logic check sound/soc/intel/skylake/skl-messages.c | 24 +++-- sound/soc/intel/skylake/skl-pcm.c | 5 +- sound/soc/intel/skylake/skl-sst-dsp.h | 6 ++ sound/soc/intel/skylake/skl-sst-utils.c | 133 ++++++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.c | 44 ++++++++- sound/soc/intel/skylake/skl-topology.h | 13 ++- sound/soc/intel/skylake/skl-tplg-interface.h | 3 +- 7 files changed, 208 insertions(+), 20 deletions(-) -- 1.9.1

2 17

[alsa-devel] Applied "ASoC: tpa6130a2: fix volume setting when no stream is running" to the asoc tree
by Mark Brown 25 Sep '16

25 Sep '16

The patch ASoC: tpa6130a2: fix volume setting when no stream is running has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a114580f8f3e5bdefc14d75d6c3ba7032210b980 Mon Sep 17 00:00:00 2001 From: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> Date: Thu, 22 Sep 2016 13:10:40 +0300 Subject: [PATCH] ASoC: tpa6130a2: fix volume setting when no stream is running After moving tpa6130a2 power management to DAPM, if chip can be physically powered off (either reset_gpio is defined, or regulator indeed removes power), then volume change no longer works unless chip is on due to a running stream. Fix that by entering regcache cache_only mode while chip is off. Move regcache calls to tpa6130a2_power() to get them at driver init time as well. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tpa6130a2.c | 49 +++++++++++++++++++++++--------------------- 1 file changed, 26 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index f1ea052a822e..3b6faed91d7e 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -52,7 +52,7 @@ struct tpa6130a2_data { static int tpa6130a2_power(struct tpa6130a2_data *data, bool enable) { - int ret; + int ret = 0, ret2; if (enable) { ret = regulator_enable(data->supply); @@ -64,20 +64,34 @@ static int tpa6130a2_power(struct tpa6130a2_data *data, bool enable) /* Power on */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); + + /* Sync registers */ + regcache_cache_only(data->regmap, false); + ret = regcache_sync(data->regmap); + if (ret != 0) { + dev_err(data->dev, + "Failed to sync registers: %d\n", ret); + goto regcache_sync_failed; + } } else { + /* Powered off device does not retain registers. While device + * is off, any register updates (i.e. volume changes) should + * happen in cache only. + */ + regcache_mark_dirty(data->regmap); +regcache_sync_failed: + regcache_cache_only(data->regmap, true); + /* Power off */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - ret = regulator_disable(data->supply); - if (ret != 0) { + ret2 = regulator_disable(data->supply); + if (ret2 != 0) { dev_err(data->dev, - "Failed to disable supply: %d\n", ret); - return ret; + "Failed to disable supply: %d\n", ret2); + return ret ? ret : ret2; } - - /* device regs does not match the cache state anymore */ - regcache_mark_dirty(data->regmap); } return ret; @@ -88,25 +102,14 @@ static int tpa6130a2_power_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); struct tpa6130a2_data *data = snd_soc_component_get_drvdata(c); - int ret; - /* before widget power up */ if (SND_SOC_DAPM_EVENT_ON(event)) { - /* Turn on the chip */ - tpa6130a2_power(data, true); - /* Sync the registers */ - ret = regcache_sync(data->regmap); - if (ret < 0) { - dev_err(c->dev, "Failed to initialize chip\n"); - tpa6130a2_power(data, false); - return ret; - } - /* after widget power down */ + /* Before widget power up: turn chip on, sync registers */ + return tpa6130a2_power(data, true); } else { - tpa6130a2_power(data, false); + /* After widget power down: turn chip off */ + return tpa6130a2_power(data, false); } - - return 0; } /* -- 2.9.3

1 0

[alsa-devel] [PATCH 0/7] Allow ADSPs to be preloaded with firmware
by Charles Keepax 25 Sep '16

25 Sep '16

As requirements to bring up audio paths are continuous getting tighter and the DSP download to most ADSP devices happens over an external bus it can become an important factor in the path bring up time. As such sometimes it is a reasonable trade off to download the firmware ahead of when it will be required and take a small hit on power consumption for keeping the core powered up. This patch chain adds such functionality to the wm_adsp driver, broadly speaking there are three parts to the chain. Firstly we setup the existing preloader widgets such that they will be easier to power up independently. Secondly, we setup the ALSA controls such that they will gracefully handle the DSP being in a low power state between loading the firmware and starting the core. Finally, we actually add the preloading features itself. Thanks, Charles Charles Keepax (7): ASoC: arizona: Attach SYSCLK to DSP preloaders ASoC: wm_adsp: Make DSP preloader a supply widget ASoC: wm_adsp: Separate concept of booted and running ASoC: wm_adsp: Move control cache sync out of preloader ASoC: wm_adsp: Put DSP into low power state between loading and running ASoC: wm_adsp: Allow preloader to do the final shutdown of the DSP ASoC: wm_adsp: Add mechanism to preload firmware on a core sound/soc/codecs/arizona.h | 18 ++++--- sound/soc/codecs/cs47l24.c | 5 +- sound/soc/codecs/wm5102.c | 3 +- sound/soc/codecs/wm5110.c | 7 ++- sound/soc/codecs/wm_adsp.c | 126 ++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm_adsp.h | 18 ++++++- 6 files changed, 138 insertions(+), 39 deletions(-) -- 2.1.4

2 12

[alsa-devel] [PATCH v3] ASoC: exynos: organize the asoc audio into a menu
by Randy Li 25 Sep '16

25 Sep '16

It is simple sound card time, we could assign different codec to a interface without making a specific driver for it. The SPDIF and I2S interface for Samsung would be possible used by simple-sound-card, but not sure about the PCM. Those S3C time entries are left alone as I don't think any new board would need them. Signed-off-by: Randy Li <ayaka(a)soulik.info> --- sound/soc/samsung/Kconfig | 57 ++++++++++++++++++++++++----------------------- 1 file changed, 29 insertions(+), 28 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7b722b0..f6023b4 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,12 +1,14 @@ -config SND_SOC_SAMSUNG +menuconfig SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on (PLAT_SAMSUNG || ARCH_EXYNOS) select SND_SOC_GENERIC_DMAENGINE_PCM - help + ---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 select the audio interfaces to support below. +if SND_SOC_SAMSUNG + config SND_S3C24XX_I2S tristate @@ -18,22 +20,22 @@ config SND_S3C2412_SOC_I2S select SND_S3C_I2SV2_SOC config SND_SAMSUNG_PCM - tristate + tristate "Samsung PCM interface support" config SND_SAMSUNG_AC97 tristate select SND_SOC_AC97_BUS config SND_SAMSUNG_SPDIF - tristate + tristate "Samsung SPDIF transmitter support" select SND_SOC_SPDIF config SND_SAMSUNG_I2S - tristate + tristate "Samsung I2S interface support" config SND_SOC_SAMSUNG_NEO1973_WM8753 tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)" - depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 + depends on MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 select SND_SOC_BT_SCO @@ -43,7 +45,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 config SND_SOC_SAMSUNG_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" - depends on SND_SOC_SAMSUNG && MACH_JIVE && I2C + depends on MACH_JIVE && I2C select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help @@ -51,7 +53,7 @@ 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_SMDKV210 || MACH_SMDKC110) + depends on MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 depends on I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -60,7 +62,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994 @@ -70,7 +71,7 @@ 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 + depends on MACH_SMDK2443 select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -80,7 +81,7 @@ 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 + depends on ARCH_S3C24XX select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -90,7 +91,7 @@ config SND_SOC_SAMSUNG_LN2440SBC_ALC650 config SND_SOC_SAMSUNG_S3C24XX_UDA134X tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on SND_SOC_SAMSUNG && ARCH_S3C24XX + depends on ARCH_S3C24XX select SND_S3C24XX_I2S select SND_SOC_L3 select SND_SOC_UDA134X @@ -102,21 +103,21 @@ 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 && I2C + depends on 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 && I2C + depends on ARCH_S3C24XX && I2C select SND_S3C24XX_I2S select SND_SOC_TLV320AIC3X select SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_H1940_UDA1380 tristate "Audio support for the HP iPAQ H1940" - depends on SND_SOC_SAMSUNG && ARCH_H1940 && I2C + depends on ARCH_H1940 && I2C select SND_S3C24XX_I2S select SND_SOC_UDA1380 help @@ -124,7 +125,7 @@ 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 && I2C + depends on MACH_RX1950 && I2C select SND_S3C24XX_I2S select SND_SOC_UDA1380 help @@ -132,7 +133,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) + depends on MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help @@ -140,20 +141,19 @@ config SND_SOC_SAMSUNG_SMDK_WM9713 config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" - depends on SND_SOC_SAMSUNG && MACH_SMARTQ && I2C + depends on MACH_SMARTQ && I2C select SND_SAMSUNG_I2S select SND_SOC_WM8750 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. config SND_SOC_SMDK_WM8580_PCM tristate "SoC PCM Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110) + depends on MACH_SMDKV210 || MACH_SMDKC110 depends on I2C select SND_SOC_WM8580 select SND_SAMSUNG_PCM @@ -162,7 +162,6 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994 @@ -172,7 +171,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && I2C && SPI_MASTER + depends on I2C && SPI_MASTER depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM8996 @@ -182,14 +181,14 @@ config SND_SOC_SPEYSIDE config SND_SOC_TOBERMORY tristate "Audio support for Wolfson Tobermory" - depends on SND_SOC_SAMSUNG && INPUT && I2C + depends on INPUT && I2C depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM8962 config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on MFD_ARIZONA && I2C && SPI_MASTER depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5102 @@ -200,7 +199,7 @@ config SND_SOC_BELLS config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" - depends on SND_SOC_SAMSUNG && I2C + depends on I2C depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5100 @@ -208,7 +207,7 @@ config SND_SOC_LOWLAND config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && I2C + depends on I2C depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select MFD_WM8994 @@ -216,7 +215,7 @@ config SND_SOC_LITTLEMILL config SND_SOC_SNOW tristate "Audio support for Google Snow boards" - depends on SND_SOC_SAMSUNG && I2C + depends on I2C select SND_SOC_MAX98090 select SND_SOC_MAX98095 select SND_SAMSUNG_I2S @@ -226,6 +225,8 @@ config SND_SOC_SNOW config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on SND_SOC_SAMSUNG && I2C + depends on I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631 + +endif #SND_SOC_SAMSUNG -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: Intel: Skylake: add missing \n to end of dev_* messages
by Colin King 25 Sep '16

25 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trival fix, some dev_* messages are missing a \n, so add it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/bxt-sst.c | 6 +++--- sound/soc/intel/skylake/skl-sst-cldma.c | 4 ++-- sound/soc/intel/skylake/skl-sst-ipc.c | 12 ++++++------ sound/soc/intel/skylake/skl-sst.c | 6 +++--- sound/soc/intel/skylake/skl-topology.c | 18 +++++++++--------- 5 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 48a4ae5..1d251d5 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -404,7 +404,7 @@ static int bxt_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id) ret = skl_dsp_disable_core(ctx, core_mask); if (ret < 0) { - dev_err(ctx->dev, "Failed to disable core %d", ret); + dev_err(ctx->dev, "Failed to disable core %d\n", ret); return ret; } skl->cores.state[core_id] = SKL_DSP_RESET; @@ -489,7 +489,7 @@ int bxt_sst_init_fw(struct device *dev, struct skl_sst *ctx) ret = sst->fw_ops.load_fw(sst); if (ret < 0) { - dev_err(dev, "Load base fw failed: %x", ret); + dev_err(dev, "Load base fw failed: %x\n", ret); return ret; } @@ -498,7 +498,7 @@ int bxt_sst_init_fw(struct device *dev, struct skl_sst *ctx) if (ctx->manifest.lib_count > 1) { ret = sst->fw_ops.load_library(sst, &ctx->manifest); if (ret < 0) { - dev_err(dev, "Load Library failed : %x", ret); + dev_err(dev, "Load Library failed : %x\n", ret); return ret; } } diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c index da2329d..efa2532 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.c +++ b/sound/soc/intel/skylake/skl-sst-cldma.c @@ -341,14 +341,14 @@ int skl_cldma_prepare(struct sst_dsp *ctx) ret = ctx->dsp_ops.alloc_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data, ctx->cl_dev.bufsize); if (ret < 0) { - dev_err(ctx->dev, "Alloc buffer for base fw failed: %x", ret); + dev_err(ctx->dev, "Alloc buffer for base fw failed: %x\n", ret); return ret; } /* Setup Code loader BDL */ ret = ctx->dsp_ops.alloc_dma_buf(ctx->dev, &ctx->cl_dev.dmab_bdl, PAGE_SIZE); if (ret < 0) { - dev_err(ctx->dev, "Alloc buffer for blde failed: %x", ret); + dev_err(ctx->dev, "Alloc buffer for blde failed: %x\n", ret); ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data); return ret; } diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 8c5d7f5..797cf40 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -344,7 +344,7 @@ static int skl_ipc_process_notification(struct sst_generic_ipc *ipc, break; default: - dev_err(ipc->dev, "ipc: Unhandled error msg=%x", + dev_err(ipc->dev, "ipc: Unhandled error msg=%x\n", header.primary); break; } @@ -385,13 +385,13 @@ static void skl_ipc_process_reply(struct sst_generic_ipc *ipc, break; default: - dev_err(ipc->dev, "Unknown ipc reply: 0x%x", reply); + dev_err(ipc->dev, "Unknown ipc reply: 0x%x\n", reply); msg->errno = -EINVAL; break; } if (reply != IPC_GLB_REPLY_SUCCESS) { - dev_err(ipc->dev, "ipc FW reply: reply=%d", reply); + dev_err(ipc->dev, "ipc FW reply: reply=%d\n", reply); dev_err(ipc->dev, "FW Error Code: %u\n", ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); } @@ -440,9 +440,9 @@ irqreturn_t skl_dsp_irq_thread_handler(int irq, void *context) hipcte = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCTE); header.primary = hipct; header.extension = hipcte; - dev_dbg(dsp->dev, "IPC irq: Firmware respond primary:%x", + dev_dbg(dsp->dev, "IPC irq: Firmware respond primary:%x\n", header.primary); - dev_dbg(dsp->dev, "IPC irq: Firmware respond extension:%x", + dev_dbg(dsp->dev, "IPC irq: Firmware respond extension:%x\n", header.extension); if (IPC_GLB_NOTIFY_RSP_TYPE(header.primary)) { @@ -749,7 +749,7 @@ int skl_ipc_bind_unbind(struct sst_generic_ipc *ipc, header.extension); ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0); if (ret < 0) { - dev_err(ipc->dev, "ipc: bind/unbind faileden"); + dev_err(ipc->dev, "ipc: bind/unbind failed\n"); return ret; } diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 064fc7e..8fc3178 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -107,13 +107,13 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) ret = skl_dsp_boot(ctx); if (ret < 0) { - dev_err(ctx->dev, "Boot dsp core failed ret: %d", ret); + dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret); goto skl_load_base_firmware_failed; } ret = skl_cldma_prepare(ctx); if (ret < 0) { - dev_err(ctx->dev, "CL dma prepare failed : %d", ret); + dev_err(ctx->dev, "CL dma prepare failed : %d\n", ret); goto skl_load_base_firmware_failed; } @@ -502,7 +502,7 @@ int skl_sst_init_fw(struct device *dev, struct skl_sst *ctx) ret = sst->fw_ops.load_fw(sst); if (ret < 0) { - dev_err(dev, "Load base fw failed : %d", ret); + dev_err(dev, "Load base fw failed : %d\n", ret); return ret; } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2d475b7..bd17996 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1588,7 +1588,7 @@ static int skl_tplg_fill_pins_info(struct device *dev, break; default: - dev_err(dev, "Invalid direction value"); + dev_err(dev, "Invalid direction value\n"); return -EINVAL; } @@ -1626,7 +1626,7 @@ static int skl_tplg_fill_fmt(struct device *dev, break; default: - dev_err(dev, "Invalid direction value"); + dev_err(dev, "Invalid direction value\n"); return -EINVAL; } @@ -1664,7 +1664,7 @@ static int skl_tplg_fill_fmt(struct device *dev, break; default: - dev_err(dev, "Invalid token %d", tkn); + dev_err(dev, "Invalid token %d\n", tkn); return -EINVAL; } @@ -1677,7 +1677,7 @@ static int skl_tplg_get_uuid(struct device *dev, struct skl_module_cfg *mconfig, if (uuid_tkn->token == SKL_TKN_UUID) memcpy(&mconfig->guid, &uuid_tkn->uuid, 16); else { - dev_err(dev, "Not an UUID token tkn %d", uuid_tkn->token); + dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token); return -EINVAL; } @@ -1913,7 +1913,7 @@ static int skl_tplg_get_tokens(struct device *dev, switch (array->type) { case SND_SOC_TPLG_TUPLE_TYPE_STRING: - dev_warn(dev, "no string tokens expected for skl tplg"); + dev_warn(dev, "no string tokens expected for skl tplg\n"); continue; case SND_SOC_TPLG_TUPLE_TYPE_UUID: @@ -1966,7 +1966,7 @@ static int skl_tplg_get_desc_blocks(struct device *dev, return tkn_elem->value; default: - dev_err(dev, "Invalid descriptor token %d", tkn_elem->token); + dev_err(dev, "Invalid descriptor token %d\n", tkn_elem->token); break; } @@ -2226,7 +2226,7 @@ static int skl_tplg_fill_str_mfest_tkn(struct device *dev, break; default: - dev_err(dev, "Not a string token %d", str_elem->token); + dev_err(dev, "Not a string token %d\n", str_elem->token); break; } @@ -2267,7 +2267,7 @@ static int skl_tplg_get_int_tkn(struct device *dev, break; default: - dev_err(dev, "Not a manifest token %d", tkn_elem->token); + dev_err(dev, "Not a manifest token %d\n", tkn_elem->token); return -EINVAL; } @@ -2306,7 +2306,7 @@ static int skl_tplg_get_manifest_tkn(struct device *dev, continue; case SND_SOC_TPLG_TUPLE_TYPE_UUID: - dev_warn(dev, "no uuid tokens for skl tplf manifest"); + dev_warn(dev, "no uuid tokens for skl tplf manifest\n"); continue; default: -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: max98926: add missing \n to end of dev_err/dev_dbg messages
by Colin King 25 Sep '16

25 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trival fix, some dev_err and deb_dbg messages are missing a \n, so add it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/max98926.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 5830a81..1eff7e0 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -347,7 +347,7 @@ static int max98926_dai_set_fmt(struct snd_soc_dai *codec_dai, max98926_set_sense_data(max98926); break; default: - dev_err(codec->dev, "DAI clock mode unsupported"); + dev_err(codec->dev, "DAI clock mode unsupported\n"); return -EINVAL; } @@ -364,7 +364,7 @@ static int max98926_dai_set_fmt(struct snd_soc_dai *codec_dai, invert = MAX98926_DAI_BCI_MASK | MAX98926_DAI_WCI_MASK; break; default: - dev_err(codec->dev, "DAI invert mode unsupported"); + dev_err(codec->dev, "DAI invert mode unsupported\n"); return -EINVAL; } @@ -408,7 +408,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream, max98926->ch_size = 32; break; default: - dev_dbg(codec->dev, "format unsupported %d", + dev_dbg(codec->dev, "format unsupported %d\n", params_format(params)); return -EINVAL; } -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: samsung: add missing \n to end of pr_err messages
by Colin King 25 Sep '16

25 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trival fix, some pr_err messages are missing a \n, so add it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/samsung/ac97.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 5eafb66..97d6700 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -74,7 +74,7 @@ static void s3c_ac97_activate(struct snd_ac97 *ac97) writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - pr_err("AC97: Unable to activate!"); + pr_err("AC97: Unable to activate!\n"); } static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, @@ -100,7 +100,7 @@ static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - pr_err("AC97: Unable to read!"); + pr_err("AC97: Unable to read!\n"); stat = readl(s3c_ac97.regs + S3C_AC97_STAT); addr = (stat >> 16) & 0x7f; @@ -137,7 +137,7 @@ static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - pr_err("AC97: Unable to write!"); + pr_err("AC97: Unable to write!\n"); ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; -- 2.9.3

3 2

[alsa-devel] [PATCH] ASoC: Intel: atom: add missing \n to end of dev_err/dev_dbg messages
by Colin King 25 Sep '16

25 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trival fix, some dev_err/deb_dbg messages are missing a \n, so add it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 773acfb..9718e82 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -166,7 +166,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) rsrc = platform_get_resource(pdev, IORESOURCE_MEM, ctx->pdata->res_info->acpi_lpe_res_index); if (!rsrc) { - dev_err(ctx->dev, "Invalid SHIM base from IFWI"); + dev_err(ctx->dev, "Invalid SHIM base from IFWI\n"); return -EIO; } dev_info(ctx->dev, "LPE base: %#x size:%#x", (unsigned int) rsrc->start, @@ -178,7 +178,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) ctx->iram = devm_ioremap_nocache(ctx->dev, ctx->iram_base, ctx->pdata->res_info->iram_size); if (!ctx->iram) { - dev_err(ctx->dev, "unable to map IRAM"); + dev_err(ctx->dev, "unable to map IRAM\n"); return -EIO; } @@ -188,7 +188,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) ctx->dram = devm_ioremap_nocache(ctx->dev, ctx->dram_base, ctx->pdata->res_info->dram_size); if (!ctx->dram) { - dev_err(ctx->dev, "unable to map DRAM"); + dev_err(ctx->dev, "unable to map DRAM\n"); return -EIO; } @@ -197,7 +197,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) ctx->shim = devm_ioremap_nocache(ctx->dev, ctx->shim_phy_add, ctx->pdata->res_info->shim_size); if (!ctx->shim) { - dev_err(ctx->dev, "unable to map SHIM"); + dev_err(ctx->dev, "unable to map SHIM\n"); return -EIO; } @@ -210,7 +210,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) ctx->mailbox = devm_ioremap_nocache(ctx->dev, ctx->mailbox_add, ctx->pdata->res_info->mbox_size); if (!ctx->mailbox) { - dev_err(ctx->dev, "unable to map mailbox"); + dev_err(ctx->dev, "unable to map mailbox\n"); return -EIO; } @@ -220,7 +220,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) rsrc = platform_get_resource(pdev, IORESOURCE_MEM, ctx->pdata->res_info->acpi_ddr_index); if (!rsrc) { - dev_err(ctx->dev, "Invalid DDR base from IFWI"); + dev_err(ctx->dev, "Invalid DDR base from IFWI\n"); return -EIO; } ctx->ddr_base = rsrc->start; @@ -229,7 +229,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) ctx->ddr = devm_ioremap_nocache(ctx->dev, ctx->ddr_base, resource_size(rsrc)); if (!ctx->ddr) { - dev_err(ctx->dev, "unable to map DDR"); + dev_err(ctx->dev, "unable to map DDR\n"); return -EIO; } @@ -296,7 +296,7 @@ static int sst_acpi_probe(struct platform_device *pdev) id = acpi_match_device(dev->driver->acpi_match_table, dev); if (!id) return -ENODEV; - dev_dbg(dev, "for %s", id->id); + dev_dbg(dev, "for %s\n", id->id); mach = (struct sst_acpi_mach *)id->driver_data; mach = sst_acpi_find_machine(mach); -- 2.9.3

2 1

[alsa-devel] [PATCH v2] ASoC: rt5660: add rt5660 codec driver
by Oder Chiou 25 Sep '16

25 Sep '16

This is the initial codec driver for rt5660 Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5660.txt | 47 + include/sound/rt5660.h | 31 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5660.c | 1353 ++++++++++++++++++++ sound/soc/codecs/rt5660.h | 847 ++++++++++++ 6 files changed, 2286 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5660.txt create mode 100644 include/sound/rt5660.h create mode 100644 sound/soc/codecs/rt5660.c create mode 100644 sound/soc/codecs/rt5660.h diff --git a/Documentation/devicetree/bindings/sound/rt5660.txt b/Documentation/devicetree/bindings/sound/rt5660.txt new file mode 100644 index 0000000..30be5f9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5660.txt @@ -0,0 +1,47 @@ +RT5660 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5660". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in3-differential + Boolean. Indicate MIC1/3 input are differential, rather than single-ended. + +- realtek,poweroff-in-suspend + Boolean. If the codec will be powered off in suspend, the resume should be + added delay time for waiting codec power ready. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using IN1P pin as dmic1 data pin + +Pins on the device (for linking into audio routes) for RT5660: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN1N + * IN2P + * IN3P + * IN3N + * SPO + * LOUTL + * LOUTR + +Example: + +rt5660 { + compatible = "realtek,rt5660"; + reg = <0x1c>; +}; diff --git a/include/sound/rt5660.h b/include/sound/rt5660.h new file mode 100644 index 0000000..065f83a --- /dev/null +++ b/include/sound/rt5660.h @@ -0,0 +1,31 @@ +/* + * linux/sound/rt5660.h -- Platform data for RT5660 + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(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 __LINUX_SND_RT5660_H +#define __LINUX_SND_RT5660_H + +enum rt5660_dmic1_data_pin { + RT5660_DMIC1_NULL, + RT5660_DMIC1_DATA_GPIO2, + RT5660_DMIC1_DATA_IN1P, +}; + +struct rt5660_platform_data { + /* IN1 & IN3 can optionally be differential */ + bool in1_diff; + bool in3_diff; + bool use_ldo2; + bool poweroff_codec_in_suspend; + + enum rt5660_dmic1_data_pin dmic1_data_pin; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3fd110..b930a1b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -113,6 +113,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5659 if I2C + select SND_SOC_RT5660 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -646,6 +647,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y + default y if SND_SOC_RT5660=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -654,6 +656,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m default m if SND_SOC_RT5659=m + default m if SND_SOC_RT5660=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -698,6 +701,9 @@ config SND_SOC_RT5651 config SND_SOC_RT5659 tristate +config SND_SOC_RT5660 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ffaa523..251f626 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -113,6 +113,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5659-objs := rt5659.o +snd-soc-rt5660-objs := rt5660.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -335,6 +336,7 @@ 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_RT5659) += snd-soc-rt5659.o +obj-$(CONFIG_SND_SOC_RT5660) += snd-soc-rt5660.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c new file mode 100644 index 0000000..9f0933c --- /dev/null +++ b/sound/soc/codecs/rt5660.c @@ -0,0 +1,1353 @@ +/* + * rt5660.c -- RT5660 ALSA SoC audio codec driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(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/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" +#include "rt5660.h" + +#define RT5660_DEVICE_ID 0x6338 + +#define RT5660_PR_RANGE_BASE (0xff + 1) +#define RT5660_PR_SPACING 0x100 + +#define RT5660_PR_BASE (RT5660_PR_RANGE_BASE + (0 * RT5660_PR_SPACING)) + +static const struct regmap_range_cfg rt5660_ranges[] = { + { .name = "PR", .range_min = RT5660_PR_BASE, + .range_max = RT5660_PR_BASE + 0xf3, + .selector_reg = RT5660_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5660_PRIV_DATA, + .window_len = 0x1, }, +}; + +static const struct reg_sequence rt5660_patch[] = { + { RT5660_ALC_PGA_CTRL2, 0x44c3 }, + { RT5660_PR_BASE + 0x3d, 0x2600 }, +}; + +static const struct reg_default rt5660_reg[] = { + { 0x00, 0x0000 }, + { 0x01, 0xc800 }, + { 0x02, 0xc8c8 }, + { 0x0d, 0x1010 }, + { 0x0e, 0x1010 }, + { 0x19, 0xafaf }, + { 0x1c, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x6060 }, + { 0x29, 0x8080 }, + { 0x2a, 0x4242 }, + { 0x2f, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe000 }, + { 0x46, 0x003e }, + { 0x48, 0xf800 }, + { 0x4a, 0x0004 }, + { 0x4d, 0x0000 }, + { 0x4e, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x70, 0x8000 }, + { 0x73, 0x7000 }, + { 0x74, 0x3c00 }, + { 0x75, 0x2800 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x8c, 0x0228 }, + { 0x8d, 0xa000 }, + { 0x8e, 0x0000 }, + { 0x92, 0x0000 }, + { 0x93, 0x3000 }, + { 0xa1, 0x0059 }, + { 0xa2, 0x0001 }, + { 0xa3, 0x5c80 }, + { 0xa4, 0x0146 }, + { 0xa5, 0x1f1f }, + { 0xa6, 0x78c6 }, + { 0xa7, 0xe5ec }, + { 0xa8, 0xba61 }, + { 0xa9, 0x3c78 }, + { 0xaa, 0x8ae2 }, + { 0xab, 0xe5ec }, + { 0xac, 0xc600 }, + { 0xad, 0xba61 }, + { 0xae, 0x17ed }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x020c }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x4000 }, + { 0xbb, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0100 }, + { 0xc0, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xd3, 0xa220 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xe0, 0x8000 }, + { 0xe1, 0x0200 }, + { 0xe2, 0x8000 }, + { 0xe3, 0x0200 }, + { 0xe4, 0x0f20 }, + { 0xe5, 0x001f }, + { 0xe6, 0x020c }, + { 0xe7, 0x1f00 }, + { 0xe8, 0x0000 }, + { 0xe9, 0x4000 }, + { 0xea, 0x00a6 }, + { 0xeb, 0x04c3 }, + { 0xec, 0x27c8 }, + { 0xed, 0x7418 }, + { 0xee, 0xbf50 }, + { 0xef, 0x0045 }, + { 0xf0, 0x0007 }, + { 0xfa, 0x0000 }, + { 0xfd, 0x0000 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6338 }, +}; + +static bool rt5660_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5660_ranges); i++) + if ((reg >= rt5660_ranges[i].window_start && + reg <= rt5660_ranges[i].window_start + + rt5660_ranges[i].window_len) || + (reg >= rt5660_ranges[i].range_min && + reg <= rt5660_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5660_RESET: + case RT5660_PRIV_DATA: + case RT5660_EQ_CTRL1: + case RT5660_IRQ_CTRL2: + case RT5660_INT_IRQ_ST: + case RT5660_VENDOR_ID: + case RT5660_VENDOR_ID1: + case RT5660_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5660_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5660_ranges); i++) + if ((reg >= rt5660_ranges[i].window_start && + reg <= rt5660_ranges[i].window_start + + rt5660_ranges[i].window_len) || + (reg >= rt5660_ranges[i].range_min && + reg <= rt5660_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5660_RESET: + case RT5660_SPK_VOL: + case RT5660_LOUT_VOL: + case RT5660_IN1_IN2: + case RT5660_IN3_IN4: + case RT5660_DAC1_DIG_VOL: + case RT5660_STO1_ADC_DIG_VOL: + case RT5660_ADC_BST_VOL1: + case RT5660_STO1_ADC_MIXER: + case RT5660_AD_DA_MIXER: + case RT5660_STO_DAC_MIXER: + case RT5660_DIG_INF1_DATA: + case RT5660_REC_L1_MIXER: + case RT5660_REC_L2_MIXER: + case RT5660_REC_R1_MIXER: + case RT5660_REC_R2_MIXER: + case RT5660_LOUT_MIXER: + case RT5660_SPK_MIXER: + case RT5660_SPO_MIXER: + case RT5660_SPO_CLSD_RATIO: + case RT5660_OUT_L_GAIN1: + case RT5660_OUT_L_GAIN2: + case RT5660_OUT_L1_MIXER: + case RT5660_OUT_R_GAIN1: + case RT5660_OUT_R_GAIN2: + case RT5660_OUT_R1_MIXER: + case RT5660_PWR_DIG1: + case RT5660_PWR_DIG2: + case RT5660_PWR_ANLG1: + case RT5660_PWR_ANLG2: + case RT5660_PWR_MIXER: + case RT5660_PWR_VOL: + case RT5660_PRIV_INDEX: + case RT5660_PRIV_DATA: + case RT5660_I2S1_SDP: + case RT5660_ADDA_CLK1: + case RT5660_ADDA_CLK2: + case RT5660_DMIC_CTRL1: + case RT5660_GLB_CLK: + case RT5660_PLL_CTRL1: + case RT5660_PLL_CTRL2: + case RT5660_CLSD_AMP_OC_CTRL: + case RT5660_CLSD_AMP_CTRL: + case RT5660_LOUT_AMP_CTRL: + case RT5660_SPK_AMP_SPKVDD: + case RT5660_MICBIAS: + case RT5660_CLSD_OUT_CTRL1: + case RT5660_CLSD_OUT_CTRL2: + case RT5660_DIPOLE_MIC_CTRL1: + case RT5660_DIPOLE_MIC_CTRL2: + case RT5660_DIPOLE_MIC_CTRL3: + case RT5660_DIPOLE_MIC_CTRL4: + case RT5660_DIPOLE_MIC_CTRL5: + case RT5660_DIPOLE_MIC_CTRL6: + case RT5660_DIPOLE_MIC_CTRL7: + case RT5660_DIPOLE_MIC_CTRL8: + case RT5660_DIPOLE_MIC_CTRL9: + case RT5660_DIPOLE_MIC_CTRL10: + case RT5660_DIPOLE_MIC_CTRL11: + case RT5660_DIPOLE_MIC_CTRL12: + case RT5660_EQ_CTRL1: + case RT5660_EQ_CTRL2: + case RT5660_DRC_AGC_CTRL1: + case RT5660_DRC_AGC_CTRL2: + case RT5660_DRC_AGC_CTRL3: + case RT5660_DRC_AGC_CTRL4: + case RT5660_DRC_AGC_CTRL5: + case RT5660_JD_CTRL: + case RT5660_IRQ_CTRL1: + case RT5660_IRQ_CTRL2: + case RT5660_INT_IRQ_ST: + case RT5660_GPIO_CTRL1: + case RT5660_GPIO_CTRL2: + case RT5660_WIND_FILTER_CTRL1: + case RT5660_SV_ZCD1: + case RT5660_SV_ZCD2: + case RT5660_DRC1_LM_CTRL1: + case RT5660_DRC1_LM_CTRL2: + case RT5660_DRC2_LM_CTRL1: + case RT5660_DRC2_LM_CTRL2: + case RT5660_MULTI_DRC_CTRL: + case RT5660_DRC2_CTRL1: + case RT5660_DRC2_CTRL2: + case RT5660_DRC2_CTRL3: + case RT5660_DRC2_CTRL4: + case RT5660_DRC2_CTRL5: + case RT5660_ALC_PGA_CTRL1: + case RT5660_ALC_PGA_CTRL2: + case RT5660_ALC_PGA_CTRL3: + case RT5660_ALC_PGA_CTRL4: + case RT5660_ALC_PGA_CTRL5: + case RT5660_ALC_PGA_CTRL6: + case RT5660_ALC_PGA_CTRL7: + case RT5660_GEN_CTRL1: + case RT5660_GEN_CTRL2: + case RT5660_GEN_CTRL3: + case RT5660_VENDOR_ID: + case RT5660_VENDOR_ID1: + case RT5660_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(rt5660_out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_adc_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_bst_tlv, -1200, 75, 0); + +static const struct snd_kcontrol_new rt5660_snd_controls[] = { + /* Speaker Output Volume */ + SOC_SINGLE("Speaker Playback Switch", RT5660_SPK_VOL, RT5660_L_MUTE_SFT, + 1, 1), + SOC_SINGLE_TLV("Speaker Playback Volume", RT5660_SPK_VOL, + RT5660_L_VOL_SFT, 39, 1, rt5660_out_vol_tlv), + + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5660_LOUT_VOL, RT5660_L_MUTE_SFT, + RT5660_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5660_LOUT_VOL, RT5660_L_VOL_SFT, + RT5660_R_VOL_SFT, 39, 1, rt5660_out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5660_DAC1_DIG_VOL, + RT5660_DAC_L1_VOL_SFT, RT5660_DAC_R1_VOL_SFT, 87, 0, + rt5660_dac_vol_tlv), + + /* IN1/IN2/IN3 Control */ + SOC_SINGLE_TLV("IN1 Boost Volume", RT5660_IN1_IN2, RT5660_BST_SFT1, 69, + 0, rt5660_bst_tlv), + SOC_SINGLE_TLV("IN2 Boost Volume", RT5660_IN1_IN2, RT5660_BST_SFT2, 69, + 0, rt5660_bst_tlv), + SOC_SINGLE_TLV("IN3 Boost Volume", RT5660_IN3_IN4, RT5660_BST_SFT3, 69, + 0, rt5660_bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5660_STO1_ADC_DIG_VOL, + RT5660_L_MUTE_SFT, RT5660_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5660_STO1_ADC_DIG_VOL, + RT5660_ADC_L_VOL_SFT, RT5660_ADC_R_VOL_SFT, 63, 0, + rt5660_adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5660_ADC_BST_VOL1, + RT5660_STO1_ADC_L_BST_SFT, RT5660_STO1_ADC_R_BST_SFT, 3, 0, + rt5660_adc_bst_tlv), +}; + +/** + * rt5660_set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + */ +static int rt5660_set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + int idx, rate; + + rate = rt5660->sysclk / rl6231_get_pre_div(rt5660->regmap, + RT5660_ADDA_CLK1, RT5660_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5660_DMIC_CTRL1, + RT5660_DMIC_CLK_MASK, idx << RT5660_DMIC_CLK_SFT); + + return idx; +} + +static int rt5660_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + unsigned int val; + + val = snd_soc_read(codec, RT5660_GLB_CLK); + val &= RT5660_SCLK_SRC_MASK; + if (val == RT5660_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5660_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5660_AD_DA_MIXER, + RT5660_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5660_AD_DA_MIXER, + RT5660_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5660_AD_DA_MIXER, + RT5660_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5660_AD_DA_MIXER, + RT5660_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5660_rec_l_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST3_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5660_REC_L2_MIXER, + RT5660_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_rec_r_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST3_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5660_REC_R2_MIXER, + RT5660_M_OM_R_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_spk_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_SPK_MIXER, + RT5660_M_BST3_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_SPK_MIXER, + RT5660_M_BST1_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_SPK_MIXER, + RT5660_M_DACL_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_SPK_MIXER, + RT5660_M_DACR_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIXL Switch", RT5660_SPK_MIXER, + RT5660_M_OM_L_SM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_out_l_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST3_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIXL Switch", RT5660_OUT_L1_MIXER, + RT5660_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_OUT_L1_MIXER, + RT5660_M_DAC_R_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_OUT_L1_MIXER, + RT5660_M_DAC_L_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5660_OUT_R1_MIXER, + RT5660_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_OUT_R1_MIXER, + RT5660_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Switch", RT5660_OUT_R1_MIXER, + RT5660_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_OUT_R1_MIXER, + RT5660_M_DAC_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_OUT_R1_MIXER, + RT5660_M_DAC_L_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_spo_mix[] = { + SOC_DAPM_SINGLE("DACR Switch", RT5660_SPO_MIXER, + RT5660_M_DAC_R_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_SPO_MIXER, + RT5660_M_DAC_L_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL Switch", RT5660_SPO_MIXER, + RT5660_M_SV_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_SPO_MIXER, + RT5660_M_BST1_SPM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_lout_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT5660_LOUT_MIXER, + RT5660_M_DAC1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX Switch", RT5660_LOUT_MIXER, + RT5660_M_LOVOL_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spk_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_SPK_VOL, + RT5660_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_l_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_LOUT_VOL, + RT5660_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_LOUT_VOL, + RT5660_VOL_R_SFT, 1, 1); + +/* Interface data select */ +static const char * const rt5660_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5660_if1_dac_enum, + RT5660_DIG_INF1_DATA, RT5660_IF1_DAC_IN_SFT, rt5660_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5660_if1_adc_enum, + RT5660_DIG_INF1_DATA, RT5660_IF1_ADC_IN_SFT, rt5660_data_select); + +static const struct snd_kcontrol_new rt5660_if1_dac_swap_mux = + SOC_DAPM_ENUM("IF1 DAC Swap Source", rt5660_if1_dac_enum); + +static const struct snd_kcontrol_new rt5660_if1_adc_swap_mux = + SOC_DAPM_ENUM("IF1 ADC Swap Source", rt5660_if1_adc_enum); + +static int rt5660_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5660_LOUT_AMP_CTRL, + RT5660_LOUT_CO_MASK | RT5660_LOUT_CB_MASK, + RT5660_LOUT_CO_EN | RT5660_LOUT_CB_PU); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5660_LOUT_AMP_CTRL, + RT5660_LOUT_CO_MASK | RT5660_LOUT_CB_MASK, + RT5660_LOUT_CO_DIS | RT5660_LOUT_CB_PD); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5660_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5660_PWR_ANLG1, + RT5660_PWR_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5660_PWR_ANLG2, + RT5660_PWR_PLL_BIT, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5660_PWR_ANLG2, + RT5660_PWR_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5660_PWR_ANLG2, + RT5660_PWR_MB2_BIT, 0, NULL, 0), + + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + rt5660_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Power", RT5660_DMIC_CTRL1, + RT5660_DMIC_1_EN_SFT, 0, NULL, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5660_PWR_ANLG2, RT5660_PWR_BST1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("BST2", RT5660_PWR_ANLG2, RT5660_PWR_BST2_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5660_PWR_ANLG2, RT5660_PWR_BST3_BIT, 0, + NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5660_PWR_MIXER, RT5660_PWR_RM_L_BIT, + 0, rt5660_rec_l_mix, ARRAY_SIZE(rt5660_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5660_PWR_MIXER, RT5660_PWR_RM_R_BIT, + 0, rt5660_rec_r_mix, ARRAY_SIZE(rt5660_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", RT5660_PWR_DIG1, + RT5660_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC R power", RT5660_PWR_DIG1, + RT5660_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", RT5660_PR_BASE + RT5660_CHOP_DAC_ADC, + 12, 0, NULL, 0), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5660_PWR_DIG2, + RT5660_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5660_sto1_adc_l_mix, ARRAY_SIZE(rt5660_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5660_sto1_adc_r_mix, ARRAY_SIZE(rt5660_sto1_adc_r_mix)), + + /* ADC */ + SND_SOC_DAPM_ADC("Stereo1 ADC MIXL", NULL, RT5660_STO1_ADC_DIG_VOL, + RT5660_L_MUTE_SFT, 1), + SND_SOC_DAPM_ADC("Stereo1 ADC MIXR", NULL, RT5660_STO1_ADC_DIG_VOL, + RT5660_R_MUTE_SFT, 1), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5660_PWR_DIG1, RT5660_PWR_I2S1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF1 DAC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5660_if1_dac_swap_mux), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF1 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5660_if1_adc_swap_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), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, rt5660_dac_l_mix, + ARRAY_SIZE(rt5660_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, rt5660_dac_r_mix, + ARRAY_SIZE(rt5660_dac_r_mix)), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5660_PWR_DIG2, + RT5660_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5660_sto_dac_l_mix, ARRAY_SIZE(rt5660_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5660_sto_dac_r_mix, ARRAY_SIZE(rt5660_sto_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5660_PWR_DIG1, + RT5660_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5660_PWR_DIG1, + RT5660_PWR_DAC_R1_BIT, 0), + + /* OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIX", RT5660_PWR_MIXER, RT5660_PWR_SM_BIT, + 0, rt5660_spk_mix, ARRAY_SIZE(rt5660_spk_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5660_PWR_MIXER, RT5660_PWR_OM_L_BIT, + 0, rt5660_out_l_mix, ARRAY_SIZE(rt5660_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5660_PWR_MIXER, RT5660_PWR_OM_R_BIT, + 0, rt5660_out_r_mix, ARRAY_SIZE(rt5660_out_r_mix)), + + /* Output Volume */ + SND_SOC_DAPM_SWITCH("SPKVOL", RT5660_PWR_VOL, + RT5660_PWR_SV_BIT, 0, &spk_vol_control), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("LOUTVOL", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("LOUTVOL L", SND_SOC_NOPM, + RT5660_PWR_LV_L_BIT, 0, &lout_l_vol_control), + SND_SOC_DAPM_SWITCH("LOUTVOL R", SND_SOC_NOPM, + RT5660_PWR_LV_R_BIT, 0, &lout_r_vol_control), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPO MIX", SND_SOC_NOPM, 0, + 0, rt5660_spo_mix, ARRAY_SIZE(rt5660_spo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, + rt5660_lout_mix, ARRAY_SIZE(rt5660_lout_mix)), + SND_SOC_DAPM_SUPPLY("VREF HP", RT5660_GEN_CTRL1, + RT5660_PWR_VREF_HP_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, RT5660_PWR_ANLG1, + RT5660_PWR_HA_BIT, 0, rt5660_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 1, RT5660_PWR_DIG1, + RT5660_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("SPO"), +}; + +static const struct snd_soc_dapm_route rt5660_dapm_routes[] = { + { "MICBIAS1", NULL, "LDO2" }, + { "MICBIAS2", NULL, "LDO2" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST2", NULL, "IN2P" }, + { "BST3", NULL, "IN3P" }, + { "BST3", NULL, "IN3N" }, + + { "RECMIXL", "BST3 Switch", "BST3" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + { "RECMIXL", "OUT MIXL Switch", "OUT MIXL" }, + + { "RECMIXR", "BST3 Switch", "BST3" }, + { "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 L", NULL, "ADC clock" }, + { "ADC R", NULL, "RECMIXR" }, + { "ADC R", NULL, "ADC R power" }, + { "ADC R", NULL, "ADC clock" }, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC Power"}, + + { "Sto1 ADC MIXL", "ADC1 Switch", "ADC L" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "DMIC L1" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "ADC R" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "DMIC R1" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "IF1 ADC", NULL, "Stereo1 ADC MIXL" }, + { "IF1 ADC", NULL, "Stereo1 ADC MIXR" }, + { "IF1 ADC", NULL, "I2S1" }, + + { "IF1 ADC Swap Mux", "L/R", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "R/L", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "L/L", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "R/R", "IF1 ADC" }, + { "AIF1TX", NULL, "IF1 ADC Swap Mux" }, + + { "IF1 DAC", NULL, "AIF1RX" }, + { "IF1 DAC", NULL, "I2S1" }, + + { "IF1 DAC Swap Mux", "L/R", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "R/L", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "L/L", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "R/R", "IF1 DAC" }, + + { "IF1 DAC L", NULL, "IF1 DAC Swap Mux" }, + { "IF1 DAC R", NULL, "IF1 DAC Swap Mux" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "IF1 DAC L" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "IF1 DAC R" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + + { "SPK MIX", "BST3 Switch", "BST3" }, + { "SPK MIX", "BST1 Switch", "BST1" }, + { "SPK MIX", "DACL Switch", "DAC L1" }, + { "SPK MIX", "DACR Switch", "DAC R1" }, + { "SPK MIX", "OUTMIXL Switch", "OUT MIXL" }, + + { "OUT MIXL", "BST3 Switch", "BST3" }, + { "OUT MIXL", "BST2 Switch", "BST2" }, + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "RECMIXL Switch", "RECMIXL" }, + { "OUT MIXL", "DACR Switch", "DAC R1" }, + { "OUT MIXL", "DACL Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "BST1 Switch", "BST1" }, + { "OUT MIXR", "RECMIXR Switch", "RECMIXR" }, + { "OUT MIXR", "DACR Switch", "DAC R1" }, + { "OUT MIXR", "DACL Switch", "DAC L1" }, + + { "SPO MIX", "DACR Switch", "DAC R1" }, + { "SPO MIX", "DACL Switch", "DAC L1" }, + { "SPO MIX", "SPKVOL Switch", "SPKVOL" }, + { "SPO MIX", "BST1 Switch", "BST1" }, + + { "SPKVOL", "Switch", "SPK MIX" }, + { "LOUTVOL L", "Switch", "OUT MIXL" }, + { "LOUTVOL R", "Switch", "OUT MIXR" }, + + { "LOUTVOL", NULL, "LOUTVOL L" }, + { "LOUTVOL", NULL, "LOUTVOL R" }, + + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + + { "LOUT MIX", "DAC Switch", "DAC 1" }, + { "LOUT MIX", "OUTMIX Switch", "LOUTVOL" }, + + { "LOUT amp", NULL, "LOUT MIX" }, + { "LOUT amp", NULL, "VREF HP" }, + { "LOUTL", NULL, "LOUT amp" }, + { "LOUTR", NULL, "LOUT amp" }, + + { "SPK amp", NULL, "SPO MIX" }, + { "SPO", NULL, "SPK amp" }, +}; + +static int rt5660_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 rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5660->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5660->sysclk, rt5660->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5660->lrck[dai->id], dai->id); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return frame_size; + } + + if (frame_size > 32) + bclk_ms = 1; + else + bclk_ms = 0; + + rt5660->bclk[dai->id] = rt5660->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5660->bclk[dai->id], rt5660->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 |= RT5660_I2S_DL_20; + break; + case 24: + val_len |= RT5660_I2S_DL_24; + break; + case 8: + val_len |= RT5660_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5660_AIF1: + mask_clk = RT5660_I2S_BCLK_MS1_MASK | RT5660_I2S_PD1_MASK; + val_clk = bclk_ms << RT5660_I2S_BCLK_MS1_SFT | + pre_div << RT5660_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5660_I2S1_SDP, RT5660_I2S_DL_MASK, + val_len); + snd_soc_update_bits(codec, RT5660_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 rt5660_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5660->master[dai->id] = 1; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5660_I2S_MS_S; + rt5660->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 |= RT5660_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 |= RT5660_I2S_DF_LEFT; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5660_I2S_DF_PCM_A; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5660_I2S_DF_PCM_B; + break; + + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5660_AIF1: + snd_soc_update_bits(codec, RT5660_I2S1_SDP, + RT5660_I2S_MS_MASK | RT5660_I2S_BP_MASK | + RT5660_I2S_DF_MASK, reg_val); + break; + + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5660_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5660->sysclk && clk_id == rt5660->sysclk_src) + return 0; + + switch (clk_id) { + case RT5660_SCLK_S_MCLK: + reg_val |= RT5660_SCLK_SRC_MCLK; + break; + + case RT5660_SCLK_S_PLL1: + reg_val |= RT5660_SCLK_SRC_PLL1; + break; + + case RT5660_SCLK_S_RCCLK: + reg_val |= RT5660_SCLK_SRC_RCCLK; + break; + + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5660_GLB_CLK, RT5660_SCLK_SRC_MASK, + reg_val); + + rt5660->sysclk = freq; + rt5660->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5660_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 rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5660->pll_src && freq_in == rt5660->pll_in && + freq_out == rt5660->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5660->pll_in = 0; + rt5660->pll_out = 0; + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_SCLK_SRC_MASK, RT5660_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5660_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_PLL1_SRC_MASK, RT5660_PLL1_SRC_MCLK); + break; + + case RT5660_PLL1_S_BCLK: + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_PLL1_SRC_MASK, RT5660_PLL1_SRC_BCLK1); + break; + + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5660_PLL_CTRL1, + pll_code.n_code << RT5660_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5660_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5660_PLL_M_SFT | + pll_code.m_bp << RT5660_PLL_M_BP_SFT); + + rt5660->pll_in = freq_in; + rt5660->pll_out = freq_out; + rt5660->pll_src = source; + + return 0; +} + +static int rt5660_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5660_priv *rt5660 = 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, RT5660_GEN_CTRL1, + RT5660_DIG_GATE_CTRL, RT5660_DIG_GATE_CTRL); + + if (IS_ERR(rt5660->mclk)) + break; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { + clk_disable_unprepare(rt5660->mclk); + } else { + ret = clk_prepare_enable(rt5660->mclk); + if (ret) + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, RT5660_PWR_ANLG1, + RT5660_PWR_VREF1 | RT5660_PWR_MB | + RT5660_PWR_BG | RT5660_PWR_VREF2, + RT5660_PWR_VREF1 | RT5660_PWR_MB | + RT5660_PWR_BG | RT5660_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5660_PWR_ANLG1, + RT5660_PWR_FV1 | RT5660_PWR_FV2, + RT5660_PWR_FV1 | RT5660_PWR_FV2); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5660_GEN_CTRL1, + RT5660_DIG_GATE_CTRL, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5660_probe(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + rt5660->codec = codec; + + return 0; +} + +static int rt5660_remove(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5660_RESET, 0); +} + +#ifdef CONFIG_PM +static int rt5660_suspend(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5660->regmap, true); + regcache_mark_dirty(rt5660->regmap); + + return 0; +} + +static int rt5660_resume(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + if (rt5660->pdata.poweroff_codec_in_suspend) + usleep_range(350000, 400000); + + regcache_cache_only(rt5660->regmap, false); + regcache_sync(rt5660->regmap); + + return 0; +} +#else +#define rt5660_suspend NULL +#define rt5660_resume NULL +#endif + +#define RT5660_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5660_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5660_aif_dai_ops = { + .hw_params = rt5660_hw_params, + .set_fmt = rt5660_set_dai_fmt, + .set_sysclk = rt5660_set_dai_sysclk, + .set_pll = rt5660_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5660_dai[] = { + { + .name = "rt5660-aif1", + .id = RT5660_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5660_STEREO_RATES, + .formats = RT5660_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5660_STEREO_RATES, + .formats = RT5660_FORMATS, + }, + .ops = &rt5660_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5660 = { + .probe = rt5660_probe, + .remove = rt5660_remove, + .suspend = rt5660_suspend, + .resume = rt5660_resume, + .set_bias_level = rt5660_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt5660_snd_controls, + .num_controls = ARRAY_SIZE(rt5660_snd_controls), + .dapm_widgets = rt5660_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5660_dapm_widgets), + .dapm_routes = rt5660_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5660_dapm_routes), + }, +}; + +static const struct regmap_config rt5660_regmap = { + .reg_bits = 8, + .val_bits = 16, + .use_single_rw = true, + + .max_register = RT5660_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5660_ranges) * + RT5660_PR_SPACING), + .volatile_reg = rt5660_volatile_register, + .readable_reg = rt5660_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5660_reg, + .num_reg_defaults = ARRAY_SIZE(rt5660_reg), + .ranges = rt5660_ranges, + .num_ranges = ARRAY_SIZE(rt5660_ranges), +}; + +static const struct i2c_device_id rt5660_i2c_id[] = { + { "rt5660", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5660_i2c_id); + +static const struct of_device_id rt5660_of_match[] = { + { .compatible = "realtek,rt5660", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5660_of_match); + +static const struct acpi_device_id rt5660_acpi_match[] = { + { "10EC5660", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); + +static int rt5660_parse_dt(struct rt5660_priv *rt5660, struct device *dev) +{ + rt5660->pdata.in1_diff = device_property_read_bool(dev, + "realtek,in1-differential"); + rt5660->pdata.in3_diff = device_property_read_bool(dev, + "realtek,in3-differential"); + rt5660->pdata.poweroff_codec_in_suspend = device_property_read_bool(dev, + "realtek,poweroff-in-suspend"); + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5660->pdata.dmic1_data_pin); + + return 0; +} + +static int rt5660_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5660_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5660_priv *rt5660; + int ret; + unsigned int val; + + rt5660 = devm_kzalloc(&i2c->dev, sizeof(struct rt5660_priv), + GFP_KERNEL); + + if (rt5660 == NULL) + return -ENOMEM; + + /* Check if MCLK provided */ + rt5660->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (PTR_ERR(rt5660->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + i2c_set_clientdata(i2c, rt5660); + + if (pdata) + rt5660->pdata = *pdata; + else if (i2c->dev.of_node) + rt5660_parse_dt(rt5660, &i2c->dev); + + rt5660->regmap = devm_regmap_init_i2c(i2c, &rt5660_regmap); + if (IS_ERR(rt5660->regmap)) { + ret = PTR_ERR(rt5660->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5660->regmap, RT5660_VENDOR_ID2, &val); + if (val != RT5660_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5660\n", val); + return -ENODEV; + } + + regmap_write(rt5660->regmap, RT5660_RESET, 0); + + ret = regmap_register_patch(rt5660->regmap, rt5660_patch, + ARRAY_SIZE(rt5660_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5660->pdata.dmic1_data_pin) { + regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, + RT5660_GP1_PIN_MASK, RT5660_GP1_PIN_DMIC1_SCL); + + if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_GPIO2) + regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, + RT5660_SEL_DMIC_DATA_MASK, + RT5660_SEL_DMIC_DATA_GPIO2); + else if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_IN1P) + regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, + RT5660_SEL_DMIC_DATA_MASK, + RT5660_SEL_DMIC_DATA_IN1P); + } + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5660, + rt5660_dai, ARRAY_SIZE(rt5660_dai)); +} + +static int rt5660_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5660_i2c_driver = { + .driver = { + .name = "rt5660", + .acpi_match_table = ACPI_PTR(rt5660_acpi_match), + .of_match_table = of_match_ptr(rt5660_of_match), + }, + .probe = rt5660_i2c_probe, + .remove = rt5660_i2c_remove, + .id_table = rt5660_i2c_id, +}; +module_i2c_driver(rt5660_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5660 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5660.h b/sound/soc/codecs/rt5660.h new file mode 100644 index 0000000..6cdb926 --- /dev/null +++ b/sound/soc/codecs/rt5660.h @@ -0,0 +1,847 @@ +/* + * rt5660.h -- RT5660 ALSA SoC audio driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(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 _RT5660_H +#define _RT5660_H + +#include <linux/clk.h> +#include <sound/rt5660.h> + +/* Info */ +#define RT5660_RESET 0x00 +#define RT5660_VENDOR_ID 0xfd +#define RT5660_VENDOR_ID1 0xfe +#define RT5660_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5660_SPK_VOL 0x01 +#define RT5660_LOUT_VOL 0x02 +/* I/O - Input */ +#define RT5660_IN1_IN2 0x0d +#define RT5660_IN3_IN4 0x0e +/* I/O - ADC/DAC/DMIC */ +#define RT5660_DAC1_DIG_VOL 0x19 +#define RT5660_STO1_ADC_DIG_VOL 0x1c +#define RT5660_ADC_BST_VOL1 0x1e +/* Mixer - D-D */ +#define RT5660_STO1_ADC_MIXER 0x27 +#define RT5660_AD_DA_MIXER 0x29 +#define RT5660_STO_DAC_MIXER 0x2a +#define RT5660_DIG_INF1_DATA 0x2f +/* Mixer - ADC */ +#define RT5660_REC_L1_MIXER 0x3b +#define RT5660_REC_L2_MIXER 0x3c +#define RT5660_REC_R1_MIXER 0x3d +#define RT5660_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5660_LOUT_MIXER 0x45 +#define RT5660_SPK_MIXER 0x46 +#define RT5660_SPO_MIXER 0x48 +#define RT5660_SPO_CLSD_RATIO 0x4a +#define RT5660_OUT_L_GAIN1 0x4d +#define RT5660_OUT_L_GAIN2 0x4e +#define RT5660_OUT_L1_MIXER 0x4f +#define RT5660_OUT_R_GAIN1 0x50 +#define RT5660_OUT_R_GAIN2 0x51 +#define RT5660_OUT_R1_MIXER 0x52 +/* Power */ +#define RT5660_PWR_DIG1 0x61 +#define RT5660_PWR_DIG2 0x62 +#define RT5660_PWR_ANLG1 0x63 +#define RT5660_PWR_ANLG2 0x64 +#define RT5660_PWR_MIXER 0x65 +#define RT5660_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5660_PRIV_INDEX 0x6a +#define RT5660_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5660_I2S1_SDP 0x70 +#define RT5660_ADDA_CLK1 0x73 +#define RT5660_ADDA_CLK2 0x74 +#define RT5660_DMIC_CTRL1 0x75 +/* Function - Analog */ +#define RT5660_GLB_CLK 0x80 +#define RT5660_PLL_CTRL1 0x81 +#define RT5660_PLL_CTRL2 0x82 +#define RT5660_CLSD_AMP_OC_CTRL 0x8c +#define RT5660_CLSD_AMP_CTRL 0x8d +#define RT5660_LOUT_AMP_CTRL 0x8e +#define RT5660_SPK_AMP_SPKVDD 0x92 +#define RT5660_MICBIAS 0x93 +#define RT5660_CLSD_OUT_CTRL1 0xa1 +#define RT5660_CLSD_OUT_CTRL2 0xa2 +#define RT5660_DIPOLE_MIC_CTRL1 0xa3 +#define RT5660_DIPOLE_MIC_CTRL2 0xa4 +#define RT5660_DIPOLE_MIC_CTRL3 0xa5 +#define RT5660_DIPOLE_MIC_CTRL4 0xa6 +#define RT5660_DIPOLE_MIC_CTRL5 0xa7 +#define RT5660_DIPOLE_MIC_CTRL6 0xa8 +#define RT5660_DIPOLE_MIC_CTRL7 0xa9 +#define RT5660_DIPOLE_MIC_CTRL8 0xaa +#define RT5660_DIPOLE_MIC_CTRL9 0xab +#define RT5660_DIPOLE_MIC_CTRL10 0xac +#define RT5660_DIPOLE_MIC_CTRL11 0xad +#define RT5660_DIPOLE_MIC_CTRL12 0xae +/* Function - Digital */ +#define RT5660_EQ_CTRL1 0xb0 +#define RT5660_EQ_CTRL2 0xb1 +#define RT5660_DRC_AGC_CTRL1 0xb3 +#define RT5660_DRC_AGC_CTRL2 0xb4 +#define RT5660_DRC_AGC_CTRL3 0xb5 +#define RT5660_DRC_AGC_CTRL4 0xb6 +#define RT5660_DRC_AGC_CTRL5 0xb7 +#define RT5660_JD_CTRL 0xbb +#define RT5660_IRQ_CTRL1 0xbd +#define RT5660_IRQ_CTRL2 0xbe +#define RT5660_INT_IRQ_ST 0xbf +#define RT5660_GPIO_CTRL1 0xc0 +#define RT5660_GPIO_CTRL2 0xc2 +#define RT5660_WIND_FILTER_CTRL1 0xd3 +#define RT5660_SV_ZCD1 0xd9 +#define RT5660_SV_ZCD2 0xda +#define RT5660_DRC1_LM_CTRL1 0xe0 +#define RT5660_DRC1_LM_CTRL2 0xe1 +#define RT5660_DRC2_LM_CTRL1 0xe2 +#define RT5660_DRC2_LM_CTRL2 0xe3 +#define RT5660_MULTI_DRC_CTRL 0xe4 +#define RT5660_DRC2_CTRL1 0xe5 +#define RT5660_DRC2_CTRL2 0xe6 +#define RT5660_DRC2_CTRL3 0xe7 +#define RT5660_DRC2_CTRL4 0xe8 +#define RT5660_DRC2_CTRL5 0xe9 +#define RT5660_ALC_PGA_CTRL1 0xea +#define RT5660_ALC_PGA_CTRL2 0xeb +#define RT5660_ALC_PGA_CTRL3 0xec +#define RT5660_ALC_PGA_CTRL4 0xed +#define RT5660_ALC_PGA_CTRL5 0xee +#define RT5660_ALC_PGA_CTRL6 0xef +#define RT5660_ALC_PGA_CTRL7 0xf0 + +/* General Control */ +#define RT5660_GEN_CTRL1 0xfa +#define RT5660_GEN_CTRL2 0xfb +#define RT5660_GEN_CTRL3 0xfc + +/* Index of Codec Private Register definition */ +#define RT5660_CHOP_DAC_ADC 0x3d + +/* Global Definition */ +#define RT5660_L_MUTE (0x1 << 15) +#define RT5660_L_MUTE_SFT 15 +#define RT5660_VOL_L_MUTE (0x1 << 14) +#define RT5660_VOL_L_SFT 14 +#define RT5660_R_MUTE (0x1 << 7) +#define RT5660_R_MUTE_SFT 7 +#define RT5660_VOL_R_MUTE (0x1 << 6) +#define RT5660_VOL_R_SFT 6 +#define RT5660_L_VOL_MASK (0x3f << 8) +#define RT5660_L_VOL_SFT 8 +#define RT5660_R_VOL_MASK (0x3f) +#define RT5660_R_VOL_SFT 0 + +/* IN1 and IN2 Control (0x0d) */ +#define RT5660_IN_DF1 (0x1 << 15) +#define RT5660_IN_SFT1 15 +#define RT5660_BST_MASK1 (0x7f << 8) +#define RT5660_BST_SFT1 8 +#define RT5660_IN_DF2 (0x1 << 7) +#define RT5660_IN_SFT2 7 +#define RT5660_BST_MASK2 (0x7f << 0) +#define RT5660_BST_SFT2 0 + +/* IN3 and IN4 Control (0x0e) */ +#define RT5660_IN_DF3 (0x1 << 15) +#define RT5660_IN_SFT3 15 +#define RT5660_BST_MASK3 (0x7f << 8) +#define RT5660_BST_SFT3 8 +#define RT5660_IN_DF4 (0x1 << 7) +#define RT5660_IN_SFT4 7 +#define RT5660_BST_MASK4 (0x7f << 0) +#define RT5660_BST_SFT4 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5660_DAC_L1_VOL_MASK (0x7f << 9) +#define RT5660_DAC_L1_VOL_SFT 9 +#define RT5660_DAC_R1_VOL_MASK (0x7f << 1) +#define RT5660_DAC_R1_VOL_SFT 1 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5660_ADC_L_VOL_MASK (0x3f << 9) +#define RT5660_ADC_L_VOL_SFT 9 +#define RT5660_ADC_R_VOL_MASK (0x3f << 1) +#define RT5660_ADC_R_VOL_SFT 1 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5660_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5660_STO1_ADC_L_BST_SFT 14 +#define RT5660_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5660_STO1_ADC_R_BST_SFT 12 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5660_M_ADC_L1 (0x1 << 14) +#define RT5660_M_ADC_L1_SFT 14 +#define RT5660_M_ADC_L2 (0x1 << 13) +#define RT5660_M_ADC_L2_SFT 13 +#define RT5660_M_ADC_R1 (0x1 << 6) +#define RT5660_M_ADC_R1_SFT 6 +#define RT5660_M_ADC_R2 (0x1 << 5) +#define RT5660_M_ADC_R2_SFT 5 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5660_M_ADCMIX_L (0x1 << 15) +#define RT5660_M_ADCMIX_L_SFT 15 +#define RT5660_M_DAC1_L (0x1 << 14) +#define RT5660_M_DAC1_L_SFT 14 +#define RT5660_M_ADCMIX_R (0x1 << 7) +#define RT5660_M_ADCMIX_R_SFT 7 +#define RT5660_M_DAC1_R (0x1 << 6) +#define RT5660_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5660_M_DAC_L1 (0x1 << 14) +#define RT5660_M_DAC_L1_SFT 14 +#define RT5660_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5660_DAC_L1_STO_L_VOL_SFT 13 +#define RT5660_M_DAC_R1_STO_L (0x1 << 9) +#define RT5660_M_DAC_R1_STO_L_SFT 9 +#define RT5660_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5660_DAC_R1_STO_L_VOL_SFT 8 +#define RT5660_M_DAC_R1 (0x1 << 6) +#define RT5660_M_DAC_R1_SFT 6 +#define RT5660_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5660_DAC_R1_STO_R_VOL_SFT 5 +#define RT5660_M_DAC_L1_STO_R (0x1 << 1) +#define RT5660_M_DAC_L1_STO_R_SFT 1 +#define RT5660_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5660_DAC_L1_STO_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5660_IF1_DAC_IN_SEL (0x3 << 14) +#define RT5660_IF1_DAC_IN_SFT 14 +#define RT5660_IF1_ADC_IN_SEL (0x3 << 12) +#define RT5660_IF1_ADC_IN_SFT 12 + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5660_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5660_G_BST3_RM_L_SFT 4 +#define RT5660_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5660_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5660_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5660_G_BST1_RM_L_SFT 13 +#define RT5660_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5660_G_OM_L_RM_L_SFT 10 +#define RT5660_M_BST3_RM_L (0x1 << 3) +#define RT5660_M_BST3_RM_L_SFT 3 +#define RT5660_M_BST2_RM_L (0x1 << 2) +#define RT5660_M_BST2_RM_L_SFT 2 +#define RT5660_M_BST1_RM_L (0x1 << 1) +#define RT5660_M_BST1_RM_L_SFT 1 +#define RT5660_M_OM_L_RM_L (0x1) +#define RT5660_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5660_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5660_G_BST3_RM_R_SFT 4 +#define RT5660_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5660_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5660_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5660_G_BST1_RM_R_SFT 13 +#define RT5660_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5660_G_OM_R_RM_R_SFT 10 +#define RT5660_M_BST3_RM_R (0x1 << 3) +#define RT5660_M_BST3_RM_R_SFT 3 +#define RT5660_M_BST2_RM_R (0x1 << 2) +#define RT5660_M_BST2_RM_R_SFT 2 +#define RT5660_M_BST1_RM_R (0x1 << 1) +#define RT5660_M_BST1_RM_R_SFT 1 +#define RT5660_M_OM_R_RM_R (0x1) +#define RT5660_M_OM_R_RM_R_SFT 0 + +/* LOUTMIX Control (0x45) */ +#define RT5660_M_DAC1_LM (0x1 << 14) +#define RT5660_M_DAC1_LM_SFT 14 +#define RT5660_M_LOVOL_M (0x1 << 13) +#define RT5660_M_LOVOL_LM_SFT 13 + +/* SPK Mixer Control (0x46) */ +#define RT5660_G_BST3_SM_MASK (0x3 << 14) +#define RT5660_G_BST3_SM_SFT 14 +#define RT5660_G_BST1_SM_MASK (0x3 << 12) +#define RT5660_G_BST1_SM_SFT 12 +#define RT5660_G_DACl_SM_MASK (0x3 << 10) +#define RT5660_G_DACl_SM_SFT 10 +#define RT5660_G_DACR_SM_MASK (0x3 << 8) +#define RT5660_G_DACR_SM_SFT 8 +#define RT5660_G_OM_L_SM_MASK (0x3 << 6) +#define RT5660_G_OM_L_SM_SFT 6 +#define RT5660_M_DACR_SM (0x1 << 5) +#define RT5660_M_DACR_SM_SFT 5 +#define RT5660_M_BST1_SM (0x1 << 4) +#define RT5660_M_BST1_SM_SFT 4 +#define RT5660_M_BST3_SM (0x1 << 3) +#define RT5660_M_BST3_SM_SFT 3 +#define RT5660_M_DACL_SM (0x1 << 2) +#define RT5660_M_DACL_SM_SFT 2 +#define RT5660_M_OM_L_SM (0x1 << 1) +#define RT5660_M_OM_L_SM_SFT 1 + +/* SPOMIX Control (0x48) */ +#define RT5660_M_DAC_R_SPM (0x1 << 14) +#define RT5660_M_DAC_R_SPM_SFT 14 +#define RT5660_M_DAC_L_SPM (0x1 << 13) +#define RT5660_M_DAC_L_SPM_SFT 13 +#define RT5660_M_SV_SPM (0x1 << 12) +#define RT5660_M_SV_SPM_SFT 12 +#define RT5660_M_BST1_SPM (0x1 << 11) +#define RT5660_M_BST1_SPM_SFT 11 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5660_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5660_G_BST3_OM_L_SFT 13 +#define RT5660_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5660_G_BST2_OM_L_SFT 10 +#define RT5660_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5660_G_BST1_OM_L_SFT 7 +#define RT5660_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5660_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5660_G_DAC_R1_OM_L_MASK (0x7 << 10) +#define RT5660_G_DAC_R1_OM_L_SFT 10 +#define RT5660_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5660_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5660_M_BST3_OM_L (0x1 << 5) +#define RT5660_M_BST3_OM_L_SFT 5 +#define RT5660_M_BST2_OM_L (0x1 << 4) +#define RT5660_M_BST2_OM_L_SFT 4 +#define RT5660_M_BST1_OM_L (0x1 << 3) +#define RT5660_M_BST1_OM_L_SFT 3 +#define RT5660_M_RM_L_OM_L (0x1 << 2) +#define RT5660_M_RM_L_OM_L_SFT 2 +#define RT5660_M_DAC_R_OM_L (0x1 << 1) +#define RT5660_M_DAC_R_OM_L_SFT 1 +#define RT5660_M_DAC_L_OM_L (0x1) +#define RT5660_M_DAC_L_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5660_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5660_G_BST2_OM_R_SFT 10 +#define RT5660_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5660_G_BST1_OM_R_SFT 7 +#define RT5660_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5660_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5660_G_DAC_L_OM_R_MASK (0x7 << 10) +#define RT5660_G_DAC_L_OM_R_SFT 10 +#define RT5660_G_DAC_R_OM_R_MASK (0x7 << 7) +#define RT5660_G_DAC_R_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5660_M_BST2_OM_R (0x1 << 4) +#define RT5660_M_BST2_OM_R_SFT 4 +#define RT5660_M_BST1_OM_R (0x1 << 3) +#define RT5660_M_BST1_OM_R_SFT 3 +#define RT5660_M_RM_R_OM_R (0x1 << 2) +#define RT5660_M_RM_R_OM_R_SFT 2 +#define RT5660_M_DAC_L_OM_R (0x1 << 1) +#define RT5660_M_DAC_L_OM_R_SFT 1 +#define RT5660_M_DAC_R_OM_R (0x1) +#define RT5660_M_DAC_R_OM_R_SFT 0 + +/* Power Management for Digital 1 (0x61) */ +#define RT5660_PWR_I2S1 (0x1 << 15) +#define RT5660_PWR_I2S1_BIT 15 +#define RT5660_PWR_DAC_L1 (0x1 << 12) +#define RT5660_PWR_DAC_L1_BIT 12 +#define RT5660_PWR_DAC_R1 (0x1 << 11) +#define RT5660_PWR_DAC_R1_BIT 11 +#define RT5660_PWR_ADC_L (0x1 << 2) +#define RT5660_PWR_ADC_L_BIT 2 +#define RT5660_PWR_ADC_R (0x1 << 1) +#define RT5660_PWR_ADC_R_BIT 1 +#define RT5660_PWR_CLS_D (0x1) +#define RT5660_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5660_PWR_ADC_S1F (0x1 << 15) +#define RT5660_PWR_ADC_S1F_BIT 15 +#define RT5660_PWR_DAC_S1F (0x1 << 11) +#define RT5660_PWR_DAC_S1F_BIT 11 + +/* Power Management for Analog 1 (0x63) */ +#define RT5660_PWR_VREF1 (0x1 << 15) +#define RT5660_PWR_VREF1_BIT 15 +#define RT5660_PWR_FV1 (0x1 << 14) +#define RT5660_PWR_FV1_BIT 14 +#define RT5660_PWR_MB (0x1 << 13) +#define RT5660_PWR_MB_BIT 13 +#define RT5660_PWR_BG (0x1 << 11) +#define RT5660_PWR_BG_BIT 11 +#define RT5660_PWR_HP_L (0x1 << 7) +#define RT5660_PWR_HP_L_BIT 7 +#define RT5660_PWR_HP_R (0x1 << 6) +#define RT5660_PWR_HP_R_BIT 6 +#define RT5660_PWR_HA (0x1 << 5) +#define RT5660_PWR_HA_BIT 5 +#define RT5660_PWR_VREF2 (0x1 << 4) +#define RT5660_PWR_VREF2_BIT 4 +#define RT5660_PWR_FV2 (0x1 << 3) +#define RT5660_PWR_FV2_BIT 3 +#define RT5660_PWR_LDO2 (0x1 << 2) +#define RT5660_PWR_LDO2_BIT 2 + +/* Power Management for Analog 2 (0x64) */ +#define RT5660_PWR_BST1 (0x1 << 15) +#define RT5660_PWR_BST1_BIT 15 +#define RT5660_PWR_BST2 (0x1 << 14) +#define RT5660_PWR_BST2_BIT 14 +#define RT5660_PWR_BST3 (0x1 << 13) +#define RT5660_PWR_BST3_BIT 13 +#define RT5660_PWR_MB1 (0x1 << 11) +#define RT5660_PWR_MB1_BIT 11 +#define RT5660_PWR_MB2 (0x1 << 10) +#define RT5660_PWR_MB2_BIT 10 +#define RT5660_PWR_PLL (0x1 << 9) +#define RT5660_PWR_PLL_BIT 9 + +/* Power Management for Mixer (0x65) */ +#define RT5660_PWR_OM_L (0x1 << 15) +#define RT5660_PWR_OM_L_BIT 15 +#define RT5660_PWR_OM_R (0x1 << 14) +#define RT5660_PWR_OM_R_BIT 14 +#define RT5660_PWR_SM (0x1 << 13) +#define RT5660_PWR_SM_BIT 13 +#define RT5660_PWR_RM_L (0x1 << 11) +#define RT5660_PWR_RM_L_BIT 11 +#define RT5660_PWR_RM_R (0x1 << 10) +#define RT5660_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5660_PWR_SV (0x1 << 15) +#define RT5660_PWR_SV_BIT 15 +#define RT5660_PWR_LV_L (0x1 << 11) +#define RT5660_PWR_LV_L_BIT 11 +#define RT5660_PWR_LV_R (0x1 << 10) +#define RT5660_PWR_LV_R_BIT 10 + +/* I2S1 Audio Serial Data Port Control (0x70) */ +#define RT5660_I2S_MS_MASK (0x1 << 15) +#define RT5660_I2S_MS_SFT 15 +#define RT5660_I2S_MS_M (0x0 << 15) +#define RT5660_I2S_MS_S (0x1 << 15) +#define RT5660_I2S_O_CP_MASK (0x3 << 10) +#define RT5660_I2S_O_CP_SFT 10 +#define RT5660_I2S_O_CP_OFF (0x0 << 10) +#define RT5660_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5660_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5660_I2S_I_CP_MASK (0x3 << 8) +#define RT5660_I2S_I_CP_SFT 8 +#define RT5660_I2S_I_CP_OFF (0x0 << 8) +#define RT5660_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5660_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5660_I2S_BP_MASK (0x1 << 7) +#define RT5660_I2S_BP_SFT 7 +#define RT5660_I2S_BP_NOR (0x0 << 7) +#define RT5660_I2S_BP_INV (0x1 << 7) +#define RT5660_I2S_DL_MASK (0x3 << 2) +#define RT5660_I2S_DL_SFT 2 +#define RT5660_I2S_DL_16 (0x0 << 2) +#define RT5660_I2S_DL_20 (0x1 << 2) +#define RT5660_I2S_DL_24 (0x2 << 2) +#define RT5660_I2S_DL_8 (0x3 << 2) +#define RT5660_I2S_DF_MASK (0x3) +#define RT5660_I2S_DF_SFT 0 +#define RT5660_I2S_DF_I2S (0x0) +#define RT5660_I2S_DF_LEFT (0x1) +#define RT5660_I2S_DF_PCM_A (0x2) +#define RT5660_I2S_DF_PCM_B (0x3) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5660_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5660_I2S_BCLK_MS1_SFT 15 +#define RT5660_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5660_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5660_I2S_PD1_MASK (0x7 << 12) +#define RT5660_I2S_PD1_SFT 12 +#define RT5660_I2S_PD1_1 (0x0 << 12) +#define RT5660_I2S_PD1_2 (0x1 << 12) +#define RT5660_I2S_PD1_3 (0x2 << 12) +#define RT5660_I2S_PD1_4 (0x3 << 12) +#define RT5660_I2S_PD1_6 (0x4 << 12) +#define RT5660_I2S_PD1_8 (0x5 << 12) +#define RT5660_I2S_PD1_12 (0x6 << 12) +#define RT5660_I2S_PD1_16 (0x7 << 12) +#define RT5660_DAC_OSR_MASK (0x3 << 2) +#define RT5660_DAC_OSR_SFT 2 +#define RT5660_DAC_OSR_128 (0x0 << 2) +#define RT5660_DAC_OSR_64 (0x1 << 2) +#define RT5660_DAC_OSR_32 (0x2 << 2) +#define RT5660_DAC_OSR_16 (0x3 << 2) +#define RT5660_ADC_OSR_MASK (0x3) +#define RT5660_ADC_OSR_SFT 0 +#define RT5660_ADC_OSR_128 (0x0) +#define RT5660_ADC_OSR_64 (0x1) +#define RT5660_ADC_OSR_32 (0x2) +#define RT5660_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5660_RESET_ADF (0x1 << 13) +#define RT5660_RESET_ADF_SFT 13 +#define RT5660_RESET_DAF (0x1 << 12) +#define RT5660_RESET_DAF_SFT 12 +#define RT5660_DAHPF_EN (0x1 << 11) +#define RT5660_DAHPF_EN_SFT 11 +#define RT5660_ADHPF_EN (0x1 << 10) +#define RT5660_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5660_DMIC_1_EN_MASK (0x1 << 15) +#define RT5660_DMIC_1_EN_SFT 15 +#define RT5660_DMIC_1_DIS (0x0 << 15) +#define RT5660_DMIC_1_EN (0x1 << 15) +#define RT5660_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5660_DMIC_1L_LH_SFT 13 +#define RT5660_DMIC_1L_LH_RISING (0x0 << 13) +#define RT5660_DMIC_1L_LH_FALLING (0x1 << 13) +#define RT5660_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5660_DMIC_1R_LH_SFT 12 +#define RT5660_DMIC_1R_LH_RISING (0x0 << 12) +#define RT5660_DMIC_1R_LH_FALLING (0x1 << 12) +#define RT5660_SEL_DMIC_DATA_MASK (0x1 << 11) +#define RT5660_SEL_DMIC_DATA_SFT 11 +#define RT5660_SEL_DMIC_DATA_GPIO2 (0x0 << 11) +#define RT5660_SEL_DMIC_DATA_IN1P (0x1 << 11) +#define RT5660_DMIC_CLK_MASK (0x7 << 5) +#define RT5660_DMIC_CLK_SFT 5 + +/* Global Clock Control (0x80) */ +#define RT5660_SCLK_SRC_MASK (0x3 << 14) +#define RT5660_SCLK_SRC_SFT 14 +#define RT5660_SCLK_SRC_MCLK (0x0 << 14) +#define RT5660_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5660_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5660_PLL1_SRC_MASK (0x3 << 12) +#define RT5660_PLL1_SRC_SFT 12 +#define RT5660_PLL1_SRC_MCLK (0x0 << 12) +#define RT5660_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5660_PLL1_SRC_RCCLK (0x2 << 12) +#define RT5660_PLL1_PD_MASK (0x1 << 3) +#define RT5660_PLL1_PD_SFT 3 +#define RT5660_PLL1_PD_1 (0x0 << 3) +#define RT5660_PLL1_PD_2 (0x1 << 3) + +#define RT5660_PLL_INP_MAX 40000000 +#define RT5660_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5660_PLL_N_MAX 0x1ff +#define RT5660_PLL_N_MASK (RT5660_PLL_N_MAX << 7) +#define RT5660_PLL_N_SFT 7 +#define RT5660_PLL_K_MAX 0x1f +#define RT5660_PLL_K_MASK (RT5660_PLL_K_MAX) +#define RT5660_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5660_PLL_M_MAX 0xf +#define RT5660_PLL_M_MASK (RT5660_PLL_M_MAX << 12) +#define RT5660_PLL_M_SFT 12 +#define RT5660_PLL_M_BP (0x1 << 11) +#define RT5660_PLL_M_BP_SFT 11 + +/* Class D Over Current Control (0x8c) */ +#define RT5660_CLSD_OC_MASK (0x1 << 9) +#define RT5660_CLSD_OC_SFT 9 +#define RT5660_CLSD_OC_PU (0x0 << 9) +#define RT5660_CLSD_OC_PD (0x1 << 9) +#define RT5660_AUTO_PD_MASK (0x1 << 8) +#define RT5660_AUTO_PD_SFT 8 +#define RT5660_AUTO_PD_DIS (0x0 << 8) +#define RT5660_AUTO_PD_EN (0x1 << 8) +#define RT5660_CLSD_OC_TH_MASK (0x3f) +#define RT5660_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5660_CLSD_RATIO_MASK (0xf << 12) +#define RT5660_CLSD_RATIO_SFT 12 + +/* Lout Amp Control 1 (0x8e) */ +#define RT5660_LOUT_CO_MASK (0x1 << 4) +#define RT5660_LOUT_CO_SFT 4 +#define RT5660_LOUT_CO_DIS (0x0 << 4) +#define RT5660_LOUT_CO_EN (0x1 << 4) +#define RT5660_LOUT_CB_MASK (0x1) +#define RT5660_LOUT_CB_SFT 0 +#define RT5660_LOUT_CB_PD (0x0) +#define RT5660_LOUT_CB_PU (0x1) + +/* SPKVDD detection control (0x92) */ +#define RT5660_SPKVDD_DET_MASK (0x1 << 15) +#define RT5660_SPKVDD_DET_SFT 15 +#define RT5660_SPKVDD_DET_DIS (0x0 << 15) +#define RT5660_SPKVDD_DET_EN (0x1 << 15) +#define RT5660_SPK_AG_MASK (0x1 << 14) +#define RT5660_SPK_AG_SFT 14 +#define RT5660_SPK_AG_DIS (0x0 << 14) +#define RT5660_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5660_MIC1_BS_MASK (0x1 << 15) +#define RT5660_MIC1_BS_SFT 15 +#define RT5660_MIC1_BS_9AV (0x0 << 15) +#define RT5660_MIC1_BS_75AV (0x1 << 15) +#define RT5660_MIC2_BS_MASK (0x1 << 14) +#define RT5660_MIC2_BS_SFT 14 +#define RT5660_MIC2_BS_9AV (0x0 << 14) +#define RT5660_MIC2_BS_75AV (0x1 << 14) +#define RT5660_MIC1_OVCD_MASK (0x1 << 11) +#define RT5660_MIC1_OVCD_SFT 11 +#define RT5660_MIC1_OVCD_DIS (0x0 << 11) +#define RT5660_MIC1_OVCD_EN (0x1 << 11) +#define RT5660_MIC1_OVTH_MASK (0x3 << 9) +#define RT5660_MIC1_OVTH_SFT 9 +#define RT5660_MIC1_OVTH_600UA (0x0 << 9) +#define RT5660_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5660_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5660_MIC2_OVCD_MASK (0x1 << 8) +#define RT5660_MIC2_OVCD_SFT 8 +#define RT5660_MIC2_OVCD_DIS (0x0 << 8) +#define RT5660_MIC2_OVCD_EN (0x1 << 8) +#define RT5660_MIC2_OVTH_MASK (0x3 << 6) +#define RT5660_MIC2_OVTH_SFT 6 +#define RT5660_MIC2_OVTH_600UA (0x0 << 6) +#define RT5660_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5660_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5660_PWR_CLK25M_MASK (0x1 << 4) +#define RT5660_PWR_CLK25M_SFT 4 +#define RT5660_PWR_CLK25M_PD (0x0 << 4) +#define RT5660_PWR_CLK25M_PU (0x1 << 4) + +/* EQ Control 1 (0xb0) */ +#define RT5660_EQ_SRC_MASK (0x1 << 15) +#define RT5660_EQ_SRC_SFT 15 +#define RT5660_EQ_SRC_DAC (0x0 << 15) +#define RT5660_EQ_SRC_ADC (0x1 << 15) +#define RT5660_EQ_UPD (0x1 << 14) +#define RT5660_EQ_UPD_BIT 14 + +/* Jack Detect Control (0xbb) */ +#define RT5660_JD_MASK (0x3 << 14) +#define RT5660_JD_SFT 14 +#define RT5660_JD_DIS (0x0 << 14) +#define RT5660_JD_GPIO1 (0x1 << 14) +#define RT5660_JD_GPIO2 (0x2 << 14) +#define RT5660_JD_LOUT_MASK (0x1 << 11) +#define RT5660_JD_LOUT_SFT 11 +#define RT5660_JD_LOUT_DIS (0x0 << 11) +#define RT5660_JD_LOUT_EN (0x1 << 11) +#define RT5660_JD_LOUT_TRG_MASK (0x1 << 10) +#define RT5660_JD_LOUT_TRG_SFT 10 +#define RT5660_JD_LOUT_TRG_LO (0x0 << 10) +#define RT5660_JD_LOUT_TRG_HI (0x1 << 10) +#define RT5660_JD_SPO_MASK (0x1 << 9) +#define RT5660_JD_SPO_SFT 9 +#define RT5660_JD_SPO_DIS (0x0 << 9) +#define RT5660_JD_SPO_EN (0x1 << 9) +#define RT5660_JD_SPO_TRG_MASK (0x1 << 8) +#define RT5660_JD_SPO_TRG_SFT 8 +#define RT5660_JD_SPO_TRG_LO (0x0 << 8) +#define RT5660_JD_SPO_TRG_HI (0x1 << 8) + +/* IRQ Control 1 (0xbd) */ +#define RT5660_IRQ_JD_MASK (0x1 << 15) +#define RT5660_IRQ_JD_SFT 15 +#define RT5660_IRQ_JD_BP (0x0 << 15) +#define RT5660_IRQ_JD_NOR (0x1 << 15) +#define RT5660_IRQ_OT_MASK (0x1 << 14) +#define RT5660_IRQ_OT_SFT 14 +#define RT5660_IRQ_OT_BP (0x0 << 14) +#define RT5660_IRQ_OT_NOR (0x1 << 14) +#define RT5660_JD_STKY_MASK (0x1 << 13) +#define RT5660_JD_STKY_SFT 13 +#define RT5660_JD_STKY_DIS (0x0 << 13) +#define RT5660_JD_STKY_EN (0x1 << 13) +#define RT5660_OT_STKY_MASK (0x1 << 12) +#define RT5660_OT_STKY_SFT 12 +#define RT5660_OT_STKY_DIS (0x0 << 12) +#define RT5660_OT_STKY_EN (0x1 << 12) +#define RT5660_JD_P_MASK (0x1 << 11) +#define RT5660_JD_P_SFT 11 +#define RT5660_JD_P_NOR (0x0 << 11) +#define RT5660_JD_P_INV (0x1 << 11) +#define RT5660_OT_P_MASK (0x1 << 10) +#define RT5660_OT_P_SFT 10 +#define RT5660_OT_P_NOR (0x0 << 10) +#define RT5660_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5660_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5660_IRQ_MB1_OC_SFT 15 +#define RT5660_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5660_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5660_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5660_IRQ_MB2_OC_SFT 14 +#define RT5660_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5660_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5660_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5660_MB1_OC_STKY_SFT 11 +#define RT5660_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5660_MB1_OC_STKY_EN (0x1 << 11) +#define RT5660_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5660_MB2_OC_STKY_SFT 10 +#define RT5660_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5660_MB2_OC_STKY_EN (0x1 << 10) +#define RT5660_MB1_OC_P_MASK (0x1 << 7) +#define RT5660_MB1_OC_P_SFT 7 +#define RT5660_MB1_OC_P_NOR (0x0 << 7) +#define RT5660_MB1_OC_P_INV (0x1 << 7) +#define RT5660_MB2_OC_P_MASK (0x1 << 6) +#define RT5660_MB2_OC_P_SFT 6 +#define RT5660_MB2_OC_P_NOR (0x0 << 6) +#define RT5660_MB2_OC_P_INV (0x1 << 6) +#define RT5660_MB1_OC_CLR (0x1 << 3) +#define RT5660_MB1_OC_CLR_SFT 3 +#define RT5660_MB2_OC_CLR (0x1 << 2) +#define RT5660_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5660_GP2_PIN_MASK (0x1 << 14) +#define RT5660_GP2_PIN_SFT 14 +#define RT5660_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5660_GP2_PIN_DMIC1_SDA (0x1 << 14) +#define RT5660_GP1_PIN_MASK (0x3 << 12) +#define RT5660_GP1_PIN_SFT 12 +#define RT5660_GP1_PIN_GPIO1 (0x0 << 12) +#define RT5660_GP1_PIN_DMIC1_SCL (0x1 << 12) +#define RT5660_GP1_PIN_IRQ (0x2 << 12) +#define RT5660_GPIO_M_MASK (0x1 << 9) +#define RT5660_GPIO_M_SFT 9 +#define RT5660_GPIO_M_FLT (0x0 << 9) +#define RT5660_GPIO_M_PH (0x1 << 9) + +/* GPIO Control 3 (0xc2) */ +#define RT5660_GP2_PF_MASK (0x1 << 5) +#define RT5660_GP2_PF_SFT 5 +#define RT5660_GP2_PF_IN (0x0 << 5) +#define RT5660_GP2_PF_OUT (0x1 << 5) +#define RT5660_GP2_OUT_MASK (0x1 << 4) +#define RT5660_GP2_OUT_SFT 4 +#define RT5660_GP2_OUT_LO (0x0 << 4) +#define RT5660_GP2_OUT_HI (0x1 << 4) +#define RT5660_GP2_P_MASK (0x1 << 3) +#define RT5660_GP2_P_SFT 3 +#define RT5660_GP2_P_NOR (0x0 << 3) +#define RT5660_GP2_P_INV (0x1 << 3) +#define RT5660_GP1_PF_MASK (0x1 << 2) +#define RT5660_GP1_PF_SFT 2 +#define RT5660_GP1_PF_IN (0x0 << 2) +#define RT5660_GP1_PF_OUT (0x1 << 2) +#define RT5660_GP1_OUT_MASK (0x1 << 1) +#define RT5660_GP1_OUT_SFT 1 +#define RT5660_GP1_OUT_LO (0x0 << 1) +#define RT5660_GP1_OUT_HI (0x1 << 1) +#define RT5660_GP1_P_MASK (0x1) +#define RT5660_GP1_P_SFT 0 +#define RT5660_GP1_P_NOR (0x0) +#define RT5660_GP1_P_INV (0x1) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5660_SV_MASK (0x1 << 15) +#define RT5660_SV_SFT 15 +#define RT5660_SV_DIS (0x0 << 15) +#define RT5660_SV_EN (0x1 << 15) +#define RT5660_SPO_SV_MASK (0x1 << 14) +#define RT5660_SPO_SV_SFT 14 +#define RT5660_SPO_SV_DIS (0x0 << 14) +#define RT5660_SPO_SV_EN (0x1 << 14) +#define RT5660_OUT_SV_MASK (0x1 << 12) +#define RT5660_OUT_SV_SFT 12 +#define RT5660_OUT_SV_DIS (0x0 << 12) +#define RT5660_OUT_SV_EN (0x1 << 12) +#define RT5660_ZCD_DIG_MASK (0x1 << 11) +#define RT5660_ZCD_DIG_SFT 11 +#define RT5660_ZCD_DIG_DIS (0x0 << 11) +#define RT5660_ZCD_DIG_EN (0x1 << 11) +#define RT5660_ZCD_MASK (0x1 << 10) +#define RT5660_ZCD_SFT 10 +#define RT5660_ZCD_PD (0x0 << 10) +#define RT5660_ZCD_PU (0x1 << 10) +#define RT5660_SV_DLY_MASK (0xf) +#define RT5660_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5660_ZCD_SPO_MASK (0x1 << 15) +#define RT5660_ZCD_SPO_SFT 15 +#define RT5660_ZCD_SPO_DIS (0x0 << 15) +#define RT5660_ZCD_SPO_EN (0x1 << 15) +#define RT5660_ZCD_OMR_MASK (0x1 << 8) +#define RT5660_ZCD_OMR_SFT 8 +#define RT5660_ZCD_OMR_DIS (0x0 << 8) +#define RT5660_ZCD_OMR_EN (0x1 << 8) +#define RT5660_ZCD_OML_MASK (0x1 << 7) +#define RT5660_ZCD_OML_SFT 7 +#define RT5660_ZCD_OML_DIS (0x0 << 7) +#define RT5660_ZCD_OML_EN (0x1 << 7) +#define RT5660_ZCD_SPM_MASK (0x1 << 6) +#define RT5660_ZCD_SPM_SFT 6 +#define RT5660_ZCD_SPM_DIS (0x0 << 6) +#define RT5660_ZCD_SPM_EN (0x1 << 6) +#define RT5660_ZCD_RMR_MASK (0x1 << 5) +#define RT5660_ZCD_RMR_SFT 5 +#define RT5660_ZCD_RMR_DIS (0x0 << 5) +#define RT5660_ZCD_RMR_EN (0x1 << 5) +#define RT5660_ZCD_RML_MASK (0x1 << 4) +#define RT5660_ZCD_RML_SFT 4 +#define RT5660_ZCD_RML_DIS (0x0 << 4) +#define RT5660_ZCD_RML_EN (0x1 << 4) + +/* General Control 1 (0xfa) */ +#define RT5660_PWR_VREF_HP (0x1 << 11) +#define RT5660_PWR_VREF_HP_SFT 11 +#define RT5660_DIG_GATE_CTRL (0x1) +#define RT5660_DIG_GATE_CTRL_SFT 0 + +/* System Clock Source */ +#define RT5660_SCLK_S_MCLK 0 +#define RT5660_SCLK_S_PLL1 1 +#define RT5660_SCLK_S_RCCLK 2 + +/* PLL1 Source */ +#define RT5660_PLL1_S_MCLK 0 +#define RT5660_PLL1_S_BCLK 1 + +enum { + RT5660_AIF1, + RT5660_AIFS, +}; + +struct rt5660_priv { + struct snd_soc_codec *codec; + struct rt5660_platform_data pdata; + struct regmap *regmap; + struct clk *mclk; + + int sysclk; + int sysclk_src; + int lrck[RT5660_AIFS]; + int bclk[RT5660_AIFS]; + int master[RT5660_AIFS]; + + int pll_src; + int pll_in; + int pll_out; +}; + +#endif -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: Intel: remove status, it is shadowing status of a higher scope
by Colin King 25 Sep '16

25 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> The second declaration of status is shadowing the status of a higher scope. This uninitialized status results in garbage being returned by the !x86_match_cpu(cpu_ids) || !iosf_mbi_available() return exit path. Fix this by removing the extraneous second declaration of status. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 773acfb..4c14215 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -249,7 +249,6 @@ static int is_byt_cr(struct device *dev, bool *bytcr) { X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */ {} }; - int status; u32 bios_status; if (!x86_match_cpu(cpu_ids) || !iosf_mbi_available()) { -- 2.9.3

4 3

[alsa-devel] [PATCH 0/2] ASoC: Chromebook Pixel 2015 support
by John Keeping 25 Sep '16

25 Sep '16

These patches add basic support for audio on the Chromebook Pixel 2015. I took the machine driver from ChromeOS commit v3.14-10186-g7e7d95df2983 and other than removing support for wake on voice (which is not supported by the mainline rt5677 codec driver), the only differences are to cope with API changes since 3.14. Only two authors have committed changes to the machine driver in the ChromeOS tree and all of the commits are signed off; both authors' SoB lines are preserved in the second commit here. The codec driver is sufficiently different that I just added ACPI support from scratch using the ChromeOS source as a reference for the property names. John Keeping (2): ASoC: rt5677: Add ACPI support ASoC: Intel: boards: Add bdw-rt5677 machine driver sound/soc/codecs/rt5677.c | 109 +++++++++--- sound/soc/intel/Kconfig | 11 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5677.c | 341 ++++++++++++++++++++++++++++++++++++ sound/soc/intel/common/sst-acpi.c | 1 + 5 files changed, 443 insertions(+), 21 deletions(-) create mode 100644 sound/soc/intel/boards/bdw-rt5677.c -- 2.9.3.728.g30b24b4.dirty

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[alsa-devel] [PATCH 0/2] Add DP dai-links to the rk3399-gru machine driver
by Chris Zhong 24 Sep '16

24 Sep '16

Hi This series patch is for the machine driver supporting for DP. It base on the DP drvier patches[0] and hdmi-notify patches[1], they are both in reviewing. [0] https://patchwork.kernel.org/patch/9315557/ https://patchwork.kernel.org/patch/9278077/ [1] https://patchwork.kernel.org/patch/8887261/ https://patchwork.kernel.org/patch/8887251/ Chris Zhong (2): drm/rockchip: cdn-dp: support audio hot-plug ASoC: rockchip: Add DP dai-links to the rk3399-gru machine driver .../bindings/sound/rockchip,rk3399-gru-sound.txt | 13 +++--- drivers/gpu/drm/rockchip/cdn-dp-core.c | 4 ++ sound/soc/rockchip/rk3399_gru_sound.c | 48 ++++++++++++++++++++++ 3 files changed, 59 insertions(+), 6 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH v2 00/11] ASoC: topology: Remaining kernel patches
by mengdong.lin＠linux.intel.com 24 Sep '16

24 Sep '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series contains all remaining kernel patches of topology, including some ABI update to PCM (FrontEnds) and link (BackEnds) objects. We don't user more series. User will be able to create BE DAIs and DAI links, configure more for FE links. Code are verified and can cover reqeust of Intel pre-release platforms for next year, so ABI should be stable. Current kernel topology code does not really touch Codec-Codec links since there is no user requst atm. We can add support for CC links later by reusing code and data structures for BE links, and no need to revise ABI. History: v2: Add the reason for creating BE DAI & DAI links by topology to commit message. Drop support for configuring DPCM trigger ordering in topology. Guneshwor Singh (1): ASoC: topology: Add FE DAIs only if not already added Mengdong Lin (10): ASoC: topology: Able to create BE DAIs ASoC: topology: ABI - Add flags to PCM ASoC: topology: ABI - Add private data to PCM ASoC: topology: ABI - Add name & component info to BE/CC links ASoC: topology: ABI - Define DAI physical PCM data formats ASoC: topology: ABI - Add HW configurations to BE/CC links ASoC: topology: ABI - Add flags and private data to BE/CC links ASoC: Define API to find a dai link by id ASoC: Probe link components after finding new links ASoC: topology: Able to create BE DAI links include/sound/soc-dai.h | 15 +- include/sound/soc-topology.h | 1 + include/sound/soc.h | 2 + include/uapi/sound/asoc.h | 90 ++++++++++- sound/soc/soc-core.c | 36 ++++- sound/soc/soc-topology.c | 355 ++++++++++++++++++++++++++++++++++++++----- 6 files changed, 446 insertions(+), 53 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 1/2] ASoC: cs35l34: Initial commit of the cs35l34 CODEC driver.
by Paul.Handrigan＠cirrus.com 24 Sep '16

24 Sep '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l34 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l34.h | 35 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l34.c | 1263 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l34.h | 269 ++++++++++ 5 files changed, 1574 insertions(+) create mode 100644 include/sound/cs35l34.h create mode 100644 sound/soc/codecs/cs35l34.c create mode 100644 sound/soc/codecs/cs35l34.h diff --git a/include/sound/cs35l34.h b/include/sound/cs35l34.h new file mode 100644 index 0000000..9c927cf --- /dev/null +++ b/include/sound/cs35l34.h @@ -0,0 +1,35 @@ +/* + * linux/sound/cs35l34.h -- Platform data for CS35l34 + * + * Copyright (c) 2016 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 __CS35L34_H +#define __CS35L34_H + +struct cs35l34_platform_data { + /* Set AIF to half drive strength */ + bool aif_half_drv; + /* Digital Soft Ramp Disable */ + bool digsft_disable; + /* Amplifier Invert */ + bool amp_inv; + /* Peak current (mA) */ + unsigned int boost_peak; + /* Boost inductor value (nH) */ + unsigned int boost_ind; + /* Boost Controller Voltage Setting (mV) */ + unsigned int boost_vtge; + /* Gain Change Zero Cross */ + bool gain_zc_disable; + /* SDIN Left/Right Selection */ + unsigned int i2s_sdinloc; + /* TDM Rising Edge */ + bool tdm_rising_edge; +}; + +#endif /* __CS35L34_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 80e89de..c974b68 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -48,6 +48,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C + select SND_SOC_CS35L34 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -398,6 +399,10 @@ config SND_SOC_CS35L33 tristate "Cirrus Logic CS35L33 CODEC" depends on I2C +config SND_SOC_CS35L34 + tristate "Cirrus Logic CS35L34 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ab19c79..731c073 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -38,6 +38,7 @@ snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o +snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -262,6 +263,7 @@ obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o +obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(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 diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c new file mode 100644 index 0000000..02f7b1b --- /dev/null +++ b/sound/soc/codecs/cs35l34.c @@ -0,0 +1,1263 @@ +/* + * cs35l34.c -- CS35l34 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(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/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/pm_runtime.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l34.h> + +#include "cs35l34.h" + +#define PDN_DONE_ATTEMPTS 10 +#define CS35L34_START_DELAY 50 + +struct cs35l34_private { + struct snd_soc_codec *codec; + struct cs35l34_platform_data pdata; + struct regmap *regmap; + void *control_data; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + int mclk_int; + bool tdm_mode; + struct gpio_desc *reset_gpio; /* Active-low reset GPIO */ +}; + +static const struct reg_default cs35l34_reg[] = { + {CS35L34_PWRCTL1, 0x01}, + {CS35L34_PWRCTL2, 0x19}, + {CS35L34_PWRCTL3, 0x01}, + {CS35L34_ADSP_CLK_CTL, 0x08}, + {CS35L34_MCLK_CTL, 0x11}, + {CS35L34_AMP_INP_DRV_CTL, 0x01}, + {CS35L34_AMP_DIG_VOL_CTL, 0x12}, + {CS35L34_AMP_DIG_VOL, 0x00}, + {CS35L34_AMP_ANLG_GAIN_CTL, 0x0F}, + {CS35L34_PROTECT_CTL, 0x06}, + {CS35L34_AMP_KEEP_ALIVE_CTL, 0x04}, + {CS35L34_BST_CVTR_V_CTL, 0x00}, + {CS35L34_BST_PEAK_I, 0x10}, + {CS35L34_BST_RAMP_CTL, 0x87}, + {CS35L34_BST_CONV_COEF_1, 0x24}, + {CS35L34_BST_CONV_COEF_2, 0x24}, + {CS35L34_BST_CONV_SLOPE_COMP, 0x4E}, + {CS35L34_BST_CONV_SW_FREQ, 0x08}, + {CS35L34_CLASS_H_CTL, 0x0D}, + {CS35L34_CLASS_H_HEADRM_CTL, 0x0D}, + {CS35L34_CLASS_H_RELEASE_RATE, 0x08}, + {CS35L34_CLASS_H_FET_DRIVE_CTL, 0x41}, + {CS35L34_CLASS_H_STATUS, 0x05}, + {CS35L34_VPBR_CTL, 0x0A}, + {CS35L34_VPBR_VOL_CTL, 0x90}, + {CS35L34_VPBR_TIMING_CTL, 0x6A}, + {CS35L34_PRED_MAX_ATTEN_SPK_LOAD, 0x95}, + {CS35L34_PRED_BROWNOUT_THRESH, 0x1C}, + {CS35L34_PRED_BROWNOUT_VOL_CTL, 0x00}, + {CS35L34_PRED_BROWNOUT_RATE_CTL, 0x10}, + {CS35L34_PRED_WAIT_CTL, 0x10}, + {CS35L34_PRED_ZVP_INIT_IMP_CTL, 0x08}, + {CS35L34_PRED_MAN_SAFE_VPI_CTL, 0x80}, + {CS35L34_VPBR_ATTEN_STATUS, 0x00}, + {CS35L34_PRED_BRWNOUT_ATT_STATUS, 0x00}, + {CS35L34_SPKR_MON_CTL, 0xC6}, + {CS35L34_ADSP_I2S_CTL, 0x00}, + {CS35L34_ADSP_TDM_CTL, 0x00}, + {CS35L34_TDM_TX_CTL_1_VMON, 0x00}, + {CS35L34_TDM_TX_CTL_2_IMON, 0x04}, + {CS35L34_TDM_TX_CTL_3_VPMON, 0x03}, + {CS35L34_TDM_TX_CTL_4_VBSTMON, 0x07}, + {CS35L34_TDM_TX_CTL_5_FLAG1, 0x08}, + {CS35L34_TDM_TX_CTL_6_FLAG2, 0x09}, + {CS35L34_TDM_TX_SLOT_EN_1, 0x00}, + {CS35L34_TDM_TX_SLOT_EN_2, 0x00}, + {CS35L34_TDM_TX_SLOT_EN_3, 0x00}, + {CS35L34_TDM_TX_SLOT_EN_4, 0x00}, + {CS35L34_TDM_RX_CTL_1_AUDIN, 0x40}, + {CS35L34_TDM_RX_CTL_3_ALIVE, 0x04}, + {CS35L34_MULT_DEV_SYNCH1, 0x00}, + {CS35L34_MULT_DEV_SYNCH2, 0x80}, + {CS35L34_PROT_RELEASE_CTL, 0x00}, + {CS35L34_DIAG_MODE_REG_LOCK, 0x00}, + {CS35L34_DIAG_MODE_CTL_1, 0x00}, + {CS35L34_DIAG_MODE_CTL_2, 0x00}, + {CS35L34_INT_MASK_1, 0xFF}, + {CS35L34_INT_MASK_2, 0xFF}, + {CS35L34_INT_MASK_3, 0xFF}, + {CS35L34_INT_MASK_4, 0xFF}, + {CS35L34_INT_STATUS_1, 0x30}, + {CS35L34_INT_STATUS_2, 0x05}, + {CS35L34_INT_STATUS_3, 0x00}, + {CS35L34_INT_STATUS_4, 0x00}, + {CS35L34_OTP_TRIM_STATUS, 0x00}, +}; + +static bool cs35l34_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L34_DEVID_AB: + case CS35L34_DEVID_CD: + case CS35L34_DEVID_E: + case CS35L34_FAB_ID: + case CS35L34_REV_ID: + case CS35L34_INT_STATUS_1: + case CS35L34_INT_STATUS_2: + case CS35L34_INT_STATUS_3: + case CS35L34_INT_STATUS_4: + case CS35L34_CLASS_H_STATUS: + case CS35L34_VPBR_ATTEN_STATUS: + case CS35L34_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l34_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L34_CHIP_ID: + case CS35L34_DEVID_AB: + case CS35L34_DEVID_CD: + case CS35L34_DEVID_E: + case CS35L34_FAB_ID: + case CS35L34_REV_ID: + case CS35L34_PWRCTL1: + case CS35L34_PWRCTL2: + case CS35L34_PWRCTL3: + case CS35L34_ADSP_CLK_CTL: + case CS35L34_MCLK_CTL: + case CS35L34_AMP_INP_DRV_CTL: + case CS35L34_AMP_DIG_VOL_CTL: + case CS35L34_AMP_DIG_VOL: + case CS35L34_AMP_ANLG_GAIN_CTL: + case CS35L34_PROTECT_CTL: + case CS35L34_AMP_KEEP_ALIVE_CTL: + case CS35L34_BST_CVTR_V_CTL: + case CS35L34_BST_PEAK_I: + case CS35L34_BST_RAMP_CTL: + case CS35L34_BST_CONV_COEF_1: + case CS35L34_BST_CONV_COEF_2: + case CS35L34_BST_CONV_SLOPE_COMP: + case CS35L34_BST_CONV_SW_FREQ: + case CS35L34_CLASS_H_CTL: + case CS35L34_CLASS_H_HEADRM_CTL: + case CS35L34_CLASS_H_RELEASE_RATE: + case CS35L34_CLASS_H_FET_DRIVE_CTL: + case CS35L34_CLASS_H_STATUS: + case CS35L34_VPBR_CTL: + case CS35L34_VPBR_VOL_CTL: + case CS35L34_VPBR_TIMING_CTL: + case CS35L34_PRED_MAX_ATTEN_SPK_LOAD: + case CS35L34_PRED_BROWNOUT_THRESH: + case CS35L34_PRED_BROWNOUT_VOL_CTL: + case CS35L34_PRED_BROWNOUT_RATE_CTL: + case CS35L34_PRED_WAIT_CTL: + case CS35L34_PRED_ZVP_INIT_IMP_CTL: + case CS35L34_PRED_MAN_SAFE_VPI_CTL: + case CS35L34_VPBR_ATTEN_STATUS: + case CS35L34_PRED_BRWNOUT_ATT_STATUS: + case CS35L34_SPKR_MON_CTL: + case CS35L34_ADSP_I2S_CTL: + case CS35L34_ADSP_TDM_CTL: + case CS35L34_TDM_TX_CTL_1_VMON: + case CS35L34_TDM_TX_CTL_2_IMON: + case CS35L34_TDM_TX_CTL_3_VPMON: + case CS35L34_TDM_TX_CTL_4_VBSTMON: + case CS35L34_TDM_TX_CTL_5_FLAG1: + case CS35L34_TDM_TX_CTL_6_FLAG2: + case CS35L34_TDM_TX_SLOT_EN_1: + case CS35L34_TDM_TX_SLOT_EN_2: + case CS35L34_TDM_TX_SLOT_EN_3: + case CS35L34_TDM_TX_SLOT_EN_4: + case CS35L34_TDM_RX_CTL_1_AUDIN: + case CS35L34_TDM_RX_CTL_3_ALIVE: + case CS35L34_MULT_DEV_SYNCH1: + case CS35L34_MULT_DEV_SYNCH2: + case CS35L34_PROT_RELEASE_CTL: + case CS35L34_DIAG_MODE_REG_LOCK: + case CS35L34_DIAG_MODE_CTL_1: + case CS35L34_DIAG_MODE_CTL_2: + case CS35L34_INT_MASK_1: + case CS35L34_INT_MASK_2: + case CS35L34_INT_MASK_3: + case CS35L34_INT_MASK_4: + case CS35L34_INT_STATUS_1: + case CS35L34_INT_STATUS_2: + case CS35L34_INT_STATUS_3: + case CS35L34_INT_STATUS_4: + case CS35L34_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l34_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L34_INT_STATUS_1: + case CS35L34_INT_STATUS_2: + case CS35L34_INT_STATUS_3: + case CS35L34_INT_STATUS_4: + return true; + default: + return false; + } +} + +static int cs35l34_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->tdm_mode) { + regmap_update_bits(priv->regmap, CS35L34_PWRCTL3, + CS35L34_PDN_TDM, 0x00); + } + ret = regmap_update_bits(priv->regmap, CS35L34_PWRCTL1, + CS35L34_PDN_ALL, 0); + if (ret < 0) { + dev_err(codec->dev, "Cannot set Power bits %d\n", ret); + return ret; + } + usleep_range(5000, 5100); + break; + case SND_SOC_DAPM_POST_PMD: + if (priv->tdm_mode) { + regmap_update_bits(priv->regmap, CS35L34_PWRCTL3, + CS35L34_PDN_TDM, CS35L34_PDN_TDM); + } + ret = regmap_update_bits(priv->regmap, CS35L34_PWRCTL1, + CS35L34_PDN_ALL, CS35L34_PDN_ALL); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + +static int cs35l34_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; + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + priv->tdm_mode = true; + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) + snd_soc_update_bits(codec, CS35L34_TDM_RX_CTL_1_AUDIN, + CS35L34_X_LOC, slot); + + /* scan tx_mask: vmon(2 slots); imon (2 slots); vpmon (1 slot) + * vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_3_VPMON, + CS35L34_X_STATE | CS35L34_X_LOC, + CS35L34_X_STATE | CS35L34_X_LOC); + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_4_VBSTMON, + CS35L34_X_STATE | CS35L34_X_LOC, + CS35L34_X_STATE | CS35L34_X_LOC); + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_1_VMON, + CS35L34_X_STATE | CS35L34_X_LOC, slot); + + /* configure IMON_TX_LOC */ + if (slot_num == 4) { + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_2_IMON, + CS35L34_X_STATE | CS35L34_X_LOC, slot); + } + /* configure VPMON_TX_LOC */ + if (slot_num == 3) { + snd_soc_update_bits(codec, CS35L34_TDM_TX_CTL_3_VPMON, + CS35L34_X_STATE | CS35L34_X_LOC, slot); + } + /* configure VBSTMON_TX_LOC */ + if (slot_num == 7) { + snd_soc_update_bits(codec, + CS35L34_TDM_TX_CTL_4_VBSTMON, + CS35L34_X_STATE | CS35L34_X_LOC, slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L34_TDM_TX_SLOT_EN_4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + snd_soc_update_bits(codec, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l34_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(priv->regmap, CS35L34_BST_CVTR_V_CTL, + CS35L34_BST_CVTL_MASK, priv->pdata.boost_vtge); + usleep_range(5000, 5100); + regmap_update_bits(priv->regmap, CS35L34_PROTECT_CTL, + CS35L34_MUTE, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L34_BST_CVTR_V_CTL, + CS35L34_BST_CVTL_MASK, 0); + regmap_update_bits(priv->regmap, CS35L34_PROTECT_CTL, + CS35L34_MUTE, CS35L34_MUTE); + usleep_range(5000, 5100); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 300, 100, 0); + + +static const struct snd_kcontrol_new cs35l34_snd_controls[] = { + SOC_SINGLE_SX_TLV("Digital Volume", CS35L34_AMP_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("Amp Gain Volume", CS35L34_AMP_ANLG_GAIN_CTL, + 0, 0xF, 0, amp_gain_tlv), +}; + + +static int cs35l34_mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + int ret, i; + unsigned int reg; + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + ret = regmap_read(priv->regmap, CS35L34_AMP_DIG_VOL_CTL, + &reg); + if (ret != 0) { + pr_err("%s regmap read failure %d\n", __func__, ret); + return ret; + } + if (reg & CS35L34_AMP_DIGSFT) + msleep(40); + else + usleep_range(2000, 2010); + + for (i = 0; i < PDN_DONE_ATTEMPTS; i++) { + ret = regmap_read(priv->regmap, CS35L34_INT_STATUS_2, + &reg); + if (ret != 0) { + pr_err("%s regmap read failure %d\n", + __func__, ret); + return ret; + } + if (reg & CS35L34_PDN_DONE) + break; + + usleep_range(5000, 5010); + } + if (i == PDN_DONE_ATTEMPTS) + pr_err("%s Device did not power down properly\n", + __func__); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget cs35l34_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L34_PWRCTL3, + 1, 1, cs35l34_sdin_event, + SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L34_PWRCTL3, 2, 1), + + SND_SOC_DAPM_SUPPLY("EXTCLK", CS35L34_PWRCTL3, 7, 1, + cs35l34_mclk_event, SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VPST"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L34_PWRCTL2, 7, 1), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L34_PWRCTL2, 6, 1), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L34_PWRCTL3, 3, 1), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L34_PWRCTL3, 4, 1), + SND_SOC_DAPM_ADC("CLASS H", NULL, CS35L34_PWRCTL2, 5, 1), + SND_SOC_DAPM_ADC("BOOST", NULL, CS35L34_PWRCTL2, 2, 1), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L34_PWRCTL2, 0, 1, NULL, 0, + cs35l34_main_amp_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cs35l34_audio_map[] = { + {"SDIN", NULL, "AMP Playback"}, + {"BOOST", NULL, "SDIN"}, + {"CLASS H", NULL, "BOOST"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, + + {"VPMON ADC", NULL, "CLASS H"}, + {"VBSTMON ADC", NULL, "CLASS H"}, + {"SPK", NULL, "VPMON ADC"}, + {"SPK", NULL, "VBSTMON ADC"}, + + {"IMON ADC", NULL, "ISENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + {"SDOUT", NULL, "IMON ADC"}, + {"SDOUT", NULL, "VMON ADC"}, + {"AMP Capture", NULL, "SDOUT"}, + + {"SDIN", NULL, "EXTCLK"}, + {"SDOUT", NULL, "EXTCLK"}, +}; + +struct cs35l34_mclk_div { + int mclk; + int srate; + u8 adsp_rate; +}; + +static struct cs35l34_mclk_div cs35l34_mclk_coeffs[] = { + + /* MCLK, Sample Rate, adsp_rate */ + + {5644800, 11025, 0x1}, + {5644800, 22050, 0x4}, + {5644800, 44100, 0x7}, + + {6000000, 8000, 0x0}, + {6000000, 11025, 0x1}, + {6000000, 12000, 0x2}, + {6000000, 16000, 0x3}, + {6000000, 22050, 0x4}, + {6000000, 24000, 0x5}, + {6000000, 32000, 0x6}, + {6000000, 44100, 0x7}, + {6000000, 48000, 0x8}, + + {6144000, 8000, 0x0}, + {6144000, 11025, 0x1}, + {6144000, 12000, 0x2}, + {6144000, 16000, 0x3}, + {6144000, 22050, 0x4}, + {6144000, 24000, 0x5}, + {6144000, 32000, 0x6}, + {6144000, 44100, 0x7}, + {6144000, 48000, 0x8}, +}; + +static int cs35l34_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l34_mclk_coeffs); i++) { + if (cs35l34_mclk_coeffs[i].mclk == mclk && + cs35l34_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l34_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L34_ADSP_CLK_CTL, + 0x80, 0x80); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L34_ADSP_CLK_CTL, + 0x80, 0x00); + break; + default: + return -EINVAL; + } + return 0; +} + +static int cs35l34_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 cs35l34_private *priv = snd_soc_codec_get_drvdata(codec); + int srate = params_rate(params); + int ret; + + int coeff = cs35l34_get_mclk_coeff(priv->mclk_int, srate); + + if (coeff < 0) { + dev_err(codec->dev, "ERROR: Invalid mclk %d and/or srate %d\n", + priv->mclk_int, srate); + return coeff; + } + + ret = regmap_update_bits(priv->regmap, CS35L34_ADSP_CLK_CTL, + CS35L34_ADSP_RATE, cs35l34_mclk_coeffs[coeff].adsp_rate); + if (ret != 0) + dev_err(codec->dev, "Failed to set clock state %d\n", ret); + + return ret; +} + +static unsigned int cs35l34_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 +}; + + +static struct snd_pcm_hw_constraint_list cs35l34_constraints = { + .count = ARRAY_SIZE(cs35l34_src_rates), + .list = cs35l34_src_rates, +}; + +static int cs35l34_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l34_constraints); + return 0; +} + + +static int cs35l34_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + + struct snd_soc_codec *codec = dai->codec; + + if (tristate) + snd_soc_update_bits(codec, CS35L34_PWRCTL3, + CS35L34_PDN_SDOUT, CS35L34_PDN_SDOUT); + else + snd_soc_update_bits(codec, CS35L34_PWRCTL3, + CS35L34_PDN_SDOUT, 0); + return 0; +} + +static int cs35l34_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l34_private *cs35l34 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L34_MCLK_5644: + snd_soc_update_bits(codec, CS35L34_MCLK_CTL, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK, + ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_5P6448); + cs35l34->mclk_int = freq; + break; + case CS35L34_MCLK_6: + snd_soc_update_bits(codec, CS35L34_MCLK_CTL, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK, + ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P0000); + cs35l34->mclk_int = freq; + break; + case CS35L34_MCLK_6144: + snd_soc_update_bits(codec, CS35L34_MCLK_CTL, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK, + ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P1440); + cs35l34->mclk_int = freq; + break; + case CS35L34_MCLK_11289: + snd_soc_update_bits(codec, CS35L34_MCLK_CTL, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_5P6448); + cs35l34->mclk_int = freq/2; + break; + case CS35L34_MCLK_12: + snd_soc_update_bits(codec, CS35L34_MCLK_CTL, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_6P0000); + cs35l34->mclk_int = freq/2; + break; + case CS35L34_MCLK_12288: + snd_soc_update_bits(codec, CS35L34_MCLK_CTL, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK, + CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_6P1440); + cs35l34->mclk_int = freq/2; + break; + default: + dev_err(codec->dev, "ERROR: Invalid Frequency %d\n", freq); + cs35l34->mclk_int = 0; + return -EINVAL; + } + return 0; +} + +static const struct snd_soc_dai_ops cs35l34_ops = { + .startup = cs35l34_pcm_startup, + .set_tristate = cs35l34_set_tristate, + .set_fmt = cs35l34_set_dai_fmt, + .hw_params = cs35l34_pcm_hw_params, + .set_sysclk = cs35l34_dai_set_sysclk, + .set_tdm_slot = cs35l34_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l34_dai = { + .name = "cs35l34", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = CS35L34_RATES, + .formats = CS35L34_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = CS35L34_RATES, + .formats = CS35L34_FORMATS, + }, + .ops = &cs35l34_ops, + .symmetric_rates = 1, +}; + +static int cs35l34_boost_inductor(struct cs35l34_private *cs35l34, + unsigned int inductor) +{ + struct snd_soc_codec *codec = cs35l34->codec; + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_1, 0x24); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_2, 0x24); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SLOPE_COMP, + 0x4E); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SW_FREQ, 0); + break; + case 1200: /* 1.2 uH */ + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_1, 0x20); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_2, 0x20); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SLOPE_COMP, + 0x47); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SW_FREQ, 1); + break; + case 1500: /* 1.5uH */ + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_1, 0x20); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_2, 0x20); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SLOPE_COMP, + 0x3C); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SW_FREQ, 2); + break; + case 2200: /* 2.2uH */ + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_1, 0x19); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_COEF_2, 0x25); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SLOPE_COMP, + 0x23); + regmap_write(cs35l34->regmap, CS35L34_BST_CONV_SW_FREQ, 3); + break; + default: + dev_err(codec->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + return 0; +} + +static int cs35l34_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct cs35l34_private *cs35l34 = snd_soc_codec_get_drvdata(codec); + + codec->control_data = cs35l34->regmap; + + pm_runtime_get_sync(codec->dev); + + /* Set over temperature warning attenuation to 6 dB */ + regmap_update_bits(cs35l34->regmap, CS35L34_PROTECT_CTL, + CS35L34_OTW_ATTN_MASK, 0x8); + + /* Set Power control registers 2 and 3 to have everything + * powered down at initialization + */ + regmap_write(cs35l34->regmap, CS35L34_PWRCTL2, 0xFD); + regmap_write(cs35l34->regmap, CS35L34_PWRCTL3, 0x1F); + + /* Set mute bit at startup */ + regmap_update_bits(cs35l34->regmap, CS35L34_PROTECT_CTL, + CS35L34_MUTE, CS35L34_MUTE); + + /* Set Platform Data */ + if (cs35l34->pdata.boost_peak) + regmap_update_bits(cs35l34->regmap, CS35L34_BST_PEAK_I, + CS35L34_BST_PEAK_MASK, + cs35l34->pdata.boost_peak); + + if (cs35l34->pdata.gain_zc_disable) + regmap_update_bits(cs35l34->regmap, CS35L34_PROTECT_CTL, + CS35L34_GAIN_ZC_MASK, 0); + else + regmap_update_bits(cs35l34->regmap, CS35L34_PROTECT_CTL, + CS35L34_GAIN_ZC_MASK, CS35L34_GAIN_ZC_MASK); + + if (cs35l34->pdata.aif_half_drv) + regmap_update_bits(cs35l34->regmap, CS35L34_ADSP_CLK_CTL, + CS35L34_ADSP_DRIVE, 0); + + if (cs35l34->pdata.digsft_disable) + regmap_update_bits(cs35l34->regmap, CS35L34_AMP_DIG_VOL_CTL, + CS35L34_AMP_DIGSFT, 0); + + if (cs35l34->pdata.amp_inv) + regmap_update_bits(cs35l34->regmap, CS35L34_AMP_DIG_VOL_CTL, + CS35L34_INV, CS35L34_INV); + + if (cs35l34->pdata.boost_ind) + ret = cs35l34_boost_inductor(cs35l34, cs35l34->pdata.boost_ind); + + if (cs35l34->pdata.i2s_sdinloc) + regmap_update_bits(cs35l34->regmap, CS35L34_ADSP_I2S_CTL, + CS35L34_I2S_LOC_MASK, + cs35l34->pdata.i2s_sdinloc << CS35L34_I2S_LOC_SHIFT); + + if (cs35l34->pdata.tdm_rising_edge) + regmap_update_bits(cs35l34->regmap, CS35L34_ADSP_TDM_CTL, + 1, 1); + + pm_runtime_put_sync(codec->dev); + + return ret; +} + + +static struct snd_soc_codec_driver soc_codec_dev_cs35l34 = { + .probe = cs35l34_probe, + + .component_driver = { + .dapm_widgets = cs35l34_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l34_dapm_widgets), + .dapm_routes = cs35l34_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l34_audio_map), + .controls = cs35l34_snd_controls, + .num_controls = ARRAY_SIZE(cs35l34_snd_controls), + }, +}; + +static struct regmap_config cs35l34_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L34_MAX_REGISTER, + .reg_defaults = cs35l34_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l34_reg), + .volatile_reg = cs35l34_volatile_register, + .readable_reg = cs35l34_readable_register, + .precious_reg = cs35l34_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs35l34_handle_of_data(struct i2c_client *i2c_client, + struct cs35l34_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + + if (of_property_read_u32(np, "cirrus,boost-vtge", &val) >= 0) { + /* Boost Voltage has a maximum of 8V */ + if (val > 8000 || (val < 3300 && val > 0)) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + if (val == 0) + pdata->boost_vtge = 0; /* Use VP */ + else + pdata->boost_vtge = ((val - 3300)/100) + 1; + } else { + dev_warn(&i2c_client->dev, + "Boost Voltage not specified. Using VP\n"); + } + + if (of_property_read_u32(np, "cirrus,boost-ind", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "cirrus,boost-peak", &val) >= 0) { + if (val > 3840 || val < 1200) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %d mA\n", val); + return -EINVAL; + } + pdata->boost_peak = ((val - 1200)/80) + 1; + } + + pdata->aif_half_drv = of_property_read_bool(np, + "cirrus,aif-half-drv"); + pdata->digsft_disable = of_property_read_bool(np, + "cirrus,digsft-disable"); + + pdata->gain_zc_disable = of_property_read_bool(np, + "cirrus,gain-zc-disable"); + pdata->amp_inv = of_property_read_bool(np, "cirrus,amp-inv"); + + if (of_property_read_u32(np, "cirrus,i2s-sdinloc", &val) >= 0) + pdata->i2s_sdinloc = val; + if (of_property_read_u32(np, "cirrus,tdm-rising-edge", &val) >= 0) + pdata->tdm_rising_edge = val; + + return 0; +} + +static irqreturn_t cs35l34_irq_thread(int irq, void *data) +{ + struct cs35l34_private *cs35l34 = data; + struct snd_soc_codec *codec = cs35l34->codec; + unsigned int sticky1, sticky2, sticky3, sticky4; + unsigned int mask1, mask2, mask3, mask4, current1; + + + /* ack the irq by reading all status registers */ + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_4, &sticky4); + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_3, &sticky3); + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_2, &sticky2); + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_1, &sticky1); + + regmap_read(cs35l34->regmap, CS35L34_INT_MASK_4, &mask4); + regmap_read(cs35l34->regmap, CS35L34_INT_MASK_3, &mask3); + regmap_read(cs35l34->regmap, CS35L34_INT_MASK_2, &mask2); + regmap_read(cs35l34->regmap, CS35L34_INT_MASK_1, &mask1); + + if (!(sticky1 & ~mask1) && !(sticky2 & ~mask2) && !(sticky3 & ~mask3) + && !(sticky4 & ~mask4)) + return IRQ_NONE; + + regmap_read(cs35l34->regmap, CS35L34_INT_STATUS_1, &current1); + + if (sticky1 & CS35L34_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L34_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_CAL_ERR_RLS, 0); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_CAL_ERR_RLS, + CS35L34_CAL_ERR_RLS); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_CAL_ERR_RLS, 0); + /* note: amp will re-calibrate on next resume */ + } + } + + if (sticky1 & CS35L34_ALIVE_ERR) + dev_err(codec->dev, "Alive error\n"); + + if (sticky1 & CS35L34_AMP_SHORT) { + dev_crit(codec->dev, "Amp short error\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L34_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_SHORT_RLS, 0); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_SHORT_RLS, + CS35L34_SHORT_RLS); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_SHORT_RLS, 0); + } + } + + if (sticky1 & CS35L34_OTW) { + dev_crit(codec->dev, "Over temperature warning\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L34_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTW_RLS, 0); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTW_RLS, + CS35L34_OTW_RLS); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTW_RLS, 0); + } + } + + if (sticky1 & CS35L34_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L34_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTE_RLS, 0); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTE_RLS, + CS35L34_OTE_RLS); + regmap_update_bits(cs35l34->regmap, + CS35L34_PROT_RELEASE_CTL, + CS35L34_OTE_RLS, 0); + } + } + + if (sticky3 & CS35L34_BST_HIGH) { + dev_crit(codec->dev, "VBST too high error; powering off!\n"); + regmap_update_bits(cs35l34->regmap, CS35L34_PWRCTL2, + CS35L34_PDN_AMP, CS35L34_PDN_AMP); + regmap_update_bits(cs35l34->regmap, CS35L34_PWRCTL1, + CS35L34_PDN_ALL, CS35L34_PDN_ALL); + } + + if (sticky3 & CS35L34_LBST_SHORT) { + dev_crit(codec->dev, "LBST short error; powering off!\n"); + regmap_update_bits(cs35l34->regmap, CS35L34_PWRCTL2, + CS35L34_PDN_AMP, CS35L34_PDN_AMP); + regmap_update_bits(cs35l34->regmap, CS35L34_PWRCTL1, + CS35L34_PDN_ALL, CS35L34_PDN_ALL); + } + + return IRQ_HANDLED; +} + +static const char * const cs35l34_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l34_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l34_private *cs35l34; + struct cs35l34_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int i; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs35l34 = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l34_private), + GFP_KERNEL); + if (!cs35l34) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs35l34); + cs35l34->regmap = devm_regmap_init_i2c(i2c_client, &cs35l34_regmap); + if (IS_ERR(cs35l34->regmap)) { + ret = PTR_ERR(cs35l34->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + cs35l34->num_core_supplies = ARRAY_SIZE(cs35l34_core_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l34_core_supplies); i++) + cs35l34->core_supplies[i].supply = cs35l34_core_supplies[i]; + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l34->num_core_supplies, + cs35l34->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l34->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l34_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 = cs35l34_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + cs35l34->pdata = *pdata; + } + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l34_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l34", cs35l34); + if (ret != 0) + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + cs35l34->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset-gpios", GPIOD_OUT_LOW); + if (IS_ERR(cs35l34->reset_gpio)) + return PTR_ERR(cs35l34->reset_gpio); + + gpiod_set_value_cansleep(cs35l34->reset_gpio, 1); + + msleep(CS35L34_START_DELAY); + + ret = regmap_read(cs35l34->regmap, CS35L34_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l34->regmap, CS35L34_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l34->regmap, CS35L34_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L34_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35l34 Device ID (%X). Expected ID %X\n", + devid, CS35L34_CHIP_ID); + ret = -ENODEV; + goto err_regulator; + } + + ret = regmap_read(cs35l34->regmap, CS35L34_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_regulator; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35l34 (%x), Revision: %02X\n", devid, + reg & 0xFF); + + /* Unmask critical interrupts */ + regmap_update_bits(cs35l34->regmap, CS35L34_INT_MASK_1, + CS35L34_M_CAL_ERR | CS35L34_M_ALIVE_ERR | + CS35L34_M_AMP_SHORT | CS35L34_M_OTW | + CS35L34_M_OTE, 0); + regmap_update_bits(cs35l34->regmap, CS35L34_INT_MASK_3, + CS35L34_M_BST_HIGH | CS35L34_M_LBST_SHORT, 0); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l34, &cs35l34_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, + "%s: Register codec failed\n", __func__); + goto err_regulator; + } + + return 0; + +err_regulator: + regulator_bulk_disable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + return ret; +} + +static int cs35l34_i2c_remove(struct i2c_client *client) +{ + struct cs35l34_private *cs35l34 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l34->reset_gpio) + gpiod_set_value_cansleep(cs35l34->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + return 0; +} + +static int __maybe_unused cs35l34_runtime_resume(struct device *dev) +{ + struct cs35l34_private *cs35l34 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", + ret); + return ret; + } + + regcache_cache_only(cs35l34->regmap, false); + + gpiod_set_value_cansleep(cs35l34->reset_gpio, 1); + msleep(CS35L34_START_DELAY); + + ret = regcache_sync(cs35l34->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + return 0; +err: + regcache_cache_only(cs35l34->regmap, true); + regulator_bulk_disable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + return ret; +} + +static int __maybe_unused cs35l34_runtime_suspend(struct device *dev) +{ + struct cs35l34_private *cs35l34 = dev_get_drvdata(dev); + + regcache_cache_only(cs35l34->regmap, true); + regcache_mark_dirty(cs35l34->regmap); + + gpiod_set_value_cansleep(cs35l34->reset_gpio, 0); + + regulator_bulk_disable(cs35l34->num_core_supplies, + cs35l34->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l34_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l34_runtime_suspend, + cs35l34_runtime_resume, + NULL) +}; + +static const struct of_device_id cs35l34_of_match[] = { + {.compatible = "cirrus,cs35l34"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l34_of_match); + +static const struct i2c_device_id cs35l34_id[] = { + {"cs35l34", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs35l34_id); + +static struct i2c_driver cs35l34_i2c_driver = { + .driver = { + .name = "cs35l34", + .pm = &cs35l34_pm_ops, + .of_match_table = cs35l34_of_match, + + }, + .id_table = cs35l34_id, + .probe = cs35l34_i2c_probe, + .remove = cs35l34_i2c_remove, + +}; + +static int __init cs35l34_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&cs35l34_i2c_driver); + if (ret != 0) { + pr_err("Failed to register CS35l34 I2C driver: %d\n", ret); + return ret; + } + return 0; +} +module_init(cs35l34_modinit); + +static void __exit cs35l34_exit(void) +{ + i2c_del_driver(&cs35l34_i2c_driver); +} +module_exit(cs35l34_exit); + +MODULE_DESCRIPTION("ASoC CS35l34 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l34.h b/sound/soc/codecs/cs35l34.h new file mode 100644 index 0000000..bcd54f1 --- /dev/null +++ b/sound/soc/codecs/cs35l34.h @@ -0,0 +1,269 @@ +/* + * cs35l34.h -- CS35L34 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(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 __CS35L34_H__ +#define __CS35L34_H__ + +#define CS35L34_CHIP_ID 0x00035A34 +#define CS35L34_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L34_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L34_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L34_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L34_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L34_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L34_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L34_PWRCTL3 0x08 /* Power Ctl 3 */ +#define CS35L34_ADSP_CLK_CTL 0x0A /* (ADSP) Clock Ctl */ +#define CS35L34_MCLK_CTL 0x0B /* Master Clocking Ctl */ +#define CS35L34_AMP_INP_DRV_CTL 0x14 /* Amp Input Drive Ctl */ +#define CS35L34_AMP_DIG_VOL_CTL 0x15 /* Amplifier Dig Volume Ctl */ +#define CS35L34_AMP_DIG_VOL 0x16 /* Amplifier Dig Volume */ +#define CS35L34_AMP_ANLG_GAIN_CTL 0x17 /* Amplifier Analog Gain Ctl */ +#define CS35L34_PROTECT_CTL 0x18 /* Amp Gain - Prot Ctl Param */ +#define CS35L34_AMP_KEEP_ALIVE_CTL 0x1A /* Amplifier Keep Alive Ctl */ +#define CS35L34_BST_CVTR_V_CTL 0x1D /* Boost Conv Voltage Ctl */ +#define CS35L34_BST_PEAK_I 0x1E /* Boost Conv Peak Current */ +#define CS35L34_BST_RAMP_CTL 0x20 /* Boost Conv Soft Ramp Ctl */ +#define CS35L34_BST_CONV_COEF_1 0x21 /* Boost Conv Coefficients 1 */ +#define CS35L34_BST_CONV_COEF_2 0x22 /* Boost Conv Coefficients 2 */ +#define CS35L34_BST_CONV_SLOPE_COMP 0x23 /* Boost Conv Slope Comp */ +#define CS35L34_BST_CONV_SW_FREQ 0x24 /* Boost Conv L BST SW Freq */ +#define CS35L34_CLASS_H_CTL 0x30 /* CLS H Control */ +#define CS35L34_CLASS_H_HEADRM_CTL 0x31 /* CLS H Headroom Ctl */ +#define CS35L34_CLASS_H_RELEASE_RATE 0x32 /* CLS H Release Rate */ +#define CS35L34_CLASS_H_FET_DRIVE_CTL 0x33 /* CLS H Weak FET Drive Ctl */ +#define CS35L34_CLASS_H_STATUS 0x38 /* CLS H Status */ +#define CS35L34_VPBR_CTL 0x3A /* VPBR Ctl */ +#define CS35L34_VPBR_VOL_CTL 0x3B /* VPBR Volume Ctl */ +#define CS35L34_VPBR_TIMING_CTL 0x3C /* VPBR Timing Ctl */ +#define CS35L34_PRED_MAX_ATTEN_SPK_LOAD 0x40 /* PRD Max Atten / Spkr Load */ +#define CS35L34_PRED_BROWNOUT_THRESH 0x41 /* PRD Brownout Threshold */ +#define CS35L34_PRED_BROWNOUT_VOL_CTL 0x42 /* PRD Brownout Volume Ctl */ +#define CS35L34_PRED_BROWNOUT_RATE_CTL 0x43 /* PRD Brownout Rate Ctl */ +#define CS35L34_PRED_WAIT_CTL 0x44 /* PRD Wait Ctl */ +#define CS35L34_PRED_ZVP_INIT_IMP_CTL 0x46 /* PRD ZVP Initial Imp Ctl */ +#define CS35L34_PRED_MAN_SAFE_VPI_CTL 0x47 /* PRD Manual Safe VPI Ctl */ +#define CS35L34_VPBR_ATTEN_STATUS 0x4B /* VPBR Attenuation Status */ +#define CS35L34_PRED_BRWNOUT_ATT_STATUS 0x4C /* PRD Brownout Atten Status */ +#define CS35L34_SPKR_MON_CTL 0x4E /* Speaker Monitoring Ctl */ +#define CS35L34_ADSP_I2S_CTL 0x50 /* ADSP I2S Ctl */ +#define CS35L34_ADSP_TDM_CTL 0x51 /* ADSP TDM Ctl */ +#define CS35L34_TDM_TX_CTL_1_VMON 0x52 /* TDM TX Ctl 1 (VMON) */ +#define CS35L34_TDM_TX_CTL_2_IMON 0x53 /* TDM TX Ctl 2 (IMON) */ +#define CS35L34_TDM_TX_CTL_3_VPMON 0x54 /* TDM TX Ctl 3 (VPMON) */ +#define CS35L34_TDM_TX_CTL_4_VBSTMON 0x55 /* TDM TX Ctl 4 (VBSTMON) */ +#define CS35L34_TDM_TX_CTL_5_FLAG1 0x56 /* TDM TX Ctl 5 (FLAG1) */ +#define CS35L34_TDM_TX_CTL_6_FLAG2 0x57 /* TDM TX Ctl 6 (FLAG2) */ +#define CS35L34_TDM_TX_SLOT_EN_1 0x5A /* TDM TX Slot Enable */ +#define CS35L34_TDM_TX_SLOT_EN_2 0x5B /* TDM TX Slot Enable */ +#define CS35L34_TDM_TX_SLOT_EN_3 0x5C /* TDM TX Slot Enable */ +#define CS35L34_TDM_TX_SLOT_EN_4 0x5D /* TDM TX Slot Enable */ +#define CS35L34_TDM_RX_CTL_1_AUDIN 0x5E /* TDM RX Ctl 1 */ +#define CS35L34_TDM_RX_CTL_3_ALIVE 0x60 /* TDM RX Ctl 3 (ALIVE) */ +#define CS35L34_MULT_DEV_SYNCH1 0x62 /* Multidevice Synch */ +#define CS35L34_MULT_DEV_SYNCH2 0x63 /* Multidevice Synch 2 */ +#define CS35L34_PROT_RELEASE_CTL 0x64 /* Protection Release Ctl */ +#define CS35L34_DIAG_MODE_REG_LOCK 0x68 /* Diagnostic Mode Reg Lock */ +#define CS35L34_DIAG_MODE_CTL_1 0x69 /* Diagnostic Mode Ctl 1 */ +#define CS35L34_DIAG_MODE_CTL_2 0x6A /* Diagnostic Mode Ctl 2 */ +#define CS35L34_INT_MASK_1 0x70 /* Interrupt Mask 1 */ +#define CS35L34_INT_MASK_2 0x71 /* Interrupt Mask 2 */ +#define CS35L34_INT_MASK_3 0x72 /* Interrupt Mask 3 */ +#define CS35L34_INT_MASK_4 0x73 /* Interrupt Mask 4 */ +#define CS35L34_INT_STATUS_1 0x74 /* Interrupt Status 1 */ +#define CS35L34_INT_STATUS_2 0x75 /* Interrupt Status 2 */ +#define CS35L34_INT_STATUS_3 0x76 /* Interrupt Status 3 */ +#define CS35L34_INT_STATUS_4 0x77 /* Interrupt Status 4 */ +#define CS35L34_OTP_TRIM_STATUS 0x7E /* OTP Trim Status */ + +#define CS35L34_MAX_REGISTER 0x7F +#define CS35L34_REGISTER_COUNT 0x4E + +#define CS35L34_MCLK_5644 5644800 +#define CS35L34_MCLK_6144 6144000 +#define CS35L34_MCLK_6 6000000 +#define CS35L34_MCLK_11289 11289600 +#define CS35L34_MCLK_12 12000000 +#define CS35L34_MCLK_12288 12288000 + +/* CS35L34_PWRCTL1 */ +#define CS35L34_SFT_RST (1 << 7) +#define CS35L34_DISCHG_FLT (1 << 1) +#define CS35L34_PDN_ALL 1 + +/* CS35L34_PWRCTL2 */ +#define CS35L34_PDN_VMON (1 << 7) +#define CS35L34_PDN_IMON (1 << 6) +#define CS35L34_PDN_CLASSH (1 << 5) +#define CS35L34_PDN_VPBR (1 << 4) +#define CS35L34_PDN_PRED (1 << 3) +#define CS35L34_PDN_BST (1 << 2) +#define CS35L34_PDN_AMP 1 + +/* CS35L34_PWRCTL3 */ +#define CS35L34_MCLK_DIS (1 << 7) +#define CS35L34_PDN_VBSTMON_OUT (1 << 4) +#define CS35L34_PDN_VMON_OUT (1 << 3) +/* Tristate the ADSP SDOUT when in I2C mode */ +#define CS35L34_PDN_SDOUT (1 << 2) +#define CS35L34_PDN_SDIN (1 << 1) +#define CS35L34_PDN_TDM 1 + +/* CS35L34_ADSP_CLK_CTL */ +#define CS35L34_ADSP_RATE 0xF +#define CS35L34_ADSP_DRIVE (1 << 4) +#define CS35L34_ADSP_M_S (1 << 7) + +/* CS35L34_MCLK_CTL */ +#define CS35L34_MCLK_DIV (1 << 4) +#define CS35L34_MCLK_RATE_MASK 0x7 +#define CS35L34_MCLK_RATE_6P1440 0x2 +#define CS35L34_MCLK_RATE_6P0000 0x1 +#define CS35L34_MCLK_RATE_5P6448 0x0 +#define CS35L34_MCLKDIS (1 << 7) +#define CS35L34_MCLKDIV2 (1 << 6) +#define CS35L34_SDOUT_3ST_TDM (1 << 5) +#define CS35L34_INT_FS_RATE (1 << 4) +#define CS35L34_ADSP_FS 0xF + +/* CS35L34_AMP_INP_DRV_CTL */ +#define CS35L34_DRV_STR_SRC (1 << 1) +#define CS35L34_DRV_STR 1 + +/* CS35L34_AMP_DIG_VOL_CTL */ +#define CS35L34_AMP_DSR_RATE_MASK 0xF0 +#define CS35L34_AMP_DSR_RATE_SHIFT (1 << 4) +#define CS35L34_NOTCH_DIS (1 << 3) +#define CS35L34_AMP_DIGSFT (1 << 1) +#define CS35L34_INV 1 + +/* CS35L34_PROTECT_CTL */ +#define CS35L34_OTW_ATTN_MASK 0xC +#define CS35L34_OTW_THRD_MASK 0x3 +#define CS35L34_MUTE (1 << 5) +#define CS35L34_GAIN_ZC (1 << 4) +#define CS35L34_GAIN_ZC_MASK 0x10 +#define CS35L34_GAIN_ZC_SHIFT 4 + +/* CS35L34_AMP_KEEP_ALIVE_CTL */ +#define CS35L34_ALIVE_WD_DIS (1 << 2) + +/* CS35L34_BST_CVTR_V_CTL */ +#define CS35L34_BST_CVTL_MASK 0x3F + +/* CS35L34_BST_PEAK_I */ +#define CS35L34_BST_PEAK_MASK 0x3F + +/* CS35L34_ADSP_I2S_CTL */ +#define CS35L34_I2S_LOC_MASK 0xC +#define CS35L34_I2S_LOC_SHIFT 2 + +/* CS35L34_MULT_DEV_SYNCH2 */ +#define CS35L34_SYNC2_MASK 0xF + +/* CS35L34_PROT_RELEASE_CTL */ +#define CS35L34_CAL_ERR_RLS (1 << 7) +#define CS35L34_SHORT_RLS (1 << 2) +#define CS35L34_OTW_RLS (1 << 1) +#define CS35L34_OTE_RLS 1 + +/* CS35L34_INT_MASK_1 */ +#define CS35L34_M_CAL_ERR_SHIFT 7 +#define CS35L34_M_CAL_ERR (1 << CS35L34_M_CAL_ERR_SHIFT) +#define CS35L34_M_ALIVE_ERR_SHIFT 5 +#define CS35L34_M_ALIVE_ERR (1 << CS35L34_M_ALIVE_ERR_SHIFT) +#define CS35L34_M_ADSP_CLK_SHIFT 4 +#define CS35L34_M_ADSP_CLK_ERR (1 << CS35L34_M_ADSP_CLK_SHIFT) +#define CS35L34_M_MCLK_SHIFT 3 +#define CS35L34_M_MCLK_ERR (1 << CS35L34_M_MCLK_SHIFT) +#define CS35L34_M_AMP_SHORT_SHIFT 2 +#define CS35L34_M_AMP_SHORT (1 << CS35L34_M_AMP_SHORT_SHIFT) +#define CS35L34_M_OTW_SHIFT 1 +#define CS35L34_M_OTW (1 << CS35L34_M_OTW_SHIFT) +#define CS35L34_M_OTE_SHIFT 0 +#define CS35L34_M_OTE (1 << CS35L34_M_OTE_SHIFT) + +/* CS35L34_INT_MASK_2 */ +#define CS35L34_M_PDN_DONE_SHIFT 4 +#define CS35L34_M_PDN_DONE (1 << CS35L34_M_PDN_DONE_SHIFT) +#define CS35L34_M_PRED_SHIFT 3 +#define CS35L34_M_PRED_ERR (1 << CS35L34_M_PRED_SHIFT) +#define CS35L34_M_PRED_CLR_SHIFT 2 +#define CS35L34_M_PRED_CLR (1 << CS35L34_M_PRED_CLR_SHIFT) +#define CS35L34_M_VPBR_SHIFT 1 +#define CS35L34_M_VPBR_ERR (1 << CS35L34_M_VPBR_SHIFT) +#define CS35L34_M_VPBR_CLR_SHIFT 0 +#define CS35L34_M_VPBR_CLR (1 << CS35L34_M_VPBR_CLR_SHIFT) + +/* CS35L34_INT_MASK_3 */ +#define CS35L34_M_BST_HIGH_SHIFT 4 +#define CS35L34_M_BST_HIGH (1 << CS35L34_M_BST_HIGH_SHIFT) +#define CS35L34_M_BST_HIGH_FLAG_SHIFT 3 +#define CS35L34_M_BST_HIGH_FLAG (1 << CS35L34_M_BST_HIGH_FLAG_SHIFT) +#define CS35L34_M_BST_IPK_FLAG_SHIFT 2 +#define CS35L34_M_BST_IPK_FLAG (1 << CS35L34_M_BST_IPK_FLAG_SHIFT) +#define CS35L34_M_LBST_SHORT_SHIFT 0 +#define CS35L34_M_LBST_SHORT (1 << CS35L34_M_LBST_SHORT_SHIFT) + +/* CS35L34_INT_MASK_4 */ +#define CS35L34_M_VMON_OVFL_SHIFT 3 +#define CS35L34_M_VMON_OVFL (1 << CS35L34_M_VMON_OVFL_SHIFT) +#define CS35L34_M_IMON_OVFL_SHIFT 2 +#define CS35L34_M_IMON_OVFL (1 << CS35L34_M_IMON_OVFL_SHIFT) +#define CS35L34_M_VPMON_OVFL_SHIFT 1 +#define CS35L34_M_VPMON_OVFL (1 << CS35L34_M_VPMON_OVFL_SHIFT) +#define CS35L34_M_VBSTMON_OVFL_SHIFT 1 +#define CS35L34_M_VBSTMON_OVFL (1 << CS35L34_M_VBSTMON_OVFL_SHIFT) + +/* CS35L34_INT_1 */ +#define CS35L34_CAL_ERR (1 << CS35L34_M_CAL_ERR_SHIFT) +#define CS35L34_ALIVE_ERR (1 << CS35L34_M_ALIVE_ERR_SHIFT) +#define CS35L34_M_ADSP_CLK_ERR (1 << CS35L34_M_ADSP_CLK_SHIFT) +#define CS35L34_MCLK_ERR (1 << CS35L34_M_MCLK_SHIFT) +#define CS35L34_AMP_SHORT (1 << CS35L34_M_AMP_SHORT_SHIFT) +#define CS35L34_OTW (1 << CS35L34_M_OTW_SHIFT) +#define CS35L34_OTE (1 << CS35L34_M_OTE_SHIFT) + +/* CS35L34_INT_2 */ +#define CS35L34_PDN_DONE (1 << CS35L34_M_PDN_DONE_SHIFT) +#define CS35L34_PRED_ERR (1 << CS35L34_M_PRED_SHIFT) +#define CS35L34_PRED_CLR (1 << CS35L34_M_PRED_CLR_SHIFT) +#define CS35L34_VPBR_ERR (1 << CS35L34_M_VPBR_SHIFT) +#define CS35L34_VPBR_CLR (1 << CS35L34_M_VPBR_CLR_SHIFT) + +/* CS35L34_INT_3 */ +#define CS35L34_BST_HIGH (1 << CS35L34_M_BST_HIGH_SHIFT) +#define CS35L34_BST_HIGH_FLAG (1 << CS35L34_M_BST_HIGH_FLAG_SHIFT) +#define CS35L34_BST_IPK_FLAG (1 << CS35L34_M_BST_IPK_FLAG_SHIFT) +#define CS35L34_LBST_SHORT (1 << CS35L34_M_LBST_SHORT_SHIFT) + +/* CS35L34_INT_4 */ +#define CS35L34_VMON_OVFL (1 << CS35L34_M_VMON_OVFL_SHIFT) +#define CS35L34_IMON_OVFL (1 << CS35L34_M_IMON_OVFL_SHIFT) +#define CS35L34_VPMON_OVFL (1 << CS35L34_M_VPMON_OVFL_SHIFT) +#define CS35L34_VBSTMON_OVFL (1 << CS35L34_M_VBSTMON_OVFL_SHIFT) + +/* CS35L34_{RX,TX}_X */ +#define CS35L34_X_STATE_SHIFT 7 +#define CS35L34_X_STATE (1 << CS35L34_X_STATE_SHIFT) +#define CS35L34_X_LOC_SHIFT 0 +#define CS35L34_X_LOC (0x1F << CS35L34_X_LOC_SHIFT) + +#define CS35L34_RATES (SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_32000) +#define CS35L34_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#endif -- 1.9.1

3 2

[alsa-devel] A small bug in soc-pcm.c
by pengdonglin137 24 Sep '16

24 Sep '16

Hi All, I found a small bug in soc-pcm.c. For example, we can see kernel logs as follows: [ 518.707401] SLIMBUS_0_TX: ASoC: trigger BE MSM8996 Media1 cmd 0 // should be SLIMBUS_0_TX other than MSM8996 Media1 [ 518.707414] MSM8996 Media1: ASoC: post trigger FE MSM8996 Media1 cmd 0 Here is the patch. diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 60d702f..8c037ef 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1790,7 +1790,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: close BE %s\n", - dpcm->fe->dai_link->name); + dpcm->be->dai_link->name); soc_pcm_close(be_substream); be_substream->runtime = NULL; @@ -1856,7 +1856,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: hw_free BE %s\n", - dpcm->fe->dai_link->name); + dpcm->be->dai_link->name); soc_pcm_hw_free(be_substream); @@ -1934,7 +1934,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: hw_params BE %s\n", - dpcm->fe->dai_link->name); + dpcm->be->dai_link->name); ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params); if (ret < 0) { @@ -2014,7 +2014,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, int ret; dev_dbg(dpcm->be->dev, "ASoC: trigger BE %s cmd %d\n", - dpcm->fe->dai_link->name, cmd); + dpcm->be->dai_link->name, cmd); ret = soc_pcm_trigger(substream, cmd); if (ret < 0) @@ -2229,7 +2229,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", - dpcm->fe->dai_link->name); + dpcm->be->dai_link->name); ret = soc_pcm_prepare(be_substream); if (ret < 0) { Thanks, Donglin Peng

2 1

[alsa-devel] [PATCH 2/2] ASoC: cs35l34: Add device tree bindings file for cs35l34
by Paul.Handrigan＠cirrus.com 23 Sep '16

23 Sep '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Add device tree bindings file for the cs35l34 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l34.txt | 64 ++++++++++++++++++++++ 1 file changed, 64 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l34.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l34.txt b/Documentation/devicetree/bindings/sound/cs35l34.txt new file mode 100644 index 0000000..df62e77 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l34.txt @@ -0,0 +1,64 @@ +CS35L34 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l34" + + - reg : the I2C address of the device for I2C. + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-vtge : Boost Voltage Value. Configures the boost converter's + output voltage in mV. The range is from VP to 8V with increments of 100mV. + + - cirrus,boost-ind: Inductor value for boost converter. The value is in nH + and they can be values of 1000nH, 1100nH, 1200nH, 1500nH, and 2200nH. + +Optional properties: + + - reset-gpios: GPIO used to reset the amplifier. + + - interrupt-parent : Specifies the phandle of the interrupt controller to + which the IRQs from CS35L34 are delivered to. + - interrupts : IRQ line info CS35L34. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-peak : Boost converter peak current limit in mA. The range + starts at 1200mA and goes to a maximum of 3840mA with increments of 80mA. + The default value is 2480mA. + + - cirrus,i2s-sdinloc : ADSP SDIN I2S channel location. Indicates whether the + received mono data is in the left or right portion of the I2S frame + according to the AD0 pin or directly via this configuration. + 0x0 (Default) = Selected by AD0 input (if AD0 = LOW, use left channel), + 0x2 = Left, + 0x1 = Selected by the inversion of the AD0 input (if AD0 = LOW, use right + channel), + 0x3 = Right. + + - cirrus,gain-zc-disable: Boolean property. If set, the gain change will take + effect without waiting for a zero cross. + + - cirrus,tdm_rising_edge: Boolean property. If set, data is on the rising edge of + SCLK. Otherwise, data is on the falling edge of SCLK. + + +Example: + +cs35l34: cs35l34@40 { + compatible = "cirrus,cs35l34"; + reg = <0x40>; + + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + reset-gpios = <&gpio 10 0>; + + cirrus,boost-vtge = <8000>; /* 8V */ + cirrus,boost-ind = <1000>; /* 1uH */ + cirrus,boost-peak = <3000>; /* 3A */ +}; + -- 1.9.1

2 1

Re: [alsa-devel] Tascam US-366 and Alsa Firmware
by Max 23 Sep '16

23 Sep '16

That's a pity. The other devices from this series seem to be working with usx2yloader, what would be neccessary to get this device running? On 2016년 09월 21일 06:09, Burton Samograd wrote: > No, I never got it running unfortunately. It also barely works under OSX without locking my system when I plug it in. > > kruhft > >> On Sep 20, 2016, at 5:39 AM, Max <abonnements(a)revolwear.com> wrote: >> >> I have this device too and was wondering if OP or anyone else ever got it running and how. >> >> Thanks. >> >> >>> Hi, >>> >>> I recently bought the Tascam US-366 and forgot to check for Linux support, >>> which it doesn't have (yet! :). Looking through the Alsa firmware sources, >>> there seems to be a firmware loader called usx2yloader which supports a >>> number of Tascam cards, but the US-366 is not listed. It looks like really >>> usx2yloader just needs the firmware and a `prepad` file. I`ve found the >>> firmware for the card (I think) buried in the OSX firmware download >>> package: >>> >>> >>> http://tascam.com/content/downloads/products/789/us-322_us-366_fw_v103_mave… >>> >>> Now I think I just need the prepad file, but there's little documentation >>> regarding exactly what that is (I'm assuming a number of 0s just to pad the >>> firmware image to alignment, but I'm not sure of what alignment). >>> >>> Any ideas if usx2yloader will work at all with the US-366? Does anyone >>> know what the pre-pad file should be/contain? Has anyone tried to get the >>> US-366 working yet and failed? >>> >>> Reference: >>> >>> http://tascam.com/product/us-366/ >>> >>> Thanks. >>> >>> Burton Samograd >>

1 0

[alsa-devel] [PATCH] ASoC: simple-card: add support for aux devices
by Nikita Yushchenko 23 Sep '16

23 Sep '16

This patch makes it possible to use simple-card in setups where separate amplifier chip is connected to codec's output. Example usage: codec: tlv320dac3100@18 { compatible = "ti,tlv320dac3100"; ... } amp: tpa6130a2@60 { compatible = "ti,tpa6130a2"; ... } sound { compatible = "simple-audio-card"; ... simple-audio-card,widgets = "Headphone", "Headphone Jack"; simple-audio-card,routing = "Headphone Jack", "HPLEFT", "Headphone Jack", "HPRIGHT", "LEFTIN", "HPL", "RIGHTIN", "HPR"; simple-audio-card,aux-devs = <&amp>; simple-audio-card,cpu { sound-dai = <&ssi2>; }; simple-audio-card,codec { sound-dai = <&codec1>; clocks = ... }; This describes audio path from IMX6 SSI2 through TLV320DAC3100 codec through TPA6130A2 amplifier to headphones. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- .../devicetree/bindings/sound/simple-card.txt | 2 ++ sound/soc/generic/simple-card.c | 34 ++++++++++++++++++++++ 2 files changed, 36 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 59d8628..5579f40 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -22,6 +22,8 @@ Optional properties: headphones are attached. - simple-audio-card,mic-det-gpio : Reference to GPIO that signals when a microphone is attached. +- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. Optional subnodes: diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 43295f0..f989f34 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -417,6 +417,36 @@ dai_link_of_err: return ret; } +static int asoc_simple_card_parse_aux_devs(struct device_node *node, + struct simple_card_data *priv) +{ + struct device *dev = simple_priv_to_dev(priv); + struct device_node *aux_node; + int i, n, len; + + if (!of_find_property(node, PREFIX "aux-devs", &len)) + return 0; /* Ok to have no aux-devs */ + + n = len / sizeof(__be32); + if (n <= 0) + return -EINVAL; + + priv->snd_card.aux_dev = devm_kzalloc(dev, + n * sizeof(*priv->snd_card.aux_dev), GFP_KERNEL); + if (!priv->snd_card.aux_dev) + return -ENOMEM; + + for (i = 0; i < n; i++) { + aux_node = of_parse_phandle(node, PREFIX "aux-devs", i); + if (!aux_node) + return -EINVAL; + priv->snd_card.aux_dev[i].codec_of_node = aux_node; + } + + priv->snd_card.num_aux_devs = n; + return 0; +} + static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv) { @@ -474,6 +504,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (ret) return ret; + ret = asoc_simple_card_parse_aux_devs(node, priv); + if (ret) + return ret; + return 0; } -- 2.1.4

2 2

[alsa-devel] [PATCH] ALSA: usb-audio: Extend DragonFly dB scale quirk to cover other variants
by Anssi Hannula 23 Sep '16

23 Sep '16

The DragonFly quirk added in 42e3121d90f4 ("ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly") applies a custom dB map on the volume control when its range is reported as 0..50 (0 .. 0.2dB). However, there exists at least one other variant (hw v1.0c, as opposed to the tested v1.2) which reports a different non-sensical volume range (0..53) and the custom map is therefore not applied for that device. This results in all of the volume change appearing close to 100% on mixer UIs that utilize the dB TLV information. Add a fallback case where no dB TLV is reported at all if the control range is not 0..50 but still 0..N where N <= 1000 (3.9 dB). Also restrict the quirk to only apply to the volume control as there is also a mute control which would match the check otherwise. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Reported-by: David W <regulars(a)d-dub.org.uk> Tested-by: David W <regulars(a)d-dub.org.uk> Cc: <stable(a)vger.kernel.org> --- sound/usb/mixer_quirks.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index f6c3bf79af9a..f0de14a293ce 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1831,6 +1831,7 @@ void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, } static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer, + struct usb_mixer_elem_info *cval, struct snd_kcontrol *kctl) { /* Approximation using 10 ranges based on output measurement on hw v1.2. @@ -1848,10 +1849,18 @@ static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer, 41, 50, TLV_DB_MINMAX_ITEM(-441, 0), ); - usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n"); - kctl->tlv.p = scale; - kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + if (cval->min == 0 && cval->max == 50) { + usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk (0-50 variant)\n"); + kctl->tlv.p = scale; + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + + } else if (cval->min == 0 && cval->max <= 1000) { + /* Some other clearly broken DragonFly variant. + * At least a 0..53 variant (hw v1.0) exists. */ + usb_audio_info(mixer->chip, "ignoring too narrow dB range on a DragonFly device"); + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } } void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, @@ -1860,8 +1869,8 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, { switch (mixer->chip->usb_id) { case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ - if (unitid == 7 && cval->min == 0 && cval->max == 50) - snd_dragonfly_quirk_db_scale(mixer, kctl); + if (unitid == 7 && cval->control == UAC_FU_VOLUME) + snd_dragonfly_quirk_db_scale(mixer, cval, kctl); break; } } -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: cq93vc: duplicated callback function goes to component
by Arnd Bergmann 23 Sep '16

23 Sep '16

A cleanup removed a couple of members from struct snd_soc_codec_driver after changing codec drivers to no longer use them, but one codec was missed in the process, giving a build error: sound/soc/codecs/cq93vc.c:134:2: error: unknown field 'controls' specified in initializer .controls = cq93vc_snd_controls, This moves the members from the cq93vc codec driver to its component driver just like the other codecs already had. Fixes: 8073aefa6082 ("ASoC: remove codec duplicated callback function") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cq93vc.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 1c895a53001d..7a2d9a2ee427 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -131,8 +131,10 @@ static struct regmap *cq93vc_get_regmap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .set_bias_level = cq93vc_set_bias_level, .get_regmap = cq93vc_get_regmap, - .controls = cq93vc_snd_controls, - .num_controls = ARRAY_SIZE(cq93vc_snd_controls), + .component_driver = { + .controls = cq93vc_snd_controls, + .num_controls = ARRAY_SIZE(cq93vc_snd_controls), + }, }; static int cq93vc_platform_probe(struct platform_device *pdev) -- 2.9.0

2 1

[alsa-devel] arecord: does hardware or software throw away unused
by Rob Nertney 22 Sep '16

22 Sep '16

My hardware is putting a stream of MAX_CHANNELS into memory, and firing the IRQs on my period boundary with respect to this full MAX_CHANNELS stream. If I apply an arecord -c <MAX_CHANNELS-x> flag, essentially saying that I only want to capture less than the maximum number of channels, will arecord expect that the *hardware* will throw away the unused channel data, and the data placed into memory is a contiguous stream of only pertinent channels? Or will arecord see the MAX_CHANNELS in my driver, and use its *software* to strip out the upper channels worth of data automatically for me? For example, if I am capturing 2 channels of u8 data, and my dma regardless puts the data into memory as 11111111222222221111111122222222_IRQ_11111111222222221111111122222222. If I apply an "arecord -c 1 ...", and my DMA remains unchanged, will the resulting wavfile have 1111111111111111_IRQ_1111111111111111? Thanks, Rob

2 10

[alsa-devel] [PATCH] ASoC: rk3399_gru_sound: fix recording pop at first attempt
by Xing Zheng 22 Sep '16

22 Sep '16

From: Wonjoon Lee <woojoo.lee(a)samsung.com> Pop happens when mclk applied but dmic's own boot-time Specify dmic delay times in dt to make sure clocks are ready earlier than dmic working Signed-off-by: Wonjoon Lee <woojoo.lee(a)samsung.com> Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- .../bindings/sound/rockchip,rk3399-gru-sound.txt | 6 ++++++ sound/soc/rockchip/rk3399_gru_sound.c | 14 ++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt index f19b6c8..b7dd3ab 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -6,6 +6,12 @@ Required properties: connected to the codecs - rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs +Optional properties: +- dmic-delay : specify delay time for DMIC ready. + If this option is specified, which means it's required dmic need + delay for DMIC to ready so that rt5514 can avoid recording before + DMIC send valid data + Example: sound { diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 164b6da..6ab838b 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -37,6 +37,8 @@ #define SOUND_FS 256 +unsigned int rt5514_dmic_delay; + static struct snd_soc_jack rockchip_sound_jack; static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { @@ -122,6 +124,9 @@ static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Wait for DMIC stable */ + msleep(rt5514_dmic_delay); + return 0; } @@ -334,6 +339,15 @@ static int rockchip_sound_probe(struct platform_device *pdev) return -ENODEV; } + /* Set DMIC delay */ + ret = device_property_read_u32(&pdev->dev, "dmic-delay", + &rt5514_dmic_delay); + if (ret) { + rt5514_dmic_delay = 0; + dev_dbg(&pdev->dev, + "no optional property 'dmic-delay' found, default: no delay\n"); + } + rockchip_dailinks[DAILINK_RT5514_DSP].cpu_name = kstrdup_const(dev_name(dev), GFP_KERNEL); rockchip_dailinks[DAILINK_RT5514_DSP].cpu_dai_name = kstrdup_const(dev_name(dev), GFP_KERNEL); rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL); -- 1.7.9.5

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[alsa-devel] USB composite audio device problem with newer kernel versions
by Pecsi, Gergely 22 Sep '16

22 Sep '16

Hi all, We are having issues with a special multiple interface audio device. The card has 3 devices with the same VID/PID, 2 of them is a mono output device and the third one has an input and an output channel also. This third device does not function properly with newer kernel versions, the audio capture is not working. Tests: -our Debian Jessie based distro with kernel 3.16 -> OK -our Debian Jessie based distro with kernel 4.1.25 -> OK -our Debian Jessie based distro with kernel 4.4.1 -> NOK -our Debian Jessie based distro with kernel 4.6 -> NOK -our Debian Jessie based distro with kernel 4.7 -> NOK -Debian testing daily snapshot with kernel 4.6.0 -> NOK In the NOK cases, when we try to capture the stream with arecord, we get: "audio open error: No such file or directory". We can also see that in the NOK cases the arecord labels the capture device with device number 0, but in the OK cases it labels it with device number 2. Aplay labels these devices with device number 2 in any case. Arecord list output in the good case: card 5: Module [Spc Module], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 and in the bad case: card 5: Module [Spc Module], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 Relevant parts of alsa-info.sh output attached. What could be the root of the problem? What areas in the alsa codebase could be involved? Should we report an ALSA bug? Best regards, Gergely Pecsi

1 0

[alsa-devel] Oxygen Possible Bug in FCM
by xerofoify 22 Sep '16

22 Sep '16

Greetings All, Do anyone known of the best way to test this function, ac97_fp_rec_volume_put. Seems there may be a bug in it. I do have the hardware so that's not a issue just wondering what the different between fp_rec and the standard get functions are in terms of testing. Further more I am assuming this has to do with the fp headphones switch but I don't have headphones that are fp so is there another way to test this particular function without checking sound levels. Cheers, Nick

3 2

[alsa-devel] [PATCH] ALSA: snd-usb-line6 depends on CONFIG_SND_HWDEP
by Valdis Kletnieks 22 Sep '16

22 Sep '16

ALSA - snd-usb-line6 depends on CONFIG_SND_HWDEP ERROR: "snd_hwdep_new" [sound/usb/line6/snd-usb-line6.ko] undefined! scripts/Makefile.modpost:91: recipe for target '__modpost' failed make[1]: *** [__modpost] Error 1 Signed-off-by: Valdis Kletnieks <valdis.kletnieks(a)vt.edu> diff --git a/sound/usb/line6/Kconfig b/sound/usb/line6/Kconfig index 8ffcf48097d9..39b400392d71 100644 --- a/sound/usb/line6/Kconfig +++ b/sound/usb/line6/Kconfig @@ -2,6 +2,7 @@ config SND_USB_LINE6 tristate select SND_RAWMIDI select SND_PCM + select SND_HWDEP config SND_USB_POD tristate "Line 6 POD USB support"

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[alsa-devel] usb: Novation Twitch not working on Linux 4.7
by Juan Pedro Bolívar Puente 21 Sep '16

21 Sep '16

Hi! I am trying to use again my Novation Twitch after some time, and audio playback is not working anymore. I have tried with Jackd and various other programs, and it seems to always fail to install the hardware params. Note that before it fails, I do get to hear a little noise in the monitor output. Here is console the output of various programs. Note that, I am using Linux 4.7.0-1-amd64 as distributed with Debian Sid. Here is the output of some commands: $ ./alsacap *** Scanning for playback devices *** ... Card 1, ID `Twitch', name `Twitch' Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available) 4 channels, sampling rate 44100..48000 Hz Sample formats: S24_3LE Subdevice 0, name `subdevice #0' $ ./alsacap -d hw:Twitch *** Exploring configuration space of device `hw:Twitch' for playback *** 4 channels Sampling rate 44100..48000 Hz Sample formats: S24_3LE Significant bits: 24 $ aplay -D hw:Twitch -f S24_3LE -c 4 -r 44100 test.wav Playing raw data 'prueba.wav' : Signed 24 bit Little Endian in 3bytes, Rate 44100 Hz, Channels 4 aplay: set_params:1361: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_3LE SUBFORMAT: STD SAMPLE_BITS: 24 FRAME_BITS: 96 CHANNELS: 4 RATE: 44100 PERIOD_TIME: (125011 125012) PERIOD_SIZE: 5513 PERIOD_BYTES: 66156 PERIODS: (3 4) BUFFER_TIME: 500000 BUFFER_SIZE: 22050 BUFFER_BYTES: 264600 TICK_TIME: 0 $ mplayer -ao alsa:device=hw=Twitch test.mp3 MPlayer 1.3.0 (Debian), built with gcc-5.4.0 (C) 2000-2016 MPlayer Team Playing test.mp3. ... [AO_ALSA] Format s16le is not supported by hardware, trying default. [AO_ALSA] Unable to set format: Invalid argument Failed to initialize audio driver 'alsa:device=hw=Twitch' $ mplayer -ao alsa:device=plughw=Twitch test.mp3 MPlayer 1.3.0 (Debian), built with gcc-5.4.0 (C) 2000-2016 MPlayer Team Playing test.mp3. ... [AO_ALSA] alsa-lib: pcm_hw.c:603:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE failed (-32): Broken pipe [AO_ALSA] alsa-lib: pcm_direct.c:1054:(snd1_pcm_direct_initialize_slave) unable to install hw params [AO_ALSA] alsa-lib: pcm_dmix.c:1053:(snd_pcm_dmix_open) unable to initialize slave [AO_ALSA] Playback open error: Broken pipe Failed to initialize audio driver 'alsa:device=hw=Twitch' $ jackd -n twitch -d alsa -d hw:Twitch jackdmp 1.9.11 ... creating alsa driver ... hw:Twitch|hw:Twitch|1024|2|48000|0|0|nomon|swmeter|-|32bit Using ALSA driver USB-Audio running on card 1 - Novation Twitch at usb-0000:00:1d.0-1.2, full speed ALSA: Cannot open PCM device alsa_pcm for capture. Falling back to playback-only mode configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods ALSA: final selected sample format for playback: 24bit little-endian in 3bytes format ALSA: use 2 periods for playback ALSA: cannot set hardware parameters for playback ALSA: cannot configure playback channel Am I doing something wrong? Is it really broken? Thanks a lot! JP -- ∿∿∿∿ https://sinusoid.es ∿∿∿∿

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[alsa-devel] [alsa-lib][PATCH] ucm: docs: typeset lists of identifiers explicitly
by Antonio Ospite 21 Sep '16

21 Sep '16

Doxygen doesn't preserve formatting that relies only on indentation, typeset lists of identifiers explicitly. This makes the HTML docs a lot more readable. This change comes along the lines of commit 72aa0f8332fb ("ucm: reformat snd_use_case_get() doc"). Some TABs has been added too in order to preserve the aligned look when reading the source code. Signed-off-by: Antonio Ospite <ao2(a)ao2.it> --- include/use-case.h | 57 +++++++++++++++++++++++++++--------------------------- 1 file changed, 29 insertions(+), 28 deletions(-) diff --git a/include/use-case.h b/include/use-case.h index 9aac6e2..1ebd945 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -192,20 +192,21 @@ int snd_use_case_free_list(const char *list[], int items); * \return Number of list entries if success, otherwise a negative error code * * Defined identifiers: - * NULL - get card list - * (in pair cardname+comment) - * _verbs - get verb list - * (in pair verb+comment) - * _devices[/{verb}] - get list of supported devices - * (in pair device+comment) - * _modifiers[/{verb}]- get list of supported modifiers - * (in pair modifier+comment) - * TQ[/{verb}] - get list of TQ identifiers - * _enadevs - get list of enabled devices - * _enamods - get list of enabled modifiers + * - NULL - get card list + * (in pair cardname+comment) + * - _verbs - get verb list + * (in pair verb+comment) + * - _devices[/{verb}] - get list of supported devices + * (in pair device+comment) + * - _modifiers[/{verb}] - get list of supported modifiers + * (in pair modifier+comment) + * - TQ[/{verb}] - get list of TQ identifiers + * - _enadevs - get list of enabled devices + * - _enamods - get list of enabled modifiers + * + * - _supporteddevs/{modifier}|{device}[/{verb}] - list of supported devices + * - _conflictingdevs/{modifier}|{device}[/{verb}] - list of conflicting devices * - * _supporteddevs/{modifier}|{device}[/{verb}] - list of supported devices - * _conflictingdevs/{modifier}|{device}[/{verb}] - list of conflicting devices * Note that at most one of the supported/conflicting devs lists has * any entries, and when neither is present, all devices are supported. * @@ -331,8 +332,8 @@ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, * \return Zero if success, otherwise a negative error code * * Known identifiers: - * _devstatus/{device} - return status for given device - * _modstatus/{modifier} - return status for given modifier + * - _devstatus/{device} - return status for given device + * - _modstatus/{modifier} - return status for given modifier */ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, const char *identifier, @@ -346,19 +347,19 @@ int snd_use_case_geti(snd_use_case_mgr_t *uc_mgr, * \return Zero if success, otherwise a negative error code * * Known identifiers: - * _verb - set current verb = value - * _enadev - enable given device = value - * _disdev - disable given device = value - * _swdev/{old_device} - new_device = value - * - disable old_device and then enable new_device - * - if old_device is not enabled just return - * - check transmit sequence firstly - * _enamod - enable given modifier = value - * _dismod - disable given modifier = value - * _swmod/{old_modifier} - new_modifier = value - * - disable old_modifier and then enable new_modifier - * - if old_modifier is not enabled just return - * - check transmit sequence firstly + * - _verb - set current verb = value + * - _enadev - enable given device = value + * - _disdev - disable given device = value + * - _swdev/{old_device} - new_device = value + * - disable old_device and then enable new_device + * - if old_device is not enabled just return + * - check transmit sequence firstly + * - _enamod - enable given modifier = value + * = _dismod - disable given modifier = value + * - _swmod/{old_modifier} - new_modifier = value + * - disable old_modifier and then enable new_modifier + * - if old_modifier is not enabled just return + * - check transmit sequence firstly */ int snd_use_case_set(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 2.9.3

1 1

[alsa-devel] [PATCH -next] ASoC: sti: fix missing clk_disable_unprepare() on error in uni_player_start()
by Wei Yongjun 21 Sep '16

21 Sep '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Fix the missing clk_disable_unprepare() before return from uni_player_start() in the error handling case. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sti/uniperif_player.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 645e415..1bc8ebc 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -893,8 +893,10 @@ static int uni_player_start(struct uniperif *player) SET_UNIPERIF_SOFT_RST_SOFT_RST(player); ret = reset_player(player); - if (ret < 0) + if (ret < 0) { + clk_disable_unprepare(player->clk); return ret; + } /* * Does not use IEC61937 features of the uniperipheral hardware.

3 2

[alsa-devel] [PATCH -next] ASoC: rt5663: fix sparse warnings
by Wei Yongjun 21 Sep '16

21 Sep '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Fixes the following sparse warnings: sound/soc/codecs/rt5663.c:1367:14: warning: duplicate const sound/soc/codecs/rt5663.c:1577:5: warning: symbol 'rt5663_button_detect' was not declared. Should it be static? sound/soc/codecs/rt5663.c:2857:24: warning: symbol 'rt5663_aif_dai_ops' was not declared. Should it be static? sound/soc/codecs/rt5663.c:2866:27: warning: symbol 'rt5663_dai' was not declared. Should it be static? sound/soc/codecs/rt5663.c:3193:6: warning: symbol 'rt5663_i2c_shutdown' was not declared. Should it be static? sound/soc/codecs/rt5663.c:3200:19: warning: symbol 'rt5663_i2c_driver' was not declared. Should it be static? Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rt5663.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 7963a27..fa79568 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1364,7 +1364,7 @@ static const char * const rt5663_if1_adc_data_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static const SOC_ENUM_SINGLE_DECL(rt5663_if1_adc_enum, RT5663_TDM_2, +static SOC_ENUM_SINGLE_DECL(rt5663_if1_adc_enum, RT5663_TDM_2, RT5663_DATA_SWAP_ADCDAT1_SHIFT, rt5663_if1_adc_data_select); static void rt5663_enable_push_button_irq(struct snd_soc_codec *codec, @@ -1574,7 +1574,7 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) return rt5663->jack_type; } -int rt5663_button_detect(struct snd_soc_codec *codec) +static int rt5663_button_detect(struct snd_soc_codec *codec) { int btn_type, val; @@ -2854,7 +2854,7 @@ static int rt5663_resume(struct snd_soc_codec *codec) #define RT5663_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 rt5663_aif_dai_ops = { +static struct snd_soc_dai_ops rt5663_aif_dai_ops = { .hw_params = rt5663_hw_params, .set_fmt = rt5663_set_dai_fmt, .set_sysclk = rt5663_set_dai_sysclk, @@ -2863,7 +2863,7 @@ struct snd_soc_dai_ops rt5663_aif_dai_ops = { .set_bclk_ratio = rt5663_set_bclk_ratio, }; -struct snd_soc_dai_driver rt5663_dai[] = { +static struct snd_soc_dai_driver rt5663_dai[] = { { .name = "rt5663-aif", .id = RT5663_AIF, @@ -3190,14 +3190,14 @@ static int rt5663_i2c_remove(struct i2c_client *i2c) return 0; } -void rt5663_i2c_shutdown(struct i2c_client *client) +static void rt5663_i2c_shutdown(struct i2c_client *client) { struct rt5663_priv *rt5663 = i2c_get_clientdata(client); regmap_write(rt5663->regmap, RT5663_RESET, 0); } -struct i2c_driver rt5663_i2c_driver = { +static struct i2c_driver rt5663_i2c_driver = { .driver = { .name = "rt5663", .owner = THIS_MODULE,

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[alsa-devel] [snd_ali5451] snd_ali5451 0000:00:06.0: BUG: , pos = 16384, buffer size = 16384, period size = 1024
by Enrico Mioso 21 Sep '16

21 Sep '16

Hello guys. The message on the subject comes from my dmesg and is generated by the said module when using mplayer. (MPlayer SVN-r37379 (C) 2000-2015 MPlayer Team) Including some devices from my lspci: if you wish to see the integral one, let me know. 00:06.0 Multimedia audio controller: ULi Electronics Inc. M5451 PCI AC-Link Controller Audio Device (rev 02) Subsystem: Hewlett-Packard Company Device 002a Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR+ <PERR+ INTx- Latency: 64 (500ns min, 6000ns max) Interrupt: pin A routed to IRQ 5 Region 0: I/O ports at 1000 [size=256] Region 1: Memory at d0001000 (32-bit, non-prefetchable) [size=4K] Capabilities: [dc] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: snd_ali5451 00:07.0 ISA bridge: ULi Electronics Inc. M1533/M1535/M1543 PCI to ISA Bridge [Aladdin IV/V/V+] Subsystem: ULi Electronics Inc. ALi M1533 Aladdin IV/V ISA Bridge Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Capabilities: [a0] Power Management version 1 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- 00:08.0 Modem: ULi Electronics Inc. M5457 AC'97 Modem Controller (prog-if 00 [Generic]) Subsystem: Hewlett-Packard Company Device 002a Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin A routed to IRQ 10 Region 0: Memory at d0002000 (32-bit, non-prefetchable) [size=4K] Region 1: I/O ports at 1400 [size=256] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Thank you. P.S.: sorry for giving no follow-up to my previous hd-audio thing; thanks.

2 4

[alsa-devel] USB Gadget audio not working
by noman pouigt 21 Sep '16

21 Sep '16

Hello, I am trying to use my device as a sound card and enabled CONFIG_USB_AUDIO. However I don't know what kind of device node should be added in the machine file? I just want to record whatever is coming into the USB played back by the host over USB. I could get my device to be detected as a sound card i.e. USB_AUDIO_CLASS in my ubuntu machine. My understanding of USB audio is very less so I thought below change should be sufficient. Below are the only changes which i have done to enable USB_AUDIO. Machine file: { .name = "something_random", .stream_name = "USB Capture", .cpu_dai_name = "snd-soc-dummy-dai", .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .ignore_suspend = 1, }, drivers/usb/gadget/u_uac1.c : #define FILE_PCM_PLAYBACK "/dev/snd/pcmC0D0p" #define FILE_PCM_CAPTURE "/dev/snd/pcmC0D0c" in defconfig CONFIG_USB_AUDIO=y Please let me know if I am doing anything wrong. Thanks,

2 1

[alsa-devel] [PATCH v5] drm/i915/dp: DP audio API changes for MST
by Dhinakaran Pandiyan 20 Sep '16

20 Sep '16

From: "Pandiyan, Dhinakaran" <dhinakaran.pandiyan(a)intel.com> DP MST provides the capability to send multiple video and audio streams through a single port. This requires the API's between i915 and audio drivers to distinguish between multiple audio capable displays that can be connected to a port. Currently only the port identity is shared in the APIs. This patch adds support for MST with an additional parameter 'int pipe'. The existing parameter 'port' does not change it's meaning. pipe = MST : display pipe that the stream originates from Non-MST : -1 Affected APIs: struct i915_audio_component_ops - int (*sync_audio_rate)(struct device *, int port, int rate); + int (*sync_audio_rate)(struct device *, int port, int pipe, + int rate); - int (*get_eld)(struct device *, int port, bool *enabled, - unsigned char *buf, int max_bytes); + int (*get_eld)(struct device *, int port, int pipe, + bool *enabled, unsigned char *buf, int max_bytes); struct i915_audio_component_audio_ops - void (*pin_eld_notify)(void *audio_ptr, int port); + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); This patch makes dummy changes in the audio drivers (thanks Libin) for build to succeed. The audio side drivers will send the right 'pipe' values for MST in patches that will follow. v2: Renamed the new API parameter from 'dev_id' to 'pipe'. (Jim, Ville) Included Asoc driver API compatibility changes from Jeeja. Added WARN_ON() for invalid pipe in get_saved_encoder(). (Takashi) Added comment for av_enc_map[] definition. (Takashi) v3: Fixed logic error introduced while renaming 'dev_id' as 'pipe' (Ville) Renamed get_saved_encoder() to get_saved_enc() to reduce line length v4: Rebased. Parameter check for pipe < -1 values in get_saved_enc() (Ville) Switched to for_each_pipe() in get_saved_enc() (Ville) Renamed 'pipe' to 'dev_id' in audio side code (Takashi) v5: Included a comment for the dev_id arg. (Libin) Signed-off-by: Dhinakaran Pandiyan <dhinakaran.pandiyan(a)intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- drivers/gpu/drm/i915/i915_drv.h | 3 +- drivers/gpu/drm/i915/intel_audio.c | 94 ++++++++++++++++++++++++++------------ include/drm/i915_component.h | 6 +-- include/sound/hda_i915.h | 11 +++-- sound/hda/hdac_i915.c | 18 +++++--- sound/pci/hda/patch_hdmi.c | 7 +-- sound/soc/codecs/hdac_hdmi.c | 2 +- 7 files changed, 94 insertions(+), 47 deletions(-) diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h index 270543c..fe14cca 100644 --- a/drivers/gpu/drm/i915/i915_drv.h +++ b/drivers/gpu/drm/i915/i915_drv.h @@ -2080,7 +2080,8 @@ struct drm_i915_private { /* perform PHY state sanity checks? */ bool chv_phy_assert[2]; - struct intel_encoder *dig_port_map[I915_MAX_PORTS]; + /* Used to save the pipe-to-encoder mapping for audio */ + struct intel_encoder *av_enc_map[I915_MAX_PIPES]; /* * NOTE: This is the dri1/ums dungeon, don't add stuff here. Your patch diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index 40fbdd8..9583f43 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -491,6 +491,7 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) struct drm_i915_private *dev_priv = to_i915(encoder->dev); struct i915_audio_component *acomp = dev_priv->audio_component; enum port port = intel_encoder->port; + enum pipe pipe = crtc->pipe; connector = drm_select_eld(encoder); if (!connector) @@ -515,12 +516,18 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) mutex_lock(&dev_priv->av_mutex); intel_encoder->audio_connector = connector; + /* referred in audio callbacks */ - dev_priv->dig_port_map[port] = intel_encoder; + dev_priv->av_enc_map[pipe] = intel_encoder; mutex_unlock(&dev_priv->av_mutex); + /* audio drivers expect pipe = -1 to indicate Non-MST cases */ + if (intel_encoder->type != INTEL_OUTPUT_DP_MST) + pipe = -1; + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + (int) port, (int) pipe); } /** @@ -536,17 +543,24 @@ void intel_audio_codec_disable(struct intel_encoder *intel_encoder) struct drm_i915_private *dev_priv = to_i915(encoder->dev); struct i915_audio_component *acomp = dev_priv->audio_component; enum port port = intel_encoder->port; + struct intel_crtc *crtc = to_intel_crtc(encoder->crtc); + enum pipe pipe = crtc->pipe; if (dev_priv->display.audio_codec_disable) dev_priv->display.audio_codec_disable(intel_encoder); mutex_lock(&dev_priv->av_mutex); intel_encoder->audio_connector = NULL; - dev_priv->dig_port_map[port] = NULL; + dev_priv->av_enc_map[pipe] = NULL; mutex_unlock(&dev_priv->av_mutex); + /* audio drivers expect pipe = -1 to indicate Non-MST cases */ + if (intel_encoder->type != INTEL_OUTPUT_DP_MST) + pipe = -1; + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + (int) port, (int) pipe); } /** @@ -621,15 +635,40 @@ static int i915_audio_component_get_cdclk_freq(struct device *kdev) return dev_priv->cdclk_freq; } -static int i915_audio_component_sync_audio_rate(struct device *kdev, - int port, int rate) +static struct intel_encoder *get_saved_enc(struct drm_i915_private *dev_priv, + int port, int pipe) +{ + + if (WARN_ON(pipe >= I915_MAX_PIPES)) + return NULL; + + /* MST */ + if (pipe >= 0) + return dev_priv->av_enc_map[pipe]; + + /* Non-MST */ + for_each_pipe(dev_priv, pipe) { + struct intel_encoder *encoder; + + encoder = dev_priv->av_enc_map[pipe]; + if (encoder == NULL) + continue; + + if (port == encoder->port) + return encoder; + } + + return NULL; +} + +static int i915_audio_component_sync_audio_rate(struct device *kdev, int port, + int pipe, int rate) { struct drm_i915_private *dev_priv = kdev_to_i915(kdev); struct intel_encoder *intel_encoder; struct intel_crtc *crtc; struct drm_display_mode *mode; struct i915_audio_component *acomp = dev_priv->audio_component; - enum pipe pipe = INVALID_PIPE; u32 tmp; int n; int err = 0; @@ -643,25 +682,20 @@ static int i915_audio_component_sync_audio_rate(struct device *kdev, i915_audio_component_get_power(kdev); mutex_lock(&dev_priv->av_mutex); + /* 1. get the pipe */ - intel_encoder = dev_priv->dig_port_map[port]; - /* intel_encoder might be NULL for DP MST */ + intel_encoder = get_saved_enc(dev_priv, port, pipe); if (!intel_encoder || !intel_encoder->base.crtc || intel_encoder->type != INTEL_OUTPUT_HDMI) { - DRM_DEBUG_KMS("no valid port %c\n", port_name(port)); + DRM_DEBUG_KMS("Not valid for port %c\n", port_name(port)); err = -ENODEV; goto unlock; } + + /* pipe passed from the audio driver will be -1 for Non-MST case */ crtc = to_intel_crtc(intel_encoder->base.crtc); pipe = crtc->pipe; - if (pipe == INVALID_PIPE) { - DRM_DEBUG_KMS("no pipe for the port %c\n", port_name(port)); - err = -ENODEV; - goto unlock; - } - DRM_DEBUG_KMS("pipe %c connects port %c\n", - pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; /* port must be valid now, otherwise the pipe will be invalid */ @@ -697,7 +731,7 @@ static int i915_audio_component_sync_audio_rate(struct device *kdev, } static int i915_audio_component_get_eld(struct device *kdev, int port, - bool *enabled, + int pipe, bool *enabled, unsigned char *buf, int max_bytes) { struct drm_i915_private *dev_priv = kdev_to_i915(kdev); @@ -706,16 +740,20 @@ static int i915_audio_component_get_eld(struct device *kdev, int port, int ret = -EINVAL; mutex_lock(&dev_priv->av_mutex); - intel_encoder = dev_priv->dig_port_map[port]; - /* intel_encoder might be NULL for DP MST */ - if (intel_encoder) { - ret = 0; - *enabled = intel_encoder->audio_connector != NULL; - if (*enabled) { - eld = intel_encoder->audio_connector->eld; - ret = drm_eld_size(eld); - memcpy(buf, eld, min(max_bytes, ret)); - } + + intel_encoder = get_saved_enc(dev_priv, port, pipe); + if (!intel_encoder) { + DRM_DEBUG_KMS("Not valid for port %c\n", port_name(port)); + mutex_unlock(&dev_priv->av_mutex); + return ret; + } + + ret = 0; + *enabled = intel_encoder->audio_connector != NULL; + if (*enabled) { + eld = intel_encoder->audio_connector->eld; + ret = drm_eld_size(eld); + memcpy(buf, eld, min(max_bytes, ret)); } mutex_unlock(&dev_priv->av_mutex); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index b46fa0e..545c6e0 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -64,7 +64,7 @@ struct i915_audio_component_ops { * Called from audio driver. After audio driver sets the * sample rate, it will call this function to set n/cts */ - int (*sync_audio_rate)(struct device *, int port, int rate); + int (*sync_audio_rate)(struct device *, int port, int pipe, int rate); /** * @get_eld: fill the audio state and ELD bytes for the given port * @@ -77,7 +77,7 @@ struct i915_audio_component_ops { * Note that the returned size may be over @max_bytes. Then it * implies that only a part of ELD has been copied to the buffer. */ - int (*get_eld)(struct device *, int port, bool *enabled, + int (*get_eld)(struct device *, int port, int pipe, bool *enabled, unsigned char *buf, int max_bytes); }; @@ -97,7 +97,7 @@ struct i915_audio_component_audio_ops { * status accordingly (even when the HDA controller is in power save * mode). */ - void (*pin_eld_notify)(void *audio_ptr, int port); + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); }; /** diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index 796cabf..5ab972e 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -10,8 +10,9 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); void snd_hdac_i915_set_bclk(struct hdac_bus *bus); -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate); -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate); +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); @@ -29,13 +30,13 @@ static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { } static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, - hda_nid_t nid, int rate) + hda_nid_t nid, int dev_id, int rate) { return 0; } static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, - bool *audio_enabled, char *buffer, - int max_bytes) + int dev_id, bool *audio_enabled, + char *buffer, int max_bytes) { return -ENODEV; } diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index c9af022..0659bf3 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -193,6 +193,7 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate * @codec: HDA codec * @nid: the pin widget NID + * @dev_id: device identifier * @rate: the sample rate to set * * This function is supposed to be used only by a HD-audio controller @@ -201,18 +202,20 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) * This function sets N/CTS value based on the given sample rate. * Returns zero for success, or a negative error code. */ -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate) { struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; - int port; + int port, pipe; if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) return -ENODEV; port = pin2port(codec, nid); if (port < 0) return -EINVAL; - return acomp->ops->sync_audio_rate(acomp->dev, port, rate); + pipe = dev_id; + return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); } EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); @@ -220,6 +223,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * snd_hdac_acomp_get_eld - Get the audio state and ELD via component * @codec: HDA codec * @nid: the pin widget NID + * @dev_id: device identifier * @audio_enabled: the pointer to store the current audio state * @buffer: the buffer pointer to store ELD bytes * @max_bytes: the max bytes to be stored on @buffer @@ -236,12 +240,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * thus it may be over @max_bytes. If it's over @max_bytes, it implies * that only a part of ELD bytes have been fetched. */ -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, bool *audio_enabled, char *buffer, int max_bytes) { struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; - int port; + int port, pipe; if (!acomp || !acomp->ops || !acomp->ops->get_eld) return -ENODEV; @@ -249,7 +253,9 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, port = pin2port(codec, nid); if (port < 0) return -EINVAL; - return acomp->ops->get_eld(acomp->dev, port, audio_enabled, + + pipe = dev_id; + return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, buffer, max_bytes); } EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 56e5204..cf9bc042 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1485,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, -1, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); if (size > 0) { @@ -1744,7 +1744,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ if (codec_has_acomp(codec)) - snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); + snd_hdac_sync_audio_rate(&codec->core, pin_nid, -1, + runtime->rate); non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -2290,7 +2291,7 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state); } -static void intel_pin_eld_notify(void *audio_ptr, int port) +static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) { struct hda_codec *codec = audio_ptr; int pin_nid; diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e181b2..dc0129b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1368,7 +1368,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, return hdac_hdmi_init_dai_map(edev); } -static void hdac_hdmi_eld_notify_cb(void *aptr, int port) +static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) { struct hdac_ext_device *edev = aptr; struct hdac_hdmi_priv *hdmi = edev->private_data; -- 2.5.0

1 0

[alsa-devel] Dell XPS One 2710 HDA Intel PCH - No S/PDIF Output
by Marve 20 Sep '16

20 Sep '16

Hi all, I've been trying stuff myself and got help from someone (debianuser) at #alsa IRC, but without success! I got the advice to explain the issue in this mailing list! I'm running Linux Mint 18 on a all-in-one Dell XPS One 2710. Most stuff works, except sound. The former Mint version also sound from the speakers worked out of the box. Now, I have to switch from the Pulse Audio volume control from "Speakers" to "Headphones (unplugged)" every time I start up to get audio from the internal speakers, which is working well then, annoying, is probably an issue that belongs in another group and not the reason I'm writing you! :) The XPS One 2710 has a S/PDIF Toslink (optical) connection. Back in the days that Microsoft wasn't trying to shove up Windows 10 to everyone, I successfully could get output from this connection to my JVC Receiver from Windows7 of the box. I find it hard to have it configured under Linux. Partly, with help from a anonymous debianuser on #alsa IRC I checked with alsamixer that the S/PDIF was not muted. Also, with the Jack Retasking for HDA tool, I enforced Pin ID: 0x1e is a SPDIF Out connection. After this, the Pulse Audio Tools shows several new interesting Built-in Audio options and I think "Digital Stereo (IEC958) Outout looks most promising. So I select this one and with Output Devices I select "Digital Output (S/PDIF). Then, when I play something on the box, the volume meters actually show that there is something playing, but I can't hear anything from the JVC receiver! The receiver supports PCM to DTS, but no sound is coming out of it! Thanks if you could help me out! I am happy to help you when you need more information or want we to test something. I had this problem on the previous Linux Mint LTS version and tried to install a newer kernel and alsa driver, which didn't work - so I decided to wait for the current LTS. I made a fresh info: http://www.alsa-project.org/db/?f=556ae2bdae66341579831231ef2d90ca0d7c8e24 If you need one with other settings, please let me know! Best wishes, M.

1 3

[alsa-devel] (no subject)
by xerofoify 20 Sep '16

20 Sep '16

1 0

[alsa-devel] DAPM over two regmaps (and a mailbox)
by Maxime Ripard 20 Sep '16

20 Sep '16

Hi, We've been working recently on a later SoC (Allwinner A33) with a codec directly embedded into it, just like sound/soc/sunxi/sun4i-codec.c, but this time using the usual i2s controller we had, instead of a custom DAI. That codec is mapped in memory, however, we have a bunch of DAPM widgets that are mapped in a separate register space, that should probably be exposed through a syscon (but isn't yet). Those are apparently used to control the analog part of the codec, including powering up the DAC, so it really feels like they should be part of DAPM. However, since we will obviously have a regmap for the main register space of the codec, that leaves us with two regmaps that we need to use, depending on the register we want to set, and DAPM doesn't really seem to be able to handle that. To make things worse, the register in the syscon behaves as a mailbox, where you actually have to set in that register the address you want to modify and the new value, in a single write. This also seem to deviate from the usual DAPM access pattern. I'm not really sure how to handle that properly. For now, we just did those writes outside of DAPM, in the startup, shutdown and prepare shutdowns. We could also have a meta-regmap, that would have custom write and read functions, and depending on the register would turn to our syscon, or do a writel. Or we could try to make DAPM able to use different regmaps depending on the register, but that seem do be very intrusive. Do you have any suggestions or preferences on how to implement this properly? Thanks, Maxime -- Maxime Ripard, Free Electrons Embedded Linux and Kernel engineering http://free-electrons.com

3 11

[alsa-devel] [PATCH] topology: Add support for included files
by mengdong.lin＠linux.intel.com 20 Sep '16

20 Sep '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this file, but only an enhancement for topology user space library. It will let a text configuration file include other files, so that some common cofigurations can be shared and we can reduce the total sizes of files. Mengdong Lin (1): topology: Add support for included files include/topology.h | 37 +++++ src/topology/parser.c | 374 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/tplg_local.h | 17 +++ 3 files changed, 423 insertions(+), 5 deletions(-) -- 2.5.0

3 3

[alsa-devel] Tascam US-366 and Alsa Firmware
by Max 20 Sep '16

20 Sep '16

I have this device too and was wondering if OP or anyone else ever got it running and how. Thanks. > Hi, > > I recently bought the Tascam US-366 and forgot to check for Linux support, > which it doesn't have (yet! :). Looking through the Alsa firmware sources, > there seems to be a firmware loader called usx2yloader which supports a > number of Tascam cards, but the US-366 is not listed. It looks like really > usx2yloader just needs the firmware and a `prepad` file. I`ve found the > firmware for the card (I think) buried in the OSX firmware download > package: > > > http://tascam.com/content/downloads/products/789/us-322_us-366_fw_v103_mave… > > Now I think I just need the prepad file, but there's little documentation > regarding exactly what that is (I'm assuming a number of 0s just to pad the > firmware image to alignment, but I'm not sure of what alignment). > > Any ideas if usx2yloader will work at all with the US-366? Does anyone > know what the pre-pad file should be/contain? Has anyone tried to get the > US-366 working yet and failed? > > Reference: > > http://tascam.com/product/us-366/ > > Thanks. > > Burton Samograd

1 0

[alsa-devel] [sound:for-next 48/48] sound/usb/line6/driver.c:484:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values.
by kbuild test robot 20 Sep '16

20 Sep '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: a16039cbf1a1ee7e76d7d100f9e613998919ab91 commit: a16039cbf1a1ee7e76d7d100f9e613998919ab91 [48/48] ALSA: line6: Add hwdep interface to access the POD control messages coccinelle warnings: (new ones prefixed by >>) >> sound/usb/line6/driver.c:484:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH 0/4][alsa-lib] ctl: sync UAPI header for TLV-related macros and apply misc changes
by Takashi Sakamoto 20 Sep '16

20 Sep '16

Hi, A few days ago, UAPI header of 4.9-pre kernel newly include existent macros related to TLV operation for user land. Then, applications are going to know layout of TLV packet payload in the header. This patchset is for related changes in ALSA user land library. The macros are put into 'include/sound/tlv.h', however they're not put into 'include/control.h' yet, because currently I have no better idea for library APIs to parse threshold level information. Takashi Sakamoto (4): Update include/sound/tlv.h from 4.9-pre kernel uapi test: use proper threshold level information for user-ctl-element-set test ctl: improve API documentation for TLV operation ctl: improve documentation about TLV-related APIs include/sound/tlv.h | 80 +++++++++++++++++++++++++++++++++++++++++++++ src/control/control.c | 21 ++++++++++-- test/user-ctl-element-set.c | 41 +++++++++++++++-------- 3 files changed, 125 insertions(+), 17 deletions(-) -- 2.7.4

2 6

[alsa-devel] [PATCH v15 0/5] Rockchip Type-C DisplayPort driver
by Chris Zhong 20 Sep '16

20 Sep '16

Hi all This series patch is for rockchip Type-C DisplayPort controller driver. The USB Type-C PHY is designed to support the USB3 and DP applications. The PHY basically has two main components: USB3 and DisplyPort. USB3 operates in SuperSpeed mode and the DP can operate at RBR, HBR and HBR2 data rates. The Type-C cable orientation detection and Power Delivery (PD) is accomplished using a PD PHY or a exernal PD chip. The DP controller is compliant with DisplayPort Specification, Version 1.3, This IP is compatible with the rockchip type-c PHY IP. There is a uCPU in DP controller, it need a firmware to work, please put the firmware file[0] rockchip/dptx.bin to /lib/firmware/rockchip/dptx.bin. The uCPU in charge of aux communication and link training, the host use mailbox to communicate with the ucpu. The DP contoller has register a notification with extcon API, to get the alt mode from PD, the PD driver need call the devm_extcon_dev_allocate to create a extcon device and use extcon_set_state to notify DP controller. And call extcon_set_cable_property to set orientation. About the DP audio, cdn-dp registered 2 DAIs: 0 is I2S, 1 is SPDIF. We can reference them in rk3399-gru machine driver. It base on the hdmi-notify patches[1], they are both in reviewing. This series is based on Mark Yao's branch[2], the extcon API backport from linux-next[3]. I test this patches on the rk3399-evb board, with a fusb302 driver, this branch has no rk3399.dtsi, so the patch about dts is not included in this series. >From V9, the Type-C PHY is split into two PHYs: DP and USB3. The PHY will be init, no matter which PHY be power_on. The DP module will enter A2 mode (standby mode) after phy_init, if DP PHY is powered on, the DP module will enter to A0 mode(running mode). Then if DP PHY is powered off, DP module will back to A2 mode. If everything is un-plugged, phy will be deinit. [0] kernel/git/firmware/linux-firmware.git [1] https://patchwork.kernel.org/patch/8887261/ https://patchwork.kernel.org/patch/8887251/ [2] https://github.com/markyzq/kernel-drm-rockchip/tree/drm-rockchip-next-2016-… [3] git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master Changes in v15: Chris Zhong's changes: - fix disable phy unbalance - support Apple Dock hot-plug in HDMI port - retraining when Apple Dock hpd irq coming Sean Paul's changes: - Restructured the worker - Synchronized between worker & drm hooks - Properly implemented enable/disable - Added error checking - Use devm variant of extcon - Use DRM_DEV_* logging - Refactored code to (hopefully) be more clear Changes in v14: - Modify some grammatical errors - remove the mutex around cdn_dp_audio_get_eld - power on the power domain after clk_enable - retry to read edid - change super speed property name to EXTCON_PROP_USB_SS - do a correct mode_valid check when bpc is 0 Changes in v13: - add dptx and apb reset - support suspend/resume - switch power domain dynamically - re-training when hpd signal is triggered Changes in v12: - use EXTCON_PROP_USB_SUPERSPEED to replace EXTCON_USB_HOST Changes in v11: - refer dp phy - add best_encoder back, since it required by drm_atomic_helper_check Changes in v10: - add pclk_vio_grf clock - remove best_encoder ops - support read sink count from DPCD - control the grf_clk in DP Changes in v9: - modify the reference phy = <&tcphy0 0>, <&tcphy1 0>; - do not need reset the phy before power_on - add a orientation information for set_capability - retry to read dpcd in 10 seconds Changes in v8: - optimization the err log Changes in v7: - support firmware standby when no dptx connection - optimization the calculation of tu size and valid symbol Changes in v6: - add assigned-clocks and assigned-clock-rates - add power-domains - add a port struct - select SND_SOC_HDMI_CODEC - force reset the phy when hpd detected Changes in v5: - alphabetical order - do not use long, use u32 or u64 - return MODE_CLOCK_HIGH when requested > actual - Optimized Coding Style - add a formula to get better tu size and symbol value. - modify according to Sean Paul's comments - fixed the fw_wait always 0 Changes in v4: - add a reset node - support 2 phys - use phy framework to control DP phy - support 2 phys Changes in v3: - add SoC specific compatible string - remove reg = <1>; - use EXTCON_DISP_DP and EXTCON_DISP_DP_ALT cable to get dp port state. - reset spdif before config it - modify the firmware clk to 100Mhz - retry load firmware if fw file is requested too early - change spdif to i2s2 Changes in v2: - Alphabetic order - remove excess error message - use define clk_rate - check all return value - remove dev_set_name(dp->dev, "cdn-dp"); - use schedule_delayed_work - remove never-called functions - remove some unnecessary () - correct the commit message Changes in v1: - add extcon node description - add #sound-dai-cells description - use extcon API - use hdmi-codec for the DP Asoc - do not initialize the "ret" - printk a err log when drm_of_encoder_active_endpoint_id - modify the dclk pin_pol to a single line Chris Zhong (5): extcon: Introduce EXTCON_PROP_DISP_HPD property Documentation: bindings: add dt documentation for cdn DP controller drm/rockchip: cdn-dp: add cdn DP support for rk3399 drm/rockchip: cdn-dp: support audio hot-plug ASoC: rockchip: Add DP dai-links to the rk3399-gru machine driver .../bindings/display/rockchip/cdn-dp-rockchip.txt | 75 ++ .../bindings/sound/rockchip,rk3399-gru-sound.txt | 11 +- drivers/gpu/drm/rockchip/Kconfig | 10 + drivers/gpu/drm/rockchip/Makefile | 1 + drivers/gpu/drm/rockchip/cdn-dp-core.c | 1165 ++++++++++++++++++++ drivers/gpu/drm/rockchip/cdn-dp-core.h | 108 ++ drivers/gpu/drm/rockchip/cdn-dp-reg.c | 956 ++++++++++++++++ drivers/gpu/drm/rockchip/cdn-dp-reg.h | 482 ++++++++ drivers/gpu/drm/rockchip/rockchip_drm_vop.c | 13 +- drivers/gpu/drm/rockchip/rockchip_drm_vop.h | 9 + drivers/gpu/drm/rockchip/rockchip_vop_reg.c | 2 + include/linux/extcon.h | 14 +- sound/soc/rockchip/rk3399_gru_sound.c | 93 ++ 13 files changed, 2930 insertions(+), 9 deletions(-) create mode 100644 Documentation/devicetree/bindings/display/rockchip/cdn-dp-rockchip.txt create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-core.c create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-core.h create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-reg.c create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-reg.h -- 1.9.1

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[alsa-devel] [PATCH 00/15] Line6 POD X3/X3Live suport
by Andrej Krutak 20 Sep '16

20 Sep '16

Hello all, attached is the first version of the driver, based on POD HD. They share a lot of similarities, but at the same time there were some generalizations needed. I think POD HD could follow up on these patches to add hwdep support, perhaps helix too... I tried to not introduce regressions to the old HW. Anyhow, this is probably not the final version of the patches, as it is my first submission to upstream - I'm sure there will be some issues. Especially the patch "Use device_create_file instead of snd_card_add_dev_attr" I have to revisit, it may not be necessarry in the end. Other than that, the patches mostly pass `make checkpatches`. I've been testing the driver for a while now, including lock debugging options etc., and there don't seem to be functional problems. There's one missing thing - the driver uses bulk USB interface of the device, but so far I wasn't able to make the usb_driver_claim_interface() work... I hope this can be added later, if someone (or I) finds time. Thanks for your inputs, greetings, Andrej Krutak (15): ALSA: line6: Make driver configuration more generic. ALSA: line6: Add LINE6_CAP_IN_NEEDS_OUT, a void playback stream during capture ALSA: line6: Distinguish device init (ctrl EP) and MIDI data transfer (int EP) ALSA: line6: Add support for POD X3 ALSA: line6: Use device_create_file instead of snd_card_add_dev_attr ALSA: line6: Allow bulk endpoints instead of interrupt endpoints ALSA: line6: Allow processing of raw incoming messages ALSA: line6: Cleanup initialization ALSA: line6: Add hwdep interface to access the POD control messages ALSA: line6: Add proper locks for hwdep open/release/read ALSA: line6: Only free buffer if it is set. ALSA: line6: Give up on the lock while URBs are released. ALSA: line6: Add support for POD X3 Live (only USB ID differs from POD X3) ALSA: line6: Give up hwdep spinlock temporarily during read operation ALSA: line6: Remove double line6_pcm_release() after failed acquire. include/uapi/sound/asound.h | 3 +- sound/usb/line6/Kconfig | 4 +- sound/usb/line6/capture.c | 49 +++++-- sound/usb/line6/driver.c | 303 ++++++++++++++++++++++++++++++++++------ sound/usb/line6/driver.h | 71 +++++++--- sound/usb/line6/midi.c | 2 +- sound/usb/line6/pcm.c | 83 +++++++---- sound/usb/line6/pcm.h | 19 +-- sound/usb/line6/playback.c | 37 +++-- sound/usb/line6/pod.c | 12 +- sound/usb/line6/podhd.c | 328 +++++++++++++++++++++++++++++++++++++++++--- sound/usb/line6/toneport.c | 6 +- sound/usb/line6/variax.c | 6 +- 13 files changed, 767 insertions(+), 156 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: samsung: make audio interface/controller explicitly
by Randy Li 19 Sep '16

19 Sep '16

It is simple sound card time, we could assign different codec to a interface without making a specific driver for it. The SPDIF and I2S interface for Samsung would be possible used by simple-sound-card, but not sure about the PCM. Signed-off-by: Randy Li <ayaka(a)soulik.info> --- sound/soc/samsung/Kconfig | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7b722b0..210ce38 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,3 +1,7 @@ +menu "SoC Audio for Samsung CPUs" + +comment "Common SoC Audio options for Samsung CPUs:" + config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on (PLAT_SAMSUNG || ARCH_EXYNOS) @@ -18,18 +22,22 @@ config SND_S3C2412_SOC_I2S select SND_S3C_I2SV2_SOC config SND_SAMSUNG_PCM - tristate + tristate "Samsung PCM interface support" + depends on SND_SOC_SAMSUNG config SND_SAMSUNG_AC97 tristate select SND_SOC_AC97_BUS + depends on SND_SOC_SAMSUNG config SND_SAMSUNG_SPDIF - tristate + tristate "Samsung SPDIF transmitter support" + depends on SND_SOC_SAMSUNG select SND_SOC_SPDIF config SND_SAMSUNG_I2S - tristate + tristate "Samsung I2S interface support" + depends on I2C && SND_SOC_SAMSUNG config SND_SOC_SAMSUNG_NEO1973_WM8753 tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)" @@ -229,3 +237,5 @@ config SND_SOC_ARNDALE_RT5631_ALC5631 depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631 + +endmenu -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: samsung: make audio interface/controller explicitly
by Krzysztof Kozlowski 19 Sep '16

19 Sep '16

On Mon, Sep 19, 2016 at 09:21:54AM +0800, Ayaka wrote: > > > 從我的 iPad 傳送 > > > Krzysztof Kozlowski <krzk(a)kernel.org> 於 2016年9月19日上午2:09 寫道： > > > >> On Sun, Sep 18, 2016 at 11:12:34PM +0800, ayaka wrote: > >> > >> > >>> On 09/18/2016 10:42 PM, Krzysztof Kozlowski wrote: > >>>> On Sun, Sep 18, 2016 at 10:09:11PM +0800, Randy Li wrote: > >>>> It is simple sound card time, we could assign different codec > >>>> to a interface without making a specific driver for it. > >>> The description does not convince me and I do not see an example using > >>> this. Could you provide one? > >> Sorry, the board TOPEET iTop 4412 for exynos 4412 I posted supported codec > >> with I2S interface using the simple sound card. Anyway, it is no harm to > >> make them explicitly right? > > > > kbuild gave you the answer... > Not sure how comes, even time I sent patches to you, I at lease build and run it once. Building only one ARM config may be sufficient for trivial changes but for more complex stuff (especially when dealing with Kconfig) needs more build coverage. On Ubuntu you can easily cross compile for x86, x86_64, arm, arm64 and powerpc. You can also use this cross-compiler collection: https://www.kernel.org/pub/tools/crosstool > > > >> Or I have to enabled those codec support for > >> SMDK, which is not needed for the other board. > > > > If I understand correctly, the i2s/pcm etc are still needed but not > > built in config choosing only simple-audio-card? I tried now such > > configuration on Odroid XU and indeed the audio is missing. > > > > The patch looks like needed but: > > 1. You need to describe the rationale in commit message, why it is > > needed. > Sorry about English. > > 2. You need to fix it... kbuild pointed build issues. > I would check that. > > > > Other solution would be to add a user-selectable option for generic > > sound on Samsung using simple audio card. The option would then select > > appropriate SND_SAMSUNG* options, just like specific drivers do. I see > > that sh does like this. Personally this approach seems simpler to me - > > the defconfig could just choose this generic sound instead of many > > SND_SAMSUNG_* sub-options. > I would just what Freescale did. It included those options entries in a sub-menu. > I don't those options should be bound either, as a board may only use one of interface or controller(like TOPEET iTOP would only use i2s. No place for SPDIF, AC97 nor PCM.) OK, sounds good for me. Thanks for explanation. Best regards, Krzysztof

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[alsa-devel] Fw: kernel panic with ALSA driver
by Pavan Krishna Guntooru 19 Sep '16

19 Sep '16

Dear All, Please find the below email and suggest us. Thanks, -Pavan G ________________________________ From: Pavan Krishna Guntooru Sent: Monday, September 19, 2016 5:40 PM To: clemens(a)ladisch.de; lrg(a)slimlogic.co.uk; broonie(a)opensource.wolfsonmicro.com Subject: kernel panic with ALSA driver Dear all, We are using ti-sdk of version 8 and linux kernel version of 3.0.14. We have ported ti-sdk-8 to our Linux embedded box. Some times kernel Panic is happening and by decoding the generated kernel panic , crash happening in kernel at place linux-3.14.26-g2489c02/sound/core/pcm_dmaengine.c:134. the issue is in PCM engine driver. After reading the some articles, people saying that there is a kernel bug and so happening. My question here are 1) If its a known kernel issue, ALSA developers are aware of this bug? 2) Even though it is a kernel issue, is there a way to avoid from application in using of ALSA API's? 3) Do you have any fix for this? 4) If it is kernel issue, so many people could have reported by this time. For every audio announcement playing, we are opening and closing the audio device in our application. Please kindly suggest and awaiting for your reply. If I sent to a wrong person, please forward to the concern person. Thanks, -Pavan G

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[alsa-devel] [PATCH v5 0/5] ASoC: Add support to Qualcomm msm8916-wcd multi codec
by Srinivas Kandagatla 19 Sep '16

19 Sep '16

This patchset aims at adding msm8916-wcd multi codec support. msm8916-wcd codec is found in Qualcomm msm8916 and apq8016 processors. This codec IP is split in to two parts(Digital & Analog), Analog part is integrated in to PMIC PM8916 and the digital part is integrated into Application processor. Register access to the analog part is done via SPMI interface to PMIC, and registers on the Application processor are memory mapped. Data transfer between Analog and Digital Die is done via a internal bus called PDM. Most of this driver is ported from downstream Andriod v3.10 kernel. Jack support and Headset button detection code is not added in to this series. First two patches add digital and analog codec support and the follow on patches add routes and multicodec support to the sound card and finally dts patch for board. This codec support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary Mic. - 2 Digital Microphones. - 2 Mic Bias Circuits. - Earpiece - Headset - Loud Speaker. - Jack Detect. - Headset Button detection circuit. Tested this code on v4.8-rc3 with for Headset (Mic and speakers), Secondary Mic and Loudspeaker on DB410C. DMIC should work too. Thanks, srini Changes since v4: - Fixed various C-style related issues reported by Mark - Added shutdown call to clear things setup in startup, spotted by Mark. - Rearranged code to make it more readble suggested by Mark. - Converted dmic clk to a dapm supply spotted by Mark. - Added example in dt bindings to show support of multicodec. - Removed two patches from this list as they are already applied in sound tree. Changes since v3: - Split the codec to 2 seperate codecs digital and analog, suggested by Mark and Kenneth. - Addressed various suggestions from Mark and Kenneth. Srinivas Kandagatla (5): ASoC: codecs: Add msm8916-wcd analog codec ASoC: codecs: Add msm8916-wcd digital codec ASoC: apq8016-sbc: dt bindings: remove incorrect property ASoC: qcom: apq8016-sbc: Add support to multi codec. arm64: dts: apq8016-sbc: add analog audio support with multicodec .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 5 +- .../bindings/sound/qcom,msm8916-wcd-analog.txt | 83 ++ .../bindings/sound/qcom,msm8916-wcd-digital.txt | 20 + arch/arm64/boot/dts/qcom/apq8016-sbc.dtsi | 63 ++ arch/arm64/boot/dts/qcom/msm8916.dtsi | 9 + arch/arm64/boot/dts/qcom/pm8916.dtsi | 45 +- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 5 +- sound/soc/codecs/msm8916-wcd-analog.c | 888 ++++++++++++++++++++ sound/soc/codecs/msm8916-wcd-digital.c | 920 +++++++++++++++++++++ sound/soc/qcom/apq8016_sbc.c | 11 +- 11 files changed, 2043 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt create mode 100644 sound/soc/codecs/msm8916-wcd-analog.c create mode 100644 sound/soc/codecs/msm8916-wcd-digital.c -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: samsung: make audio interface/controller explicitly
by Mark Brown 19 Sep '16

19 Sep '16

On Mon, Sep 19, 2016 at 05:41:50PM +0800, Ayaka wrote: > 從我的 iPad 傳送 Please fix your mail client to word wrap within paragraphs at something substantially less than 80 columns. Doing this makes your messages much easier to read and reply to. > > Are you *sure* simple-card works for AC'97? > Not sure. I could hardly find a AC97 device in embedded market. If it > didn't I could disable it. I have only tested simple sound card for > i2s and SPDIF. I would be very surprised if it does and AC'97 is supposed to be enumerable and not need a card binding anyway so best to remove this one for the time being.

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Re: [alsa-devel] [PATCH] ASoC: rt5660: add rt5660 codec driver
by Mark Brown 19 Sep '16

19 Sep '16

On Mon, Sep 19, 2016 at 03:11:00AM +0000, Oder Chiou wrote: > > On Tue, Aug 23, 2016 at 01:33:43PM +0800, Oder Chiou wrote: > > > +- realtek,use-ldo2 > > This would more normally be done by making LDO2 a regulator via the > > normal regulator framework - look at how the arizona drivers handle > > DCVDD for an example, it provides a default supply mapping which makes > > things really easy for users. Right now it's not clear to me looking at > > the code that the driver works for systems that don't use an internal > > LDO as there's nothing that enables MICVDD. > The LDO2 is not a regulator, and it is an internal LDO of codec that > controlled by register setting. Normally, the MICVDD is connected to 3.3V It's an internal regulator in the CODEC. > VDD directly, and it isn't controlled by regulator. The option > "realtek,use-ldo2" is depend on the customer that can decide that the MICVDD > is provided by external 3.3V VDD or internal LDO2 for it. If they used the > internal LDO2, the pin of MICVDD can be floated. Sure, that's exactly what it looks like so my comments above apply. > > We do a reset when we unregister at the CODEC level - that'll overwrite > > these settings won't it? > We have done it in the function "rt5660_remove". Right, and if the user removes and reloads the driver then the reset will happen.

1 0

[alsa-devel] [PATCH] make samsung interface entries explicitly
by Randy Li 19 Sep '16

19 Sep '16

I found those modules should be selected directly, as I only use the simple sound card. Randy Li (1): ASoC: samsung: make audio interface/controller explicitly sound/soc/samsung/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v4 0/3] ASoC: sun4i-codec: Distinguish sun4i from sun7i
by Danny Milosavljevic 18 Sep '16

18 Sep '16

ASoC: sun4i-codec: Introduce mechanism to detect sun7i and provide a different regmap different compared to sun4i Allwinner A10. The controls will be extended in a forthcoming patch - it is necessary to distinguish between sun4i and sun7i controls because they have different registers. Renamed SUN4I_CODEC_AC_SYS_VERI to SUN7I_CODEC_AC_DAC_CAL and renamed SUN4I_CODEC_AC_MIC_PHONE_CAL to SUN7I_CODEC_AC_MIC_PHONE_CAL because these are actually not present on Allwinner A10. Handle quirks by regmap config and codec and select the correct quirks automatically. Danny Milosavljevic (3): ASoC: sun4i-codec: rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field name. ASoC: rename "SUN4I_CODEC_SYS_VERI" to "SUN7I_CODEC_AC_DAC_CAL"; rename "SUN4I_CODEC_AC_MIC_PHONE_CAL" to "SUN7I_CODEC_AC_MIC_PHONE_CAL". ASoC: sun4i-codec: Add custom regmap configs for the A10 and A20 variants. sound/soc/sunxi/sun4i-codec.c | 45 ++++++++++++++++++++++++++++++++++--------- 1 file changed, 36 insertions(+), 9 deletions(-)

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[alsa-devel] alsa-tools FTBFS on armel: gcc-6 narrowing conversions
by Elimar Riesebieter 17 Sep '16

17 Sep '16

Hi all, in Debian a FTBFS was discovered in building alsa-tools: | arm-linux-gnueabi-g++ -DPACKAGE_NAME=\"hdspmixer\" -DPACKAGE_TARNAME=\"hdspmixer\" -DPACKAGE_VERSION=\"1.11\" -DPACKAGE_STRING=\"hdspmixer\ 1.11\" -DPACKAGE_BUGREPORT=\"\" -DPACKAGE_URL=\"\" -DPACKAGE=\"hdspmixer\" -DVERSION=\"1.11\" -DSTDC_HEADERS=1 -DHAVE_LIBASOUND=1 -I. -Wdate-time -D_FORTIFY_SOURCE=2 -g -O2 -fdebug-prefix-map=/<<PKGBUILDDIR>>=. -fstack-protector-strong -Wformat -Werror=format-security -I/usr/include/freetype2 -D_THREAD_SAFE -D_REENTRANT -fno-exceptions -MT channelmap.o -MD -MP -MF .deps/channelmap.Tpo -c -o channelmap.o channelmap.cxx | channelmap.cxx:59:1: error: narrowing conversion of '-1' from 'int' to 'char' inside { } [-Wnarrowing] | }; | ^ More Details can be found at: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=837590 Could this be fixed for next release? Thanks in advance Elimar -- Experience is something you don't get until just after you need it!

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[alsa-devel] [PATCH v4] drm/i915/dp: DP audio API changes for MST
by Dhinakaran Pandiyan 16 Sep '16

16 Sep '16

DP MST provides the capability to send multiple video and audio streams through a single port. This requires the API's between i915 and audio drivers to distinguish between multiple audio capable displays that can be connected to a port. Currently only the port identity is shared in the APIs. This patch adds support for MST with an additional parameter 'int pipe'. The existing parameter 'port' does not change it's meaning. pipe = MST : display pipe that the stream originates from Non-MST : -1 Affected APIs: struct i915_audio_component_ops - int (*sync_audio_rate)(struct device *, int port, int rate); + int (*sync_audio_rate)(struct device *, int port, int pipe, + int rate); - int (*get_eld)(struct device *, int port, bool *enabled, - unsigned char *buf, int max_bytes); + int (*get_eld)(struct device *, int port, int pipe, + bool *enabled, unsigned char *buf, int max_bytes); struct i915_audio_component_audio_ops - void (*pin_eld_notify)(void *audio_ptr, int port); + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); This patch makes dummy changes in the audio drivers (thanks Libin) for build to succeed. The audio side drivers will send the right 'pipe' values for MST in patches that will follow. v2: Renamed the new API parameter from 'dev_id' to 'pipe'. (Jim, Ville) Included Asoc driver API compatibility changes from Jeeja. Added WARN_ON() for invalid pipe in get_saved_encoder(). (Takashi) Added comment for av_enc_map[] definition. (Takashi) v3: Fixed logic error introduced while renaming 'dev_id' as 'pipe' (Ville) Renamed get_saved_encoder() to get_saved_enc() to reduce line length v4: Rebased. Parameter check for pipe < -1 values in get_saved_enc() (Ville) Switched to for_each_pipe() in get_saved_enc() (Ville) Renamed 'pipe' to 'dev_id' in audio side code (Takashi) Signed-off-by: Dhinakaran Pandiyan <dhinakaran.pandiyan(a)intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/drm/i915/i915_drv.h | 3 +- drivers/gpu/drm/i915/intel_audio.c | 94 ++++++++++++++++++++++++++------------ include/drm/i915_component.h | 6 +-- include/sound/hda_i915.h | 11 +++-- sound/hda/hdac_i915.c | 16 ++++--- sound/pci/hda/patch_hdmi.c | 7 +-- sound/soc/codecs/hdac_hdmi.c | 2 +- 7 files changed, 92 insertions(+), 47 deletions(-) diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h index aaa9c60..dac81c7 100644 --- a/drivers/gpu/drm/i915/i915_drv.h +++ b/drivers/gpu/drm/i915/i915_drv.h @@ -2056,7 +2056,8 @@ struct drm_i915_private { /* perform PHY state sanity checks? */ bool chv_phy_assert[2]; - struct intel_encoder *dig_port_map[I915_MAX_PORTS]; + /* Used to save the pipe-to-encoder mapping for audio */ + struct intel_encoder *av_enc_map[I915_MAX_PIPES]; /* * NOTE: This is the dri1/ums dungeon, don't add stuff here. Your patch diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index 40fbdd8..9583f43 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -491,6 +491,7 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) struct drm_i915_private *dev_priv = to_i915(encoder->dev); struct i915_audio_component *acomp = dev_priv->audio_component; enum port port = intel_encoder->port; + enum pipe pipe = crtc->pipe; connector = drm_select_eld(encoder); if (!connector) @@ -515,12 +516,18 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) mutex_lock(&dev_priv->av_mutex); intel_encoder->audio_connector = connector; + /* referred in audio callbacks */ - dev_priv->dig_port_map[port] = intel_encoder; + dev_priv->av_enc_map[pipe] = intel_encoder; mutex_unlock(&dev_priv->av_mutex); + /* audio drivers expect pipe = -1 to indicate Non-MST cases */ + if (intel_encoder->type != INTEL_OUTPUT_DP_MST) + pipe = -1; + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + (int) port, (int) pipe); } /** @@ -536,17 +543,24 @@ void intel_audio_codec_disable(struct intel_encoder *intel_encoder) struct drm_i915_private *dev_priv = to_i915(encoder->dev); struct i915_audio_component *acomp = dev_priv->audio_component; enum port port = intel_encoder->port; + struct intel_crtc *crtc = to_intel_crtc(encoder->crtc); + enum pipe pipe = crtc->pipe; if (dev_priv->display.audio_codec_disable) dev_priv->display.audio_codec_disable(intel_encoder); mutex_lock(&dev_priv->av_mutex); intel_encoder->audio_connector = NULL; - dev_priv->dig_port_map[port] = NULL; + dev_priv->av_enc_map[pipe] = NULL; mutex_unlock(&dev_priv->av_mutex); + /* audio drivers expect pipe = -1 to indicate Non-MST cases */ + if (intel_encoder->type != INTEL_OUTPUT_DP_MST) + pipe = -1; + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + (int) port, (int) pipe); } /** @@ -621,15 +635,40 @@ static int i915_audio_component_get_cdclk_freq(struct device *kdev) return dev_priv->cdclk_freq; } -static int i915_audio_component_sync_audio_rate(struct device *kdev, - int port, int rate) +static struct intel_encoder *get_saved_enc(struct drm_i915_private *dev_priv, + int port, int pipe) +{ + + if (WARN_ON(pipe >= I915_MAX_PIPES)) + return NULL; + + /* MST */ + if (pipe >= 0) + return dev_priv->av_enc_map[pipe]; + + /* Non-MST */ + for_each_pipe(dev_priv, pipe) { + struct intel_encoder *encoder; + + encoder = dev_priv->av_enc_map[pipe]; + if (encoder == NULL) + continue; + + if (port == encoder->port) + return encoder; + } + + return NULL; +} + +static int i915_audio_component_sync_audio_rate(struct device *kdev, int port, + int pipe, int rate) { struct drm_i915_private *dev_priv = kdev_to_i915(kdev); struct intel_encoder *intel_encoder; struct intel_crtc *crtc; struct drm_display_mode *mode; struct i915_audio_component *acomp = dev_priv->audio_component; - enum pipe pipe = INVALID_PIPE; u32 tmp; int n; int err = 0; @@ -643,25 +682,20 @@ static int i915_audio_component_sync_audio_rate(struct device *kdev, i915_audio_component_get_power(kdev); mutex_lock(&dev_priv->av_mutex); + /* 1. get the pipe */ - intel_encoder = dev_priv->dig_port_map[port]; - /* intel_encoder might be NULL for DP MST */ + intel_encoder = get_saved_enc(dev_priv, port, pipe); if (!intel_encoder || !intel_encoder->base.crtc || intel_encoder->type != INTEL_OUTPUT_HDMI) { - DRM_DEBUG_KMS("no valid port %c\n", port_name(port)); + DRM_DEBUG_KMS("Not valid for port %c\n", port_name(port)); err = -ENODEV; goto unlock; } + + /* pipe passed from the audio driver will be -1 for Non-MST case */ crtc = to_intel_crtc(intel_encoder->base.crtc); pipe = crtc->pipe; - if (pipe == INVALID_PIPE) { - DRM_DEBUG_KMS("no pipe for the port %c\n", port_name(port)); - err = -ENODEV; - goto unlock; - } - DRM_DEBUG_KMS("pipe %c connects port %c\n", - pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; /* port must be valid now, otherwise the pipe will be invalid */ @@ -697,7 +731,7 @@ static int i915_audio_component_sync_audio_rate(struct device *kdev, } static int i915_audio_component_get_eld(struct device *kdev, int port, - bool *enabled, + int pipe, bool *enabled, unsigned char *buf, int max_bytes) { struct drm_i915_private *dev_priv = kdev_to_i915(kdev); @@ -706,16 +740,20 @@ static int i915_audio_component_get_eld(struct device *kdev, int port, int ret = -EINVAL; mutex_lock(&dev_priv->av_mutex); - intel_encoder = dev_priv->dig_port_map[port]; - /* intel_encoder might be NULL for DP MST */ - if (intel_encoder) { - ret = 0; - *enabled = intel_encoder->audio_connector != NULL; - if (*enabled) { - eld = intel_encoder->audio_connector->eld; - ret = drm_eld_size(eld); - memcpy(buf, eld, min(max_bytes, ret)); - } + + intel_encoder = get_saved_enc(dev_priv, port, pipe); + if (!intel_encoder) { + DRM_DEBUG_KMS("Not valid for port %c\n", port_name(port)); + mutex_unlock(&dev_priv->av_mutex); + return ret; + } + + ret = 0; + *enabled = intel_encoder->audio_connector != NULL; + if (*enabled) { + eld = intel_encoder->audio_connector->eld; + ret = drm_eld_size(eld); + memcpy(buf, eld, min(max_bytes, ret)); } mutex_unlock(&dev_priv->av_mutex); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index b46fa0e..545c6e0 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -64,7 +64,7 @@ struct i915_audio_component_ops { * Called from audio driver. After audio driver sets the * sample rate, it will call this function to set n/cts */ - int (*sync_audio_rate)(struct device *, int port, int rate); + int (*sync_audio_rate)(struct device *, int port, int pipe, int rate); /** * @get_eld: fill the audio state and ELD bytes for the given port * @@ -77,7 +77,7 @@ struct i915_audio_component_ops { * Note that the returned size may be over @max_bytes. Then it * implies that only a part of ELD has been copied to the buffer. */ - int (*get_eld)(struct device *, int port, bool *enabled, + int (*get_eld)(struct device *, int port, int pipe, bool *enabled, unsigned char *buf, int max_bytes); }; @@ -97,7 +97,7 @@ struct i915_audio_component_audio_ops { * status accordingly (even when the HDA controller is in power save * mode). */ - void (*pin_eld_notify)(void *audio_ptr, int port); + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); }; /** diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index 796cabf..5ab972e 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -10,8 +10,9 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); void snd_hdac_i915_set_bclk(struct hdac_bus *bus); -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate); -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate); +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); @@ -29,13 +30,13 @@ static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { } static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, - hda_nid_t nid, int rate) + hda_nid_t nid, int dev_id, int rate) { return 0; } static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, - bool *audio_enabled, char *buffer, - int max_bytes) + int dev_id, bool *audio_enabled, + char *buffer, int max_bytes) { return -ENODEV; } diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index c9af022..1979b49 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -201,18 +201,20 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) * This function sets N/CTS value based on the given sample rate. * Returns zero for success, or a negative error code. */ -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate) { struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; - int port; + int port, pipe; if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) return -ENODEV; port = pin2port(codec, nid); if (port < 0) return -EINVAL; - return acomp->ops->sync_audio_rate(acomp->dev, port, rate); + pipe = dev_id; + return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); } EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); @@ -236,12 +238,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * thus it may be over @max_bytes. If it's over @max_bytes, it implies * that only a part of ELD bytes have been fetched. */ -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, bool *audio_enabled, char *buffer, int max_bytes) { struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; - int port; + int port, pipe; if (!acomp || !acomp->ops || !acomp->ops->get_eld) return -ENODEV; @@ -249,7 +251,9 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, port = pin2port(codec, nid); if (port < 0) return -EINVAL; - return acomp->ops->get_eld(acomp->dev, port, audio_enabled, + + pipe = dev_id; + return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, buffer, max_bytes); } EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 56e5204..cf9bc042 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1485,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, -1, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); if (size > 0) { @@ -1744,7 +1744,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ if (codec_has_acomp(codec)) - snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); + snd_hdac_sync_audio_rate(&codec->core, pin_nid, -1, + runtime->rate); non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -2290,7 +2291,7 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state); } -static void intel_pin_eld_notify(void *audio_ptr, int port) +static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) { struct hda_codec *codec = audio_ptr; int pin_nid; diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e181b2..dc0129b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1368,7 +1368,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, return hdac_hdmi_init_dai_map(edev); } -static void hdac_hdmi_eld_notify_cb(void *aptr, int port) +static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) { struct hdac_ext_device *edev = aptr; struct hdac_hdmi_priv *hdmi = edev->private_data; -- 2.5.0

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[alsa-devel] [PATCH 2/2] ASoC: cs4341: Add DT bindings documentation for CS4341 CODEC
by Alexander Shiyan 16 Sep '16

16 Sep '16

This patch adds DT bindings documentation for Cirrus Logic CS4341 CODEC. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- Documentation/devicetree/bindings/sound/cs4341.txt | 15 +++++++++++++++ 1 file changed, 15 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4341.txt diff --git a/Documentation/devicetree/bindings/sound/cs4341.txt b/Documentation/devicetree/bindings/sound/cs4341.txt new file mode 100644 index 0000000..78d45be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4341.txt @@ -0,0 +1,15 @@ +Cirrus Logic CS4341 DT bindings + +Required properties: + - compatible: "cirrus,cs4341a" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Example: + codec: cs4341@0 { + #sound-dai-cells = <0>; + compatible = "cirrus,cs4341a"; + reg = <0>; + spi-max-frequency = <6000000>; + }; -- 2.7.3

2 1

Re: [alsa-devel] Novation Twitch broken on Linux 4.7?
by Juan Pedro Bolívar Puente 16 Sep '16

16 Sep '16

Hi, Mark Hills <mark(a)xwax.org> writes: > On Fri, 16 Sep 2016, Juan Pedro Bolívar Puente wrote: > >> I am trying to use again my Novation Twitch after some time, and audio >> playback is not working anymore. I have tried with Jackd and various >> other programs, and it seems to always fail to install the hardware >> params. > > [...] > > If it doesn't work then maybe the attached alsacap.c is useful and you > could post the output back (and CC alsa-devel); it should show us the > advertised rates and formats. Here is the ouput: $ ./alsacap *** Scanning for playback devices *** ... Card 1, ID `Twitch', name `Twitch' Device 0, ID `USB Audio', name `USB Audio', 1 subdevices (1 available) 4 channels, sampling rate 44100..48000 Hz Sample formats: S24_3LE Subdevice 0, name `subdevice #0' $ ./alsacap -d hw:Twitch *** Exploring configuration space of device `hw:Twitch' for playback *** 4 channels Sampling rate 44100..48000 Hz Sample formats: S24_3LE Significant bits: 24 For people reading this message for the first time in `alsa-devel`, here are the error output of various commands when trying to use the device: $ aplay -D hw:Twitch -f S24_3LE -c 4 -r 44100 test.wav Playing raw data 'prueba.wav' : Signed 24 bit Little Endian in 3bytes, Rate 44100 Hz, Channels 4 aplay: set_params:1361: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_3LE SUBFORMAT: STD SAMPLE_BITS: 24 FRAME_BITS: 96 CHANNELS: 4 RATE: 44100 PERIOD_TIME: (125011 125012) PERIOD_SIZE: 5513 PERIOD_BYTES: 66156 PERIODS: (3 4) BUFFER_TIME: 500000 BUFFER_SIZE: 22050 BUFFER_BYTES: 264600 TICK_TIME: 0 $ mplayer -ao alsa:device=hw=Twitch test.mp3 MPlayer 1.3.0 (Debian), built with gcc-5.4.0 (C) 2000-2016 MPlayer Team Playing test.mp3. ... [AO_ALSA] Format s16le is not supported by hardware, trying default. [AO_ALSA] Unable to set format: Invalid argument Failed to initialize audio driver 'alsa:device=hw=Twitch' $ mplayer -ao alsa:device=plughw=Twitch test.mp3 MPlayer 1.3.0 (Debian), built with gcc-5.4.0 (C) 2000-2016 MPlayer Team Playing test.mp3. ... [AO_ALSA] alsa-lib: pcm_hw.c:603:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE failed (-32): Broken pipe [AO_ALSA] alsa-lib: pcm_direct.c:1054:(snd1_pcm_direct_initialize_slave) unable to install hw params [AO_ALSA] alsa-lib: pcm_dmix.c:1053:(snd_pcm_dmix_open) unable to initialize slave [AO_ALSA] Playback open error: Broken pipe Failed to initialize audio driver 'alsa:device=hw=Twitch' $ jackd -n twitch -d alsa -d hw:Twitch jackdmp 1.9.11 ... creating alsa driver ... hw:Twitch|hw:Twitch|1024|2|48000|0|0|nomon|swmeter|-|32bit Using ALSA driver USB-Audio running on card 1 - Novation Twitch at usb-0000:00:1d.0-1.2, full speed ALSA: Cannot open PCM device alsa_pcm for capture. Falling back to playback-only mode configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods ALSA: final selected sample format for playback: 24bit little-endian in 3bytes format ALSA: use 2 periods for playback ALSA: cannot set hardware parameters for playback ALSA: cannot configure playback channel Thanks a lot! -- ∿∿∿∿ https://sinusoid.es ∿∿∿∿

1 0

[alsa-devel] [PATCH] ASoC: rockchip: Support headset button function on rk3399
by Cheng-Yi Chiang 16 Sep '16

16 Sep '16

Set the mapping between button and media key event. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 164b6da..ee06489 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -23,6 +23,7 @@ #include <linux/of_gpio.h> #include <linux/delay.h> #include <linux/spi/spi.h> +#include <linux/input.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -209,6 +210,14 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + snd_jack_set_key(rockchip_sound_jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key( + rockchip_sound_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key( + rockchip_sound_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key( + rockchip_sound_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(codec, &rockchip_sound_jack); return 0; -- 2.6.6

2 1

[alsa-devel] [PATCH] ASoC: rt5663: refine error handling
by Arnd Bergmann 16 Sep '16

16 Sep '16

The newly added rt5663 codec driver introduces a couple of warnings when built with -Wmaybe-uninitialized: codecs/rt5663.c: In function 'rt5663_set_tdm_slot': codecs/rt5663.c:2680:2: error: 'reg' may be used uninitialized in this function [-Werror=maybe-uninitialized] codecs/rt5663.c: In function 'rt5663_set_dai_pll': codecs/rt5663.c:2556:12: error: 'shift' may be used uninitialized in this function [-Werror=maybe-uninitialized] codecs/rt5663.c:2596:2: error: 'mask' may be used uninitialized in this function [-Werror=maybe-uninitialized] We can avoid those warnings by always returning an error from the switch() statements instead of falling through with an uninitialized variable when we hit an unexpected case. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: df7c52168ee1 ("ASoC: add rt5663 codec driver") --- sound/soc/codecs/rt5663.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 7963a2736a91..8078abcaf728 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1831,6 +1831,7 @@ static int rt5663_i2s_use_asrc(struct snd_soc_dapm_widget *source, break; default: dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + return 1; } if (da_asrc_en || ad_asrc_en) @@ -2579,6 +2580,7 @@ static int rt5663_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + return -EINVAL; } switch (source) { @@ -2674,6 +2676,7 @@ static int rt5663_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; default: dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + return -EINVAL; } snd_soc_update_bits(codec, reg, RT5668_TDM_MODE_MASK | -- 2.9.0

2 1

[alsa-devel] [PATCH] ALSA: compress: fix some missing and misplaced \n in messages
by Colin King 16 Sep '16

16 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Fix a missing \n in a pr_debug message and move the \n to the end of a pr_err message. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/compress_offload.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 583d407..fec1dfd 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -780,7 +780,7 @@ static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) ret = wait_event_interruptible(stream->runtime->sleep, (stream->runtime->state != SNDRV_PCM_STATE_DRAINING)); if (ret == -ERESTARTSYS) - pr_debug("wait aborted by a signal"); + pr_debug("wait aborted by a signal\n"); else if (ret) pr_debug("wait for drain failed with %d\n", ret); @@ -962,7 +962,7 @@ static int snd_compress_dev_register(struct snd_device *device) compr->card, compr->device, &snd_compr_file_ops, compr, &compr->dev); if (ret < 0) { - pr_err("snd_register_device failed\n %d", ret); + pr_err("snd_register_device failed %d\n", ret); return ret; } return ret; -- 2.9.3

2 1

[alsa-devel] [PATCH] ALSA: hdac: add missing \n to end of dev_err messages
by Colin King 16 Sep '16

16 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trival fix, some dev_err messages are missing a \n, so add it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/hda/ext/hdac_ext_stream.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 2441273..3be051a 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -424,7 +424,7 @@ void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { - dev_err(bus->dev, "Address of SPB capability is NULL"); + dev_err(bus->dev, "Address of SPB capability is NULL\n"); return; } @@ -453,7 +453,7 @@ int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { - dev_err(bus->dev, "Address of SPB capability is NULL"); + dev_err(bus->dev, "Address of SPB capability is NULL\n"); return -EINVAL; } @@ -476,7 +476,7 @@ int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { - dev_err(bus->dev, "Address of SPB capability is NULL"); + dev_err(bus->dev, "Address of SPB capability is NULL\n"); return -EINVAL; } @@ -516,7 +516,7 @@ void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { - dev_err(bus->dev, "Address of DRSM capability is NULL"); + dev_err(bus->dev, "Address of DRSM capability is NULL\n"); return; } @@ -545,7 +545,7 @@ int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus, struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { - dev_err(bus->dev, "Address of DRSM capability is NULL"); + dev_err(bus->dev, "Address of DRSM capability is NULL\n"); return -EINVAL; } -- 2.9.3

2 1

[alsa-devel] [PATCH] Revert "ASoC: core: mark SND_SOC_BYTES_EXT as deprecated"
by Takashi Sakamoto 16 Sep '16

16 Sep '16

SND_SOC_BYTES_EXT was added by commit d98812082c87 ("ASoC: add SND_SOC_BYTES_EXT") to allow drivers to transfer arbitrary byte-aligned data to/from devices via ALSA control interface. Later, SND_SOC_BYTES_TLV was added by commit 7523a271682f ("ASoC: core: add a helper for extended byte controls using TLV") was added for the same purpose and developers expected this to obsolete the former. However, SND_SOC_BYTES_TLV brought confusions to user land because it doesn't follow to a protocol of ALSA control interface. At least, this macro and related kernel APIs include two misunderstandings: - 'struct snd_ctl_elem_info.count' can also represent the length of TLV packet paylaod, snd_soc_bytes_info_ext() performs in this way. - 'struct snd_ctl_tlv.tlv' can include arbitrary data regardless of TLV packet structure, snd_soc_bytes_tlv_callback() performs in this way. In a policy of kernel land development, it's quite worse to break protocols for applications. Therefore, developers are discouraged to use these kernel APIs. In the first place, SND_SOC_BYTES_TLV was added to satisfy a request of developers who need to add control elements which transfer data larger than the size which 'struct snd_ctl_elem_value' can represent; e.g. over 512 bytes. As long as the size is less than the size, SND_SOC_BYTES_EXT is still available. Although there is the limitation of maximum data size, it's better to use this API for stable application interface till better alternative ways are implemented in future. For these reasons, this commit reverts the previous commit which lead developers to the worse behaviour. Unfortunately, SND_SOC_BYTES_TLV was already merged in development cycle for kernel 3.17 and published, so this commit doesn't touch it for any compatibilities. Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069483.h… Reference: commit d98812082c87 ("ASoC: add SND_SOC_BYTES_EXT") Reference: commit 7523a271682f ("ASoC: core: add a helper for extended byte controls using TLV") Cc: Vinod Koul <vinod.koul(a)intel.com> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/soc.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6144882..18f7d21 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -311,9 +311,6 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } -/* - * SND_SOC_BYTES_EXT is deprecated, please USE SND_SOC_BYTES_TLV instead - */ #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, \ -- 2.7.4

2 3

[alsa-devel] [PATCH] ALSA: au88x0: Add missing \n to end of dev_err message
by Colin King 16 Sep '16

16 Sep '16

From: Colin Ian King <colin.king(a)canonical.com> Trival fix, the dev_err message is missing a \n so add it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/au88x0/au88x0_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index d3125c1..e1af24f 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1043,7 +1043,7 @@ static void vortex_fifo_init(vortex_t * vortex) for (x = NR_ADB - 1; x >= 0; x--) { hwwrite(vortex->mmio, addr, (FIFO_U0 | FIFO_U1)); if (hwread(vortex->mmio, addr) != (FIFO_U0 | FIFO_U1)) - dev_err(vortex->card->dev, "bad adb fifo reset!"); + dev_err(vortex->card->dev, "bad adb fifo reset!\n"); vortex_fifo_clearadbdata(vortex, x, FIFO_SIZE); addr -= 4; } -- 2.9.3

2 1

[alsa-devel] [PATCH v2 0/9] ASoC: simple-card: tidyup and synchronize patches
by Kuninori Morimoto 16 Sep '16

16 Sep '16

Hi Mark These are v2 of cleanup/tidyup and synchronize patches for simple-card. We have now simple-card and simple-scu-card. These are almost same, difference should be very small. But, because of historical reason, these have "code" difference. This patch cleanup and synchronize these. Kuninori Morimoto (9): ASoC: simple-card: call of_node_put() for dai_link ASoC: simple-card: tidyup mclk-fs of_property_read_u32() usage ASoC: simple-card: code sync: tidyup simple_priv_to_xxx() macro ASoC: simple-card: code sync: use simple_priv_to_props() macro ASoC: simple-card: is GPL v2 ASoC: simple-card: code sync: tidyup white line ASoC: simple-card: code sync: use tag ASoC: simple-card: code sync: rename num_link to num ASoC: simple-card: use kzalloc() for dai_props / dai_link sound/soc/generic/simple-card.c | 87 ++++++++++++++++++++--------------------- 1 file changed, 43 insertions(+), 44 deletions(-) -- 1.9.1

2 11

[alsa-devel] [PATCH] Revert "ASoC: core: mark SND_SOC_BYTES_EXT as deprecated"
by Takashi Sakamoto 16 Sep '16

16 Sep '16

SND_SOC_BYTES_TLV brings confusions to user land because it doesn't follow to a protocol of ctl and tlv operation. At least, this macro and related kernel APIs include two misunderstandings: - 'struct snd_ctl_elem_info.count' can also represent the length of TLV packet paylaod, snd_soc_bytes_info_ext() performs in this way. - 'struct snd_ctl_tlv.tlv' can include arbitrary data regardless of TLV packet structure, snd_soc_bytes_tlv_callback() performs in this way. In a policy of kernel land development, it's quite worse to break protocols for applications. Therefore, developers are discouraged to use these kernel APIs. In the first place, SND_SOC_BYTES_TLV was added to satisfy a request of developers who need to add control elements which transfer data larger than the size which 'struct snd_ctl_elem_value' can represent; e.g. over 512 bytes. However, as long as the size is less than the size; e.g. 512 bytes, SND_SOC_BYTES_EXT is still available. Although there is actually the limitation of maximum data size, it's better to use this API for stable application interface till better alternative ways are implemented in future. For these reasons, this commit reverts the previous commit which lead developers to the worse behaviour. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/soc.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6144882..18f7d21 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -311,9 +311,6 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } -/* - * SND_SOC_BYTES_EXT is deprecated, please USE SND_SOC_BYTES_TLV instead - */ #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, \ -- 2.7.4

3 5

[alsa-devel] Audio miniconference at LPC/KS confirmed
by Mark Brown 15 Sep '16

15 Sep '16

On Thu, Aug 25, 2016 at 04:39:40PM +0100, Mark Brown wrote: > As in previous years we're organizing an audio miniconference so we can > get together and talk through issues, especially design decisons, face > to face. This year we're holding it as part of the workshops day shared > between Kernel Summit and Linux Plumbers Conference 1st to 4th November > (the workshops will be on the 2nd). This has now been confirmed and I > hope to see many of you there! > To attend you will NEED to have either a Kernel Summit invite (these > are going out now) or a Linux Plumbers Conference ticket. The number of > attendees at Plumbers is limited so I'd strongly encoruage you to sign > up early. The next batch of tickets go on sale on Saturday, the > registration site is here: If anyone wants to attend but has not yet got a ticket can you please contact me? I have a limited number of passes and discounted tickets I can allocate.

1 0

[alsa-devel] [PATCH v4] clk: x86: Add Atom PMC platform clocks
by Irina Tirdea 15 Sep '16

15 Sep '16

The BayTrail and CherryTrail platforms provide platform clocks through their Power Management Controller (PMC). The SoC supports up to 6 clocks (PMC_PLT_CLK[5:0]) with a frequency of either 19.2 MHz (PLL) or 25 MHz (XTAL) for BayTrail an a frequency of 19.2 MHz (XTAL) for CherryTrail. These clocks are available for general system use, where appropriate, and each have Control & Frequency register fields associated with them. For example, the usage for platform clocks suggested in the datasheet is the following: PLT_CLK[2:0] - Camera PLT_CLK[3] - Audio Codec PLT_CLK[4] - PLT_CLK[5] - COMMs Signed-off-by: Irina Tirdea <irina.tirdea(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Notes: Submitting this patch through the clock tree as requested by Mark Brown. This patch specifically enables the audio MCLK required by Baytrail CR devices (support already merged in Mark's tree) Changes from v3: - replace devm_kzalloc with devm_kcalloc - add x86 architecture maintainers Changes from v2: - move clk platform data structures to a separate include file - store clk_hw pointer for the fixed rate clocks Changes from v1: - register the clk device as child of pmc device - pass iomem pointer from pmc driver to clk driver to avoid using pmc_atom_read()/write() and use readl/writel API instead - use devm_clk_hw_register/clkdev_hw_create instead of clk_register/clkdev_create arch/x86/Kconfig | 1 + arch/x86/include/asm/pmc_atom.h | 3 + arch/x86/platform/atom/pmc_atom.c | 78 +++++- drivers/clk/x86/Makefile | 1 + drivers/clk/x86/clk-byt-plt.c | 380 ++++++++++++++++++++++++++++++ include/linux/platform_data/clk-byt-plt.h | 31 +++ 6 files changed, 491 insertions(+), 3 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c create mode 100644 include/linux/platform_data/clk-byt-plt.h diff --git a/arch/x86/Kconfig b/arch/x86/Kconfig index 2a1f0ce..3e603a8 100644 --- a/arch/x86/Kconfig +++ b/arch/x86/Kconfig @@ -2744,6 +2744,7 @@ config X86_DMA_REMAP config PMC_ATOM def_bool y depends on PCI + select COMMON_CLK config VMD depends on PCI_MSI diff --git a/arch/x86/include/asm/pmc_atom.h b/arch/x86/include/asm/pmc_atom.h index aa8744c..2d310cf 100644 --- a/arch/x86/include/asm/pmc_atom.h +++ b/arch/x86/include/asm/pmc_atom.h @@ -50,6 +50,9 @@ BIT_ORED_DEDICATED_IRQ_GPSC | \ BIT_SHARED_IRQ_GPSS) +/* Platform clock control registers */ +#define PMC_CLK_CTL_0 0x60 + /* The timers acumulate time spent in sleep state */ #define PMC_S0IR_TMR 0x80 #define PMC_S0I1_TMR 0x84 diff --git a/arch/x86/platform/atom/pmc_atom.c b/arch/x86/platform/atom/pmc_atom.c index 964ff4f..b353893 100644 --- a/arch/x86/platform/atom/pmc_atom.c +++ b/arch/x86/platform/atom/pmc_atom.c @@ -21,6 +21,8 @@ #include <linux/debugfs.h> #include <linux/seq_file.h> #include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/platform_data/clk-byt-plt.h> #include <asm/pmc_atom.h> @@ -37,6 +39,11 @@ struct pmc_reg_map { const struct pmc_bit_map *pss; }; +struct pmc_data { + const struct pmc_reg_map *map; + const struct pmc_clk *clks; +}; + struct pmc_dev { u32 base_addr; void __iomem *regmap; @@ -50,6 +57,29 @@ struct pmc_dev { static struct pmc_dev pmc_device; static u32 acpi_base_addr; +static const struct pmc_clk byt_clks[] = { + { + .name = "xtal", + .freq = 25000000, + .parent_name = NULL, + }, + { + .name = "pll", + .freq = 19200000, + .parent_name = "xtal", + }, + {}, +}; + +static const struct pmc_clk cht_clks[] = { + { + .name = "xtal", + .freq = 19200000, + .parent_name = NULL, + }, + {}, +}; + static const struct pmc_bit_map d3_sts_0_map[] = { {"LPSS1_F0_DMA", BIT_LPSS1_F0_DMA}, {"LPSS1_F1_PWM1", BIT_LPSS1_F1_PWM1}, @@ -169,6 +199,16 @@ static const struct pmc_reg_map cht_reg_map = { .pss = cht_pss_map, }; +static const struct pmc_data byt_data = { + .map = &byt_reg_map, + .clks = byt_clks, +}; + +static const struct pmc_data cht_data = { + .map = &cht_reg_map, + .clks = cht_clks, +}; + static inline u32 pmc_reg_read(struct pmc_dev *pmc, int reg_offset) { return readl(pmc->regmap + reg_offset); @@ -382,10 +422,36 @@ static int pmc_dbgfs_register(struct pmc_dev *pmc) } #endif /* CONFIG_DEBUG_FS */ +static int pmc_setup_clks(struct pci_dev *pdev, void __iomem *pmc_regmap, + const struct pmc_data *pmc_data) +{ + struct platform_device *clkdev; + struct pmc_clk_data *clk_data; + + clk_data = kzalloc(sizeof(*clk_data), GFP_KERNEL); + if (!clk_data) + return -ENOMEM; + + clk_data->base = pmc_regmap + PMC_CLK_CTL_0; + clk_data->clks = pmc_data->clks; + + clkdev = platform_device_register_data(&pdev->dev, "clk-byt-plt", -1, + clk_data, sizeof(*clk_data)); + if (IS_ERR(clkdev)) { + kfree(clk_data); + return PTR_ERR(clkdev); + } + + kfree(clk_data); + + return 0; +} + static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) { struct pmc_dev *pmc = &pmc_device; - const struct pmc_reg_map *map = (struct pmc_reg_map *)ent->driver_data; + const struct pmc_data *data = (struct pmc_data *)ent->driver_data; + const struct pmc_reg_map *map = data->map; int ret; /* Obtain ACPI base address */ @@ -414,6 +480,12 @@ static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) if (ret) dev_warn(&pdev->dev, "debugfs register failed\n"); + /* Register platform clocks - PMC_PLT_CLK [5:0] */ + ret = pmc_setup_clks(pdev, pmc->regmap, data); + if (ret) + dev_warn(&pdev->dev, "platform clocks register failed: %d\n", + ret); + pmc->init = true; return ret; } @@ -424,8 +496,8 @@ static int pmc_setup_dev(struct pci_dev *pdev, const struct pci_device_id *ent) * used by pci_match_id() call below. */ static const struct pci_device_id pmc_pci_ids[] = { - { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_VLV_PMC), (kernel_ulong_t)&byt_reg_map }, - { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_CHT_PMC), (kernel_ulong_t)&cht_reg_map }, + { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_VLV_PMC), (kernel_ulong_t)&byt_data }, + { PCI_VDEVICE(INTEL, PCI_DEVICE_ID_CHT_PMC), (kernel_ulong_t)&cht_data }, { 0, }, }; diff --git a/drivers/clk/x86/Makefile b/drivers/clk/x86/Makefile index 0478138..cbdc8cc 100644 --- a/drivers/clk/x86/Makefile +++ b/drivers/clk/x86/Makefile @@ -1,2 +1,3 @@ clk-x86-lpss-objs := clk-lpt.o obj-$(CONFIG_X86_INTEL_LPSS) += clk-x86-lpss.o +obj-$(CONFIG_PMC_ATOM) += clk-byt-plt.o diff --git a/drivers/clk/x86/clk-byt-plt.c b/drivers/clk/x86/clk-byt-plt.c new file mode 100644 index 0000000..7bd6c0d --- /dev/null +++ b/drivers/clk/x86/clk-byt-plt.c @@ -0,0 +1,380 @@ +/* + * Intel Atom platform clocks driver for BayTrail and CherryTrail SoC. + * + * Copyright (C) 2016, Intel Corporation + * Author: Irina Tirdea <irina.tirdea(a)intel.com> + * + * 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/clk-provider.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/clkdev.h> +#include <linux/platform_data/clk-byt-plt.h> + +#define PLT_CLK_NAME_BASE "pmc_plt_clk_" +#define PLT_CLK_DRIVER_NAME "clk-byt-plt" + +#define PMC_CLK_CTL_SIZE 4 +#define PMC_CLK_NUM 6 +#define PMC_MASK_CLK_CTL GENMASK(1, 0) +#define PMC_MASK_CLK_FREQ BIT(2) +#define PMC_CLK_CTL_GATED_ON_D3 0x0 +#define PMC_CLK_CTL_FORCE_ON 0x1 +#define PMC_CLK_CTL_FORCE_OFF 0x2 +#define PMC_CLK_CTL_RESERVED 0x3 +#define PMC_CLK_FREQ_XTAL 0x0 /* 25 MHz */ +#define PMC_CLK_FREQ_PLL 0x4 /* 19.2 MHz */ + +struct clk_plt_fixed { + struct clk_hw *clk; + struct clk_lookup *lookup; +}; + +struct clk_plt { + struct clk_hw hw; + void __iomem *reg; + struct clk_lookup *lookup; + spinlock_t lock; +}; + +#define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) + +struct clk_plt_data { + struct clk_plt_fixed **parents; + u8 nparents; + struct clk_plt *clks[PMC_CLK_NUM]; +}; + +static inline int plt_reg_to_parent(int reg) +{ + switch (reg & PMC_MASK_CLK_FREQ) { + case PMC_CLK_FREQ_XTAL: + return 0; /* index 0 in parents[] */ + case PMC_CLK_FREQ_PLL: + return 1; /* index 1 in parents[] */ + } + + return 0; +} + +static inline int plt_parent_to_reg(int index) +{ + switch (index) { + case 0: /* index 0 in parents[] */ + return PMC_CLK_FREQ_XTAL; + case 1: /* index 0 in parents[] */ + return PMC_CLK_FREQ_PLL; + } + + return PMC_CLK_FREQ_XTAL; +} + +static inline int plt_reg_to_enabled(int reg) +{ + switch (reg & PMC_MASK_CLK_CTL) { + case PMC_CLK_CTL_GATED_ON_D3: + case PMC_CLK_CTL_FORCE_ON: + return 1; /* enabled */ + case PMC_CLK_CTL_FORCE_OFF: + case PMC_CLK_CTL_RESERVED: + default: + return 0; /* disabled */ + } +} + +static void plt_clk_reg_update(struct clk_plt *clk, u32 mask, u32 val) +{ + u32 orig, tmp; + unsigned long flags = 0; + + spin_lock_irqsave(&clk->lock, flags); + + orig = readl(clk->reg); + + tmp = orig & ~mask; + tmp |= val & mask; + + if (tmp != orig) + writel(tmp, clk->reg); + + spin_unlock_irqrestore(&clk->lock, flags); +} + +static int plt_clk_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_plt *clk = to_clk_plt(hw); + + plt_clk_reg_update(clk, PMC_MASK_CLK_FREQ, plt_parent_to_reg(index)); + + return 0; +} + +static u8 plt_clk_get_parent(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + u32 value; + + value = readl(clk->reg); + + return plt_reg_to_parent(value); +} + +static int plt_clk_enable(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + + plt_clk_reg_update(clk, PMC_MASK_CLK_CTL, PMC_CLK_CTL_FORCE_ON); + + return 0; +} + +static void plt_clk_disable(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + + plt_clk_reg_update(clk, PMC_MASK_CLK_CTL, PMC_CLK_CTL_FORCE_OFF); +} + +static int plt_clk_is_enabled(struct clk_hw *hw) +{ + struct clk_plt *clk = to_clk_plt(hw); + u32 value; + + value = readl(clk->reg); + + return plt_reg_to_enabled(value); +} + +static const struct clk_ops plt_clk_ops = { + .enable = plt_clk_enable, + .disable = plt_clk_disable, + .is_enabled = plt_clk_is_enabled, + .get_parent = plt_clk_get_parent, + .set_parent = plt_clk_set_parent, + .determine_rate = __clk_mux_determine_rate, +}; + +static struct clk_plt *plt_clk_register(struct platform_device *pdev, int id, + void __iomem *base, + const char **parent_names, + int num_parents) +{ + struct clk_plt *pclk; + struct clk_init_data init; + int ret; + + pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); + if (!pclk) + return ERR_PTR(-ENOMEM); + + init.name = kasprintf(GFP_KERNEL, "%s%d", PLT_CLK_NAME_BASE, id); + init.ops = &plt_clk_ops; + init.flags = 0; + init.parent_names = parent_names; + init.num_parents = num_parents; + + pclk->hw.init = &init; + pclk->reg = base + id * PMC_CLK_CTL_SIZE; + spin_lock_init(&pclk->lock); + + ret = devm_clk_hw_register(&pdev->dev, &pclk->hw); + if (ret) + goto err_free_init; + + pclk->lookup = clkdev_hw_create(&pclk->hw, init.name, NULL); + if (!pclk->lookup) { + ret = -ENOMEM; + goto err_free_init; + } + + kfree(init.name); + + return pclk; + +err_free_init: + kfree(init.name); + return ERR_PTR(ret); +} + +static void plt_clk_unregister(struct clk_plt *pclk) +{ + clkdev_drop(pclk->lookup); +} + +static struct clk_plt_fixed *plt_clk_register_fixed_rate(struct platform_device *pdev, + const char *name, + const char *parent_name, + unsigned long fixed_rate) +{ + struct clk_plt_fixed *pclk; + int ret = 0; + + pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); + if (!pclk) + return ERR_PTR(-ENOMEM); + + pclk->clk = clk_hw_register_fixed_rate(&pdev->dev, name, parent_name, + 0, fixed_rate); + if (IS_ERR(pclk->clk)) + return ERR_CAST(pclk->clk); + + pclk->lookup = clkdev_hw_create(pclk->clk, name, NULL); + if (!pclk->lookup) { + ret = -ENOMEM; + goto err_clk_unregister; + } + + return pclk; + +err_clk_unregister: + clk_hw_unregister_fixed_rate(pclk->clk); + return ERR_PTR(ret); +} + +static void plt_clk_unregister_fixed_rate(struct clk_plt_fixed *pclk) +{ + clkdev_drop(pclk->lookup); + clk_hw_unregister_fixed_rate(pclk->clk); +} + +static const char **plt_clk_register_parents(struct platform_device *pdev, + struct clk_plt_data *data, + const struct pmc_clk *clks) +{ + const char **parent_names; + int i, err; + + data->nparents = 0; + while (clks[data->nparents].name) + data->nparents++; + + data->parents = devm_kcalloc(&pdev->dev, data->nparents, + sizeof(*data->parents), GFP_KERNEL); + if (!data->parents) { + err = -ENOMEM; + goto err_out; + } + + parent_names = kcalloc(data->nparents, sizeof(*parent_names), + GFP_KERNEL); + if (!parent_names) { + err = -ENOMEM; + goto err_out; + } + + for (i = 0; i < data->nparents; i++) { + data->parents[i] = + plt_clk_register_fixed_rate(pdev, clks[i].name, + clks[i].parent_name, + clks[i].freq); + if (IS_ERR(data->parents[i])) { + err = PTR_ERR(data->parents[i]); + goto err_unreg; + } + parent_names[i] = kstrdup_const(clks[i].name, GFP_KERNEL); + } + + return parent_names; + +err_unreg: + for (i--; i >= 0; i--) { + plt_clk_unregister_fixed_rate(data->parents[i]); + kfree_const(parent_names[i]); + } + kfree(parent_names); +err_out: + data->nparents = 0; + return ERR_PTR(err); +} + +static void plt_clk_unregister_parents(struct clk_plt_data *data) +{ + int i; + + for (i = 0; i < data->nparents; i++) + plt_clk_unregister_fixed_rate(data->parents[i]); +} + +static int plt_clk_probe(struct platform_device *pdev) +{ + struct clk_plt_data *data; + int i, err; + const char **parent_names; + const struct pmc_clk_data *pmc_data; + + pmc_data = dev_get_platdata(&pdev->dev); + if (!pmc_data || !pmc_data->clks) + return -EINVAL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + parent_names = plt_clk_register_parents(pdev, data, pmc_data->clks); + if (IS_ERR(parent_names)) + return PTR_ERR(parent_names); + + for (i = 0; i < PMC_CLK_NUM; i++) { + data->clks[i] = plt_clk_register(pdev, i, pmc_data->base, + parent_names, data->nparents); + if (IS_ERR(data->clks[i])) { + err = PTR_ERR(data->clks[i]); + goto err_unreg_clk_plt; + } + } + + for (i = 0; i < data->nparents; i++) + kfree_const(parent_names[i]); + kfree(parent_names); + + dev_set_drvdata(&pdev->dev, data); + return 0; + +err_unreg_clk_plt: + for (i--; i >= 0; i--) + plt_clk_unregister(data->clks[i]); + plt_clk_unregister_parents(data); + for (i = 0; i < data->nparents; i++) + kfree_const(parent_names[i]); + kfree(parent_names); + return err; +} + +static int plt_clk_remove(struct platform_device *pdev) +{ + struct clk_plt_data *data; + int i; + + data = dev_get_drvdata(&pdev->dev); + if (!data) + return 0; + + for (i = 0; i < PMC_CLK_NUM; i++) + plt_clk_unregister(data->clks[i]); + plt_clk_unregister_parents(data); + return 0; +} + +static struct platform_driver plt_clk_driver = { + .driver = { + .name = PLT_CLK_DRIVER_NAME, + }, + .probe = plt_clk_probe, + .remove = plt_clk_remove, +}; +module_platform_driver(plt_clk_driver); + +MODULE_DESCRIPTION("Intel Atom platform clocks driver"); +MODULE_AUTHOR("Irina Tirdea <irina.tirdea(a)intel.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/include/linux/platform_data/clk-byt-plt.h b/include/linux/platform_data/clk-byt-plt.h new file mode 100644 index 0000000..e6bca9c --- /dev/null +++ b/include/linux/platform_data/clk-byt-plt.h @@ -0,0 +1,31 @@ +/* + * Intel Atom platform clocks for BayTrail and CherryTrail SoC. + * + * Copyright (C) 2016, Intel Corporation + * Author: Irina Tirdea <irina.tirdea(a)intel.com> + * + * 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. + */ + +#ifndef __CLK_BYT_PLT_H +#define __CLK_BYT_PLT_H + +struct pmc_clk { + const char *name; + unsigned long freq; + const char *parent_name; +}; + +struct pmc_clk_data { + void __iomem *base; + const struct pmc_clk *clks; +}; + +#endif /* __CLK_BYT_PLT_H */ -- 1.9.1

1 0

[alsa-devel] [PATCH v2 2/2] hda: thinkpad_helper: Add support for hid-lenovo LED control
by Dennis Wassenberg 15 Sep '16

15 Sep '16

Make the thinkpad_helper able to support not only led control over acpi with thinkpad_acpi driver but also led control over hid-lenovo. The hid-lenovo driver adapted the led control api of thinkpad_acpi. Make the thinkpad_acpi and hid-lenovo able to work combined to support connected Lenovo USB keyboards at Lenovo Thinkpad devices. Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- Changes v1 -> v2 (suggested by Takashi Iwai): * Adapt to not rename fixup_thinkpad_acpi to fixup_thinkpad * Made sure that it works even with one of kconfigs is disabled sound/pci/hda/thinkpad_helper.c | 182 +++++++++++++++++++++++++++++++--------- 1 file changed, 144 insertions(+), 38 deletions(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index f0955fd..0d98624 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -1,83 +1,188 @@ /* Helper functions for Thinkpad LED control; * to be included from codec driver + * + * These helper functions include both LED control over thinkpad_acpi and + * hid-lenovo */ -#if IS_ENABLED(CONFIG_THINKPAD_ACPI) +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) || IS_ENABLED(CONFIG_HID_LENOVO) #include <linux/acpi.h> +#include <linux/hid-lenovo.h> #include <linux/thinkpad_acpi.h> -static int (*led_set_func)(int, bool); +static int (*led_set_func_tpacpi)(int, bool); +static int (*led_set_func_hid_lenovo)(int, bool); static void (*old_vmaster_hook)(void *, int); static bool is_thinkpad(struct hda_codec *codec) { + return (codec->core.subsystem_id >> 16 == 0x17aa); +} + +static bool is_thinkpad_acpi(struct hda_codec *codec) +{ return (codec->core.subsystem_id >> 16 == 0x17aa) && (acpi_dev_found("LEN0068") || acpi_dev_found("IBM0068")); } -static void update_tpacpi_mute_led(void *private_data, int enabled) +static void update_thinkpad_mute_led(void *private_data, int enabled) { if (old_vmaster_hook) old_vmaster_hook(private_data, enabled); - if (led_set_func) - led_set_func(TPACPI_LED_MUTE, !enabled); +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) + if (led_set_func_tpacpi) + led_set_func_tpacpi(TPACPI_LED_MUTE, !enabled); +#endif +#if IS_ENABLED(CONFIG_HID_LENOVO) + if (led_set_func_hid_lenovo) + led_set_func_hid_lenovo(HID_LENOVO_LED_MUTE, !enabled); +#endif } -static void update_tpacpi_micmute_led(struct hda_codec *codec, + + +static void update_thinkpad_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - if (!ucontrol || !led_set_func) + if (!ucontrol) return; if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { /* TODO: How do I verify if it's a mono or stereo here? */ bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) + if (led_set_func_tpacpi) + led_set_func_tpacpi(TPACPI_LED_MICMUTE, !val); +#endif +#if IS_ENABLED(CONFIG_HID_LENOVO) + if (led_set_func_hid_lenovo) + led_set_func_hid_lenovo(HID_LENOVO_LED_MICMUTE, !val); +#endif } } -static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) +static int hda_fixup_thinkpad_acpi_prepare(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; - bool removefunc = false; + int ret = -ENXIO; - if (action == HDA_FIXUP_ACT_PROBE) { - if (!is_thinkpad(codec)) - return; - if (!led_set_func) - led_set_func = symbol_request(tpacpi_led_set); - if (!led_set_func) { - codec_warn(codec, - "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); - return; - } + if (!is_thinkpad(codec)) + return -ENODEV; + if (!is_thinkpad_acpi(codec)) + return -ENODEV; + if (!led_set_func_tpacpi) + led_set_func_tpacpi = symbol_request(tpacpi_led_set); + if (!led_set_func_tpacpi) { + codec_warn(codec, + "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); + return -ENOENT; + } - removefunc = true; - if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { - old_vmaster_hook = spec->vmaster_mute.hook; - spec->vmaster_mute.hook = update_tpacpi_mute_led; - removefunc = false; - } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->num_adc_nids > 1) - codec_dbg(codec, - "Skipping micmute LED control due to several ADCs"); - else { - spec->cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } + if (led_set_func_tpacpi(TPACPI_LED_MUTE, false) >= 0) { + old_vmaster_hook = spec->vmaster_mute.hook; + spec->vmaster_mute.hook = update_thinkpad_mute_led; + ret = 0; + } + + if (led_set_func_tpacpi(TPACPI_LED_MICMUTE, false) >= 0) { + if (spec->num_adc_nids > 1) + codec_dbg(codec, + "Skipping micmute LED control due to several ADCs"); + else { + spec->cap_sync_hook = update_thinkpad_micmute_led; + ret = 0; } } - if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + return ret; +} + +static void hda_fixup_thinkpad_acpi_setup(struct hda_codec *codec, int action) +{ + int ret = 0; + + if (action == HDA_FIXUP_ACT_PROBE) { + ret = hda_fixup_thinkpad_acpi_prepare(codec); + } + + if (led_set_func_tpacpi && + (action == HDA_FIXUP_ACT_FREE || ret)) { + symbol_put(tpacpi_led_set); - led_set_func = NULL; + led_set_func_tpacpi = NULL; + old_vmaster_hook = NULL; + } +} +#else +static void hda_fixup_thinkpad_acpi_setup(struct hda_codec *codec, int action) +{ + return 0; +} +#endif + +#if IS_ENABLED(CONFIG_HID_LENOVO) +static int hda_fixup_thinkpad_hid_prepare(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + int ret = 0; + + if (!is_thinkpad(codec)) + return -ENODEV; + if (!led_set_func_hid_lenovo) + led_set_func_hid_lenovo = symbol_request(hid_lenovo_led_set); + if (!led_set_func_hid_lenovo) { + codec_warn(codec, + "Failed to find hid-lenovo symbol hid_lenovo_led_set\n"); + return -ENOENT; + } + + if (update_thinkpad_mute_led != spec->vmaster_mute.hook) + old_vmaster_hook = spec->vmaster_mute.hook; + + /* do not remove hook if setting delay does not work currently because + * it is a usb hid devices which is not connected right now + * maybe is will be connected later + */ + led_set_func_hid_lenovo(HID_LENOVO_LED_MUTE, false); + spec->vmaster_mute.hook = update_thinkpad_mute_led; + + led_set_func_hid_lenovo(HID_LENOVO_LED_MICMUTE, false); + spec->cap_sync_hook = update_thinkpad_micmute_led; + + return ret; +} + +static void hda_fixup_thinkpad_hid_setup(struct hda_codec *codec, int action) +{ + int ret = 0; + + if (action == HDA_FIXUP_ACT_PROBE) { + ret = hda_fixup_thinkpad_hid_prepare(codec); + } + + if (led_set_func_hid_lenovo && + (action == HDA_FIXUP_ACT_FREE || ret)) { + + symbol_put(hid_lenovo_led_set); + led_set_func_hid_lenovo = NULL; old_vmaster_hook = NULL; } } +#else +static void hda_fixup_thinkpad_hid_setup(struct hda_codec *codec, int action) +{ +} +#endif + +/** this fixup is not for acpi case only, it will handle hid-lenovo as well */ +static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + hda_fixup_thinkpad_acpi_setup(codec, action); + hda_fixup_thinkpad_hid_setup(codec, action); +} #else /* CONFIG_THINKPAD_ACPI */ @@ -87,3 +192,4 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, } #endif /* CONFIG_THINKPAD_ACPI */ + -- 1.9.1

2 2

[alsa-devel] [PATCH 0/2] ALSA: control: export all of TLV related macros to user land
by Takashi Sakamoto 15 Sep '16

15 Sep '16

Hi, This patchset updates my previous RFCs, and they're now for merging to upstream. [alsa-devel] [RFC][PATCH 0/2] ALSA: control: export all of TLV related macros to user land http://mailman.alsa-project.org/pipermail/alsa-devel/2016-September/112645.… Currently, TLV related protocol is not shared to user land. This is not good in a point of application interfaces, because application developers can't realize the protocol just to see UAPI headers, and they're forced to look for TLV-related codes in kernel land. This patchset moves all of macros related to TLV into UAPI header. To avoid confusions such as name conflict in user land, these macros have prefixes of 'SNDRV_CTL_TLVT_' and 'SNDRV_CTL_TLVD_'. When adding new items to the protocol, developers should use these prefixes and add them to the UAPI header. Changes from RFC: - Drop second patch to replace include statements - Rename each macros to prevent applications from name conflict Takashi Sakamoto (2): ALSA: control: export layout of TLV payload to UAPI header ALSA: rename TLV-related macros so that they're friendly to user applications include/sound/tlv.h | 73 +++++++++++++----------------------------------- include/uapi/sound/tlv.h | 72 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 91 insertions(+), 54 deletions(-) -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: da7219: software reset codec at probe
by Xing Zheng 14 Sep '16

14 Sep '16

From: Hsin-Yu Chao <hychao(a)chromium.org> On some platform da7219 codec has persistent power across reboot so it doesn't reset and cause abnormal jack detection. Workaround this issue by doing software reset at probe. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- sound/soc/codecs/da7219.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 737e914..9d08c11 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1941,6 +1941,14 @@ static int da7219_i2c_probe(struct i2c_client *i2c, return ret; } + /* Software reset codec. */ + regmap_write_bits(da7219->regmap, DA7219_ACCDET_CONFIG_1, + DA7219_ACCDET_EN_MASK, 0); + regmap_write_bits(da7219->regmap, DA7219_CIF_CTRL, + DA7219_CIF_REG_SOFT_RESET_MASK, 0); + regmap_write_bits(da7219->regmap, DA7219_SYSTEM_ACTIVE, + DA7219_SYSTEM_ACTIVE_MASK, 0); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219, &da7219_dai, 1); if (ret < 0) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: nau8825: fix bug in FLL parameter
by John Hsu 14 Sep '16

14 Sep '16

In the FLL parameter calculation, the FVCO should choose the maximum one. The patch is to fix the bug about the wrong FVCO chosen. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 057785b..e643be9 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1907,7 +1907,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional * input based on FDCO, FREF and FLL ratio. */ - fvco = div_u64(fvco << 16, fref * fll_param->ratio); + fvco = div_u64(fvco_max << 16, fref * fll_param->ratio); fll_param->fll_int = (fvco >> 16) & 0x3FF; fll_param->fll_frac = fvco & 0xFFFF; return 0; -- 2.6.4

2 1

[alsa-devel] [PATCH v2] ASoC: add rt5663 codec driver
by Bard Liao 14 Sep '16

14 Sep '16

This is the initial codec driver for both rt5663 rt5668 codec. Signed-off-by: John Lin <john.lin(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * correct the codec name in the documentation file. --- Documentation/devicetree/bindings/sound/rt5663.txt | 30 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5663.c | 3216 ++++++++++++++++++++ sound/soc/codecs/rt5663.h | 1121 +++++++ 5 files changed, 4376 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5663.txt create mode 100644 sound/soc/codecs/rt5663.c create mode 100644 sound/soc/codecs/rt5663.h diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt new file mode 100644 index 0000000..7d3c974 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -0,0 +1,30 @@ +RT5663/RT5668 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5663" or "realtek,rt5668". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +Pins on the device (for linking into audio routes) for RT5663/RT5668: + + * IN1P + * IN1N + * IN2P + * IN2N + * HPOL + * HPOR + +Example: + +codec: rt5663@12 { + compatible = "realtek,rt5663"; + reg = <0x12>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3fd110..80e89de 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -113,6 +113,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5659 if I2C + select SND_SOC_RT5663 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -646,6 +647,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y + default y if SND_SOC_RT5663=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -654,6 +656,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m default m if SND_SOC_RT5659=m + default m if SND_SOC_RT5663=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -666,6 +669,7 @@ config SND_SOC_RL6347A config SND_SOC_RT286 tristate + select SND_SOC_RT5663 depends on I2C config SND_SOC_RT298 @@ -698,6 +702,9 @@ config SND_SOC_RT5651 config SND_SOC_RT5659 tristate +config SND_SOC_RT5663 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ffaa523..ab19c79 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -113,6 +113,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5659-objs := rt5659.o +snd-soc-rt5663-objs := rt5663.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -335,6 +336,7 @@ 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_RT5659) += snd-soc-rt5659.o +obj-$(CONFIG_SND_SOC_RT5663) += snd-soc-rt5663.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c new file mode 100644 index 0000000..7963a27 --- /dev/null +++ b/sound/soc/codecs/rt5663.c @@ -0,0 +1,3216 @@ +/* + * rt5663.c -- RT5668/RT5663 ALSA SoC audio codec driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Jack Yu <jack.yu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/workqueue.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 "rt5663.h" +#include "rl6231.h" + +#define RT5668_DEVICE_ID 0x6451 +#define RT5663_DEVICE_ID 0x6406 + +enum { + CODEC_TYPE_RT5668, + CODEC_TYPE_RT5663, +}; + +struct rt5663_priv { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct delayed_work jack_detect_work; + struct snd_soc_jack *hs_jack; + struct timer_list btn_check_timer; + + int codec_type; + int sysclk; + int sysclk_src; + int lrck; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_default rt5668_reg[] = { + { 0x0000, 0x0000 }, + { 0x0001, 0xc8c8 }, + { 0x0002, 0x8080 }, + { 0x0003, 0x8000 }, + { 0x0004, 0xc80a }, + { 0x0005, 0x0000 }, + { 0x0006, 0x0000 }, + { 0x0007, 0x0000 }, + { 0x000a, 0x0000 }, + { 0x000b, 0x0000 }, + { 0x000c, 0x0000 }, + { 0x000d, 0x0000 }, + { 0x000f, 0x0808 }, + { 0x0010, 0x4000 }, + { 0x0011, 0x0000 }, + { 0x0012, 0x1404 }, + { 0x0013, 0x1000 }, + { 0x0014, 0xa00a }, + { 0x0015, 0x0404 }, + { 0x0016, 0x0404 }, + { 0x0017, 0x0011 }, + { 0x0018, 0xafaf }, + { 0x0019, 0xafaf }, + { 0x001a, 0xafaf }, + { 0x001b, 0x0011 }, + { 0x001c, 0x2f2f }, + { 0x001d, 0x2f2f }, + { 0x001e, 0x2f2f }, + { 0x001f, 0x0000 }, + { 0x0020, 0x0000 }, + { 0x0021, 0x0000 }, + { 0x0022, 0x5757 }, + { 0x0023, 0x0039 }, + { 0x0024, 0x000b }, + { 0x0026, 0xc0c0 }, + { 0x0027, 0xc0c0 }, + { 0x0028, 0xc0c0 }, + { 0x0029, 0x8080 }, + { 0x002a, 0xaaaa }, + { 0x002b, 0xaaaa }, + { 0x002c, 0xaba8 }, + { 0x002d, 0x0000 }, + { 0x002e, 0x0000 }, + { 0x002f, 0x0000 }, + { 0x0030, 0x0000 }, + { 0x0031, 0x5000 }, + { 0x0032, 0x0000 }, + { 0x0033, 0x0000 }, + { 0x0034, 0x0000 }, + { 0x0035, 0x0000 }, + { 0x003a, 0x0000 }, + { 0x003b, 0x0000 }, + { 0x003c, 0x00ff }, + { 0x003d, 0x0000 }, + { 0x003e, 0x00ff }, + { 0x003f, 0x0000 }, + { 0x0040, 0x0000 }, + { 0x0041, 0x00ff }, + { 0x0042, 0x0000 }, + { 0x0043, 0x00ff }, + { 0x0044, 0x0c0c }, + { 0x0049, 0xc00b }, + { 0x004a, 0x0000 }, + { 0x004b, 0x031f }, + { 0x004d, 0x0000 }, + { 0x004e, 0x001f }, + { 0x004f, 0x0000 }, + { 0x0050, 0x001f }, + { 0x0052, 0xf000 }, + { 0x0061, 0x0000 }, + { 0x0062, 0x0000 }, + { 0x0063, 0x003e }, + { 0x0064, 0x0000 }, + { 0x0065, 0x0000 }, + { 0x0066, 0x003f }, + { 0x0067, 0x0000 }, + { 0x006b, 0x0000 }, + { 0x006d, 0xff00 }, + { 0x006e, 0x2808 }, + { 0x006f, 0x000a }, + { 0x0070, 0x8000 }, + { 0x0071, 0x8000 }, + { 0x0072, 0x8000 }, + { 0x0073, 0x7000 }, + { 0x0074, 0x7770 }, + { 0x0075, 0x0002 }, + { 0x0076, 0x0001 }, + { 0x0078, 0x00f0 }, + { 0x0079, 0x0000 }, + { 0x007a, 0x0000 }, + { 0x007b, 0x0000 }, + { 0x007c, 0x0000 }, + { 0x007d, 0x0123 }, + { 0x007e, 0x4500 }, + { 0x007f, 0x8003 }, + { 0x0080, 0x0000 }, + { 0x0081, 0x0000 }, + { 0x0082, 0x0000 }, + { 0x0083, 0x0000 }, + { 0x0084, 0x0000 }, + { 0x0085, 0x0000 }, + { 0x0086, 0x0008 }, + { 0x0087, 0x0000 }, + { 0x0088, 0x0000 }, + { 0x0089, 0x0000 }, + { 0x008a, 0x0000 }, + { 0x008b, 0x0000 }, + { 0x008c, 0x0003 }, + { 0x008e, 0x0060 }, + { 0x008f, 0x1000 }, + { 0x0091, 0x0c26 }, + { 0x0092, 0x0073 }, + { 0x0093, 0x0000 }, + { 0x0094, 0x0080 }, + { 0x0098, 0x0000 }, + { 0x0099, 0x0000 }, + { 0x009a, 0x0007 }, + { 0x009f, 0x0000 }, + { 0x00a0, 0x0000 }, + { 0x00a1, 0x0002 }, + { 0x00a2, 0x0001 }, + { 0x00a3, 0x0002 }, + { 0x00a4, 0x0001 }, + { 0x00ae, 0x2040 }, + { 0x00af, 0x0000 }, + { 0x00b6, 0x0000 }, + { 0x00b7, 0x0000 }, + { 0x00b8, 0x0000 }, + { 0x00b9, 0x0000 }, + { 0x00ba, 0x0002 }, + { 0x00bb, 0x0000 }, + { 0x00be, 0x0000 }, + { 0x00c0, 0x0000 }, + { 0x00c1, 0x0aaa }, + { 0x00c2, 0xaa80 }, + { 0x00c3, 0x0003 }, + { 0x00c4, 0x0000 }, + { 0x00d0, 0x0000 }, + { 0x00d1, 0x2244 }, + { 0x00d2, 0x0000 }, + { 0x00d3, 0x3300 }, + { 0x00d4, 0x2200 }, + { 0x00d9, 0x0809 }, + { 0x00da, 0x0000 }, + { 0x00db, 0x0008 }, + { 0x00dc, 0x00c0 }, + { 0x00dd, 0x6724 }, + { 0x00de, 0x3131 }, + { 0x00df, 0x0008 }, + { 0x00e0, 0x4000 }, + { 0x00e1, 0x3131 }, + { 0x00e2, 0x600c }, + { 0x00ea, 0xb320 }, + { 0x00eb, 0x0000 }, + { 0x00ec, 0xb300 }, + { 0x00ed, 0x0000 }, + { 0x00ee, 0xb320 }, + { 0x00ef, 0x0000 }, + { 0x00f0, 0x0201 }, + { 0x00f1, 0x0ddd }, + { 0x00f2, 0x0ddd }, + { 0x00f6, 0x0000 }, + { 0x00f7, 0x0000 }, + { 0x00f8, 0x0000 }, + { 0x00fa, 0x0000 }, + { 0x00fb, 0x0000 }, + { 0x00fc, 0x0000 }, + { 0x00fd, 0x0000 }, + { 0x00fe, 0x10ec }, + { 0x00ff, 0x6451 }, + { 0x0100, 0xaaaa }, + { 0x0101, 0x000a }, + { 0x010a, 0xaaaa }, + { 0x010b, 0xa0a0 }, + { 0x010c, 0xaeae }, + { 0x010d, 0xaaaa }, + { 0x010e, 0xaaaa }, + { 0x010f, 0xaaaa }, + { 0x0110, 0xe002 }, + { 0x0111, 0xa602 }, + { 0x0112, 0xaaaa }, + { 0x0113, 0x2000 }, + { 0x0117, 0x0f00 }, + { 0x0125, 0x0420 }, + { 0x0132, 0x0000 }, + { 0x0133, 0x0000 }, + { 0x0136, 0x5555 }, + { 0x0137, 0x5540 }, + { 0x0138, 0x3700 }, + { 0x0139, 0x79a1 }, + { 0x013a, 0x2020 }, + { 0x013b, 0x2020 }, + { 0x013c, 0x2005 }, + { 0x013f, 0x0000 }, + { 0x0145, 0x0002 }, + { 0x0146, 0x0000 }, + { 0x0147, 0x0000 }, + { 0x0148, 0x0000 }, + { 0x0160, 0x4ec0 }, + { 0x0161, 0x0080 }, + { 0x0162, 0x0200 }, + { 0x0163, 0x0800 }, + { 0x0164, 0x0000 }, + { 0x0165, 0x0000 }, + { 0x0166, 0x0000 }, + { 0x0167, 0x000f }, + { 0x0168, 0x000f }, + { 0x0170, 0x4e80 }, + { 0x0171, 0x0080 }, + { 0x0172, 0x0200 }, + { 0x0173, 0x0800 }, + { 0x0174, 0x00ff }, + { 0x0175, 0x0000 }, + { 0x0190, 0x4131 }, + { 0x0191, 0x4131 }, + { 0x0192, 0x4131 }, + { 0x0193, 0x4131 }, + { 0x0194, 0x0000 }, + { 0x0195, 0x0000 }, + { 0x0196, 0x0000 }, + { 0x0197, 0x0000 }, + { 0x0198, 0x0000 }, + { 0x0199, 0x0000 }, + { 0x01a0, 0x1e64 }, + { 0x01a1, 0x06a3 }, + { 0x01a2, 0x0000 }, + { 0x01a3, 0x0000 }, + { 0x01a4, 0x0000 }, + { 0x01a5, 0x0000 }, + { 0x01a6, 0x0000 }, + { 0x01a7, 0x0000 }, + { 0x01a8, 0x0000 }, + { 0x01a9, 0x0000 }, + { 0x01aa, 0x0000 }, + { 0x01ab, 0x0000 }, + { 0x01b5, 0x0000 }, + { 0x01b6, 0x01c3 }, + { 0x01b7, 0x02a0 }, + { 0x01b8, 0x03e9 }, + { 0x01b9, 0x1389 }, + { 0x01ba, 0xc351 }, + { 0x01bb, 0x0009 }, + { 0x01bc, 0x0018 }, + { 0x01bd, 0x002a }, + { 0x01be, 0x004c }, + { 0x01bf, 0x0097 }, + { 0x01c0, 0x433d }, + { 0x01c1, 0x0000 }, + { 0x01c2, 0x0000 }, + { 0x01c3, 0x0000 }, + { 0x01c4, 0x0000 }, + { 0x01c5, 0x0000 }, + { 0x01c6, 0x0000 }, + { 0x01c7, 0x0000 }, + { 0x01c8, 0x40af }, + { 0x01c9, 0x0702 }, + { 0x01ca, 0x0000 }, + { 0x01cb, 0x0000 }, + { 0x01cc, 0x5757 }, + { 0x01cd, 0x5757 }, + { 0x01ce, 0x5757 }, + { 0x01cf, 0x5757 }, + { 0x01d0, 0x5757 }, + { 0x01d1, 0x5757 }, + { 0x01d2, 0x5757 }, + { 0x01d3, 0x5757 }, + { 0x01d4, 0x5757 }, + { 0x01d5, 0x5757 }, + { 0x01d6, 0x003c }, + { 0x01da, 0x0000 }, + { 0x01db, 0x0000 }, + { 0x01dc, 0x0000 }, + { 0x01de, 0x7c00 }, + { 0x01df, 0x0320 }, + { 0x01e0, 0x06a1 }, + { 0x01e1, 0x0000 }, + { 0x01e2, 0x0000 }, + { 0x01e3, 0x0000 }, + { 0x01e4, 0x0000 }, + { 0x01e5, 0x0000 }, + { 0x01e6, 0x0001 }, + { 0x01e7, 0x0000 }, + { 0x01e8, 0x0000 }, + { 0x01ea, 0x0000 }, + { 0x01eb, 0x0000 }, + { 0x01ec, 0x0000 }, + { 0x01ed, 0x0000 }, + { 0x01ee, 0x0000 }, + { 0x01ef, 0x0000 }, + { 0x01f0, 0x0000 }, + { 0x01f1, 0x0000 }, + { 0x01f2, 0x0000 }, + { 0x01f3, 0x0000 }, + { 0x01f4, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x0000 }, + { 0x0202, 0x0000 }, + { 0x0203, 0x0000 }, + { 0x0204, 0x0000 }, + { 0x0205, 0x0000 }, + { 0x0206, 0x0000 }, + { 0x0207, 0x0000 }, + { 0x0208, 0x0000 }, + { 0x0210, 0x60b1 }, + { 0x0211, 0xa000 }, + { 0x0212, 0x024c }, + { 0x0213, 0xf7ff }, + { 0x0214, 0x024c }, + { 0x0215, 0x0102 }, + { 0x0216, 0x00a3 }, + { 0x0217, 0x0048 }, + { 0x0218, 0x92c0 }, + { 0x0219, 0x0000 }, + { 0x021a, 0x00c8 }, + { 0x021b, 0x0020 }, + { 0x02fa, 0x0000 }, + { 0x02fb, 0x0000 }, + { 0x02fc, 0x0000 }, + { 0x02ff, 0x0110 }, + { 0x0300, 0x001f }, + { 0x0301, 0x032c }, + { 0x0302, 0x5f21 }, + { 0x0303, 0x4000 }, + { 0x0304, 0x4000 }, + { 0x0305, 0x06d5 }, + { 0x0306, 0x8000 }, + { 0x0307, 0x0700 }, + { 0x0310, 0x4560 }, + { 0x0311, 0xa4a8 }, + { 0x0312, 0x7418 }, + { 0x0313, 0x0000 }, + { 0x0314, 0x0006 }, + { 0x0315, 0xffff }, + { 0x0316, 0xc400 }, + { 0x0317, 0x0000 }, + { 0x0330, 0x00a6 }, + { 0x0331, 0x04c3 }, + { 0x0332, 0x27c8 }, + { 0x0333, 0xbf50 }, + { 0x0334, 0x0045 }, + { 0x0335, 0x0007 }, + { 0x0336, 0x7418 }, + { 0x0337, 0x0501 }, + { 0x0338, 0x0000 }, + { 0x0339, 0x0010 }, + { 0x033a, 0x1010 }, + { 0x03c0, 0x7e00 }, + { 0x03c1, 0x8000 }, + { 0x03c2, 0x8000 }, + { 0x03c3, 0x8000 }, + { 0x03c4, 0x8000 }, + { 0x03c5, 0x8000 }, + { 0x03c6, 0x8000 }, + { 0x03c7, 0x8000 }, + { 0x03c8, 0x8000 }, + { 0x03c9, 0x8000 }, + { 0x03ca, 0x8000 }, + { 0x03cb, 0x8000 }, + { 0x03cc, 0x8000 }, + { 0x03d0, 0x0000 }, + { 0x03d1, 0x0000 }, + { 0x03d2, 0x0000 }, + { 0x03d3, 0x0000 }, + { 0x03d4, 0x2000 }, + { 0x03d5, 0x2000 }, + { 0x03d6, 0x0000 }, + { 0x03d7, 0x0000 }, + { 0x03d8, 0x2000 }, + { 0x03d9, 0x2000 }, + { 0x03da, 0x2000 }, + { 0x03db, 0x2000 }, + { 0x03dc, 0x0000 }, + { 0x03dd, 0x0000 }, + { 0x03de, 0x0000 }, + { 0x03df, 0x2000 }, + { 0x03e0, 0x0000 }, + { 0x03e1, 0x0000 }, + { 0x03e2, 0x0000 }, + { 0x03e3, 0x0000 }, + { 0x03e4, 0x0000 }, + { 0x03e5, 0x0000 }, + { 0x03e6, 0x0000 }, + { 0x03e7, 0x0000 }, + { 0x03e8, 0x0000 }, + { 0x03e9, 0x0000 }, + { 0x03ea, 0x0000 }, + { 0x03eb, 0x0000 }, + { 0x03ec, 0x0000 }, + { 0x03ed, 0x0000 }, + { 0x03ee, 0x0000 }, + { 0x03ef, 0x0000 }, + { 0x03f0, 0x0800 }, + { 0x03f1, 0x0800 }, + { 0x03f2, 0x0800 }, + { 0x03f3, 0x0800 }, + { 0x03fe, 0x0000 }, + { 0x03ff, 0x0000 }, + { 0x07f0, 0x0000 }, + { 0x07fa, 0x0000 }, +}; + +static const struct reg_default rt5663_reg[] = { + { 0x0000, 0x0000 }, + { 0x0002, 0x0008 }, + { 0x0005, 0x1000 }, + { 0x0006, 0x1000 }, + { 0x000a, 0x0000 }, + { 0x0010, 0x000f }, + { 0x0015, 0x42c1 }, + { 0x0016, 0x0000 }, + { 0x0018, 0x000b }, + { 0x0019, 0xafaf }, + { 0x001c, 0x2f2f }, + { 0x001f, 0x0000 }, + { 0x0022, 0x5757 }, + { 0x0023, 0x0039 }, + { 0x0026, 0xc0c0 }, + { 0x0029, 0x8080 }, + { 0x002a, 0xa0a0 }, + { 0x002c, 0x000c }, + { 0x002d, 0x0000 }, + { 0x0040, 0x0808 }, + { 0x0061, 0x0000 }, + { 0x0062, 0x0000 }, + { 0x0063, 0x003e }, + { 0x0064, 0x0000 }, + { 0x0065, 0x0000 }, + { 0x0066, 0x0000 }, + { 0x006b, 0x0000 }, + { 0x006e, 0x0000 }, + { 0x006f, 0x0000 }, + { 0x0070, 0x8020 }, + { 0x0073, 0x1000 }, + { 0x0074, 0xe400 }, + { 0x0075, 0x0002 }, + { 0x0076, 0x0001 }, + { 0x0077, 0x00f0 }, + { 0x0078, 0x0000 }, + { 0x0079, 0x0000 }, + { 0x007a, 0x0123 }, + { 0x007b, 0x8003 }, + { 0x0080, 0x0000 }, + { 0x0081, 0x0000 }, + { 0x0082, 0x0000 }, + { 0x0083, 0x0000 }, + { 0x0084, 0x0000 }, + { 0x0086, 0x0008 }, + { 0x0087, 0x0000 }, + { 0x008a, 0x0000 }, + { 0x008b, 0x0000 }, + { 0x008c, 0x0003 }, + { 0x008e, 0x0004 }, + { 0x008f, 0x1000 }, + { 0x0090, 0x0646 }, + { 0x0091, 0x0e3e }, + { 0x0092, 0x1071 }, + { 0x0093, 0x0000 }, + { 0x0094, 0x0080 }, + { 0x0097, 0x0000 }, + { 0x0098, 0x0000 }, + { 0x009a, 0x0000 }, + { 0x009f, 0x0000 }, + { 0x00ae, 0x2000 }, + { 0x00af, 0x0000 }, + { 0x00b6, 0x0000 }, + { 0x00b7, 0x0000 }, + { 0x00b8, 0x0000 }, + { 0x00ba, 0x0000 }, + { 0x00bb, 0x0000 }, + { 0x00be, 0x0000 }, + { 0x00bf, 0x0000 }, + { 0x00c0, 0x0000 }, + { 0x00c1, 0x0000 }, + { 0x00c5, 0x0000 }, + { 0x00cb, 0xa02f }, + { 0x00cc, 0x0000 }, + { 0x00cd, 0x0e02 }, + { 0x00d9, 0x08f9 }, + { 0x00db, 0x0008 }, + { 0x00dc, 0x00c0 }, + { 0x00dd, 0x6724 }, + { 0x00de, 0x3131 }, + { 0x00df, 0x0008 }, + { 0x00e0, 0x4000 }, + { 0x00e1, 0x3131 }, + { 0x00e2, 0x0043 }, + { 0x00e4, 0x400b }, + { 0x00e5, 0x8031 }, + { 0x00e6, 0x3080 }, + { 0x00e7, 0x4100 }, + { 0x00e8, 0x1400 }, + { 0x00e9, 0xe00a }, + { 0x00ea, 0x0404 }, + { 0x00eb, 0x0404 }, + { 0x00ec, 0xb320 }, + { 0x00ed, 0x0000 }, + { 0x00f4, 0x0000 }, + { 0x00f6, 0x0000 }, + { 0x00f8, 0x0000 }, + { 0x00fa, 0x8000 }, + { 0x00fd, 0x0001 }, + { 0x00fe, 0x10ec }, + { 0x00ff, 0x6406 }, + { 0x0100, 0xa0a0 }, + { 0x0108, 0x4444 }, + { 0x0109, 0x4444 }, + { 0x010a, 0xaaaa }, + { 0x010b, 0x00a0 }, + { 0x010c, 0x8aaa }, + { 0x010d, 0xaaaa }, + { 0x010e, 0x2aaa }, + { 0x010f, 0x002a }, + { 0x0110, 0xa0a4 }, + { 0x0111, 0x4602 }, + { 0x0112, 0x0101 }, + { 0x0113, 0x2000 }, + { 0x0114, 0x0000 }, + { 0x0116, 0x0000 }, + { 0x0117, 0x0f00 }, + { 0x0118, 0x0006 }, + { 0x0125, 0x2224 }, + { 0x0126, 0x5550 }, + { 0x0127, 0x0400 }, + { 0x0128, 0x7711 }, + { 0x0132, 0x0004 }, + { 0x0137, 0x5441 }, + { 0x0139, 0x79a1 }, + { 0x013a, 0x30c0 }, + { 0x013b, 0x2000 }, + { 0x013c, 0x2005 }, + { 0x013d, 0x30c0 }, + { 0x013e, 0x0000 }, + { 0x0140, 0x3700 }, + { 0x0141, 0x1f00 }, + { 0x0144, 0x0000 }, + { 0x0145, 0x0002 }, + { 0x0146, 0x0000 }, + { 0x0160, 0x0e80 }, + { 0x0161, 0x0020 }, + { 0x0162, 0x0080 }, + { 0x0163, 0x0800 }, + { 0x0164, 0x0000 }, + { 0x0165, 0x0000 }, + { 0x0166, 0x0000 }, + { 0x0167, 0x1417 }, + { 0x0168, 0x0017 }, + { 0x0169, 0x0017 }, + { 0x0180, 0x2000 }, + { 0x0181, 0x0000 }, + { 0x0182, 0x0000 }, + { 0x0183, 0x2000 }, + { 0x0184, 0x0000 }, + { 0x0185, 0x0000 }, + { 0x01b0, 0x4b30 }, + { 0x01b1, 0x0000 }, + { 0x01b2, 0xd870 }, + { 0x01b3, 0x0000 }, + { 0x01b4, 0x0030 }, + { 0x01b5, 0x5757 }, + { 0x01b6, 0x5757 }, + { 0x01b7, 0x5757 }, + { 0x01b8, 0x5757 }, + { 0x01c0, 0x433d }, + { 0x01c1, 0x0540 }, + { 0x01c2, 0x0000 }, + { 0x01c3, 0x0000 }, + { 0x01c4, 0x0000 }, + { 0x01c5, 0x0009 }, + { 0x01c6, 0x0018 }, + { 0x01c7, 0x002a }, + { 0x01c8, 0x004c }, + { 0x01c9, 0x0097 }, + { 0x01ca, 0x01c3 }, + { 0x01cb, 0x03e9 }, + { 0x01cc, 0x1389 }, + { 0x01cd, 0xc351 }, + { 0x01ce, 0x0000 }, + { 0x01cf, 0x0000 }, + { 0x01d0, 0x0000 }, + { 0x01d1, 0x0000 }, + { 0x01d2, 0x0000 }, + { 0x01d3, 0x003c }, + { 0x01d4, 0x5757 }, + { 0x01d5, 0x5757 }, + { 0x01d6, 0x5757 }, + { 0x01d7, 0x5757 }, + { 0x01d8, 0x5757 }, + { 0x01d9, 0x5757 }, + { 0x01da, 0x0000 }, + { 0x01db, 0x0000 }, + { 0x01dd, 0x0009 }, + { 0x01de, 0x7f00 }, + { 0x01df, 0x00c8 }, + { 0x01e0, 0x0691 }, + { 0x01e1, 0x0000 }, + { 0x01e2, 0x0000 }, + { 0x01e3, 0x0000 }, + { 0x01e4, 0x0000 }, + { 0x01e5, 0x0040 }, + { 0x01e6, 0x0000 }, + { 0x01e7, 0x0000 }, + { 0x01e8, 0x0000 }, + { 0x01ea, 0x0000 }, + { 0x01eb, 0x0000 }, + { 0x01ec, 0x0000 }, + { 0x01ed, 0x0000 }, + { 0x01ee, 0x0000 }, + { 0x01ef, 0x0000 }, + { 0x01f0, 0x0000 }, + { 0x01f1, 0x0000 }, + { 0x01f2, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x2244 }, + { 0x0202, 0xaaaa }, + { 0x0250, 0x8010 }, + { 0x0251, 0x0000 }, + { 0x0252, 0x028a }, + { 0x02fa, 0x0000 }, + { 0x02fb, 0x0000 }, + { 0x02fc, 0x0000 }, + { 0x0300, 0x0000 }, + { 0x03d0, 0x0000 }, + { 0x03d1, 0x0000 }, + { 0x03d2, 0x0000 }, + { 0x03d3, 0x0000 }, + { 0x03d4, 0x2000 }, + { 0x03d5, 0x2000 }, + { 0x03d6, 0x0000 }, + { 0x03d7, 0x0000 }, + { 0x03d8, 0x2000 }, + { 0x03d9, 0x2000 }, + { 0x03da, 0x2000 }, + { 0x03db, 0x2000 }, + { 0x03dc, 0x0000 }, + { 0x03dd, 0x0000 }, + { 0x03de, 0x0000 }, + { 0x03df, 0x2000 }, + { 0x03e0, 0x0000 }, + { 0x03e1, 0x0000 }, + { 0x03e2, 0x0000 }, + { 0x03e3, 0x0000 }, + { 0x03e4, 0x0000 }, + { 0x03e5, 0x0000 }, + { 0x03e6, 0x0000 }, + { 0x03e7, 0x0000 }, + { 0x03e8, 0x0000 }, + { 0x03e9, 0x0000 }, + { 0x03ea, 0x0000 }, + { 0x03eb, 0x0000 }, + { 0x03ec, 0x0000 }, + { 0x03ed, 0x0000 }, + { 0x03ee, 0x0000 }, + { 0x03ef, 0x0000 }, + { 0x03f0, 0x0800 }, + { 0x03f1, 0x0800 }, + { 0x03f2, 0x0800 }, + { 0x03f3, 0x0800 }, +}; + +static bool rt5663_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5663_RESET: + case RT5663_SIL_DET_CTL: + case RT5663_HP_IMP_GAIN_2: + case RT5663_AD_DA_MIXER: + case RT5663_FRAC_DIV_2: + case RT5663_MICBIAS_1: + case RT5663_ASRC_11_2: + case RT5663_ADC_EQ_1: + case RT5663_INT_ST_1: + case RT5663_INT_ST_2: + case RT5663_GPIO_STA: + case RT5663_SIN_GEN_1: + case RT5663_IL_CMD_1: + case RT5663_IL_CMD_5: + case RT5663_IL_CMD_PWRSAV1: + case RT5663_EM_JACK_TYPE_1: + case RT5663_EM_JACK_TYPE_2: + case RT5663_EM_JACK_TYPE_3: + case RT5663_JD_CTRL2: + case RT5663_VENDOR_ID: + case RT5663_VENDOR_ID_1: + case RT5663_VENDOR_ID_2: + case RT5663_PLL_INT_REG: + case RT5663_SOFT_RAMP: + case RT5663_STO_DRE_1: + case RT5663_STO_DRE_5: + case RT5663_STO_DRE_6: + case RT5663_STO_DRE_7: + case RT5663_MIC_DECRO_1: + case RT5663_MIC_DECRO_4: + case RT5663_HP_IMP_SEN_1: + case RT5663_HP_IMP_SEN_3: + case RT5663_HP_IMP_SEN_4: + case RT5663_HP_IMP_SEN_5: + case RT5663_HP_CALIB_1_1: + case RT5663_HP_CALIB_9: + case RT5663_HP_CALIB_ST1: + case RT5663_HP_CALIB_ST2: + case RT5663_HP_CALIB_ST3: + case RT5663_HP_CALIB_ST4: + case RT5663_HP_CALIB_ST5: + case RT5663_HP_CALIB_ST6: + case RT5663_HP_CALIB_ST7: + case RT5663_HP_CALIB_ST8: + case RT5663_HP_CALIB_ST9: + case RT5663_ANA_JD: + return true; + default: + return false; + } +} + +static bool rt5663_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5663_RESET: + case RT5663_HP_OUT_EN: + case RT5663_HP_LCH_DRE: + case RT5663_HP_RCH_DRE: + case RT5663_CALIB_BST: + case RT5663_RECMIX: + case RT5663_SIL_DET_CTL: + case RT5663_PWR_SAV_SILDET: + case RT5663_SIDETONE_CTL: + case RT5663_STO1_DAC_DIG_VOL: + case RT5663_STO1_ADC_DIG_VOL: + case RT5663_STO1_BOOST: + case RT5663_HP_IMP_GAIN_1: + case RT5663_HP_IMP_GAIN_2: + case RT5663_STO1_ADC_MIXER: + case RT5663_AD_DA_MIXER: + case RT5663_STO_DAC_MIXER: + case RT5663_DIG_SIDE_MIXER: + case RT5663_BYPASS_STO_DAC: + case RT5663_CALIB_REC_MIX: + case RT5663_PWR_DIG_1: + case RT5663_PWR_DIG_2: + case RT5663_PWR_ANLG_1: + case RT5663_PWR_ANLG_2: + case RT5663_PWR_ANLG_3: + case RT5663_PWR_MIXER: + case RT5663_SIG_CLK_DET: + case RT5663_PRE_DIV_GATING_1: + case RT5663_PRE_DIV_GATING_2: + case RT5663_I2S1_SDP: + case RT5663_ADDA_CLK_1: + case RT5663_ADDA_RST: + case RT5663_FRAC_DIV_1: + case RT5663_FRAC_DIV_2: + case RT5663_TDM_1: + case RT5663_TDM_2: + case RT5663_TDM_3: + case RT5663_TDM_4: + case RT5663_TDM_5: + case RT5663_GLB_CLK: + case RT5663_PLL_1: + case RT5663_PLL_2: + case RT5663_ASRC_1: + case RT5663_ASRC_2: + case RT5663_ASRC_4: + case RT5663_DUMMY_REG: + case RT5663_ASRC_8: + case RT5663_ASRC_9: + case RT5663_ASRC_11: + case RT5663_DEPOP_1: + case RT5663_DEPOP_2: + case RT5663_DEPOP_3: + case RT5663_HP_CHARGE_PUMP_1: + case RT5663_HP_CHARGE_PUMP_2: + case RT5663_MICBIAS_1: + case RT5663_RC_CLK: + case RT5663_ASRC_11_2: + case RT5663_DUMMY_REG_2: + case RT5663_REC_PATH_GAIN: + case RT5663_AUTO_1MRC_CLK: + case RT5663_ADC_EQ_1: + case RT5663_ADC_EQ_2: + case RT5663_IRQ_1: + case RT5663_IRQ_2: + case RT5663_IRQ_3: + case RT5663_IRQ_4: + case RT5663_IRQ_5: + case RT5663_INT_ST_1: + case RT5663_INT_ST_2: + case RT5663_GPIO_1: + case RT5663_GPIO_2: + case RT5663_GPIO_STA: + case RT5663_SIN_GEN_1: + case RT5663_SIN_GEN_2: + case RT5663_SIN_GEN_3: + case RT5663_SOF_VOL_ZC1: + case RT5663_IL_CMD_1: + case RT5663_IL_CMD_2: + case RT5663_IL_CMD_3: + case RT5663_IL_CMD_4: + case RT5663_IL_CMD_5: + case RT5663_IL_CMD_6: + case RT5663_IL_CMD_7: + case RT5663_IL_CMD_8: + case RT5663_IL_CMD_PWRSAV1: + case RT5663_IL_CMD_PWRSAV2: + case RT5663_EM_JACK_TYPE_1: + case RT5663_EM_JACK_TYPE_2: + case RT5663_EM_JACK_TYPE_3: + case RT5663_EM_JACK_TYPE_4: + case RT5663_EM_JACK_TYPE_5: + case RT5663_EM_JACK_TYPE_6: + case RT5663_STO1_HPF_ADJ1: + case RT5663_STO1_HPF_ADJ2: + case RT5663_FAST_OFF_MICBIAS: + case RT5663_JD_CTRL1: + case RT5663_JD_CTRL2: + case RT5663_DIG_MISC: + case RT5663_VENDOR_ID: + case RT5663_VENDOR_ID_1: + case RT5663_VENDOR_ID_2: + case RT5663_DIG_VOL_ZCD: + case RT5663_ANA_BIAS_CUR_1: + case RT5663_ANA_BIAS_CUR_2: + case RT5663_ANA_BIAS_CUR_3: + case RT5663_ANA_BIAS_CUR_4: + case RT5663_ANA_BIAS_CUR_5: + case RT5663_ANA_BIAS_CUR_6: + case RT5663_BIAS_CUR_5: + case RT5663_BIAS_CUR_6: + case RT5663_BIAS_CUR_7: + case RT5663_BIAS_CUR_8: + case RT5663_DACREF_LDO: + case RT5663_DUMMY_REG_3: + case RT5663_BIAS_CUR_9: + case RT5663_DUMMY_REG_4: + case RT5663_VREFADJ_OP: + case RT5663_VREF_RECMIX: + case RT5663_CHARGE_PUMP_1: + case RT5663_CHARGE_PUMP_1_2: + case RT5663_CHARGE_PUMP_1_3: + case RT5663_CHARGE_PUMP_2: + case RT5663_DIG_IN_PIN1: + case RT5663_PAD_DRV_CTL: + case RT5663_PLL_INT_REG: + case RT5663_CHOP_DAC_L: + case RT5663_CHOP_ADC: + case RT5663_CALIB_ADC: + case RT5663_CHOP_DAC_R: + case RT5663_DUMMY_CTL_DACLR: + case RT5663_DUMMY_REG_5: + case RT5663_SOFT_RAMP: + case RT5663_TEST_MODE_1: + case RT5663_TEST_MODE_2: + case RT5663_TEST_MODE_3: + case RT5663_STO_DRE_1: + case RT5663_STO_DRE_2: + case RT5663_STO_DRE_3: + case RT5663_STO_DRE_4: + case RT5663_STO_DRE_5: + case RT5663_STO_DRE_6: + case RT5663_STO_DRE_7: + case RT5663_STO_DRE_8: + case RT5663_STO_DRE_9: + case RT5663_STO_DRE_10: + case RT5663_MIC_DECRO_1: + case RT5663_MIC_DECRO_2: + case RT5663_MIC_DECRO_3: + case RT5663_MIC_DECRO_4: + case RT5663_MIC_DECRO_5: + case RT5663_MIC_DECRO_6: + case RT5663_HP_DECRO_1: + case RT5663_HP_DECRO_2: + case RT5663_HP_DECRO_3: + case RT5663_HP_DECRO_4: + case RT5663_HP_DECOUP: + case RT5663_HP_IMP_SEN_MAP8: + case RT5663_HP_IMP_SEN_MAP9: + case RT5663_HP_IMP_SEN_MAP10: + case RT5663_HP_IMP_SEN_MAP11: + case RT5663_HP_IMP_SEN_1: + case RT5663_HP_IMP_SEN_2: + case RT5663_HP_IMP_SEN_3: + case RT5663_HP_IMP_SEN_4: + case RT5663_HP_IMP_SEN_5: + case RT5663_HP_IMP_SEN_6: + case RT5663_HP_IMP_SEN_7: + case RT5663_HP_IMP_SEN_8: + case RT5663_HP_IMP_SEN_9: + case RT5663_HP_IMP_SEN_10: + case RT5663_HP_IMP_SEN_11: + case RT5663_HP_IMP_SEN_12: + case RT5663_HP_IMP_SEN_13: + case RT5663_HP_IMP_SEN_14: + case RT5663_HP_IMP_SEN_15: + case RT5663_HP_IMP_SEN_16: + case RT5663_HP_IMP_SEN_17: + case RT5663_HP_IMP_SEN_18: + case RT5663_HP_IMP_SEN_19: + case RT5663_HP_IMPSEN_DIG5: + case RT5663_HP_IMPSEN_MAP1: + case RT5663_HP_IMPSEN_MAP2: + case RT5663_HP_IMPSEN_MAP3: + case RT5663_HP_IMPSEN_MAP4: + case RT5663_HP_IMPSEN_MAP5: + case RT5663_HP_IMPSEN_MAP7: + case RT5663_HP_LOGIC_1: + case RT5663_HP_LOGIC_2: + case RT5663_HP_CALIB_1: + case RT5663_HP_CALIB_1_1: + case RT5663_HP_CALIB_2: + case RT5663_HP_CALIB_3: + case RT5663_HP_CALIB_4: + case RT5663_HP_CALIB_5: + case RT5663_HP_CALIB_5_1: + case RT5663_HP_CALIB_6: + case RT5663_HP_CALIB_7: + case RT5663_HP_CALIB_9: + case RT5663_HP_CALIB_10: + case RT5663_HP_CALIB_11: + case RT5663_HP_CALIB_ST1: + case RT5663_HP_CALIB_ST2: + case RT5663_HP_CALIB_ST3: + case RT5663_HP_CALIB_ST4: + case RT5663_HP_CALIB_ST5: + case RT5663_HP_CALIB_ST6: + case RT5663_HP_CALIB_ST7: + case RT5663_HP_CALIB_ST8: + case RT5663_HP_CALIB_ST9: + case RT5663_HP_AMP_DET: + case RT5663_DUMMY_REG_6: + case RT5663_HP_BIAS: + case RT5663_CBJ_1: + case RT5663_CBJ_2: + case RT5663_CBJ_3: + case RT5663_DUMMY_1: + case RT5663_DUMMY_2: + case RT5663_DUMMY_3: + case RT5663_ANA_JD: + case RT5663_ADC_LCH_LPF1_A1: + case RT5663_ADC_RCH_LPF1_A1: + case RT5663_ADC_LCH_LPF1_H0: + case RT5663_ADC_RCH_LPF1_H0: + case RT5663_ADC_LCH_BPF1_A1: + case RT5663_ADC_RCH_BPF1_A1: + case RT5663_ADC_LCH_BPF1_A2: + case RT5663_ADC_RCH_BPF1_A2: + case RT5663_ADC_LCH_BPF1_H0: + case RT5663_ADC_RCH_BPF1_H0: + case RT5663_ADC_LCH_BPF2_A1: + case RT5663_ADC_RCH_BPF2_A1: + case RT5663_ADC_LCH_BPF2_A2: + case RT5663_ADC_RCH_BPF2_A2: + case RT5663_ADC_LCH_BPF2_H0: + case RT5663_ADC_RCH_BPF2_H0: + case RT5663_ADC_LCH_BPF3_A1: + case RT5663_ADC_RCH_BPF3_A1: + case RT5663_ADC_LCH_BPF3_A2: + case RT5663_ADC_RCH_BPF3_A2: + case RT5663_ADC_LCH_BPF3_H0: + case RT5663_ADC_RCH_BPF3_H0: + case RT5663_ADC_LCH_BPF4_A1: + case RT5663_ADC_RCH_BPF4_A1: + case RT5663_ADC_LCH_BPF4_A2: + case RT5663_ADC_RCH_BPF4_A2: + case RT5663_ADC_LCH_BPF4_H0: + case RT5663_ADC_RCH_BPF4_H0: + case RT5663_ADC_LCH_HPF1_A1: + case RT5663_ADC_RCH_HPF1_A1: + case RT5663_ADC_LCH_HPF1_H0: + case RT5663_ADC_RCH_HPF1_H0: + case RT5663_ADC_EQ_PRE_VOL_L: + case RT5663_ADC_EQ_PRE_VOL_R: + case RT5663_ADC_EQ_POST_VOL_L: + case RT5663_ADC_EQ_POST_VOL_R: + return true; + default: + return false; + } +} + +static bool rt5668_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5663_RESET: + case RT5668_CBJ_TYPE_2: + case RT5668_PDM_OUT_CTL: + case RT5668_PDM_I2C_DATA_CTL1: + case RT5668_PDM_I2C_DATA_CTL4: + case RT5668_ALC_BK_GAIN: + case RT5663_PLL_2: + case RT5663_MICBIAS_1: + case RT5663_ADC_EQ_1: + case RT5663_INT_ST_1: + case RT5668_GPIO_STA: + case RT5663_IL_CMD_1: + case RT5663_IL_CMD_5: + case RT5668_A_JD_CTRL: + case RT5663_JD_CTRL2: + case RT5663_VENDOR_ID: + case RT5663_VENDOR_ID_1: + case RT5663_VENDOR_ID_2: + case RT5663_STO_DRE_1: + case RT5663_STO_DRE_5: + case RT5663_STO_DRE_6: + case RT5663_STO_DRE_7: + case RT5668_MONO_DYNA_6: + case RT5668_STO1_SIL_DET: + case RT5668_MONOL_SIL_DET: + case RT5668_MONOR_SIL_DET: + case RT5668_STO2_DAC_SIL: + case RT5668_MONO_AMP_CAL_ST1: + case RT5668_MONO_AMP_CAL_ST2: + case RT5668_MONO_AMP_CAL_ST3: + case RT5668_MONO_AMP_CAL_ST4: + case RT5663_HP_IMP_SEN_2: + case RT5663_HP_IMP_SEN_3: + case RT5663_HP_IMP_SEN_4: + case RT5663_HP_IMP_SEN_10: + case RT5663_HP_CALIB_1: + case RT5663_HP_CALIB_10: + case RT5663_HP_CALIB_ST1: + case RT5663_HP_CALIB_ST4: + case RT5663_HP_CALIB_ST5: + case RT5663_HP_CALIB_ST6: + case RT5663_HP_CALIB_ST7: + case RT5663_HP_CALIB_ST8: + case RT5663_HP_CALIB_ST9: + case RT5668_HP_CALIB_ST10: + case RT5668_HP_CALIB_ST11: + return true; + default: + return false; + } +} + +static bool rt5668_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5668_LOUT_CTRL: + case RT5668_HP_AMP_2: + case RT5668_MONO_OUT: + case RT5668_MONO_GAIN: + case RT5668_AEC_BST: + case RT5668_IN1_IN2: + case RT5668_IN3_IN4: + case RT5668_INL1_INR1: + case RT5668_CBJ_TYPE_2: + case RT5668_CBJ_TYPE_3: + case RT5668_CBJ_TYPE_4: + case RT5668_CBJ_TYPE_5: + case RT5668_CBJ_TYPE_8: + case RT5668_DAC3_DIG_VOL: + case RT5668_DAC3_CTRL: + case RT5668_MONO_ADC_DIG_VOL: + case RT5668_STO2_ADC_DIG_VOL: + case RT5668_MONO_ADC_BST_GAIN: + case RT5668_STO2_ADC_BST_GAIN: + case RT5668_SIDETONE_CTRL: + case RT5668_MONO1_ADC_MIXER: + case RT5668_STO2_ADC_MIXER: + case RT5668_MONO_DAC_MIXER: + case RT5668_DAC2_SRC_CTRL: + case RT5668_IF_3_4_DATA_CTL: + case RT5668_IF_5_DATA_CTL: + case RT5668_PDM_OUT_CTL: + case RT5668_PDM_I2C_DATA_CTL1: + case RT5668_PDM_I2C_DATA_CTL2: + case RT5668_PDM_I2C_DATA_CTL3: + case RT5668_PDM_I2C_DATA_CTL4: + case RT5668_RECMIX1_NEW: + case RT5668_RECMIX1L_0: + case RT5668_RECMIX1L: + case RT5668_RECMIX1R_0: + case RT5668_RECMIX1R: + case RT5668_RECMIX2_NEW: + case RT5668_RECMIX2_L_2: + case RT5668_RECMIX2_R: + case RT5668_RECMIX2_R_2: + case RT5668_CALIB_REC_LR: + case RT5668_ALC_BK_GAIN: + case RT5668_MONOMIX_GAIN: + case RT5668_MONOMIX_IN_GAIN: + case RT5668_OUT_MIXL_GAIN: + case RT5668_OUT_LMIX_IN_GAIN: + case RT5668_OUT_RMIX_IN_GAIN: + case RT5668_OUT_RMIX_IN_GAIN1: + case RT5668_LOUT_MIXER_CTRL: + case RT5668_PWR_VOL: + case RT5668_ADCDAC_RST: + case RT5668_I2S34_SDP: + case RT5668_I2S5_SDP: + case RT5668_TDM_5: + case RT5668_TDM_6: + case RT5668_TDM_7: + case RT5668_TDM_8: + case RT5668_ASRC_3: + case RT5668_ASRC_6: + case RT5668_ASRC_7: + case RT5668_PLL_TRK_13: + case RT5668_I2S_M_CLK_CTL: + case RT5668_FDIV_I2S34_M_CLK: + case RT5668_FDIV_I2S34_M_CLK2: + case RT5668_FDIV_I2S5_M_CLK: + case RT5668_FDIV_I2S5_M_CLK2: + case RT5668_IRQ_4: + case RT5668_GPIO_3: + case RT5668_GPIO_4: + case RT5668_GPIO_STA: + case RT5668_HP_AMP_DET1: + case RT5668_HP_AMP_DET2: + case RT5668_HP_AMP_DET3: + case RT5668_MID_BD_HP_AMP: + case RT5668_LOW_BD_HP_AMP: + case RT5668_SOF_VOL_ZC2: + case RT5668_ADC_STO2_ADJ1: + case RT5668_ADC_STO2_ADJ2: + case RT5668_A_JD_CTRL: + case RT5668_JD1_TRES_CTRL: + case RT5668_JD2_TRES_CTRL: + case RT5668_JD_CTRL2: + case RT5668_DUM_REG_2: + case RT5668_DUM_REG_3: + case RT5663_VENDOR_ID: + case RT5663_VENDOR_ID_1: + case RT5663_VENDOR_ID_2: + case RT5668_DACADC_DIG_VOL2: + case RT5668_DIG_IN_PIN2: + case RT5668_PAD_DRV_CTL1: + case RT5668_SOF_RAM_DEPOP: + case RT5668_VOL_TEST: + case RT5668_TEST_MODE_3: + case RT5668_TEST_MODE_4: + case RT5663_STO_DRE_9: + case RT5668_MONO_DYNA_1: + case RT5668_MONO_DYNA_2: + case RT5668_MONO_DYNA_3: + case RT5668_MONO_DYNA_4: + case RT5668_MONO_DYNA_5: + case RT5668_MONO_DYNA_6: + case RT5668_STO1_SIL_DET: + case RT5668_MONOL_SIL_DET: + case RT5668_MONOR_SIL_DET: + case RT5668_STO2_DAC_SIL: + case RT5668_PWR_SAV_CTL1: + case RT5668_PWR_SAV_CTL2: + case RT5668_PWR_SAV_CTL3: + case RT5668_PWR_SAV_CTL4: + case RT5668_PWR_SAV_CTL5: + case RT5668_PWR_SAV_CTL6: + case RT5668_MONO_AMP_CAL1: + case RT5668_MONO_AMP_CAL2: + case RT5668_MONO_AMP_CAL3: + case RT5668_MONO_AMP_CAL4: + case RT5668_MONO_AMP_CAL5: + case RT5668_MONO_AMP_CAL6: + case RT5668_MONO_AMP_CAL7: + case RT5668_MONO_AMP_CAL_ST1: + case RT5668_MONO_AMP_CAL_ST2: + case RT5668_MONO_AMP_CAL_ST3: + case RT5668_MONO_AMP_CAL_ST4: + case RT5668_MONO_AMP_CAL_ST5: + case RT5668_HP_IMP_SEN_13: + case RT5668_HP_IMP_SEN_14: + case RT5668_HP_IMP_SEN_6: + case RT5668_HP_IMP_SEN_7: + case RT5668_HP_IMP_SEN_8: + case RT5668_HP_IMP_SEN_9: + case RT5668_HP_IMP_SEN_10: + case RT5668_HP_LOGIC_3: + case RT5668_HP_CALIB_ST10: + case RT5668_HP_CALIB_ST11: + case RT5668_PRO_REG_TBL_4: + case RT5668_PRO_REG_TBL_5: + case RT5668_PRO_REG_TBL_6: + case RT5668_PRO_REG_TBL_7: + case RT5668_PRO_REG_TBL_8: + case RT5668_PRO_REG_TBL_9: + case RT5668_SAR_ADC_INL_1: + case RT5668_SAR_ADC_INL_2: + case RT5668_SAR_ADC_INL_3: + case RT5668_SAR_ADC_INL_4: + case RT5668_SAR_ADC_INL_5: + case RT5668_SAR_ADC_INL_6: + case RT5668_SAR_ADC_INL_7: + case RT5668_SAR_ADC_INL_8: + case RT5668_SAR_ADC_INL_9: + case RT5668_SAR_ADC_INL_10: + case RT5668_SAR_ADC_INL_11: + case RT5668_SAR_ADC_INL_12: + case RT5668_DRC_CTRL_1: + case RT5668_DRC1_CTRL_2: + case RT5668_DRC1_CTRL_3: + case RT5668_DRC1_CTRL_4: + case RT5668_DRC1_CTRL_5: + case RT5668_DRC1_CTRL_6: + case RT5668_DRC1_HD_CTRL_1: + case RT5668_DRC1_HD_CTRL_2: + case RT5668_DRC1_PRI_REG_1: + case RT5668_DRC1_PRI_REG_2: + case RT5668_DRC1_PRI_REG_3: + case RT5668_DRC1_PRI_REG_4: + case RT5668_DRC1_PRI_REG_5: + case RT5668_DRC1_PRI_REG_6: + case RT5668_DRC1_PRI_REG_7: + case RT5668_DRC1_PRI_REG_8: + case RT5668_ALC_PGA_CTL_1: + case RT5668_ALC_PGA_CTL_2: + case RT5668_ALC_PGA_CTL_3: + case RT5668_ALC_PGA_CTL_4: + case RT5668_ALC_PGA_CTL_5: + case RT5668_ALC_PGA_CTL_6: + case RT5668_ALC_PGA_CTL_7: + case RT5668_ALC_PGA_CTL_8: + case RT5668_ALC_PGA_REG_1: + case RT5668_ALC_PGA_REG_2: + case RT5668_ALC_PGA_REG_3: + case RT5668_ADC_EQ_RECOV_1: + case RT5668_ADC_EQ_RECOV_2: + case RT5668_ADC_EQ_RECOV_3: + case RT5668_ADC_EQ_RECOV_4: + case RT5668_ADC_EQ_RECOV_5: + case RT5668_ADC_EQ_RECOV_6: + case RT5668_ADC_EQ_RECOV_7: + case RT5668_ADC_EQ_RECOV_8: + case RT5668_ADC_EQ_RECOV_9: + case RT5668_ADC_EQ_RECOV_10: + case RT5668_ADC_EQ_RECOV_11: + case RT5668_ADC_EQ_RECOV_12: + case RT5668_ADC_EQ_RECOV_13: + case RT5668_VID_HIDDEN: + case RT5668_VID_CUSTOMER: + case RT5668_SCAN_MODE: + case RT5668_I2C_BYPA: + return true; + case RT5663_TDM_1: + case RT5663_DEPOP_3: + case RT5663_ASRC_11_2: + case RT5663_INT_ST_2: + case RT5663_GPIO_STA: + case RT5663_SIN_GEN_1: + case RT5663_SIN_GEN_2: + case RT5663_SIN_GEN_3: + case RT5663_IL_CMD_PWRSAV1: + case RT5663_IL_CMD_PWRSAV2: + case RT5663_EM_JACK_TYPE_1: + case RT5663_EM_JACK_TYPE_2: + case RT5663_EM_JACK_TYPE_3: + case RT5663_EM_JACK_TYPE_4: + case RT5663_FAST_OFF_MICBIAS: + case RT5663_ANA_BIAS_CUR_1: + case RT5663_ANA_BIAS_CUR_2: + case RT5663_BIAS_CUR_9: + case RT5663_DUMMY_REG_4: + case RT5663_VREF_RECMIX: + case RT5663_CHARGE_PUMP_1_2: + case RT5663_CHARGE_PUMP_1_3: + case RT5663_CHARGE_PUMP_2: + case RT5663_CHOP_DAC_R: + case RT5663_DUMMY_CTL_DACLR: + case RT5663_DUMMY_REG_5: + case RT5663_SOFT_RAMP: + case RT5663_TEST_MODE_1: + case RT5663_STO_DRE_10: + case RT5663_MIC_DECRO_1: + case RT5663_MIC_DECRO_2: + case RT5663_MIC_DECRO_3: + case RT5663_MIC_DECRO_4: + case RT5663_MIC_DECRO_5: + case RT5663_MIC_DECRO_6: + case RT5663_HP_DECRO_1: + case RT5663_HP_DECRO_2: + case RT5663_HP_DECRO_3: + case RT5663_HP_DECRO_4: + case RT5663_HP_DECOUP: + case RT5663_HP_IMPSEN_MAP4: + case RT5663_HP_IMPSEN_MAP5: + case RT5663_HP_IMPSEN_MAP7: + case RT5663_HP_CALIB_1: + case RT5663_CBJ_1: + case RT5663_CBJ_2: + case RT5663_CBJ_3: + return false; + default: + return rt5663_readable_register(dev, reg); + } +} + +static const DECLARE_TLV_DB_SCALE(rt5663_hp_vol_tlv, -2400, 150, 0); +static const DECLARE_TLV_DB_SCALE(rt5668_hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(in_bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5663_if1_adc_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5663_if1_adc_enum, RT5663_TDM_2, + RT5663_DATA_SWAP_ADCDAT1_SHIFT, rt5663_if1_adc_data_select); + +static void rt5663_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + if (enable) { + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_EN_4BTN_INL_MASK, RT5668_EN_4BTN_INL_EN); + /* reset in-line command */ + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_RESET); + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_NOR); + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + snd_soc_update_bits(codec, RT5663_IRQ_3, + RT5668_EN_IRQ_INLINE_MASK, + RT5668_EN_IRQ_INLINE_NOR); + break; + case CODEC_TYPE_RT5663: + snd_soc_update_bits(codec, RT5663_IRQ_2, + RT5663_EN_IRQ_INLINE_MASK, + RT5663_EN_IRQ_INLINE_NOR); + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + } else { + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + snd_soc_update_bits(codec, RT5663_IRQ_3, + RT5668_EN_IRQ_INLINE_MASK, + RT5668_EN_IRQ_INLINE_BYP); + break; + case CODEC_TYPE_RT5663: + snd_soc_update_bits(codec, RT5663_IRQ_2, + RT5663_EN_IRQ_INLINE_MASK, + RT5663_EN_IRQ_INLINE_BYP); + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_EN_4BTN_INL_MASK, RT5668_EN_4BTN_INL_DIS); + /* reset in-line command */ + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_RESET); + snd_soc_update_bits(codec, RT5663_IL_CMD_6, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_NOR); + } +} + +/** + * rt5668_jack_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5668_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5663_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30}; + + dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert); + if (jack_insert) { + snd_soc_write(codec, RT5668_CBJ_TYPE_2, 0x8040); + snd_soc_write(codec, RT5668_CBJ_TYPE_3, 0x1484); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_update_bits(codec, RT5663_RC_CLK, + RT5668_DIG_1M_CLK_MASK, RT5668_DIG_1M_CLK_EN); + snd_soc_update_bits(codec, RT5663_RECMIX, 0x8, 0x8); + + while (i < 5) { + msleep(sleep_time[i]); + val = snd_soc_read(codec, RT5668_CBJ_TYPE_2) & 0x0003; + if (val == 0x1 || val == 0x2 || val == 0x3) + break; + dev_dbg(codec->dev, "%s: MX-0011 val=%x sleep %d\n", + __func__, val, sleep_time[i]); + i++; + } + dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + switch (val) { + case 1: + case 2: + rt5668->jack_type = SND_JACK_HEADSET; + rt5663_enable_push_button_irq(codec, true); + break; + default: + snd_soc_dapm_disable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + rt5668->jack_type = SND_JACK_HEADPHONE; + break; + } + } else { + snd_soc_update_bits(codec, RT5663_RECMIX, 0x8, 0x0); + + if (rt5668->jack_type == SND_JACK_HEADSET) { + rt5663_enable_push_button_irq(codec, false); + snd_soc_dapm_disable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + } + rt5668->jack_type = 0; + } + + dev_dbg(codec->dev, "jack_type = %d\n", rt5668->jack_type); + return rt5668->jack_type; +} + +/** + * rt5663_jack_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30}; + + dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert); + + if (jack_insert) { + snd_soc_update_bits(codec, RT5663_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, RT5668_DIG_GATE_CTRL_EN); + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5663_SI_HP_MASK | RT5668_OSW_HP_L_MASK | + RT5668_OSW_HP_R_MASK, RT5663_SI_HP_EN | + RT5668_OSW_HP_L_DIS | RT5668_OSW_HP_R_DIS); + snd_soc_update_bits(codec, RT5663_DUMMY_1, + RT5663_EMB_CLK_MASK | RT5663_HPA_CPL_BIAS_MASK | + RT5663_HPA_CPR_BIAS_MASK, RT5663_EMB_CLK_EN | + RT5663_HPA_CPL_BIAS_1 | RT5663_HPA_CPR_BIAS_1); + snd_soc_update_bits(codec, RT5663_CBJ_1, + RT5663_INBUF_CBJ_BST1_MASK | RT5663_CBJ_SENSE_BST1_MASK, + RT5663_INBUF_CBJ_BST1_ON | RT5663_CBJ_SENSE_BST1_L); + snd_soc_update_bits(codec, RT5663_IL_CMD_2, + RT5663_PWR_MIC_DET_MASK, RT5663_PWR_MIC_DET_ON); + /* BST1 power on for JD */ + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5668_PWR_BST1_MASK, RT5668_PWR_BST1_ON); + snd_soc_update_bits(codec, RT5663_EM_JACK_TYPE_1, + RT5663_CBJ_DET_MASK | RT5663_EXT_JD_MASK | + RT5663_POL_EXT_JD_MASK, RT5663_CBJ_DET_EN | + RT5663_EXT_JD_EN | RT5663_POL_EXT_JD_EN); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5668_PWR_MB_MASK | RT5668_LDO1_DVO_MASK | + RT5668_AMP_HP_MASK, RT5668_PWR_MB | + RT5668_LDO1_DVO_0_9V | RT5668_AMP_HP_3X); + snd_soc_update_bits(codec, RT5663_AUTO_1MRC_CLK, + RT5668_IRQ_POW_SAV_MASK, RT5668_IRQ_POW_SAV_EN); + snd_soc_update_bits(codec, RT5663_IRQ_1, + RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN); + while (i < 5) { + msleep(sleep_time[i]); + val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) & + 0x0003; + i++; + if (val == 0x1 || val == 0x2 || val == 0x3) + break; + dev_dbg(codec->dev, "%s: MX-00e7 val=%x sleep %d\n", + __func__, val, sleep_time[i]); + } + dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + switch (val) { + case 1: + case 2: + rt5663->jack_type = SND_JACK_HEADSET; + rt5663_enable_push_button_irq(codec, true); + break; + default: + rt5663->jack_type = SND_JACK_HEADPHONE; + break; + } + } else { + if (rt5663->jack_type == SND_JACK_HEADSET) + rt5663_enable_push_button_irq(codec, false); + rt5663->jack_type = 0; + } + + dev_dbg(codec->dev, "jack_type = %d\n", rt5663->jack_type); + return rt5663->jack_type; +} + +int rt5663_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5663_IL_CMD_5); + dev_dbg(codec->dev, "%s: val=0x%x\n", __func__, val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5663_IL_CMD_5, val); + + return btn_type; +} + +static irqreturn_t rt5663_irq(int irq, void *data) +{ + struct rt5663_priv *rt5663 = data; + + dev_dbg(rt5663->codec->dev, "%s IRQ queue work\n", __func__); + + queue_delayed_work(system_wq, &rt5663->jack_detect_work, + msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +int rt5663_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + rt5663->hs_jack = hs_jack; + + rt5663_irq(0, rt5663); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5663_set_jack_detect); + +static bool rt5663_check_jd_status(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + int val = snd_soc_read(codec, RT5663_INT_ST_1); + + dev_dbg(codec->dev, "%s val=%x\n", __func__, val); + + /* JD1 */ + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + return !(val & 0x2000); + case CODEC_TYPE_RT5663: + return !(val & 0x1000); + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + + return false; +} + +static void rt5663_jack_detect_work(struct work_struct *work) +{ + struct rt5663_priv *rt5663 = + container_of(work, struct rt5663_priv, jack_detect_work.work); + struct snd_soc_codec *codec = rt5663->codec; + int btn_type, report = 0; + + if (!codec) + return; + + if (rt5663_check_jd_status(codec)) { + /* jack in */ + if (rt5663->jack_type == 0) { + /* jack was out, report jack type */ + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + report = rt5668_jack_detect(rt5663->codec, 1); + break; + case CODEC_TYPE_RT5663: + report = rt5663_jack_detect(rt5663->codec, 1); + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + } else { + /* jack is already in, report button event */ + report = SND_JACK_HEADSET; + btn_type = rt5663_button_detect(rt5663->codec); + /** + * rt5663 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5663->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + /* button release or spurious interrput*/ + if (btn_type == 0) + report = rt5663->jack_type; + } + } else { + /* jack out */ + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + report = rt5668_jack_detect(rt5663->codec, 0); + break; + case CODEC_TYPE_RT5663: + report = rt5663_jack_detect(rt5663->codec, 0); + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + } + dev_dbg(codec->dev, "%s jack report: 0x%04x\n", __func__, report); + snd_soc_jack_report(rt5663->hs_jack, report, SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); +} + +static const struct snd_kcontrol_new rt5663_snd_controls[] = { + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC Playback Volume", RT5663_STO1_DAC_DIG_VOL, + RT5668_DAC_L1_VOL_SHIFT + 1, RT5668_DAC_R1_VOL_SHIFT + 1, + 87, 0, dac_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5663_STO1_ADC_DIG_VOL, + RT5668_ADC_L_MUTE_SHIFT, RT5668_ADC_R_MUTE_SHIFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5663_STO1_ADC_DIG_VOL, + RT5668_ADC_L_VOL_SHIFT + 1, RT5668_ADC_R_VOL_SHIFT + 1, + 63, 0, adc_vol_tlv), +}; + +static const struct snd_kcontrol_new rt5668_specific_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_HP_LCH_DRE, + RT5663_HP_RCH_DRE, RT5668_GAIN_HP_SHIFT, 15, 1, + rt5668_hp_vol_tlv), + /* Mic Boost Volume */ + SOC_SINGLE_TLV("IN1 Capture Volume", RT5668_AEC_BST, + RT5668_GAIN_CBJ_SHIFT, 8, 0, in_bst_tlv), +}; + +static const struct snd_kcontrol_new rt5663_specific_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, + RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, + rt5663_hp_vol_tlv), + /* Mic Boost Volume*/ + SOC_SINGLE_TLV("IN1 Capture Volume", RT5663_CBJ_2, + RT5663_GAIN_BST1_SHIFT, 8, 0, in_bst_tlv), + /* Data Swap for Slot0/1 in ADCDAT1 */ + SOC_ENUM("IF1 ADC Data Swap", rt5663_if1_adc_enum), +}; + +static int rt5663_is_sys_clk_from_pll(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + val = snd_soc_read(codec, RT5663_GLB_CLK); + val &= RT5663_SCLK_SRC_MASK; + if (val == RT5663_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int rt5663_is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + switch (w->shift) { + case RT5668_ADC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_3; + shift = RT5668_AD_STO1_TRACK_SHIFT; + break; + case RT5668_DAC_STO1_ASRC_SHIFT: + reg = RT5663_ASRC_2; + shift = RT5668_DA_STO1_TRACK_SHIFT; + break; + default: + return 0; + } + } else { + switch (w->shift) { + case RT5663_ADC_STO1_ASRC_SHIFT: + reg = RT5663_ASRC_2; + shift = RT5663_AD_STO1_TRACK_SHIFT; + break; + case RT5663_DAC_STO1_ASRC_SHIFT: + reg = RT5663_ASRC_2; + shift = RT5663_DA_STO1_TRACK_SHIFT; + break; + default: + return 0; + } + } + + val = (snd_soc_read(codec, reg) >> shift) & 0x7; + + if (val) + return 1; + + return 0; +} + +static int rt5663_i2s_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + int da_asrc_en, ad_asrc_en; + + da_asrc_en = (snd_soc_read(codec, RT5663_ASRC_2) & + RT5663_DA_STO1_TRACK_MASK) ? 1 : 0; + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + ad_asrc_en = (snd_soc_read(codec, RT5668_ASRC_3) & + RT5668_AD_STO1_TRACK_MASK) ? 1 : 0; + break; + case CODEC_TYPE_RT5663: + ad_asrc_en = (snd_soc_read(codec, RT5663_ASRC_2) & + RT5663_AD_STO1_TRACK_MASK) ? 1 : 0; + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + + if (da_asrc_en || ad_asrc_en) + if (rt5663->sysclk > rt5663->lrck * 384) + return 1; + + dev_err(codec->dev, "sysclk < 384 x fs, disable i2s asrc\n"); + + return 0; +} + +/** + * rt5663_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5668 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5663_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + struct rt5663_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc2_mask = 0; + unsigned int asrc2_value = 0; + unsigned int asrc3_mask = 0; + unsigned int asrc3_value = 0; + + switch (clk_src) { + case RT5663_CLK_SEL_SYS: + case RT5663_CLK_SEL_I2S1_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5663_DA_STEREO_FILTER) { + asrc2_mask |= RT5668_DA_STO1_TRACK_MASK; + asrc2_value |= clk_src << RT5668_DA_STO1_TRACK_SHIFT; + } + + if (filter_mask & RT5663_AD_STEREO_FILTER) { + switch (rt5668->codec_type) { + case CODEC_TYPE_RT5668: + asrc3_mask |= RT5668_AD_STO1_TRACK_MASK; + asrc3_value |= clk_src << RT5668_AD_STO1_TRACK_SHIFT; + break; + case CODEC_TYPE_RT5663: + asrc2_mask |= RT5663_AD_STO1_TRACK_MASK; + asrc2_value |= clk_src << RT5663_AD_STO1_TRACK_SHIFT; + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5663_ASRC_2, asrc2_mask, + asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5668_ASRC_3, asrc3_mask, + asrc3_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5663_sel_asrc_clk_src); + +/* Analog Mixer */ +static const struct snd_kcontrol_new rt5668_recmix1l[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5668_RECMIX1L, + RT5668_RECMIX1L_BST2_SHIFT, 1, 1), + SOC_DAPM_SINGLE("BST1 CBJ Switch", RT5668_RECMIX1L, + RT5668_RECMIX1L_BST1_CBJ_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_recmix1r[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5668_RECMIX1R, + RT5668_RECMIX1R_BST2_SHIFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5663_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5663_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L1_SHIFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5663_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L2_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5663_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R1_SHIFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5663_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R2_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5663_adda_l_mix[] = { + SOC_DAPM_SINGLE("ADC L Switch", RT5663_AD_DA_MIXER, + RT5668_M_ADCMIX_L_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC L Switch", RT5663_AD_DA_MIXER, + RT5668_M_DAC1_L_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5663_adda_r_mix[] = { + SOC_DAPM_SINGLE("ADC R Switch", RT5663_AD_DA_MIXER, + RT5668_M_ADCMIX_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5663_AD_DA_MIXER, + RT5668_M_DAC1_R_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, + RT5668_M_DAC_L1_STO_L_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, + RT5668_M_DAC_R1_STO_L_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5663_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, + RT5668_M_DAC_L1_STO_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, + RT5668_M_DAC_R1_STO_R_SHIFT, 1, 1), +}; + +/* Out Switch */ +static const struct snd_kcontrol_new rt5668_hpo_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5668_HP_AMP_2, + RT5668_EN_DAC_HPO_SHIFT, 1, 0); + +/* Stereo ADC source */ +static const char * const rt5668_sto1_adc_src[] = { + "ADC L", "ADC R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5668_sto1_adcl_enum, RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_L_SRC_SHIFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcl_mux = + SOC_DAPM_ENUM("STO1 ADC L Mux", rt5668_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5668_sto1_adcr_enum, RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_R_SRC_SHIFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcr_mux = + SOC_DAPM_ENUM("STO1 ADC R Mux", rt5668_sto1_adcr_enum); + +/* RT5663: Analog DACL1 input source */ +static const char * const rt5663_alg_dacl_src[] = { + "DAC L", "STO DAC MIXL" +}; + +static SOC_ENUM_SINGLE_DECL(rt5663_alg_dacl_enum, RT5663_BYPASS_STO_DAC, + RT5663_DACL1_SRC_SHIFT, rt5663_alg_dacl_src); + +static const struct snd_kcontrol_new rt5663_alg_dacl_mux = + SOC_DAPM_ENUM("DAC L Mux", rt5663_alg_dacl_enum); + +/* RT5663: Analog DACR1 input source */ +static const char * const rt5663_alg_dacr_src[] = { + "DAC R", "STO DAC MIXR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5663_alg_dacr_enum, RT5663_BYPASS_STO_DAC, + RT5663_DACR1_SRC_SHIFT, rt5663_alg_dacr_src); + +static const struct snd_kcontrol_new rt5663_alg_dacr_mux = + SOC_DAPM_ENUM("DAC R Mux", rt5663_alg_dacr_enum); + +static int rt5663_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5668_SEL_PM_HP_SHIFT, RT5668_SEL_PM_HP_HIGH); + snd_soc_update_bits(codec, RT5663_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, + RT5668_HP_SIG_SRC1_SILENCE); + } else { + snd_soc_write(codec, RT5663_DEPOP_2, 0x3003); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x000b, + 0x000b); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, + 0x0030); + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5668_OVCD_HP_MASK, RT5668_OVCD_HP_DIS); + snd_soc_write(codec, RT5663_HP_CHARGE_PUMP_2, 0x1371); + snd_soc_write(codec, RT5663_HP_BIAS, 0xabba); + snd_soc_write(codec, RT5663_CHARGE_PUMP_1, 0x2224); + snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7766); + snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa); + snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, + 0x3000); + } + break; + + case SND_SOC_DAPM_PRE_PMD: + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5663_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, + RT5668_HP_SIG_SRC1_REG); + } else { + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x3000, 0x0); + snd_soc_update_bits(codec, RT5663_HP_CHARGE_PUMP_1, + RT5668_OVCD_HP_MASK, RT5668_OVCD_HP_EN); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x0030, 0x0); + snd_soc_update_bits(codec, RT5663_DEPOP_1, 0x000b, + 0x000b); + } + break; + + default: + return 0; + } + + return 0; +} + +static int rt5668_bst2_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5668_PWR_BST2_MASK | RT5668_PWR_BST2_OP_MASK, + RT5668_PWR_BST2 | RT5668_PWR_BST2_OP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5663_PWR_ANLG_2, + RT5668_PWR_BST2_MASK | RT5668_PWR_BST2_OP_MASK, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5663_pre_div_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0xff00); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0xfffc); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0x0000); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0x0000); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5663_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5663_PWR_ANLG_3, RT5668_PWR_PLL_SHIFT, 0, + NULL, 0), + + /* micbias */ + SND_SOC_DAPM_MICBIAS("MICBIAS1", RT5663_PWR_ANLG_2, + RT5668_PWR_MB1_SHIFT, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS2", RT5663_PWR_ANLG_2, + RT5668_PWR_MB2_SHIFT, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + + /* REC Mixer Power */ + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5663_PWR_ANLG_2, + RT5668_PWR_RECMIX1_SHIFT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC L Power", RT5663_PWR_DIG_1, + RT5668_PWR_ADC_L1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", RT5663_CHOP_ADC, + RT5668_CKGEN_ADCC_SHIFT, 0, NULL, 0), + + /* ADC Mixer */ + SND_SOC_DAPM_MIXER("STO1 ADC MIXL", SND_SOC_NOPM, + 0, 0, rt5663_sto1_adc_l_mix, + ARRAY_SIZE(rt5663_sto1_adc_l_mix)), + + /* ADC Filter Power */ + SND_SOC_DAPM_SUPPLY("STO1 ADC Filter", RT5663_PWR_DIG_2, + RT5668_PWR_ADC_S1F_SHIFT, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S", RT5663_PWR_DIG_1, RT5668_PWR_I2S1_SHIFT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("IF DAC", 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 ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFTX", "AIF Capture", 0, SND_SOC_NOPM, 0, 0), + + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("ADDA MIXL", SND_SOC_NOPM, 0, 0, rt5663_adda_l_mix, + ARRAY_SIZE(rt5663_adda_l_mix)), + SND_SOC_DAPM_MIXER("ADDA MIXR", SND_SOC_NOPM, 0, 0, rt5663_adda_r_mix, + ARRAY_SIZE(rt5663_adda_r_mix)), + SND_SOC_DAPM_PGA("DAC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("STO1 DAC Filter", RT5663_PWR_DIG_2, + RT5668_PWR_DAC_S1F_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("STO1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5663_sto1_dac_l_mix, ARRAY_SIZE(rt5663_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("STO1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5663_sto1_dac_r_mix, ARRAY_SIZE(rt5663_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("STO1 DAC L Power", RT5663_PWR_DIG_1, + RT5668_PWR_DAC_L1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("STO1 DAC R Power", RT5663_PWR_DIG_1, + RT5668_PWR_DAC_R1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Headphone*/ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5663_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_widget rt5668_specific_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5663_PWR_ANLG_3, + RT5668_PWR_LDO2_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5668_PWR_VOL, + RT5668_PWR_MIC_DET_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO DAC", RT5663_PWR_DIG_1, + RT5668_PWR_LDO_DACREF_SHIFT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY("I2S ASRC", RT5663_ASRC_1, + RT5668_I2S1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC ASRC", RT5663_ASRC_1, + RT5668_DAC_STO1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC ASRC", RT5663_ASRC_1, + RT5668_ADC_STO1_ASRC_SHIFT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5663_PWR_ANLG_3, + RT5668_PWR_CBJ_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BST2 Power", SND_SOC_NOPM, 0, 0, + rt5668_bst2_power, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5668_recmix1l, + ARRAY_SIZE(rt5668_recmix1l)), + SND_SOC_DAPM_MIXER("RECMIX1R", SND_SOC_NOPM, 0, 0, rt5668_recmix1r, + ARRAY_SIZE(rt5668_recmix1r)), + SND_SOC_DAPM_SUPPLY("RECMIX1R Power", RT5663_PWR_ANLG_2, + RT5668_PWR_RECMIX2_SHIFT, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC R Power", RT5663_PWR_DIG_1, + RT5668_PWR_ADC_R1_SHIFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_PGA("STO1 ADC L1", RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_L1_SRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC R1", RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_R1_SRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC L2", RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_L2_SRC_SHIFT, 1, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC R2", RT5663_STO1_ADC_MIXER, + RT5668_STO1_ADC_R2_SRC_SHIFT, 1, NULL, 0), + + SND_SOC_DAPM_MUX("STO1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcl_mux), + SND_SOC_DAPM_MUX("STO1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcr_mux), + + /* ADC Mix */ + SND_SOC_DAPM_MIXER("STO1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5668_sto1_adc_r_mix, ARRAY_SIZE(rt5668_sto1_adc_r_mix)), + + /* Analog DAC Clock */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RT5663_CHOP_DAC_L, + RT5668_CKGEN_DAC1_SHIFT, 0, NULL, 0), + + /* Headphone out */ + SND_SOC_DAPM_SWITCH("HPO Playback", SND_SOC_NOPM, 0, 0, + &rt5668_hpo_switch), +}; + +static const struct snd_soc_dapm_widget rt5663_specific_dapm_widgets[] = { + /* System Clock Pre Divider Gating */ + SND_SOC_DAPM_SUPPLY("Pre Div Power", SND_SOC_NOPM, 0, 0, + rt5663_pre_div_power, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + /* LDO */ + SND_SOC_DAPM_SUPPLY("LDO ADC", RT5663_PWR_DIG_1, + RT5668_PWR_LDO_DACREF_SHIFT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY("I2S ASRC", RT5663_ASRC_1, + RT5663_I2S1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC ASRC", RT5663_ASRC_1, + RT5663_DAC_STO1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC ASRC", RT5663_ASRC_1, + RT5663_ADC_STO1_ASRC_SHIFT, 0, NULL, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* STO ADC */ + SND_SOC_DAPM_PGA("STO1 ADC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC L2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog DAC source */ + SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0, &rt5663_alg_dacl_mux), + SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0, &rt5663_alg_dacr_mux), +}; + +static const struct snd_soc_dapm_route rt5663_dapm_routes[] = { + /* PLL */ + { "I2S", NULL, "PLL", rt5663_is_sys_clk_from_pll }, + + /* ASRC */ + { "STO1 ADC Filter", NULL, "ADC ASRC", rt5663_is_using_asrc }, + { "STO1 DAC Filter", NULL, "DAC ASRC", rt5663_is_using_asrc }, + { "I2S", NULL, "I2S ASRC", rt5663_i2s_use_asrc }, + + { "ADC L", NULL, "ADC L Power" }, + { "ADC L", NULL, "ADC Clock" }, + + { "STO1 ADC L2", NULL, "STO1 DAC MIXL" }, + + { "STO1 ADC MIXL", "ADC1 Switch", "STO1 ADC L1" }, + { "STO1 ADC MIXL", "ADC2 Switch", "STO1 ADC L2" }, + { "STO1 ADC MIXL", NULL, "STO1 ADC Filter" }, + + { "IF1 ADC1", NULL, "STO1 ADC MIXL" }, + { "IF ADC", NULL, "IF1 ADC1" }, + { "AIFTX", NULL, "IF ADC" }, + { "AIFTX", NULL, "I2S" }, + + { "AIFRX", NULL, "I2S" }, + { "IF DAC", NULL, "AIFRX" }, + { "IF1 DAC1 L", NULL, "IF DAC" }, + { "IF1 DAC1 R", NULL, "IF DAC" }, + + { "ADDA MIXL", "ADC L Switch", "STO1 ADC MIXL" }, + { "ADDA MIXL", "DAC L Switch", "IF1 DAC1 L" }, + { "ADDA MIXL", NULL, "STO1 DAC Filter" }, + { "ADDA MIXL", NULL, "STO1 DAC L Power" }, + { "ADDA MIXR", "DAC R Switch", "IF1 DAC1 R" }, + { "ADDA MIXR", NULL, "STO1 DAC Filter" }, + { "ADDA MIXR", NULL, "STO1 DAC R Power" }, + + { "DAC L1", NULL, "ADDA MIXL" }, + { "DAC R1", NULL, "ADDA MIXR" }, + + { "STO1 DAC MIXL", "DAC L Switch", "DAC L1" }, + { "STO1 DAC MIXL", "DAC R Switch", "DAC R1" }, + { "STO1 DAC MIXL", NULL, "STO1 DAC L Power" }, + { "STO1 DAC MIXL", NULL, "STO1 DAC Filter" }, + { "STO1 DAC MIXR", "DAC R Switch", "DAC R1" }, + { "STO1 DAC MIXR", "DAC L Switch", "DAC L1" }, + { "STO1 DAC MIXR", NULL, "STO1 DAC R Power" }, + { "STO1 DAC MIXR", NULL, "STO1 DAC Filter" }, + + { "HP Amp", NULL, "DAC L" }, + { "HP Amp", NULL, "DAC R" }, +}; + +static const struct snd_soc_dapm_route rt5668_specific_dapm_routes[] = { + { "MICBIAS1", NULL, "LDO2" }, + { "MICBIAS2", NULL, "LDO2" }, + + { "BST1 CBJ", NULL, "IN1P" }, + { "BST1 CBJ", NULL, "IN1N" }, + { "BST1 CBJ", NULL, "CBJ Power" }, + + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + { "BST2", NULL, "BST2 Power" }, + + { "RECMIX1L", "BST2 Switch", "BST2" }, + { "RECMIX1L", "BST1 CBJ Switch", "BST1 CBJ" }, + { "RECMIX1L", NULL, "RECMIX1L Power" }, + { "RECMIX1R", "BST2 Switch", "BST2" }, + { "RECMIX1R", NULL, "RECMIX1R Power" }, + + { "ADC L", NULL, "RECMIX1L" }, + { "ADC R", NULL, "RECMIX1R" }, + { "ADC R", NULL, "ADC R Power" }, + { "ADC R", NULL, "ADC Clock" }, + + { "STO1 ADC L Mux", "ADC L", "ADC L" }, + { "STO1 ADC L Mux", "ADC R", "ADC R" }, + { "STO1 ADC L1", NULL, "STO1 ADC L Mux" }, + + { "STO1 ADC R Mux", "ADC L", "ADC L" }, + { "STO1 ADC R Mux", "ADC R", "ADC R" }, + { "STO1 ADC R1", NULL, "STO1 ADC R Mux" }, + { "STO1 ADC R2", NULL, "STO1 DAC MIXR" }, + + { "STO1 ADC MIXR", "ADC1 Switch", "STO1 ADC R1" }, + { "STO1 ADC MIXR", "ADC2 Switch", "STO1 ADC R2" }, + { "STO1 ADC MIXR", NULL, "STO1 ADC Filter" }, + + { "IF1 ADC1", NULL, "STO1 ADC MIXR" }, + + { "ADDA MIXR", "ADC R Switch", "STO1 ADC MIXR" }, + + { "DAC L", NULL, "STO1 DAC MIXL" }, + { "DAC L", NULL, "LDO DAC" }, + { "DAC L", NULL, "DAC Clock" }, + { "DAC R", NULL, "STO1 DAC MIXR" }, + { "DAC R", NULL, "LDO DAC" }, + { "DAC R", NULL, "DAC Clock" }, + + { "HPO Playback", "Switch", "HP Amp" }, + { "HPOL", NULL, "HPO Playback" }, + { "HPOR", NULL, "HPO Playback" }, +}; + +static const struct snd_soc_dapm_route rt5663_specific_dapm_routes[] = { + { "I2S", NULL, "Pre Div Power" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "RECMIX1L Power" }, + + { "ADC L", NULL, "BST1" }, + + { "STO1 ADC L1", NULL, "ADC L" }, + + { "DAC L Mux", "DAC L", "DAC L1" }, + { "DAC L Mux", "STO DAC MIXL", "STO1 DAC MIXL" }, + { "DAC R Mux", "DAC R", "DAC R1"}, + { "DAC R Mux", "STO DAC MIXR", "STO1 DAC MIXR" }, + + { "DAC L", NULL, "DAC L Mux" }, + { "DAC R", NULL, "DAC R Mux" }, + + { "HPOL", NULL, "HP Amp" }, + { "HPOR", NULL, "HP Amp" }, +}; + +static int rt5663_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 rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + int pre_div; + + rt5663->lrck = params_rate(params); + + dev_dbg(dai->dev, "bclk is %dHz and sysclk is %dHz\n", + rt5663->lrck, rt5663->sysclk); + + pre_div = rl6231_get_clk_info(rt5663->sysclk, rt5663->lrck); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5663->lrck, dai->id); + return -EINVAL; + } + + dev_dbg(dai->dev, "pre_div is %d for iis %d\n", pre_div, dai->id); + + switch (params_width(params)) { + case 8: + val_len = RT5668_I2S_DL_8; + break; + case 16: + val_len = RT5668_I2S_DL_16; + break; + case 20: + val_len = RT5668_I2S_DL_20; + break; + case 24: + val_len = RT5668_I2S_DL_24; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5663_I2S1_SDP, + RT5668_I2S_DL_MASK, val_len); + + snd_soc_update_bits(codec, RT5663_ADDA_CLK_1, + RT5668_I2S_PD1_MASK, pre_div << RT5668_I2S_PD1_SHIFT); + + return 0; +} + +static int rt5663_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5668_I2S_MS_S; + 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 |= RT5668_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 |= RT5668_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5668_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5668_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5663_I2S1_SDP, RT5668_I2S_MS_MASK | + RT5668_I2S_BP_MASK | RT5668_I2S_DF_MASK, reg_val); + + return 0; +} + +static int rt5663_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5663->sysclk && clk_id == rt5663->sysclk_src) + return 0; + + switch (clk_id) { + case RT5663_SCLK_S_MCLK: + reg_val |= RT5663_SCLK_SRC_MCLK; + break; + case RT5663_SCLK_S_PLL1: + reg_val |= RT5663_SCLK_SRC_PLL1; + break; + case RT5663_SCLK_S_RCCLK: + reg_val |= RT5663_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5663_GLB_CLK, RT5668_SCLK_SRC_MASK, + reg_val); + rt5663->sysclk = freq; + rt5663->sysclk_src = clk_id; + + dev_dbg(codec->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5663_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 rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + int mask, shift, val; + + if (source == rt5663->pll_src && freq_in == rt5663->pll_in && + freq_out == rt5663->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5663->pll_in = 0; + rt5663->pll_out = 0; + snd_soc_update_bits(codec, RT5663_GLB_CLK, + RT5663_SCLK_SRC_MASK, RT5663_SCLK_SRC_MCLK); + return 0; + } + + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + mask = RT5668_PLL1_SRC_MASK; + shift = RT5668_PLL1_SRC_SHIFT; + break; + case CODEC_TYPE_RT5663: + mask = RT5663_PLL1_SRC_MASK; + shift = RT5663_PLL1_SRC_SHIFT; + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + + switch (source) { + case RT5663_PLL1_S_MCLK: + val = 0x0; + break; + case RT5663_PLL1_S_BCLK1: + val = 0x1; + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5663_GLB_CLK, mask, (val << shift)); + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", pll_code.m_bp, + (pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code, + pll_code.k_code); + + snd_soc_write(codec, RT5663_PLL_1, + pll_code.n_code << RT5668_PLL_N_SHIFT | pll_code.k_code); + snd_soc_write(codec, RT5663_PLL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5668_PLL_M_SHIFT | + pll_code.m_bp << RT5668_PLL_M_BP_SHIFT); + + rt5663->pll_in = freq_in; + rt5663->pll_out = freq_out; + rt5663->pll_src = source; + + return 0; +} + +static int rt5663_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; + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, reg; + + if (rx_mask || tx_mask) + val |= RT5668_TDM_MODE_TDM; + + switch (slots) { + case 4: + val |= RT5668_TDM_IN_CH_4; + val |= RT5668_TDM_OUT_CH_4; + break; + case 6: + val |= RT5668_TDM_IN_CH_6; + val |= RT5668_TDM_OUT_CH_6; + break; + case 8: + val |= RT5668_TDM_IN_CH_8; + val |= RT5668_TDM_OUT_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= RT5668_TDM_IN_LEN_20; + val |= RT5668_TDM_OUT_LEN_20; + break; + case 24: + val |= RT5668_TDM_IN_LEN_24; + val |= RT5668_TDM_OUT_LEN_24; + break; + case 32: + val |= RT5668_TDM_IN_LEN_32; + val |= RT5668_TDM_OUT_LEN_32; + break; + case 16: + break; + default: + return -EINVAL; + } + + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + reg = RT5663_TDM_2; + break; + case CODEC_TYPE_RT5663: + reg = RT5663_TDM_1; + break; + default: + dev_err(codec->dev, "Unknown CODEC_TYPE\n"); + } + + snd_soc_update_bits(codec, reg, RT5668_TDM_MODE_MASK | + RT5668_TDM_IN_CH_MASK | RT5668_TDM_OUT_CH_MASK | + RT5668_TDM_IN_LEN_MASK | RT5668_TDM_OUT_LEN_MASK, val); + + return 0; +} + +static int rt5663_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + dev_dbg(codec->dev, "%s ratio = %d\n", __func__, ratio); + + if (rt5663->codec_type == CODEC_TYPE_RT5668) + reg = RT5668_TDM_8; + else + reg = RT5663_TDM_5; + + switch (ratio) { + case 32: + snd_soc_update_bits(codec, reg, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_16); + break; + case 40: + snd_soc_update_bits(codec, reg, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_20); + break; + case 48: + snd_soc_update_bits(codec, reg, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_24); + break; + case 64: + snd_soc_update_bits(codec, reg, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_32); + break; + default: + dev_err(codec->dev, "Invalid ratio!\n"); + return -EINVAL; + } + + return 0; +} + +static int rt5663_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5668_PWR_FV1_MASK | RT5668_PWR_FV2_MASK, + RT5668_PWR_FV1 | RT5668_PWR_FV2); + break; + + case SND_SOC_BIAS_PREPARE: + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5663_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, + RT5668_DIG_GATE_CTRL_EN); + snd_soc_update_bits(codec, RT5663_SIG_CLK_DET, + RT5668_EN_ANA_CLK_DET_MASK | + RT5668_PWR_CLK_DET_MASK, + RT5668_EN_ANA_CLK_DET_AUTO | + RT5668_PWR_CLK_DET_EN); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (rt5663->codec_type == CODEC_TYPE_RT5668) + snd_soc_update_bits(codec, RT5663_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, + RT5668_DIG_GATE_CTRL_DIS); + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5668_PWR_VREF1_MASK | RT5668_PWR_VREF2_MASK | + RT5668_PWR_FV1_MASK | RT5668_PWR_FV2_MASK | + RT5668_PWR_MB_MASK, RT5668_PWR_VREF1 | + RT5668_PWR_VREF2 | RT5668_PWR_MB); + usleep_range(10000, 10005); + if (rt5663->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5663_SIG_CLK_DET, + RT5668_EN_ANA_CLK_DET_MASK | + RT5668_PWR_CLK_DET_MASK, + RT5668_EN_ANA_CLK_DET_DIS | + RT5668_PWR_CLK_DET_DIS); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5663_PWR_ANLG_1, + RT5668_PWR_VREF1_MASK | RT5668_PWR_VREF2_MASK | + RT5668_PWR_FV1 | RT5668_PWR_FV2, 0x0); + break; + + default: + break; + } + + return 0; +} + +static int rt5663_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + rt5663->codec = codec; + + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + snd_soc_dapm_new_controls(dapm, + rt5668_specific_dapm_widgets, + ARRAY_SIZE(rt5668_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, + rt5668_specific_dapm_routes, + ARRAY_SIZE(rt5668_specific_dapm_routes)); + snd_soc_add_codec_controls(codec, rt5668_specific_controls, + ARRAY_SIZE(rt5668_specific_controls)); + break; + case CODEC_TYPE_RT5663: + snd_soc_dapm_new_controls(dapm, + rt5663_specific_dapm_widgets, + ARRAY_SIZE(rt5663_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, + rt5663_specific_dapm_routes, + ARRAY_SIZE(rt5663_specific_dapm_routes)); + snd_soc_add_codec_controls(codec, rt5663_specific_controls, + ARRAY_SIZE(rt5663_specific_controls)); + break; + } + + return 0; +} + +static int rt5663_remove(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5663->regmap, RT5663_RESET, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5663_suspend(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5663->regmap, true); + regcache_mark_dirty(rt5663->regmap); + + return 0; +} + +static int rt5663_resume(struct snd_soc_codec *codec) +{ + struct rt5663_priv *rt5663 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5663->regmap, false); + regcache_sync(rt5663->regmap); + + return 0; +} +#else +#define rt5663_suspend NULL +#define rt5663_resume NULL +#endif + +#define RT5663_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5663_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 rt5663_aif_dai_ops = { + .hw_params = rt5663_hw_params, + .set_fmt = rt5663_set_dai_fmt, + .set_sysclk = rt5663_set_dai_sysclk, + .set_pll = rt5663_set_dai_pll, + .set_tdm_slot = rt5663_set_tdm_slot, + .set_bclk_ratio = rt5663_set_bclk_ratio, +}; + +struct snd_soc_dai_driver rt5663_dai[] = { + { + .name = "rt5663-aif", + .id = RT5663_AIF, + .playback = { + .stream_name = "AIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5663_STEREO_RATES, + .formats = RT5663_FORMATS, + }, + .capture = { + .stream_name = "AIF Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5663_STEREO_RATES, + .formats = RT5663_FORMATS, + }, + .ops = &rt5663_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5663 = { + .probe = rt5663_probe, + .remove = rt5663_remove, + .suspend = rt5663_suspend, + .resume = rt5663_resume, + .set_bias_level = rt5663_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt5663_snd_controls, + .num_controls = ARRAY_SIZE(rt5663_snd_controls), + .dapm_widgets = rt5663_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5663_dapm_widgets), + .dapm_routes = rt5663_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5663_dapm_routes), + } +}; + +static const struct regmap_config rt5668_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x07fa, + .volatile_reg = rt5668_volatile_register, + .readable_reg = rt5668_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5668_reg, + .num_reg_defaults = ARRAY_SIZE(rt5668_reg), +}; + +static const struct regmap_config rt5663_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x03f3, + .volatile_reg = rt5663_volatile_register, + .readable_reg = rt5663_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5663_reg, + .num_reg_defaults = ARRAY_SIZE(rt5663_reg), +}; + +static const struct regmap_config temp_regmap = { + .name = "nocache", + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x03f3, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt5663_i2c_id[] = { + { "rt5668", 0 }, + { "rt5663", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5663_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt5663_of_match[] = { + { .compatible = "realtek,rt5668", }, + { .compatible = "realtek,rt5663", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5663_of_match); +#endif + +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5663_acpi_match[] = { + { "10EC5668", 0}, + { "10EC5663", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5663_acpi_match); +#endif + +static void rt5668_calibrate(struct rt5663_priv *rt5668) +{ + regmap_write(rt5668->regmap, RT5663_BIAS_CUR_8, 0xa402); + regmap_write(rt5668->regmap, RT5663_PWR_DIG_1, 0x0100); + regmap_write(rt5668->regmap, RT5663_RECMIX, 0x4040); + regmap_write(rt5668->regmap, RT5663_DIG_MISC, 0x0001); + regmap_write(rt5668->regmap, RT5663_RC_CLK, 0x0380); + regmap_write(rt5668->regmap, RT5663_GLB_CLK, 0x8000); + regmap_write(rt5668->regmap, RT5663_ADDA_CLK_1, 0x1000); + regmap_write(rt5668->regmap, RT5663_CHOP_DAC_L, 0x3030); + regmap_write(rt5668->regmap, RT5663_CALIB_ADC, 0x3c05); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_1, 0xa23e); + msleep(40); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_1, 0xf23e); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_2, 0x0321); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_1, 0xfc00); + msleep(500); +} + +static void rt5663_calibrate(struct rt5663_priv *rt5668) +{ + int value, count; + + regmap_write(rt5668->regmap, RT5663_RC_CLK, 0x0280); + regmap_write(rt5668->regmap, RT5663_GLB_CLK, 0x8000); + regmap_write(rt5668->regmap, RT5663_DIG_MISC, 0x8001); + regmap_write(rt5668->regmap, RT5663_VREF_RECMIX, 0x0032); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_1, 0xa2be); + msleep(20); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_1, 0xf2be); + regmap_write(rt5668->regmap, RT5663_PWR_DIG_2, 0x8400); + regmap_write(rt5668->regmap, RT5663_CHOP_ADC, 0x3000); + regmap_write(rt5668->regmap, RT5663_DEPOP_1, 0x003b); + regmap_write(rt5668->regmap, RT5663_PWR_DIG_1, 0x8df8); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_2, 0x0003); + regmap_write(rt5668->regmap, RT5663_PWR_ANLG_3, 0x018c); + regmap_write(rt5668->regmap, RT5663_ADDA_CLK_1, 0x1111); + regmap_write(rt5668->regmap, RT5663_PRE_DIV_GATING_1, 0xffff); + regmap_write(rt5668->regmap, RT5663_PRE_DIV_GATING_2, 0xffff); + regmap_write(rt5668->regmap, RT5663_DEPOP_2, 0x3003); + regmap_write(rt5668->regmap, RT5663_DEPOP_1, 0x003b); + regmap_write(rt5668->regmap, RT5663_HP_CHARGE_PUMP_1, 0x1e32); + regmap_write(rt5668->regmap, RT5663_HP_CHARGE_PUMP_2, 0x1371); + regmap_write(rt5668->regmap, RT5663_DACREF_LDO, 0x3b0b); + regmap_write(rt5668->regmap, RT5663_STO_DAC_MIXER, 0x2080); + regmap_write(rt5668->regmap, RT5663_BYPASS_STO_DAC, 0x000c); + regmap_write(rt5668->regmap, RT5663_HP_BIAS, 0xabba); + regmap_write(rt5668->regmap, RT5663_CHARGE_PUMP_1, 0x2224); + regmap_write(rt5668->regmap, RT5663_HP_OUT_EN, 0x8088); + regmap_write(rt5668->regmap, RT5663_STO_DRE_9, 0x0017); + regmap_write(rt5668->regmap, RT5663_STO_DRE_10, 0x0017); + regmap_write(rt5668->regmap, RT5663_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5668->regmap, RT5663_RECMIX, 0x0005); + regmap_write(rt5668->regmap, RT5663_ADDA_RST, 0xc000); + regmap_write(rt5668->regmap, RT5663_STO1_HPF_ADJ1, 0x3320); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_2, 0x00c9); + regmap_write(rt5668->regmap, RT5663_DUMMY_1, 0x004c); + regmap_write(rt5668->regmap, RT5663_ANA_BIAS_CUR_1, 0x7766); + regmap_write(rt5668->regmap, RT5663_BIAS_CUR_8, 0x4702); + msleep(200); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_1, 0x0069); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_3, 0x06c2); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_1_1, 0x7b00); + regmap_write(rt5668->regmap, RT5663_HP_CALIB_1_1, 0xfb00); + count = 0; + while (true) { + regmap_read(rt5668->regmap, RT5663_HP_CALIB_1_1, &value); + if (value & 0x8000) + usleep_range(10000, 10005); + else + break; + + if (count > 200) + return; + count++; + } +} + +static int rt5663_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5663_priv *rt5663; + int ret; + unsigned int val; + struct regmap *regmap; + + rt5663 = devm_kzalloc(&i2c->dev, sizeof(struct rt5663_priv), + GFP_KERNEL); + + if (rt5663 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5663); + + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&i2c->dev, "Failed to allocate temp register map: %d\n", + ret); + return ret; + } + regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + switch (val) { + case RT5668_DEVICE_ID: + rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5668_regmap); + rt5663->codec_type = CODEC_TYPE_RT5668; + break; + case RT5663_DEVICE_ID: + rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5663_regmap); + rt5663->codec_type = CODEC_TYPE_RT5663; + break; + default: + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663 or rt5668\n", + val); + return -ENODEV; + } + + if (IS_ERR(rt5663->regmap)) { + ret = PTR_ERR(rt5663->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + /* reset and calibrate */ + regmap_write(rt5663->regmap, RT5663_RESET, 0); + regcache_cache_bypass(rt5663->regmap, true); + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + rt5668_calibrate(rt5663); + break; + case CODEC_TYPE_RT5663: + rt5663_calibrate(rt5663); + break; + default: + dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); + } + regcache_cache_bypass(rt5663->regmap, false); + regmap_write(rt5663->regmap, RT5663_RESET, 0); + dev_dbg(&i2c->dev, "calibrate done\n"); + + /* GPIO1 as IRQ */ + regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, RT5668_GP1_PIN_MASK, + RT5668_GP1_PIN_IRQ); + /* 4btn inline command debounce */ + regmap_update_bits(rt5663->regmap, RT5663_IL_CMD_5, + RT5668_4BTN_CLK_DEB_MASK, RT5668_4BTN_CLK_DEB_65MS); + + switch (rt5663->codec_type) { + case CODEC_TYPE_RT5668: + regmap_write(rt5663->regmap, RT5663_BIAS_CUR_8, 0xa402); + /* JD1 */ + regmap_update_bits(rt5663->regmap, RT5663_AUTO_1MRC_CLK, + RT5668_IRQ_POW_SAV_MASK | RT5668_IRQ_POW_SAV_JD1_MASK, + RT5668_IRQ_POW_SAV_EN | RT5668_IRQ_POW_SAV_JD1_EN); + regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_2, + RT5668_PWR_JD1_MASK, RT5668_PWR_JD1); + regmap_update_bits(rt5663->regmap, RT5663_IRQ_1, + RT5668_EN_CB_JD_MASK, RT5668_EN_CB_JD_EN); + + regmap_update_bits(rt5663->regmap, RT5663_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, RT5668_HP_SIG_SRC1_REG); + regmap_update_bits(rt5663->regmap, RT5663_RECMIX, + RT5668_VREF_BIAS_MASK | RT5668_CBJ_DET_MASK | + RT5668_DET_TYPE_MASK, RT5668_VREF_BIAS_REG | + RT5668_CBJ_DET_EN | RT5668_DET_TYPE_QFN); + /* Set GPIO4 and GPIO8 as input for combo jack */ + regmap_update_bits(rt5663->regmap, RT5663_GPIO_2, + RT5668_GP4_PIN_CONF_MASK, RT5668_GP4_PIN_CONF_INPUT); + regmap_update_bits(rt5663->regmap, RT5668_GPIO_3, + RT5668_GP8_PIN_CONF_MASK, RT5668_GP8_PIN_CONF_INPUT); + regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_1, + RT5668_LDO1_DVO_MASK | RT5668_AMP_HP_MASK, + RT5668_LDO1_DVO_0_9V | RT5668_AMP_HP_3X); + break; + case CODEC_TYPE_RT5663: + regmap_write(rt5663->regmap, RT5663_VREF_RECMIX, 0x0032); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xa2be); + msleep(20); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xf2be); + regmap_update_bits(rt5663->regmap, RT5663_GPIO_2, + RT5663_GP1_PIN_CONF_MASK, RT5663_GP1_PIN_CONF_OUTPUT); + /* DACREF LDO control */ + regmap_update_bits(rt5663->regmap, RT5663_DACREF_LDO, 0x3e0e, + 0x3a0a); + regmap_update_bits(rt5663->regmap, RT5663_RECMIX, + RT5663_RECMIX1_BST1_MASK, RT5663_RECMIX1_BST1_ON); + regmap_update_bits(rt5663->regmap, RT5663_TDM_2, + RT5663_DATA_SWAP_ADCDAT1_MASK, + RT5663_DATA_SWAP_ADCDAT1_LL); + break; + default: + dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); + } + + INIT_DELAYED_WORK(&rt5663->jack_detect_work, rt5663_jack_detect_work); + + if (i2c->irq) { + ret = request_irq(i2c->irq, rt5663_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5663", rt5663); + if (ret) + dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", + __func__, ret); + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5663, + rt5663_dai, ARRAY_SIZE(rt5663_dai)); + + if (ret) { + if (i2c->irq) + free_irq(i2c->irq, rt5663); + } + + return ret; +} + +static int rt5663_i2c_remove(struct i2c_client *i2c) +{ + struct rt5663_priv *rt5663 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt5663); + + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +void rt5663_i2c_shutdown(struct i2c_client *client) +{ + struct rt5663_priv *rt5663 = i2c_get_clientdata(client); + + regmap_write(rt5663->regmap, RT5663_RESET, 0); +} + +struct i2c_driver rt5663_i2c_driver = { + .driver = { + .name = "rt5663", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5663_acpi_match), + .of_match_table = of_match_ptr(rt5663_of_match), + }, + .probe = rt5663_i2c_probe, + .remove = rt5663_i2c_remove, + .shutdown = rt5663_i2c_shutdown, + .id_table = rt5663_i2c_id, +}; +module_i2c_driver(rt5663_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5663 driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h new file mode 100644 index 0000000..2cc8f28 --- /dev/null +++ b/sound/soc/codecs/rt5663.h @@ -0,0 +1,1121 @@ +/* + * rt5663.h -- RT5663 ALSA SoC audio driver + * + * Copyright 2016 Realtek Microelectronics + * Author: Jack Yu <jack.yu(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 __RT5663_H__ +#define __RT5663_H__ + +/* Info */ +#define RT5663_RESET 0x0000 +#define RT5663_VENDOR_ID 0x00fd +#define RT5663_VENDOR_ID_1 0x00fe +#define RT5663_VENDOR_ID_2 0x00ff + +#define RT5668_LOUT_CTRL 0x0001 +#define RT5668_HP_AMP_2 0x0003 +#define RT5668_MONO_OUT 0x0004 +#define RT5668_MONO_GAIN 0x0007 + +#define RT5668_AEC_BST 0x000b +#define RT5668_IN1_IN2 0x000c +#define RT5668_IN3_IN4 0x000d +#define RT5668_INL1_INR1 0x000f +#define RT5668_CBJ_TYPE_2 0x0011 +#define RT5668_CBJ_TYPE_3 0x0012 +#define RT5668_CBJ_TYPE_4 0x0013 +#define RT5668_CBJ_TYPE_5 0x0014 +#define RT5668_CBJ_TYPE_8 0x0017 + +/* I/O - ADC/DAC/DMIC */ +#define RT5668_DAC3_DIG_VOL 0x001a +#define RT5668_DAC3_CTRL 0x001b +#define RT5668_MONO_ADC_DIG_VOL 0x001d +#define RT5668_STO2_ADC_DIG_VOL 0x001e +#define RT5668_MONO_ADC_BST_GAIN 0x0020 +#define RT5668_STO2_ADC_BST_GAIN 0x0021 +#define RT5668_SIDETONE_CTRL 0x0024 +/* Mixer - D-D */ +#define RT5668_MONO1_ADC_MIXER 0x0027 +#define RT5668_STO2_ADC_MIXER 0x0028 +#define RT5668_MONO_DAC_MIXER 0x002b +#define RT5668_DAC2_SRC_CTRL 0x002e +#define RT5668_IF_3_4_DATA_CTL 0x002f +#define RT5668_IF_5_DATA_CTL 0x0030 +#define RT5668_PDM_OUT_CTL 0x0031 +#define RT5668_PDM_I2C_DATA_CTL1 0x0032 +#define RT5668_PDM_I2C_DATA_CTL2 0x0033 +#define RT5668_PDM_I2C_DATA_CTL3 0x0034 +#define RT5668_PDM_I2C_DATA_CTL4 0x0035 + +/*Mixer - Analog*/ +#define RT5668_RECMIX1_NEW 0x003a +#define RT5668_RECMIX1L_0 0x003b +#define RT5668_RECMIX1L 0x003c +#define RT5668_RECMIX1R_0 0x003d +#define RT5668_RECMIX1R 0x003e +#define RT5668_RECMIX2_NEW 0x003f +#define RT5668_RECMIX2_L_2 0x0041 +#define RT5668_RECMIX2_R 0x0042 +#define RT5668_RECMIX2_R_2 0x0043 +#define RT5668_CALIB_REC_LR 0x0044 +#define RT5668_ALC_BK_GAIN 0x0049 +#define RT5668_MONOMIX_GAIN 0x004a +#define RT5668_MONOMIX_IN_GAIN 0x004b +#define RT5668_OUT_MIXL_GAIN 0x004d +#define RT5668_OUT_LMIX_IN_GAIN 0x004e +#define RT5668_OUT_RMIX_IN_GAIN 0x004f +#define RT5668_OUT_RMIX_IN_GAIN1 0x0050 +#define RT5668_LOUT_MIXER_CTRL 0x0052 +/* Power */ +#define RT5668_PWR_VOL 0x0067 + +#define RT5668_ADCDAC_RST 0x006d +/* Format - ADC/DAC */ +#define RT5668_I2S34_SDP 0x0071 +#define RT5668_I2S5_SDP 0x0072 +/* Format - TDM Control */ +#define RT5668_TDM_5 0x007c +#define RT5668_TDM_6 0x007d +#define RT5668_TDM_7 0x007e +#define RT5668_TDM_8 0x007f + +/* Function - Analog */ +#define RT5668_ASRC_3 0x0085 +#define RT5668_ASRC_6 0x0088 +#define RT5668_ASRC_7 0x0089 +#define RT5668_PLL_TRK_13 0x0099 +#define RT5668_I2S_M_CLK_CTL 0x00a0 +#define RT5668_FDIV_I2S34_M_CLK 0x00a1 +#define RT5668_FDIV_I2S34_M_CLK2 0x00a2 +#define RT5668_FDIV_I2S5_M_CLK 0x00a3 +#define RT5668_FDIV_I2S5_M_CLK2 0x00a4 + +/* Function - Digital */ +#define RT5668_IRQ_4 0x00b9 +#define RT5668_GPIO_3 0x00c2 +#define RT5668_GPIO_4 0x00c3 +#define RT5668_GPIO_STA 0x00c4 +#define RT5668_HP_AMP_DET1 0x00d0 +#define RT5668_HP_AMP_DET2 0x00d1 +#define RT5668_HP_AMP_DET3 0x00d2 +#define RT5668_MID_BD_HP_AMP 0x00d3 +#define RT5668_LOW_BD_HP_AMP 0x00d4 +#define RT5668_SOF_VOL_ZC2 0x00da +#define RT5668_ADC_STO2_ADJ1 0x00ee +#define RT5668_ADC_STO2_ADJ2 0x00ef +/* General Control */ +#define RT5668_A_JD_CTRL 0x00f0 +#define RT5668_JD1_TRES_CTRL 0x00f1 +#define RT5668_JD2_TRES_CTRL 0x00f2 +#define RT5668_JD_CTRL2 0x00f7 +#define RT5668_DUM_REG_2 0x00fb +#define RT5668_DUM_REG_3 0x00fc + + +#define RT5668_DACADC_DIG_VOL2 0x0101 +#define RT5668_DIG_IN_PIN2 0x0133 +#define RT5668_PAD_DRV_CTL1 0x0136 +#define RT5668_SOF_RAM_DEPOP 0x0138 +#define RT5668_VOL_TEST 0x013f +#define RT5668_TEST_MODE_3 0x0147 +#define RT5668_TEST_MODE_4 0x0148 +#define RT5668_MONO_DYNA_1 0x0170 +#define RT5668_MONO_DYNA_2 0x0171 +#define RT5668_MONO_DYNA_3 0x0172 +#define RT5668_MONO_DYNA_4 0x0173 +#define RT5668_MONO_DYNA_5 0x0174 +#define RT5668_MONO_DYNA_6 0x0175 +#define RT5668_STO1_SIL_DET 0x0190 +#define RT5668_MONOL_SIL_DET 0x0191 +#define RT5668_MONOR_SIL_DET 0x0192 +#define RT5668_STO2_DAC_SIL 0x0193 +#define RT5668_PWR_SAV_CTL1 0x0194 +#define RT5668_PWR_SAV_CTL2 0x0195 +#define RT5668_PWR_SAV_CTL3 0x0196 +#define RT5668_PWR_SAV_CTL4 0x0197 +#define RT5668_PWR_SAV_CTL5 0x0198 +#define RT5668_PWR_SAV_CTL6 0x0199 +#define RT5668_MONO_AMP_CAL1 0x01a0 +#define RT5668_MONO_AMP_CAL2 0x01a1 +#define RT5668_MONO_AMP_CAL3 0x01a2 +#define RT5668_MONO_AMP_CAL4 0x01a3 +#define RT5668_MONO_AMP_CAL5 0x01a4 +#define RT5668_MONO_AMP_CAL6 0x01a5 +#define RT5668_MONO_AMP_CAL7 0x01a6 +#define RT5668_MONO_AMP_CAL_ST1 0x01a7 +#define RT5668_MONO_AMP_CAL_ST2 0x01a8 +#define RT5668_MONO_AMP_CAL_ST3 0x01a9 +#define RT5668_MONO_AMP_CAL_ST4 0x01aa +#define RT5668_MONO_AMP_CAL_ST5 0x01ab +#define RT5668_HP_IMP_SEN_13 0x01b9 +#define RT5668_HP_IMP_SEN_14 0x01ba +#define RT5668_HP_IMP_SEN_6 0x01bb +#define RT5668_HP_IMP_SEN_7 0x01bc +#define RT5668_HP_IMP_SEN_8 0x01bd +#define RT5668_HP_IMP_SEN_9 0x01be +#define RT5668_HP_IMP_SEN_10 0x01bf +#define RT5668_HP_LOGIC_3 0x01dc +#define RT5668_HP_CALIB_ST10 0x01f3 +#define RT5668_HP_CALIB_ST11 0x01f4 +#define RT5668_PRO_REG_TBL_4 0x0203 +#define RT5668_PRO_REG_TBL_5 0x0204 +#define RT5668_PRO_REG_TBL_6 0x0205 +#define RT5668_PRO_REG_TBL_7 0x0206 +#define RT5668_PRO_REG_TBL_8 0x0207 +#define RT5668_PRO_REG_TBL_9 0x0208 +#define RT5668_SAR_ADC_INL_1 0x0210 +#define RT5668_SAR_ADC_INL_2 0x0211 +#define RT5668_SAR_ADC_INL_3 0x0212 +#define RT5668_SAR_ADC_INL_4 0x0213 +#define RT5668_SAR_ADC_INL_5 0x0214 +#define RT5668_SAR_ADC_INL_6 0x0215 +#define RT5668_SAR_ADC_INL_7 0x0216 +#define RT5668_SAR_ADC_INL_8 0x0217 +#define RT5668_SAR_ADC_INL_9 0x0218 +#define RT5668_SAR_ADC_INL_10 0x0219 +#define RT5668_SAR_ADC_INL_11 0x021a +#define RT5668_SAR_ADC_INL_12 0x021b +#define RT5668_DRC_CTRL_1 0x02ff +#define RT5668_DRC1_CTRL_2 0x0301 +#define RT5668_DRC1_CTRL_3 0x0302 +#define RT5668_DRC1_CTRL_4 0x0303 +#define RT5668_DRC1_CTRL_5 0x0304 +#define RT5668_DRC1_CTRL_6 0x0305 +#define RT5668_DRC1_HD_CTRL_1 0x0306 +#define RT5668_DRC1_HD_CTRL_2 0x0307 +#define RT5668_DRC1_PRI_REG_1 0x0310 +#define RT5668_DRC1_PRI_REG_2 0x0311 +#define RT5668_DRC1_PRI_REG_3 0x0312 +#define RT5668_DRC1_PRI_REG_4 0x0313 +#define RT5668_DRC1_PRI_REG_5 0x0314 +#define RT5668_DRC1_PRI_REG_6 0x0315 +#define RT5668_DRC1_PRI_REG_7 0x0316 +#define RT5668_DRC1_PRI_REG_8 0x0317 +#define RT5668_ALC_PGA_CTL_1 0x0330 +#define RT5668_ALC_PGA_CTL_2 0x0331 +#define RT5668_ALC_PGA_CTL_3 0x0332 +#define RT5668_ALC_PGA_CTL_4 0x0333 +#define RT5668_ALC_PGA_CTL_5 0x0334 +#define RT5668_ALC_PGA_CTL_6 0x0335 +#define RT5668_ALC_PGA_CTL_7 0x0336 +#define RT5668_ALC_PGA_CTL_8 0x0337 +#define RT5668_ALC_PGA_REG_1 0x0338 +#define RT5668_ALC_PGA_REG_2 0x0339 +#define RT5668_ALC_PGA_REG_3 0x033a +#define RT5668_ADC_EQ_RECOV_1 0x03c0 +#define RT5668_ADC_EQ_RECOV_2 0x03c1 +#define RT5668_ADC_EQ_RECOV_3 0x03c2 +#define RT5668_ADC_EQ_RECOV_4 0x03c3 +#define RT5668_ADC_EQ_RECOV_5 0x03c4 +#define RT5668_ADC_EQ_RECOV_6 0x03c5 +#define RT5668_ADC_EQ_RECOV_7 0x03c6 +#define RT5668_ADC_EQ_RECOV_8 0x03c7 +#define RT5668_ADC_EQ_RECOV_9 0x03c8 +#define RT5668_ADC_EQ_RECOV_10 0x03c9 +#define RT5668_ADC_EQ_RECOV_11 0x03ca +#define RT5668_ADC_EQ_RECOV_12 0x03cb +#define RT5668_ADC_EQ_RECOV_13 0x03cc +#define RT5668_VID_HIDDEN 0x03fe +#define RT5668_VID_CUSTOMER 0x03ff +#define RT5668_SCAN_MODE 0x07f0 +#define RT5668_I2C_BYPA 0x07fa + +/* Headphone Amp Control 2 (0x0003) */ +#define RT5668_EN_DAC_HPO_MASK (0x1 << 14) +#define RT5668_EN_DAC_HPO_SHIFT 14 +#define RT5668_EN_DAC_HPO_DIS (0x0 << 14) +#define RT5668_EN_DAC_HPO_EN (0x1 << 14) + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5668_GAIN_HP (0x1f << 8) +#define RT5668_GAIN_HP_SHIFT 8 + +/* AEC BST Control (0x000b) */ +#define RT5668_GAIN_CBJ_MASK (0xf << 8) +#define RT5668_GAIN_CBJ_SHIFT 8 + +/* IN1 Control / MIC GND REF (0x000c) */ +#define RT5668_IN1_DF_MASK (0x1 << 15) +#define RT5668_IN1_DF_SHIFT 15 + +/* Combo Jack and Type Detection Control 1 (0x0010) */ +#define RT5668_CBJ_DET_MASK (0x1 << 15) +#define RT5668_CBJ_DET_SHIFT 15 +#define RT5668_CBJ_DET_DIS (0x0 << 15) +#define RT5668_CBJ_DET_EN (0x1 << 15) +#define RT5668_DET_TYPE_MASK (0x1 << 12) +#define RT5668_DET_TYPE_SHIFT 12 +#define RT5668_DET_TYPE_WLCSP (0x0 << 12) +#define RT5668_DET_TYPE_QFN (0x1 << 12) +#define RT5668_VREF_BIAS_MASK (0x1 << 6) +#define RT5668_VREF_BIAS_SHIFT 6 +#define RT5668_VREF_BIAS_FSM (0x0 << 6) +#define RT5668_VREF_BIAS_REG (0x1 << 6) + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5668_RECMIX1L_BST1_CBJ (0x1 << 7) +#define RT5668_RECMIX1L_BST1_CBJ_SHIFT 7 +#define RT5668_RECMIX1L_BST2 (0x1 << 4) +#define RT5668_RECMIX1L_BST2_SHIFT 4 + +/* REC Right Mixer Control 2 (0x003e) */ +#define RT5668_RECMIX1R_BST2 (0x1 << 4) +#define RT5668_RECMIX1R_BST2_SHIFT 4 + +/* DAC1 Digital Volume (0x0019) */ +#define RT5668_DAC_L1_VOL_MASK (0xff << 8) +#define RT5668_DAC_L1_VOL_SHIFT 8 +#define RT5668_DAC_R1_VOL_MASK (0xff) +#define RT5668_DAC_R1_VOL_SHIFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5668_ADC_L_MUTE_MASK (0x1 << 15) +#define RT5668_ADC_L_MUTE_SHIFT 15 +#define RT5668_ADC_L_VOL_MASK (0x7f << 8) +#define RT5668_ADC_L_VOL_SHIFT 8 +#define RT5668_ADC_R_MUTE_MASK (0x1 << 7) +#define RT5668_ADC_R_MUTE_SHIFT 7 +#define RT5668_ADC_R_VOL_MASK (0x7f) +#define RT5668_ADC_R_VOL_SHIFT 0 + +/* Stereo ADC Mixer Control (0x0026) */ +#define RT5668_M_STO1_ADC_L1 (0x1 << 15) +#define RT5668_M_STO1_ADC_L1_SHIFT 15 +#define RT5668_M_STO1_ADC_L2 (0x1 << 14) +#define RT5668_M_STO1_ADC_L2_SHIFT 14 +#define RT5668_STO1_ADC_L1_SRC (0x1 << 13) +#define RT5668_STO1_ADC_L1_SRC_SHIFT 13 +#define RT5668_STO1_ADC_L2_SRC (0x1 << 12) +#define RT5668_STO1_ADC_L2_SRC_SHIFT 12 +#define RT5668_STO1_ADC_L_SRC (0x3 << 10) +#define RT5668_STO1_ADC_L_SRC_SHIFT 10 +#define RT5668_M_STO1_ADC_R1 (0x1 << 7) +#define RT5668_M_STO1_ADC_R1_SHIFT 7 +#define RT5668_M_STO1_ADC_R2 (0x1 << 6) +#define RT5668_M_STO1_ADC_R2_SHIFT 6 +#define RT5668_STO1_ADC_R1_SRC (0x1 << 5) +#define RT5668_STO1_ADC_R1_SRC_SHIFT 5 +#define RT5668_STO1_ADC_R2_SRC (0x1 << 4) +#define RT5668_STO1_ADC_R2_SRC_SHIFT 4 +#define RT5668_STO1_ADC_R_SRC (0x3 << 2) +#define RT5668_STO1_ADC_R_SRC_SHIFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5668_M_ADCMIX_L (0x1 << 15) +#define RT5668_M_ADCMIX_L_SHIFT 15 +#define RT5668_M_DAC1_L (0x1 << 14) +#define RT5668_M_DAC1_L_SHIFT 14 +#define RT5668_M_ADCMIX_R (0x1 << 7) +#define RT5668_M_ADCMIX_R_SHIFT 7 +#define RT5668_M_DAC1_R (0x1 << 6) +#define RT5668_M_DAC1_R_SHIFT 6 + +/* Stereo DAC Mixer Control (0x002a) */ +#define RT5668_M_DAC_L1_STO_L (0x1 << 15) +#define RT5668_M_DAC_L1_STO_L_SHIFT 15 +#define RT5668_M_DAC_R1_STO_L (0x1 << 13) +#define RT5668_M_DAC_R1_STO_L_SHIFT 13 +#define RT5668_M_DAC_L1_STO_R (0x1 << 7) +#define RT5668_M_DAC_L1_STO_R_SHIFT 7 +#define RT5668_M_DAC_R1_STO_R (0x1 << 5) +#define RT5668_M_DAC_R1_STO_R_SHIFT 5 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5668_PWR_I2S1 (0x1 << 15) +#define RT5668_PWR_I2S1_SHIFT 15 +#define RT5668_PWR_DAC_L1 (0x1 << 11) +#define RT5668_PWR_DAC_L1_SHIFT 11 +#define RT5668_PWR_DAC_R1 (0x1 << 10) +#define RT5668_PWR_DAC_R1_SHIFT 10 +#define RT5668_PWR_LDO_DACREF_MASK (0x1 << 8) +#define RT5668_PWR_LDO_DACREF_SHIFT 8 +#define RT5668_PWR_LDO_DACREF_ON (0x1 << 8) +#define RT5668_PWR_LDO_DACREF_DOWN (0x0 << 8) +#define RT5668_PWR_LDO_SHIFT 8 +#define RT5668_PWR_ADC_L1 (0x1 << 4) +#define RT5668_PWR_ADC_L1_SHIFT 4 +#define RT5668_PWR_ADC_R1 (0x1 << 3) +#define RT5668_PWR_ADC_R1_SHIFT 3 + +/* Power Management for Digital 2 (0x0062) */ +#define RT5668_PWR_ADC_S1F (0x1 << 15) +#define RT5668_PWR_ADC_S1F_SHIFT 15 +#define RT5668_PWR_DAC_S1F (0x1 << 10) +#define RT5668_PWR_DAC_S1F_SHIFT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5668_PWR_VREF1 (0x1 << 15) +#define RT5668_PWR_VREF1_MASK (0x1 << 15) +#define RT5668_PWR_VREF1_SHIFT 15 +#define RT5668_PWR_FV1 (0x1 << 14) +#define RT5668_PWR_FV1_MASK (0x1 << 14) +#define RT5668_PWR_FV1_SHIFT 14 +#define RT5668_PWR_VREF2 (0x1 << 13) +#define RT5668_PWR_VREF2_MASK (0x1 << 13) +#define RT5668_PWR_VREF2_SHIFT 13 +#define RT5668_PWR_FV2 (0x1 << 12) +#define RT5668_PWR_FV2_MASK (0x1 << 12) +#define RT5668_PWR_FV2_SHIFT 12 +#define RT5668_PWR_MB (0x1 << 9) +#define RT5668_PWR_MB_MASK (0x1 << 9) +#define RT5668_PWR_MB_SHIFT 9 +#define RT5668_AMP_HP_MASK (0x3 << 2) +#define RT5668_AMP_HP_SHIFT 2 +#define RT5668_AMP_HP_1X (0x0 << 2) +#define RT5668_AMP_HP_3X (0x1 << 2) +#define RT5668_AMP_HP_5X (0x3 << 2) +#define RT5668_LDO1_DVO_MASK (0x3) +#define RT5668_LDO1_DVO_SHIFT 0 +#define RT5668_LDO1_DVO_0_9V (0x0) +#define RT5668_LDO1_DVO_1_0V (0x1) +#define RT5668_LDO1_DVO_1_2V (0x2) +#define RT5668_LDO1_DVO_1_4V (0x3) + +/* Power Management for Analog 2 (0x0064) */ +#define RT5668_PWR_BST1 (0x1 << 15) +#define RT5668_PWR_BST1_MASK (0x1 << 15) +#define RT5668_PWR_BST1_SHIFT 15 +#define RT5668_PWR_BST1_OFF (0x0 << 15) +#define RT5668_PWR_BST1_ON (0x1 << 15) +#define RT5668_PWR_BST2 (0x1 << 14) +#define RT5668_PWR_BST2_MASK (0x1 << 14) +#define RT5668_PWR_BST2_SHIFT 14 +#define RT5668_PWR_MB1 (0x1 << 11) +#define RT5668_PWR_MB1_SHIFT 11 +#define RT5668_PWR_MB2 (0x1 << 10) +#define RT5668_PWR_MB2_SHIFT 10 +#define RT5668_PWR_BST2_OP (0x1 << 6) +#define RT5668_PWR_BST2_OP_MASK (0x1 << 6) +#define RT5668_PWR_BST2_OP_SHIFT 6 +#define RT5668_PWR_JD1 (0x1 << 3) +#define RT5668_PWR_JD1_MASK (0x1 << 3) +#define RT5668_PWR_JD1_SHIFT 3 +#define RT5668_PWR_JD2 (0x1 << 2) +#define RT5668_PWR_JD2_MASK (0x1 << 2) +#define RT5668_PWR_JD2_SHIFT 2 +#define RT5668_PWR_RECMIX1 (0x1 << 1) +#define RT5668_PWR_RECMIX1_SHIFT 1 +#define RT5668_PWR_RECMIX2 (0x1) +#define RT5668_PWR_RECMIX2_SHIFT 0 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5668_PWR_CBJ_MASK (0x1 << 9) +#define RT5668_PWR_CBJ_SHIFT 9 +#define RT5668_PWR_CBJ_OFF (0x0 << 9) +#define RT5668_PWR_CBJ_ON (0x1 << 9) +#define RT5668_PWR_PLL (0x1 << 6) +#define RT5668_PWR_PLL_SHIFT 6 +#define RT5668_PWR_LDO2 (0x1 << 2) +#define RT5668_PWR_LDO2_SHIFT 2 + +/* Power Management for Volume (0x0067) */ +#define RT5668_PWR_MIC_DET (0x1 << 5) +#define RT5668_PWR_MIC_DET_SHIFT 5 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5668_EN_ANA_CLK_DET_MASK (0x1 << 15) +#define RT5668_EN_ANA_CLK_DET_SHIFT 15 +#define RT5668_EN_ANA_CLK_DET_DIS (0x0 << 15) +#define RT5668_EN_ANA_CLK_DET_AUTO (0x1 << 15) +#define RT5668_PWR_CLK_DET_MASK (0x1) +#define RT5668_PWR_CLK_DET_SHIFT 0 +#define RT5668_PWR_CLK_DET_DIS (0x0) +#define RT5668_PWR_CLK_DET_EN (0x1) + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5668_I2S_MS_MASK (0x1 << 15) +#define RT5668_I2S_MS_SHIFT 15 +#define RT5668_I2S_MS_M (0x0 << 15) +#define RT5668_I2S_MS_S (0x1 << 15) +#define RT5668_I2S_BP_MASK (0x1 << 8) +#define RT5668_I2S_BP_SHIFT 8 +#define RT5668_I2S_BP_NOR (0x0 << 8) +#define RT5668_I2S_BP_INV (0x1 << 8) +#define RT5668_I2S_DL_MASK (0x3 << 4) +#define RT5668_I2S_DL_SHIFT 4 +#define RT5668_I2S_DL_16 (0x0 << 4) +#define RT5668_I2S_DL_20 (0x1 << 4) +#define RT5668_I2S_DL_24 (0x2 << 4) +#define RT5668_I2S_DL_8 (0x3 << 4) +#define RT5668_I2S_DF_MASK (0x7) +#define RT5668_I2S_DF_SHIFT 0 +#define RT5668_I2S_DF_I2S (0x0) +#define RT5668_I2S_DF_LEFT (0x1) +#define RT5668_I2S_DF_PCM_A (0x2) +#define RT5668_I2S_DF_PCM_B (0x3) +#define RT5668_I2S_DF_PCM_A_N (0x6) +#define RT5668_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5668_I2S_PD1_MASK (0x7 << 12) +#define RT5668_I2S_PD1_SHIFT 12 +#define RT5668_M_I2S_DIV_MASK (0x7 << 8) +#define RT5668_M_I2S_DIV_SHIFT 8 +#define RT5668_CLK_SRC_MASK (0x3 << 4) +#define RT5668_CLK_SRC_MCLK (0x0 << 4) +#define RT5668_CLK_SRC_PLL_OUT (0x1 << 4) +#define RT5668_CLK_SRC_DIV (0x2 << 4) +#define RT5668_CLK_SRC_RC (0x3 << 4) +#define RT5668_DAC_OSR_MASK (0x3 << 2) +#define RT5668_DAC_OSR_SHIFT 2 +#define RT5668_DAC_OSR_128 (0x0 << 2) +#define RT5668_DAC_OSR_64 (0x1 << 2) +#define RT5668_DAC_OSR_32 (0x2 << 2) +#define RT5668_ADC_OSR_MASK (0x3) +#define RT5668_ADC_OSR_SHIFT 0 +#define RT5668_ADC_OSR_128 (0x0) +#define RT5668_ADC_OSR_64 (0x1) +#define RT5668_ADC_OSR_32 (0x2) + +/* TDM1 control 1 (0x0078) */ +#define RT5668_TDM_MODE_MASK (0x1 << 15) +#define RT5668_TDM_MODE_SHIFT 15 +#define RT5668_TDM_MODE_I2S (0x0 << 15) +#define RT5668_TDM_MODE_TDM (0x1 << 15) +#define RT5668_TDM_IN_CH_MASK (0x3 << 10) +#define RT5668_TDM_IN_CH_SHIFT 10 +#define RT5668_TDM_IN_CH_2 (0x0 << 10) +#define RT5668_TDM_IN_CH_4 (0x1 << 10) +#define RT5668_TDM_IN_CH_6 (0x2 << 10) +#define RT5668_TDM_IN_CH_8 (0x3 << 10) +#define RT5668_TDM_OUT_CH_MASK (0x3 << 8) +#define RT5668_TDM_OUT_CH_SHIFT 8 +#define RT5668_TDM_OUT_CH_2 (0x0 << 8) +#define RT5668_TDM_OUT_CH_4 (0x1 << 8) +#define RT5668_TDM_OUT_CH_6 (0x2 << 8) +#define RT5668_TDM_OUT_CH_8 (0x3 << 8) +#define RT5668_TDM_IN_LEN_MASK (0x3 << 6) +#define RT5668_TDM_IN_LEN_SHIFT 6 +#define RT5668_TDM_IN_LEN_16 (0x0 << 6) +#define RT5668_TDM_IN_LEN_20 (0x1 << 6) +#define RT5668_TDM_IN_LEN_24 (0x2 << 6) +#define RT5668_TDM_IN_LEN_32 (0x3 << 6) +#define RT5668_TDM_OUT_LEN_MASK (0x3 << 4) +#define RT5668_TDM_OUT_LEN_SHIFT 4 +#define RT5668_TDM_OUT_LEN_16 (0x0 << 4) +#define RT5668_TDM_OUT_LEN_20 (0x1 << 4) +#define RT5668_TDM_OUT_LEN_24 (0x2 << 4) +#define RT5668_TDM_OUT_LEN_32 (0x3 << 4) + +/* Global Clock Control (0x0080) */ +#define RT5668_SCLK_SRC_MASK (0x3 << 14) +#define RT5668_SCLK_SRC_SHIFT 14 +#define RT5668_SCLK_SRC_MCLK (0x0 << 14) +#define RT5668_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5668_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5668_PLL1_SRC_MASK (0x7 << 8) +#define RT5668_PLL1_SRC_SHIFT 8 +#define RT5668_PLL1_SRC_MCLK (0x0 << 8) +#define RT5668_PLL1_SRC_BCLK1 (0x1 << 8) +#define RT5668_PLL1_PD_MASK (0x1 << 4) +#define RT5668_PLL1_PD_SHIFT 4 + +#define RT5668_PLL_INP_MAX 40000000 +#define RT5668_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5668_PLL_N_MAX 0x001ff +#define RT5668_PLL_N_MASK (RT5668_PLL_N_MAX << 7) +#define RT5668_PLL_N_SHIFT 7 +#define RT5668_PLL_K_MAX 0x001f +#define RT5668_PLL_K_MASK (RT5668_PLL_K_MAX) +#define RT5668_PLL_K_SHIFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5668_PLL_M_MAX 0x00f +#define RT5668_PLL_M_MASK (RT5668_PLL_M_MAX << 12) +#define RT5668_PLL_M_SHIFT 12 +#define RT5668_PLL_M_BP (0x1 << 11) +#define RT5668_PLL_M_BP_SHIFT 11 + +/* PLL tracking mode 1 (0x0083) */ +#define RT5668_I2S1_ASRC_MASK (0x1 << 13) +#define RT5668_I2S1_ASRC_SHIFT 13 +#define RT5668_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5668_DAC_STO1_ASRC_SHIFT 12 +#define RT5668_ADC_STO1_ASRC_MASK (0x1 << 4) +#define RT5668_ADC_STO1_ASRC_SHIFT 4 + +/* PLL tracking mode 2 (0x0084)*/ +#define RT5668_DA_STO1_TRACK_MASK (0x7 << 12) +#define RT5668_DA_STO1_TRACK_SHIFT 12 +#define RT5668_DA_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5668_DA_STO1_TRACK_I2S1 (0x1 << 12) + +/* PLL tracking mode 3 (0x0085)*/ +#define RT5668_AD_STO1_TRACK_MASK (0x7 << 12) +#define RT5668_AD_STO1_TRACK_SHIFT 12 +#define RT5668_AD_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5668_AD_STO1_TRACK_I2S1 (0x1 << 12) + +/* HPOUT Charge pump control 1 (0x0091) */ +#define RT5668_OSW_HP_L_MASK (0x1 << 11) +#define RT5668_OSW_HP_L_SHIFT 11 +#define RT5668_OSW_HP_L_EN (0x1 << 11) +#define RT5668_OSW_HP_L_DIS (0x0 << 11) +#define RT5668_OSW_HP_R_MASK (0x1 << 10) +#define RT5668_OSW_HP_R_SHIFT 10 +#define RT5668_OSW_HP_R_EN (0x1 << 10) +#define RT5668_OSW_HP_R_DIS (0x0 << 10) +#define RT5668_SEL_PM_HP_MASK (0x3 << 8) +#define RT5668_SEL_PM_HP_SHIFT 8 +#define RT5668_SEL_PM_HP_0_6 (0x0 << 8) +#define RT5668_SEL_PM_HP_0_9 (0x1 << 8) +#define RT5668_SEL_PM_HP_1_8 (0x2 << 8) +#define RT5668_SEL_PM_HP_HIGH (0x3 << 8) +#define RT5668_OVCD_HP_MASK (0x1 << 2) +#define RT5668_OVCD_HP_SHIFT 2 +#define RT5668_OVCD_HP_EN (0x1 << 2) +#define RT5668_OVCD_HP_DIS (0x0 << 2) + +/* RC Clock Control (0x0094) */ +#define RT5668_DIG_25M_CLK_MASK (0x1 << 9) +#define RT5668_DIG_25M_CLK_SHIFT 9 +#define RT5668_DIG_25M_CLK_DIS (0x0 << 9) +#define RT5668_DIG_25M_CLK_EN (0x1 << 9) +#define RT5668_DIG_1M_CLK_MASK (0x1 << 8) +#define RT5668_DIG_1M_CLK_SHIFT 8 +#define RT5668_DIG_1M_CLK_DIS (0x0 << 8) +#define RT5668_DIG_1M_CLK_EN (0x1 << 8) + +/* Auto Turn On 1M RC CLK (0x009f) */ +#define RT5668_IRQ_POW_SAV_MASK (0x1 << 15) +#define RT5668_IRQ_POW_SAV_SHIFT 15 +#define RT5668_IRQ_POW_SAV_DIS (0x0 << 15) +#define RT5668_IRQ_POW_SAV_EN (0x1 << 15) +#define RT5668_IRQ_POW_SAV_JD1_MASK (0x1 << 14) +#define RT5668_IRQ_POW_SAV_JD1_SHIFT 14 +#define RT5668_IRQ_POW_SAV_JD1_DIS (0x0 << 14) +#define RT5668_IRQ_POW_SAV_JD1_EN (0x1 << 14) + +/* IRQ Control 1 (0x00b6) */ +#define RT5668_EN_CB_JD_MASK (0x1 << 3) +#define RT5668_EN_CB_JD_SHIFT 3 +#define RT5668_EN_CB_JD_EN (0x1 << 3) +#define RT5668_EN_CB_JD_DIS (0x0 << 3) + +/* IRQ Control 3 (0x00b8) */ +#define RT5668_EN_IRQ_INLINE_MASK (0x1 << 6) +#define RT5668_EN_IRQ_INLINE_SHIFT 6 +#define RT5668_EN_IRQ_INLINE_BYP (0x0 << 6) +#define RT5668_EN_IRQ_INLINE_NOR (0x1 << 6) + +/* GPIO Control 1 (0x00c0) */ +#define RT5668_GP1_PIN_MASK (0x1 << 15) +#define RT5668_GP1_PIN_SHIFT 15 +#define RT5668_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5668_GP1_PIN_IRQ (0x1 << 15) + +/* GPIO Control 2 (0x00c1) */ +#define RT5668_GP4_PIN_CONF_MASK (0x1 << 5) +#define RT5668_GP4_PIN_CONF_SHIFT 5 +#define RT5668_GP4_PIN_CONF_INPUT (0x0 << 5) +#define RT5668_GP4_PIN_CONF_OUTPUT (0x1 << 5) + +/* GPIO Control 2 (0x00c2) */ +#define RT5668_GP8_PIN_CONF_MASK (0x1 << 13) +#define RT5668_GP8_PIN_CONF_SHIFT 13 +#define RT5668_GP8_PIN_CONF_INPUT (0x0 << 13) +#define RT5668_GP8_PIN_CONF_OUTPUT (0x1 << 13) + +/* 4 Buttons Inline Command Function 1 (0x00df) */ +#define RT5668_4BTN_CLK_DEB_MASK (0x3 << 2) +#define RT5668_4BTN_CLK_DEB_SHIFT 2 +#define RT5668_4BTN_CLK_DEB_8MS (0x0 << 2) +#define RT5668_4BTN_CLK_DEB_16MS (0x1 << 2) +#define RT5668_4BTN_CLK_DEB_32MS (0x2 << 2) +#define RT5668_4BTN_CLK_DEB_65MS (0x3 << 2) + +/* Inline Command Function 6 (0x00e0) */ +#define RT5668_EN_4BTN_INL_MASK (0x1 << 15) +#define RT5668_EN_4BTN_INL_SHIFT 15 +#define RT5668_EN_4BTN_INL_DIS (0x0 << 15) +#define RT5668_EN_4BTN_INL_EN (0x1 << 15) +#define RT5668_RESET_4BTN_INL_MASK (0x1 << 14) +#define RT5668_RESET_4BTN_INL_SHIFT 14 +#define RT5668_RESET_4BTN_INL_RESET (0x0 << 14) +#define RT5668_RESET_4BTN_INL_NOR (0x1 << 14) + +/* Digital Misc Control (0x00fa) */ +#define RT5668_DIG_GATE_CTRL_MASK 0x1 +#define RT5668_DIG_GATE_CTRL_SHIFT (0) +#define RT5668_DIG_GATE_CTRL_DIS 0x0 +#define RT5668_DIG_GATE_CTRL_EN 0x1 + +/* Chopper and Clock control for DAC L (0x013a)*/ +#define RT5668_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5668_CKXEN_DAC1_SHIFT 13 +#define RT5668_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5668_CKGEN_DAC1_SHIFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5668_CKXEN_ADCC_MASK (0x1 << 13) +#define RT5668_CKXEN_ADCC_SHIFT 13 +#define RT5668_CKGEN_ADCC_MASK (0x1 << 12) +#define RT5668_CKGEN_ADCC_SHIFT 12 + +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5668_HP_SIG_SRC1_MASK (0x3) +#define RT5668_HP_SIG_SRC1_SHIFT 0 +#define RT5668_HP_SIG_SRC1_HP_DC (0x0) +#define RT5668_HP_SIG_SRC1_HP_CALIB (0x1) +#define RT5668_HP_SIG_SRC1_REG (0x2) +#define RT5668_HP_SIG_SRC1_SILENCE (0x3) + +/* RT5663 specific register */ +#define RT5663_HP_OUT_EN 0x0002 +#define RT5663_HP_LCH_DRE 0x0005 +#define RT5663_HP_RCH_DRE 0x0006 +#define RT5663_CALIB_BST 0x000a +#define RT5663_RECMIX 0x0010 +#define RT5663_SIL_DET_CTL 0x0015 +#define RT5663_PWR_SAV_SILDET 0x0016 +#define RT5663_SIDETONE_CTL 0x0018 +#define RT5663_STO1_DAC_DIG_VOL 0x0019 +#define RT5663_STO1_ADC_DIG_VOL 0x001c +#define RT5663_STO1_BOOST 0x001f +#define RT5663_HP_IMP_GAIN_1 0x0022 +#define RT5663_HP_IMP_GAIN_2 0x0023 +#define RT5663_STO1_ADC_MIXER 0x0026 +#define RT5663_AD_DA_MIXER 0x0029 +#define RT5663_STO_DAC_MIXER 0x002a +#define RT5663_DIG_SIDE_MIXER 0x002c +#define RT5663_BYPASS_STO_DAC 0x002d +#define RT5663_CALIB_REC_MIX 0x0040 +#define RT5663_PWR_DIG_1 0x0061 +#define RT5663_PWR_DIG_2 0x0062 +#define RT5663_PWR_ANLG_1 0x0063 +#define RT5663_PWR_ANLG_2 0x0064 +#define RT5663_PWR_ANLG_3 0x0065 +#define RT5663_PWR_MIXER 0x0066 +#define RT5663_SIG_CLK_DET 0x006b +#define RT5663_PRE_DIV_GATING_1 0x006e +#define RT5663_PRE_DIV_GATING_2 0x006f +#define RT5663_I2S1_SDP 0x0070 +#define RT5663_ADDA_CLK_1 0x0073 +#define RT5663_ADDA_RST 0x0074 +#define RT5663_FRAC_DIV_1 0x0075 +#define RT5663_FRAC_DIV_2 0x0076 +#define RT5663_TDM_1 0x0077 +#define RT5663_TDM_2 0x0078 +#define RT5663_TDM_3 0x0079 +#define RT5663_TDM_4 0x007a +#define RT5663_TDM_5 0x007b +#define RT5663_GLB_CLK 0x0080 +#define RT5663_PLL_1 0x0081 +#define RT5663_PLL_2 0x0082 +#define RT5663_ASRC_1 0x0083 +#define RT5663_ASRC_2 0x0084 +#define RT5663_ASRC_4 0x0086 +#define RT5663_DUMMY_REG 0x0087 +#define RT5663_ASRC_8 0x008a +#define RT5663_ASRC_9 0x008b +#define RT5663_ASRC_11 0x008c +#define RT5663_DEPOP_1 0x008e +#define RT5663_DEPOP_2 0x008f +#define RT5663_DEPOP_3 0x0090 +#define RT5663_HP_CHARGE_PUMP_1 0x0091 +#define RT5663_HP_CHARGE_PUMP_2 0x0092 +#define RT5663_MICBIAS_1 0x0093 +#define RT5663_RC_CLK 0x0094 +#define RT5663_ASRC_11_2 0x0097 +#define RT5663_DUMMY_REG_2 0x0098 +#define RT5663_REC_PATH_GAIN 0x009a +#define RT5663_AUTO_1MRC_CLK 0x009f +#define RT5663_ADC_EQ_1 0x00ae +#define RT5663_ADC_EQ_2 0x00af +#define RT5663_IRQ_1 0x00b6 +#define RT5663_IRQ_2 0x00b7 +#define RT5663_IRQ_3 0x00b8 +#define RT5663_IRQ_4 0x00ba +#define RT5663_IRQ_5 0x00bb +#define RT5663_INT_ST_1 0x00be +#define RT5663_INT_ST_2 0x00bf +#define RT5663_GPIO_1 0x00c0 +#define RT5663_GPIO_2 0x00c1 +#define RT5663_GPIO_STA 0x00c5 +#define RT5663_SIN_GEN_1 0x00cb +#define RT5663_SIN_GEN_2 0x00cc +#define RT5663_SIN_GEN_3 0x00cd +#define RT5663_SOF_VOL_ZC1 0x00d9 +#define RT5663_IL_CMD_1 0x00db +#define RT5663_IL_CMD_2 0x00dc +#define RT5663_IL_CMD_3 0x00dd +#define RT5663_IL_CMD_4 0x00de +#define RT5663_IL_CMD_5 0x00df +#define RT5663_IL_CMD_6 0x00e0 +#define RT5663_IL_CMD_7 0x00e1 +#define RT5663_IL_CMD_8 0x00e2 +#define RT5663_IL_CMD_PWRSAV1 0x00e4 +#define RT5663_IL_CMD_PWRSAV2 0x00e5 +#define RT5663_EM_JACK_TYPE_1 0x00e6 +#define RT5663_EM_JACK_TYPE_2 0x00e7 +#define RT5663_EM_JACK_TYPE_3 0x00e8 +#define RT5663_EM_JACK_TYPE_4 0x00e9 +#define RT5663_EM_JACK_TYPE_5 0x00ea +#define RT5663_EM_JACK_TYPE_6 0x00eb +#define RT5663_STO1_HPF_ADJ1 0x00ec +#define RT5663_STO1_HPF_ADJ2 0x00ed +#define RT5663_FAST_OFF_MICBIAS 0x00f4 +#define RT5663_JD_CTRL1 0x00f6 +#define RT5663_JD_CTRL2 0x00f8 +#define RT5663_DIG_MISC 0x00fa +#define RT5663_DIG_VOL_ZCD 0x0100 +#define RT5663_ANA_BIAS_CUR_1 0x0108 +#define RT5663_ANA_BIAS_CUR_2 0x0109 +#define RT5663_ANA_BIAS_CUR_3 0x010a +#define RT5663_ANA_BIAS_CUR_4 0x010b +#define RT5663_ANA_BIAS_CUR_5 0x010c +#define RT5663_ANA_BIAS_CUR_6 0x010d +#define RT5663_BIAS_CUR_5 0x010e +#define RT5663_BIAS_CUR_6 0x010f +#define RT5663_BIAS_CUR_7 0x0110 +#define RT5663_BIAS_CUR_8 0x0111 +#define RT5663_DACREF_LDO 0x0112 +#define RT5663_DUMMY_REG_3 0x0113 +#define RT5663_BIAS_CUR_9 0x0114 +#define RT5663_DUMMY_REG_4 0x0116 +#define RT5663_VREFADJ_OP 0x0117 +#define RT5663_VREF_RECMIX 0x0118 +#define RT5663_CHARGE_PUMP_1 0x0125 +#define RT5663_CHARGE_PUMP_1_2 0x0126 +#define RT5663_CHARGE_PUMP_1_3 0x0127 +#define RT5663_CHARGE_PUMP_2 0x0128 +#define RT5663_DIG_IN_PIN1 0x0132 +#define RT5663_PAD_DRV_CTL 0x0137 +#define RT5663_PLL_INT_REG 0x0139 +#define RT5663_CHOP_DAC_L 0x013a +#define RT5663_CHOP_ADC 0x013b +#define RT5663_CALIB_ADC 0x013c +#define RT5663_CHOP_DAC_R 0x013d +#define RT5663_DUMMY_CTL_DACLR 0x013e +#define RT5663_DUMMY_REG_5 0x0140 +#define RT5663_SOFT_RAMP 0x0141 +#define RT5663_TEST_MODE_1 0x0144 +#define RT5663_TEST_MODE_2 0x0145 +#define RT5663_TEST_MODE_3 0x0146 +#define RT5663_STO_DRE_1 0x0160 +#define RT5663_STO_DRE_2 0x0161 +#define RT5663_STO_DRE_3 0x0162 +#define RT5663_STO_DRE_4 0x0163 +#define RT5663_STO_DRE_5 0x0164 +#define RT5663_STO_DRE_6 0x0165 +#define RT5663_STO_DRE_7 0x0166 +#define RT5663_STO_DRE_8 0x0167 +#define RT5663_STO_DRE_9 0x0168 +#define RT5663_STO_DRE_10 0x0169 +#define RT5663_MIC_DECRO_1 0x0180 +#define RT5663_MIC_DECRO_2 0x0181 +#define RT5663_MIC_DECRO_3 0x0182 +#define RT5663_MIC_DECRO_4 0x0183 +#define RT5663_MIC_DECRO_5 0x0184 +#define RT5663_MIC_DECRO_6 0x0185 +#define RT5663_HP_DECRO_1 0x01b0 +#define RT5663_HP_DECRO_2 0x01b1 +#define RT5663_HP_DECRO_3 0x01b2 +#define RT5663_HP_DECRO_4 0x01b3 +#define RT5663_HP_DECOUP 0x01b4 +#define RT5663_HP_IMP_SEN_MAP8 0x01b5 +#define RT5663_HP_IMP_SEN_MAP9 0x01b6 +#define RT5663_HP_IMP_SEN_MAP10 0x01b7 +#define RT5663_HP_IMP_SEN_MAP11 0x01b8 +#define RT5663_HP_IMP_SEN_1 0x01c0 +#define RT5663_HP_IMP_SEN_2 0x01c1 +#define RT5663_HP_IMP_SEN_3 0x01c2 +#define RT5663_HP_IMP_SEN_4 0x01c3 +#define RT5663_HP_IMP_SEN_5 0x01c4 +#define RT5663_HP_IMP_SEN_6 0x01c5 +#define RT5663_HP_IMP_SEN_7 0x01c6 +#define RT5663_HP_IMP_SEN_8 0x01c7 +#define RT5663_HP_IMP_SEN_9 0x01c8 +#define RT5663_HP_IMP_SEN_10 0x01c9 +#define RT5663_HP_IMP_SEN_11 0x01ca +#define RT5663_HP_IMP_SEN_12 0x01cb +#define RT5663_HP_IMP_SEN_13 0x01cc +#define RT5663_HP_IMP_SEN_14 0x01cd +#define RT5663_HP_IMP_SEN_15 0x01ce +#define RT5663_HP_IMP_SEN_16 0x01cf +#define RT5663_HP_IMP_SEN_17 0x01d0 +#define RT5663_HP_IMP_SEN_18 0x01d1 +#define RT5663_HP_IMP_SEN_19 0x01d2 +#define RT5663_HP_IMPSEN_DIG5 0x01d3 +#define RT5663_HP_IMPSEN_MAP1 0x01d4 +#define RT5663_HP_IMPSEN_MAP2 0x01d5 +#define RT5663_HP_IMPSEN_MAP3 0x01d6 +#define RT5663_HP_IMPSEN_MAP4 0x01d7 +#define RT5663_HP_IMPSEN_MAP5 0x01d8 +#define RT5663_HP_IMPSEN_MAP7 0x01d9 +#define RT5663_HP_LOGIC_1 0x01da +#define RT5663_HP_LOGIC_2 0x01db +#define RT5663_HP_CALIB_1 0x01dd +#define RT5663_HP_CALIB_1_1 0x01de +#define RT5663_HP_CALIB_2 0x01df +#define RT5663_HP_CALIB_3 0x01e0 +#define RT5663_HP_CALIB_4 0x01e1 +#define RT5663_HP_CALIB_5 0x01e2 +#define RT5663_HP_CALIB_5_1 0x01e3 +#define RT5663_HP_CALIB_6 0x01e4 +#define RT5663_HP_CALIB_7 0x01e5 +#define RT5663_HP_CALIB_9 0x01e6 +#define RT5663_HP_CALIB_10 0x01e7 +#define RT5663_HP_CALIB_11 0x01e8 +#define RT5663_HP_CALIB_ST1 0x01ea +#define RT5663_HP_CALIB_ST2 0x01eb +#define RT5663_HP_CALIB_ST3 0x01ec +#define RT5663_HP_CALIB_ST4 0x01ed +#define RT5663_HP_CALIB_ST5 0x01ee +#define RT5663_HP_CALIB_ST6 0x01ef +#define RT5663_HP_CALIB_ST7 0x01f0 +#define RT5663_HP_CALIB_ST8 0x01f1 +#define RT5663_HP_CALIB_ST9 0x01f2 +#define RT5663_HP_AMP_DET 0x0200 +#define RT5663_DUMMY_REG_6 0x0201 +#define RT5663_HP_BIAS 0x0202 +#define RT5663_CBJ_1 0x0250 +#define RT5663_CBJ_2 0x0251 +#define RT5663_CBJ_3 0x0252 +#define RT5663_DUMMY_1 0x02fa +#define RT5663_DUMMY_2 0x02fb +#define RT5663_DUMMY_3 0x02fc +#define RT5663_ANA_JD 0x0300 +#define RT5663_ADC_LCH_LPF1_A1 0x03d0 +#define RT5663_ADC_RCH_LPF1_A1 0x03d1 +#define RT5663_ADC_LCH_LPF1_H0 0x03d2 +#define RT5663_ADC_RCH_LPF1_H0 0x03d3 +#define RT5663_ADC_LCH_BPF1_A1 0x03d4 +#define RT5663_ADC_RCH_BPF1_A1 0x03d5 +#define RT5663_ADC_LCH_BPF1_A2 0x03d6 +#define RT5663_ADC_RCH_BPF1_A2 0x03d7 +#define RT5663_ADC_LCH_BPF1_H0 0x03d8 +#define RT5663_ADC_RCH_BPF1_H0 0x03d9 +#define RT5663_ADC_LCH_BPF2_A1 0x03da +#define RT5663_ADC_RCH_BPF2_A1 0x03db +#define RT5663_ADC_LCH_BPF2_A2 0x03dc +#define RT5663_ADC_RCH_BPF2_A2 0x03dd +#define RT5663_ADC_LCH_BPF2_H0 0x03de +#define RT5663_ADC_RCH_BPF2_H0 0x03df +#define RT5663_ADC_LCH_BPF3_A1 0x03e0 +#define RT5663_ADC_RCH_BPF3_A1 0x03e1 +#define RT5663_ADC_LCH_BPF3_A2 0x03e2 +#define RT5663_ADC_RCH_BPF3_A2 0x03e3 +#define RT5663_ADC_LCH_BPF3_H0 0x03e4 +#define RT5663_ADC_RCH_BPF3_H0 0x03e5 +#define RT5663_ADC_LCH_BPF4_A1 0x03e6 +#define RT5663_ADC_RCH_BPF4_A1 0x03e7 +#define RT5663_ADC_LCH_BPF4_A2 0x03e8 +#define RT5663_ADC_RCH_BPF4_A2 0x03e9 +#define RT5663_ADC_LCH_BPF4_H0 0x03ea +#define RT5663_ADC_RCH_BPF4_H0 0x03eb +#define RT5663_ADC_LCH_HPF1_A1 0x03ec +#define RT5663_ADC_RCH_HPF1_A1 0x03ed +#define RT5663_ADC_LCH_HPF1_H0 0x03ee +#define RT5663_ADC_RCH_HPF1_H0 0x03ef +#define RT5663_ADC_EQ_PRE_VOL_L 0x03f0 +#define RT5663_ADC_EQ_PRE_VOL_R 0x03f1 +#define RT5663_ADC_EQ_POST_VOL_L 0x03f2 +#define RT5663_ADC_EQ_POST_VOL_R 0x03f3 + +/* RT5663: RECMIX Control (0x0010) */ +#define RT5663_RECMIX1_BST1_MASK (0x1) +#define RT5663_RECMIX1_BST1_SHIFT 0 +#define RT5663_RECMIX1_BST1_ON (0x0) +#define RT5663_RECMIX1_BST1_OFF (0x1) + +/* RT5663: Bypass Stereo1 DAC Mixer Control (0x002d) */ +#define RT5663_DACL1_SRC_MASK (0x1 << 3) +#define RT5663_DACL1_SRC_SHIFT 3 +#define RT5663_DACR1_SRC_MASK (0x1 << 2) +#define RT5663_DACR1_SRC_SHIFT 2 + +/* RT5663: TDM control 2 (0x0078) */ +#define RT5663_DATA_SWAP_ADCDAT1_MASK (0x3 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_SHIFT 14 +#define RT5663_DATA_SWAP_ADCDAT1_LR (0x0 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_RL (0x1 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_LL (0x2 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_RR (0x3 << 14) + +/* RT5663: TDM control 5 (0x007b) */ +#define RT5663_TDM_LENGTN_MASK (0x3) +#define RT5663_TDM_LENGTN_SHIFT 0 +#define RT5663_TDM_LENGTN_16 (0x0) +#define RT5663_TDM_LENGTN_20 (0x1) +#define RT5663_TDM_LENGTN_24 (0x2) +#define RT5663_TDM_LENGTN_32 (0x3) + +/* RT5663: Global Clock Control (0x0080) */ +#define RT5663_SCLK_SRC_MASK (0x3 << 14) +#define RT5663_SCLK_SRC_SHIFT 14 +#define RT5663_SCLK_SRC_MCLK (0x0 << 14) +#define RT5663_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5663_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5663_PLL1_SRC_MASK (0x7 << 11) +#define RT5663_PLL1_SRC_SHIFT 11 +#define RT5663_PLL1_SRC_MCLK (0x0 << 11) +#define RT5663_PLL1_SRC_BCLK1 (0x1 << 11) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5663_I2S1_ASRC_MASK (0x1 << 11) +#define RT5663_I2S1_ASRC_SHIFT 11 +#define RT5663_DAC_STO1_ASRC_MASK (0x1 << 10) +#define RT5663_DAC_STO1_ASRC_SHIFT 10 +#define RT5663_ADC_STO1_ASRC_MASK (0x1 << 3) +#define RT5663_ADC_STO1_ASRC_SHIFT 3 + +/* PLL tracking mode 2 (0x0084)*/ +#define RT5663_DA_STO1_TRACK_MASK (0x7 << 12) +#define RT5663_DA_STO1_TRACK_SHIFT 12 +#define RT5663_DA_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5663_DA_STO1_TRACK_I2S1 (0x1 << 12) +#define RT5663_AD_STO1_TRACK_MASK (0x7) +#define RT5663_AD_STO1_TRACK_SHIFT 0 +#define RT5663_AD_STO1_TRACK_SYSCLK (0x0) +#define RT5663_AD_STO1_TRACK_I2S1 (0x1) + +/* RT5663: HPOUT Charge pump control 1 (0x0091) */ +#define RT5663_SI_HP_MASK (0x1 << 12) +#define RT5663_SI_HP_SHIFT 12 +#define RT5663_SI_HP_EN (0x1 << 12) +#define RT5663_SI_HP_DIS (0x0 << 12) + +/* RT5663: GPIO Control 2 (0x00b6) */ +#define RT5663_GP1_PIN_CONF_MASK (0x1 << 2) +#define RT5663_GP1_PIN_CONF_SHIFT 2 +#define RT5663_GP1_PIN_CONF_OUTPUT (0x1 << 2) +#define RT5663_GP1_PIN_CONF_INPUT (0x0 << 2) + +/* RT5663: GPIO Control 2 (0x00b7) */ +#define RT5663_EN_IRQ_INLINE_MASK (0x1 << 3) +#define RT5663_EN_IRQ_INLINE_SHIFT 3 +#define RT5663_EN_IRQ_INLINE_NOR (0x1 << 3) +#define RT5663_EN_IRQ_INLINE_BYP (0x0 << 3) + +/* RT5663: IRQ Control 1 (0x00c1) */ +#define RT5663_EN_IRQ_JD1_MASK (0x1 << 6) +#define RT5663_EN_IRQ_JD1_SHIFT 6 +#define RT5663_EN_IRQ_JD1_EN (0x1 << 6) +#define RT5663_EN_IRQ_JD1_DIS (0x0 << 6) + +/* RT5663: Inline Command Function 2 (0x00dc) */ +#define RT5663_PWR_MIC_DET_MASK (0x1) +#define RT5663_PWR_MIC_DET_SHIFT 0 +#define RT5663_PWR_MIC_DET_ON (0x1) +#define RT5663_PWR_MIC_DET_OFF (0x0) + +/* RT5663: Embeeded Jack and Type Detection Control 1 (0x00e6)*/ +#define RT5663_CBJ_DET_MASK (0x1 << 15) +#define RT5663_CBJ_DET_SHIFT 15 +#define RT5663_CBJ_DET_DIS (0x0 << 15) +#define RT5663_CBJ_DET_EN (0x1 << 15) +#define RT5663_EXT_JD_MASK (0x1 << 11) +#define RT5663_EXT_JD_SHIFT 11 +#define RT5663_EXT_JD_EN (0x1 << 11) +#define RT5663_EXT_JD_DIS (0x0 << 11) +#define RT5663_POL_EXT_JD_MASK (0x1 << 10) +#define RT5663_POL_EXT_JD_SHIFT 10 +#define RT5663_POL_EXT_JD_EN (0x1 << 10) +#define RT5663_POL_EXT_JD_DIS (0x0 << 10) + +/* RT5663: DACREF LDO Control (0x0112)*/ +#define RT5663_PWR_LDO_DACREFL_MASK (0x1 << 9) +#define RT5663_PWR_LDO_DACREFL_SHIFT 9 +#define RT5663_PWR_LDO_DACREFR_MASK (0x1 << 1) +#define RT5663_PWR_LDO_DACREFR_SHIFT 1 + +/* RT5663: Stereo Dynamic Range Enhancement Control 9 (0x0168, 0x0169)*/ +#define RT5663_DRE_GAIN_HP_MASK (0x1f) +#define RT5663_DRE_GAIN_HP_SHIFT 0 + +/* RT5663: Combo Jack Control (0x0250) */ +#define RT5663_INBUF_CBJ_BST1_MASK (0x1 << 11) +#define RT5663_INBUF_CBJ_BST1_SHIFT 11 +#define RT5663_INBUF_CBJ_BST1_ON (0x1 << 11) +#define RT5663_INBUF_CBJ_BST1_OFF (0x0 << 11) +#define RT5663_CBJ_SENSE_BST1_MASK (0x1 << 10) +#define RT5663_CBJ_SENSE_BST1_SHIFT 10 +#define RT5663_CBJ_SENSE_BST1_L (0x1 << 10) +#define RT5663_CBJ_SENSE_BST1_R (0x0 << 10) + +/* RT5663: Combo Jack Control (0x0251) */ +#define RT5663_GAIN_BST1_MASK (0xf) +#define RT5663_GAIN_BST1_SHIFT 0 + +/* RT5663: Dummy register 1 (0x02fa) */ +#define RT5663_EMB_CLK_MASK (0x1 << 9) +#define RT5663_EMB_CLK_SHIFT 9 +#define RT5663_EMB_CLK_EN (0x1 << 9) +#define RT5663_EMB_CLK_DIS (0x0 << 9) +#define RT5663_HPA_CPL_BIAS_MASK (0x7 << 6) +#define RT5663_HPA_CPL_BIAS_SHIFT 6 +#define RT5663_HPA_CPL_BIAS_0_5 (0x0 << 6) +#define RT5663_HPA_CPL_BIAS_1 (0x1 << 6) +#define RT5663_HPA_CPL_BIAS_2 (0x2 << 6) +#define RT5663_HPA_CPL_BIAS_3 (0x3 << 6) +#define RT5663_HPA_CPL_BIAS_4_1 (0x4 << 6) +#define RT5663_HPA_CPL_BIAS_4_2 (0x5 << 6) +#define RT5663_HPA_CPL_BIAS_6 (0x6 << 6) +#define RT5663_HPA_CPL_BIAS_8 (0x7 << 6) +#define RT5663_HPA_CPR_BIAS_MASK (0x7 << 3) +#define RT5663_HPA_CPR_BIAS_SHIFT 3 +#define RT5663_HPA_CPR_BIAS_0_5 (0x0 << 3) +#define RT5663_HPA_CPR_BIAS_1 (0x1 << 3) +#define RT5663_HPA_CPR_BIAS_2 (0x2 << 3) +#define RT5663_HPA_CPR_BIAS_3 (0x3 << 3) +#define RT5663_HPA_CPR_BIAS_4_1 (0x4 << 3) +#define RT5663_HPA_CPR_BIAS_4_2 (0x5 << 3) +#define RT5663_HPA_CPR_BIAS_6 (0x6 << 3) +#define RT5663_HPA_CPR_BIAS_8 (0x7 << 3) +#define RT5663_DUMMY_BIAS_MASK (0x7) +#define RT5663_DUMMY_BIAS_SHIFT 0 +#define RT5663_DUMMY_BIAS_0_5 (0x0) +#define RT5663_DUMMY_BIAS_1 (0x1) +#define RT5663_DUMMY_BIAS_2 (0x2) +#define RT5663_DUMMY_BIAS_3 (0x3) +#define RT5663_DUMMY_BIAS_4_1 (0x4) +#define RT5663_DUMMY_BIAS_4_2 (0x5) +#define RT5663_DUMMY_BIAS_6 (0x6) +#define RT5663_DUMMY_BIAS_8 (0x7) + + +/* System Clock Source */ +enum { + RT5663_SCLK_S_MCLK, + RT5663_SCLK_S_PLL1, + RT5663_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5663_PLL1_S_MCLK, + RT5663_PLL1_S_BCLK1, +}; + +enum { + RT5663_AIF, + RT5663_AIFS, +}; + +/* asrc clock source */ +enum { + RT5663_CLK_SEL_SYS = 0x0, + RT5663_CLK_SEL_I2S1_ASRC = 0x1, +}; + +/* filter mask */ +enum { + RT5663_DA_STEREO_FILTER = 0x1, + RT5663_AD_STEREO_FILTER = 0x2, +}; + +int rt5663_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack); +int rt5663_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5663_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: da7219: Support HP detect procedure when MCLK not present
by Adam Thomson 14 Sep '16

14 Sep '16

Currently the driver code assumes that MCLK will be present during the procedure, or if it has control of MCLK then it can make sure it's enabled. Some platforms however do not provide MCLK to the driver in this way, and will not leave MCLK enabled when Codec is in STANDBY state. This patch adds support for the HP detect procedure when running from the device's internal oscillator, to ensure all platforms can make use of this feature correctly. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219-aad.c | 34 ++++++++++++++++++++++++++++++++-- sound/soc/codecs/da7219-aad.h | 6 ++++-- sound/soc/codecs/da7219.h | 2 ++ 3 files changed, 38 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 4e369a1..fc27dab 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -115,7 +115,7 @@ static void da7219_aad_hptest_work(struct work_struct *work) struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); u16 tonegen_freq_hptest; - u8 accdet_cfg8; + u8 pll_srm_sts, gain_ramp_ctrl, accdet_cfg8; int report = 0, ret = 0; /* Lock DAPM and any Kcontrols that are affected by this test */ @@ -133,6 +133,20 @@ static void da7219_aad_hptest_work(struct work_struct *work) } } + /* + * If MCLK not present, then we're using the internal oscillator and + * require different frequency settings to achieve the same result. + */ + pll_srm_sts = snd_soc_read(codec, DA7219_PLL_SRM_STS); + if (pll_srm_sts & DA7219_PLL_SRM_STS_MCLK) + tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ); + else + tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ_INT_OSC); + + /* Ensure gain ramping at fastest rate */ + gain_ramp_ctrl = snd_soc_read(codec, DA7219_GAIN_RAMP_CTRL); + snd_soc_write(codec, DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_X8); + /* Bypass cache so it saves current settings */ regcache_cache_bypass(da7219->regmap, true); @@ -195,9 +209,15 @@ static void da7219_aad_hptest_work(struct work_struct *work) snd_soc_write(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + /* + * If we're running from the internal oscillator then give audio paths + * time to settle before running test. + */ + if (!(pll_srm_sts & DA7219_PLL_SRM_STS_MCLK)) + msleep(DA7219_AAD_HPTEST_INT_OSC_PATH_DELAY); + /* Configure & start Tone Generator */ snd_soc_write(codec, DA7219_TONE_GEN_ON_PER, DA7219_BEEP_ON_PER_MASK); - tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ); regmap_raw_write(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, &tonegen_freq_hptest, sizeof(tonegen_freq_hptest)); snd_soc_update_bits(codec, DA7219_TONE_GEN_CFG2, @@ -256,6 +276,16 @@ static void da7219_aad_hptest_work(struct work_struct *work) snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, DA7219_HPTEST_EN_MASK, 0); + /* + * If we're running from the internal oscillator then give audio paths + * time to settle before allowing headphones to be driven as required. + */ + if (!(pll_srm_sts & DA7219_PLL_SRM_STS_MCLK)) + msleep(DA7219_AAD_HPTEST_INT_OSC_PATH_DELAY); + + /* Restore gain ramping rate */ + snd_soc_write(codec, DA7219_GAIN_RAMP_CTRL, gain_ramp_ctrl); + /* Drive Headphones/lineout */ snd_soc_update_bits(codec, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK, DA7219_HP_L_AMP_OE_MASK); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h index 4fccf67..a34be48 100644 --- a/sound/soc/codecs/da7219-aad.h +++ b/sound/soc/codecs/da7219-aad.h @@ -176,8 +176,10 @@ #define DA7219_AAD_MICBIAS_CHK_DELAY 10 #define DA7219_AAD_MICBIAS_CHK_RETRIES 5 -#define DA7219_AAD_HPTEST_RAMP_FREQ 0x28 -#define DA7219_AAD_HPTEST_PERIOD 65 +#define DA7219_AAD_HPTEST_RAMP_FREQ 0x28 +#define DA7219_AAD_HPTEST_RAMP_FREQ_INT_OSC 0x4D +#define DA7219_AAD_HPTEST_PERIOD 65 +#define DA7219_AAD_HPTEST_INT_OSC_PATH_DELAY 20 enum da7219_aad_event_regs { DA7219_AAD_IRQ_REG_A = 0, diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index ff2a2f0..545576d 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -224,6 +224,7 @@ #define DA7219_PLL_SRM_STATE_MASK (0xF << 0) #define DA7219_PLL_SRM_STATUS_SHIFT 4 #define DA7219_PLL_SRM_STATUS_MASK (0xF << 4) +#define DA7219_PLL_SRM_STS_MCLK (0x1 << 4) #define DA7219_PLL_SRM_STS_SRM_LOCK (0x1 << 7) /* DA7219_DIG_ROUTING_DAI = 0x2A */ @@ -576,6 +577,7 @@ /* DA7219_GAIN_RAMP_CTRL = 0x92 */ #define DA7219_GAIN_RAMP_RATE_SHIFT 0 #define DA7219_GAIN_RAMP_RATE_MASK (0x3 << 0) +#define DA7219_GAIN_RAMP_RATE_X8 (0x0 << 0) #define DA7219_GAIN_RAMP_RATE_MAX 4 /* DA7219_PC_COUNT = 0x94 */ -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: dpcm: Don't apply symmetry for BE with hw param fixup
by jeeja.kp＠intel.com 14 Sep '16

14 Sep '16

From: Jeeja KP <jeeja.kp(a)intel.com> If be_hw_param_fixup is defined for BE then it will force the BE to a specific configuration supported by HW. In this case don't apply symmetry. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 60d702f..c1bfe9d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1694,6 +1694,9 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_soc_pcm_runtime *rtd = be_substream->private_data; int i; + if (rtd->dai_link->be_hw_params_fixup) + continue; + if (soc_pcm_has_symmetry(be_substream)) be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; -- 2.5.0

2 1

[alsa-devel] [RFC][PATCH] ASoC: rt5659: Fix incorrect register addresses
by Nicolin Chen 14 Sep '16

14 Sep '16

According to the datasheets of Realtek ALC5658 (Rev. 0.9) and ALC5659 (Rev. 0.13), the "IRQ Control 4" register is located at MX-00BAh while the "IRQ control 5" register is at MX-00BBh. There is no "IRQ Control 6" register in the datasheets but it is supposed to be behind the "IRQ Control 5" register. So this patch corrects these addresses. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Hi Mark, This patch needs the confirmation from Realtek side (hoping that at least Bard or Oder would give an Ack) as the datasheets that I have might not reflect the latest register map. And the datasheets don't seem to be directly downloadable from Realtek website. sound/soc/codecs/rt5659.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h index d69b0eb..8f1aeef 100644 --- a/sound/soc/codecs/rt5659.h +++ b/sound/soc/codecs/rt5659.h @@ -180,9 +180,9 @@ #define RT5659_IRQ_CTRL_1 0x00b6 #define RT5659_IRQ_CTRL_2 0x00b7 #define RT5659_IRQ_CTRL_3 0x00b8 -#define RT5659_IRQ_CTRL_4 0x00b9 -#define RT5659_IRQ_CTRL_5 0x00ba -#define RT5659_IRQ_CTRL_6 0x00bb +#define RT5659_IRQ_CTRL_4 0x00ba +#define RT5659_IRQ_CTRL_5 0x00bb +#define RT5659_IRQ_CTRL_6 0x00bc #define RT5659_INT_ST_1 0x00be #define RT5659_INT_ST_2 0x00bf #define RT5659_GPIO_CTRL_1 0x00c0 -- 2.1.4

2 1

[alsa-devel] [[Patch] 0/2] ASoC sti: suppress inappropriate DT fields
by Arnaud Pouliquen 14 Sep '16

14 Sep '16

Aim of this patchset is to clean some "dirty" fields in devicetree. As Values are linked to the chip version and the IP usage, they can be handled by device data associated to the compatibility string. Arnaud Pouliquen (2): ASoC: sti: suppress inappropriate DT fields Documentation: dt-bindings: suppress inappropriate fields for sti sound card .../devicetree/bindings/sound/st,sti-asoc-card.txt | 52 +++------ sound/soc/sti/sti_uniperif.c | 130 +++++++++++++++++---- sound/soc/sti/uniperif.h | 28 ++--- sound/soc/sti/uniperif_player.c | 79 +++---------- sound/soc/sti/uniperif_reader.c | 46 +------- 5 files changed, 156 insertions(+), 179 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH] ASoC: rt5660: add rt5660 codec driver
by Oder Chiou 14 Sep '16

14 Sep '16

This is the initial codec driver for rt5660 Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5660.txt | 50 + include/sound/rt5660.h | 31 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5660.c | 1364 ++++++++++++++++++++ sound/soc/codecs/rt5660.h | 847 ++++++++++++ 6 files changed, 2300 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5660.txt create mode 100644 include/sound/rt5660.h create mode 100644 sound/soc/codecs/rt5660.c create mode 100644 sound/soc/codecs/rt5660.h diff --git a/Documentation/devicetree/bindings/sound/rt5660.txt b/Documentation/devicetree/bindings/sound/rt5660.txt new file mode 100644 index 0000000..c6cdb0d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5660.txt @@ -0,0 +1,50 @@ +RT5660 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5660". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in3-differential + Boolean. Indicate MIC1/3 input are differential, rather than single-ended. + +- realtek,use-ldo2 + Boolean. Use LDO2 for MICVDD + +- realtek,poweroff-in-suspend + Boolean. If the codec will be powered off in suspend, the resume should be + added delay time for waiting codec power ready. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using IN1P pin as dmic1 data pin + +Pins on the device (for linking into audio routes) for RT5660: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN1N + * IN2P + * IN3P + * IN3N + * SPO + * LOUTL + * LOUTR + +Example: + +rt5660 { + compatible = "realtek,rt5660"; + reg = <0x1c>; +}; diff --git a/include/sound/rt5660.h b/include/sound/rt5660.h new file mode 100644 index 0000000..065f83a --- /dev/null +++ b/include/sound/rt5660.h @@ -0,0 +1,31 @@ +/* + * linux/sound/rt5660.h -- Platform data for RT5660 + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(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 __LINUX_SND_RT5660_H +#define __LINUX_SND_RT5660_H + +enum rt5660_dmic1_data_pin { + RT5660_DMIC1_NULL, + RT5660_DMIC1_DATA_GPIO2, + RT5660_DMIC1_DATA_IN1P, +}; + +struct rt5660_platform_data { + /* IN1 & IN3 can optionally be differential */ + bool in1_diff; + bool in3_diff; + bool use_ldo2; + bool poweroff_codec_in_suspend; + + enum rt5660_dmic1_data_pin dmic1_data_pin; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1cd6ab3..890affe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -112,6 +112,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5659 if I2C + select SND_SOC_RT5660 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -645,6 +646,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y + default y if SND_SOC_RT5660=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -653,6 +655,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m default m if SND_SOC_RT5659=m + default m if SND_SOC_RT5660=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -697,6 +700,9 @@ config SND_SOC_RT5651 config SND_SOC_RT5659 tristate +config SND_SOC_RT5660 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 58036af..68fed20 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5659-objs := rt5659.o +snd-soc-rt5660-objs := rt5660.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -333,6 +334,7 @@ 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_RT5659) += snd-soc-rt5659.o +obj-$(CONFIG_SND_SOC_RT5660) += snd-soc-rt5660.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c new file mode 100644 index 0000000..4c2c65e --- /dev/null +++ b/sound/soc/codecs/rt5660.c @@ -0,0 +1,1364 @@ +/* + * rt5660.c -- RT5660 ALSA SoC audio codec driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(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/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" +#include "rt5660.h" + +#define RT5660_DEVICE_ID 0x6338 + +#define RT5660_PR_RANGE_BASE (0xff + 1) +#define RT5660_PR_SPACING 0x100 + +#define RT5660_PR_BASE (RT5660_PR_RANGE_BASE + (0 * RT5660_PR_SPACING)) + +static const struct regmap_range_cfg rt5660_ranges[] = { + { .name = "PR", .range_min = RT5660_PR_BASE, + .range_max = RT5660_PR_BASE + 0xf3, + .selector_reg = RT5660_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5660_PRIV_DATA, + .window_len = 0x1, }, +}; + +static const struct reg_sequence rt5660_patch[] = { + { RT5660_ALC_PGA_CTRL2, 0x44c3 }, + { RT5660_PR_BASE + 0x3d, 0x2600 }, +}; + +static const struct reg_default rt5660_reg[] = { + { 0x00, 0x0000 }, + { 0x01, 0xc800 }, + { 0x02, 0xc8c8 }, + { 0x0d, 0x1010 }, + { 0x0e, 0x1010 }, + { 0x19, 0xafaf }, + { 0x1c, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x6060 }, + { 0x29, 0x8080 }, + { 0x2a, 0x4242 }, + { 0x2f, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe000 }, + { 0x46, 0x003e }, + { 0x48, 0xf800 }, + { 0x4a, 0x0004 }, + { 0x4d, 0x0000 }, + { 0x4e, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x70, 0x8000 }, + { 0x73, 0x7000 }, + { 0x74, 0x3c00 }, + { 0x75, 0x2800 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x8c, 0x0228 }, + { 0x8d, 0xa000 }, + { 0x8e, 0x0000 }, + { 0x92, 0x0000 }, + { 0x93, 0x3000 }, + { 0xa1, 0x0059 }, + { 0xa2, 0x0001 }, + { 0xa3, 0x5c80 }, + { 0xa4, 0x0146 }, + { 0xa5, 0x1f1f }, + { 0xa6, 0x78c6 }, + { 0xa7, 0xe5ec }, + { 0xa8, 0xba61 }, + { 0xa9, 0x3c78 }, + { 0xaa, 0x8ae2 }, + { 0xab, 0xe5ec }, + { 0xac, 0xc600 }, + { 0xad, 0xba61 }, + { 0xae, 0x17ed }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x020c }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x4000 }, + { 0xbb, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0100 }, + { 0xc0, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xd3, 0xa220 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xe0, 0x8000 }, + { 0xe1, 0x0200 }, + { 0xe2, 0x8000 }, + { 0xe3, 0x0200 }, + { 0xe4, 0x0f20 }, + { 0xe5, 0x001f }, + { 0xe6, 0x020c }, + { 0xe7, 0x1f00 }, + { 0xe8, 0x0000 }, + { 0xe9, 0x4000 }, + { 0xea, 0x00a6 }, + { 0xeb, 0x04c3 }, + { 0xec, 0x27c8 }, + { 0xed, 0x7418 }, + { 0xee, 0xbf50 }, + { 0xef, 0x0045 }, + { 0xf0, 0x0007 }, + { 0xfa, 0x0000 }, + { 0xfd, 0x0000 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6338 }, +}; + +static bool rt5660_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5660_ranges); i++) + if ((reg >= rt5660_ranges[i].window_start && + reg <= rt5660_ranges[i].window_start + + rt5660_ranges[i].window_len) || + (reg >= rt5660_ranges[i].range_min && + reg <= rt5660_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5660_RESET: + case RT5660_PRIV_DATA: + case RT5660_EQ_CTRL1: + case RT5660_IRQ_CTRL2: + case RT5660_INT_IRQ_ST: + case RT5660_VENDOR_ID: + case RT5660_VENDOR_ID1: + case RT5660_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5660_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5660_ranges); i++) + if ((reg >= rt5660_ranges[i].window_start && + reg <= rt5660_ranges[i].window_start + + rt5660_ranges[i].window_len) || + (reg >= rt5660_ranges[i].range_min && + reg <= rt5660_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5660_RESET: + case RT5660_SPK_VOL: + case RT5660_LOUT_VOL: + case RT5660_IN1_IN2: + case RT5660_IN3_IN4: + case RT5660_DAC1_DIG_VOL: + case RT5660_STO1_ADC_DIG_VOL: + case RT5660_ADC_BST_VOL1: + case RT5660_STO1_ADC_MIXER: + case RT5660_AD_DA_MIXER: + case RT5660_STO_DAC_MIXER: + case RT5660_DIG_INF1_DATA: + case RT5660_REC_L1_MIXER: + case RT5660_REC_L2_MIXER: + case RT5660_REC_R1_MIXER: + case RT5660_REC_R2_MIXER: + case RT5660_LOUT_MIXER: + case RT5660_SPK_MIXER: + case RT5660_SPO_MIXER: + case RT5660_SPO_CLSD_RATIO: + case RT5660_OUT_L_GAIN1: + case RT5660_OUT_L_GAIN2: + case RT5660_OUT_L1_MIXER: + case RT5660_OUT_R_GAIN1: + case RT5660_OUT_R_GAIN2: + case RT5660_OUT_R1_MIXER: + case RT5660_PWR_DIG1: + case RT5660_PWR_DIG2: + case RT5660_PWR_ANLG1: + case RT5660_PWR_ANLG2: + case RT5660_PWR_MIXER: + case RT5660_PWR_VOL: + case RT5660_PRIV_INDEX: + case RT5660_PRIV_DATA: + case RT5660_I2S1_SDP: + case RT5660_ADDA_CLK1: + case RT5660_ADDA_CLK2: + case RT5660_DMIC_CTRL1: + case RT5660_GLB_CLK: + case RT5660_PLL_CTRL1: + case RT5660_PLL_CTRL2: + case RT5660_CLSD_AMP_OC_CTRL: + case RT5660_CLSD_AMP_CTRL: + case RT5660_LOUT_AMP_CTRL: + case RT5660_SPK_AMP_SPKVDD: + case RT5660_MICBIAS: + case RT5660_CLSD_OUT_CTRL1: + case RT5660_CLSD_OUT_CTRL2: + case RT5660_DIPOLE_MIC_CTRL1: + case RT5660_DIPOLE_MIC_CTRL2: + case RT5660_DIPOLE_MIC_CTRL3: + case RT5660_DIPOLE_MIC_CTRL4: + case RT5660_DIPOLE_MIC_CTRL5: + case RT5660_DIPOLE_MIC_CTRL6: + case RT5660_DIPOLE_MIC_CTRL7: + case RT5660_DIPOLE_MIC_CTRL8: + case RT5660_DIPOLE_MIC_CTRL9: + case RT5660_DIPOLE_MIC_CTRL10: + case RT5660_DIPOLE_MIC_CTRL11: + case RT5660_DIPOLE_MIC_CTRL12: + case RT5660_EQ_CTRL1: + case RT5660_EQ_CTRL2: + case RT5660_DRC_AGC_CTRL1: + case RT5660_DRC_AGC_CTRL2: + case RT5660_DRC_AGC_CTRL3: + case RT5660_DRC_AGC_CTRL4: + case RT5660_DRC_AGC_CTRL5: + case RT5660_JD_CTRL: + case RT5660_IRQ_CTRL1: + case RT5660_IRQ_CTRL2: + case RT5660_INT_IRQ_ST: + case RT5660_GPIO_CTRL1: + case RT5660_GPIO_CTRL2: + case RT5660_WIND_FILTER_CTRL1: + case RT5660_SV_ZCD1: + case RT5660_SV_ZCD2: + case RT5660_DRC1_LM_CTRL1: + case RT5660_DRC1_LM_CTRL2: + case RT5660_DRC2_LM_CTRL1: + case RT5660_DRC2_LM_CTRL2: + case RT5660_MULTI_DRC_CTRL: + case RT5660_DRC2_CTRL1: + case RT5660_DRC2_CTRL2: + case RT5660_DRC2_CTRL3: + case RT5660_DRC2_CTRL4: + case RT5660_DRC2_CTRL5: + case RT5660_ALC_PGA_CTRL1: + case RT5660_ALC_PGA_CTRL2: + case RT5660_ALC_PGA_CTRL3: + case RT5660_ALC_PGA_CTRL4: + case RT5660_ALC_PGA_CTRL5: + case RT5660_ALC_PGA_CTRL6: + case RT5660_ALC_PGA_CTRL7: + case RT5660_GEN_CTRL1: + case RT5660_GEN_CTRL2: + case RT5660_GEN_CTRL3: + case RT5660_VENDOR_ID: + case RT5660_VENDOR_ID1: + case RT5660_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(rt5660_out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_adc_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(rt5660_bst_tlv, -1200, 75, 0); + +static const struct snd_kcontrol_new rt5660_snd_controls[] = { + /* Speaker Output Volume */ + SOC_SINGLE("Speaker Playback Switch", RT5660_SPK_VOL, RT5660_L_MUTE_SFT, + 1, 1), + SOC_SINGLE_TLV("Speaker Playback Volume", RT5660_SPK_VOL, + RT5660_L_VOL_SFT, 39, 1, rt5660_out_vol_tlv), + + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5660_LOUT_VOL, RT5660_L_MUTE_SFT, + RT5660_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5660_LOUT_VOL, RT5660_L_VOL_SFT, + RT5660_R_VOL_SFT, 39, 1, rt5660_out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5660_DAC1_DIG_VOL, + RT5660_DAC_L1_VOL_SFT, RT5660_DAC_R1_VOL_SFT, 87, 0, + rt5660_dac_vol_tlv), + + /* IN1/IN2/IN3 Control */ + SOC_SINGLE_TLV("IN1 Boost Volume", RT5660_IN1_IN2, RT5660_BST_SFT1, 69, + 0, rt5660_bst_tlv), + SOC_SINGLE_TLV("IN2 Boost Volume", RT5660_IN1_IN2, RT5660_BST_SFT2, 69, + 0, rt5660_bst_tlv), + SOC_SINGLE_TLV("IN3 Boost Volume", RT5660_IN3_IN4, RT5660_BST_SFT3, 69, + 0, rt5660_bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5660_STO1_ADC_DIG_VOL, + RT5660_L_MUTE_SFT, RT5660_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5660_STO1_ADC_DIG_VOL, + RT5660_ADC_L_VOL_SFT, RT5660_ADC_R_VOL_SFT, 63, 0, + rt5660_adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5660_ADC_BST_VOL1, + RT5660_STO1_ADC_L_BST_SFT, RT5660_STO1_ADC_R_BST_SFT, 3, 0, + rt5660_adc_bst_tlv), +}; + +/** + * rt5660_set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + */ +static int rt5660_set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + int idx, rate; + + rate = rt5660->sysclk / rl6231_get_pre_div(rt5660->regmap, + RT5660_ADDA_CLK1, RT5660_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5660_DMIC_CTRL1, + RT5660_DMIC_CLK_MASK, idx << RT5660_DMIC_CLK_SFT); + + return idx; +} + +static int rt5660_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + unsigned int val; + + val = snd_soc_read(codec, RT5660_GLB_CLK); + val &= RT5660_SCLK_SRC_MASK; + if (val == RT5660_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int rt5660_is_using_ldo2(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + return rt5660->pdata.use_ldo2; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5660_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5660_STO1_ADC_MIXER, + RT5660_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5660_AD_DA_MIXER, + RT5660_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5660_AD_DA_MIXER, + RT5660_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5660_AD_DA_MIXER, + RT5660_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5660_AD_DA_MIXER, + RT5660_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5660_STO_DAC_MIXER, + RT5660_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5660_rec_l_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST3_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_REC_L2_MIXER, + RT5660_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5660_REC_L2_MIXER, + RT5660_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_rec_r_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST3_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_REC_R2_MIXER, + RT5660_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5660_REC_R2_MIXER, + RT5660_M_OM_R_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_spk_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_SPK_MIXER, + RT5660_M_BST3_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_SPK_MIXER, + RT5660_M_BST1_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_SPK_MIXER, + RT5660_M_DACL_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_SPK_MIXER, + RT5660_M_DACR_SM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIXL Switch", RT5660_SPK_MIXER, + RT5660_M_OM_L_SM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_out_l_mix[] = { + SOC_DAPM_SINGLE("BST3 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST3_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_OUT_L1_MIXER, + RT5660_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIXL Switch", RT5660_OUT_L1_MIXER, + RT5660_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_OUT_L1_MIXER, + RT5660_M_DAC_R_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_OUT_L1_MIXER, + RT5660_M_DAC_L_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5660_OUT_R1_MIXER, + RT5660_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_OUT_R1_MIXER, + RT5660_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIXR Switch", RT5660_OUT_R1_MIXER, + RT5660_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DACR Switch", RT5660_OUT_R1_MIXER, + RT5660_M_DAC_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_OUT_R1_MIXER, + RT5660_M_DAC_L_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_spo_mix[] = { + SOC_DAPM_SINGLE("DACR Switch", RT5660_SPO_MIXER, + RT5660_M_DAC_R_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("DACL Switch", RT5660_SPO_MIXER, + RT5660_M_DAC_L_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL Switch", RT5660_SPO_MIXER, + RT5660_M_SV_SPM_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5660_SPO_MIXER, + RT5660_M_BST1_SPM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5660_lout_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT5660_LOUT_MIXER, + RT5660_M_DAC1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX Switch", RT5660_LOUT_MIXER, + RT5660_M_LOVOL_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spk_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_SPK_VOL, + RT5660_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_l_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_LOUT_VOL, + RT5660_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_vol_control = + SOC_DAPM_SINGLE("Switch", RT5660_LOUT_VOL, + RT5660_VOL_R_SFT, 1, 1); + +/* Interface data select */ +static const char * const rt5660_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5660_if1_dac_enum, + RT5660_DIG_INF1_DATA, RT5660_IF1_DAC_IN_SFT, rt5660_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5660_if1_adc_enum, + RT5660_DIG_INF1_DATA, RT5660_IF1_ADC_IN_SFT, rt5660_data_select); + +static const struct snd_kcontrol_new rt5660_if1_dac_swap_mux = + SOC_DAPM_ENUM("IF1 DAC Swap Source", rt5660_if1_dac_enum); + +static const struct snd_kcontrol_new rt5660_if1_adc_swap_mux = + SOC_DAPM_ENUM("IF1 ADC Swap Source", rt5660_if1_adc_enum); + +static int rt5660_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5660_LOUT_AMP_CTRL, + RT5660_LOUT_CO_MASK | RT5660_LOUT_CB_MASK, + RT5660_LOUT_CO_EN | RT5660_LOUT_CB_PU); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5660_LOUT_AMP_CTRL, + RT5660_LOUT_CO_MASK | RT5660_LOUT_CB_MASK, + RT5660_LOUT_CO_DIS | RT5660_LOUT_CB_PD); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5660_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5660_PWR_ANLG1, + RT5660_PWR_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5660_PWR_ANLG2, + RT5660_PWR_PLL_BIT, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5660_PWR_ANLG2, + RT5660_PWR_MB1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5660_PWR_ANLG2, + RT5660_PWR_MB2_BIT, 0, NULL, 0), + + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + rt5660_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Power", RT5660_DMIC_CTRL1, + RT5660_DMIC_1_EN_SFT, 0, NULL, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5660_PWR_ANLG2, RT5660_PWR_BST1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("BST2", RT5660_PWR_ANLG2, RT5660_PWR_BST2_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5660_PWR_ANLG2, RT5660_PWR_BST3_BIT, 0, + NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5660_PWR_MIXER, RT5660_PWR_RM_L_BIT, + 0, rt5660_rec_l_mix, ARRAY_SIZE(rt5660_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5660_PWR_MIXER, RT5660_PWR_RM_R_BIT, + 0, rt5660_rec_r_mix, ARRAY_SIZE(rt5660_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", RT5660_PWR_DIG1, + RT5660_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC R power", RT5660_PWR_DIG1, + RT5660_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", RT5660_PR_BASE + RT5660_CHOP_DAC_ADC, + 12, 0, NULL, 0), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5660_PWR_DIG2, + RT5660_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5660_sto1_adc_l_mix, ARRAY_SIZE(rt5660_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5660_sto1_adc_r_mix, ARRAY_SIZE(rt5660_sto1_adc_r_mix)), + + /* ADC */ + SND_SOC_DAPM_ADC("Stereo1 ADC MIXL", NULL, RT5660_STO1_ADC_DIG_VOL, + RT5660_L_MUTE_SFT, 1), + SND_SOC_DAPM_ADC("Stereo1 ADC MIXR", NULL, RT5660_STO1_ADC_DIG_VOL, + RT5660_R_MUTE_SFT, 1), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5660_PWR_DIG1, RT5660_PWR_I2S1_BIT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF1 DAC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5660_if1_dac_swap_mux), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF1 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5660_if1_adc_swap_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), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, rt5660_dac_l_mix, + ARRAY_SIZE(rt5660_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, rt5660_dac_r_mix, + ARRAY_SIZE(rt5660_dac_r_mix)), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5660_PWR_DIG2, + RT5660_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5660_sto_dac_l_mix, ARRAY_SIZE(rt5660_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5660_sto_dac_r_mix, ARRAY_SIZE(rt5660_sto_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5660_PWR_DIG1, + RT5660_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5660_PWR_DIG1, + RT5660_PWR_DAC_R1_BIT, 0), + + /* OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIX", RT5660_PWR_MIXER, RT5660_PWR_SM_BIT, + 0, rt5660_spk_mix, ARRAY_SIZE(rt5660_spk_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5660_PWR_MIXER, RT5660_PWR_OM_L_BIT, + 0, rt5660_out_l_mix, ARRAY_SIZE(rt5660_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5660_PWR_MIXER, RT5660_PWR_OM_R_BIT, + 0, rt5660_out_r_mix, ARRAY_SIZE(rt5660_out_r_mix)), + + /* Output Volume */ + SND_SOC_DAPM_SWITCH("SPKVOL", RT5660_PWR_VOL, + RT5660_PWR_SV_BIT, 0, &spk_vol_control), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("LOUTVOL", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("LOUTVOL L", SND_SOC_NOPM, + RT5660_PWR_LV_L_BIT, 0, &lout_l_vol_control), + SND_SOC_DAPM_SWITCH("LOUTVOL R", SND_SOC_NOPM, + RT5660_PWR_LV_R_BIT, 0, &lout_r_vol_control), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPO MIX", SND_SOC_NOPM, 0, + 0, rt5660_spo_mix, ARRAY_SIZE(rt5660_spo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, + rt5660_lout_mix, ARRAY_SIZE(rt5660_lout_mix)), + SND_SOC_DAPM_SUPPLY("VREF HP", RT5660_GEN_CTRL1, + RT5660_PWR_VREF_HP_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, RT5660_PWR_ANLG1, + RT5660_PWR_HA_BIT, 0, rt5660_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 1, RT5660_PWR_DIG1, + RT5660_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("SPO"), +}; + +static const struct snd_soc_dapm_route rt5660_dapm_routes[] = { + { "MICBIAS1", NULL, "LDO2", rt5660_is_using_ldo2 }, + { "MICBIAS2", NULL, "LDO2", rt5660_is_using_ldo2 }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST2", NULL, "IN2P" }, + { "BST3", NULL, "IN3P" }, + { "BST3", NULL, "IN3N" }, + + { "RECMIXL", "BST3 Switch", "BST3" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + { "RECMIXL", "OUT MIXL Switch", "OUT MIXL" }, + + { "RECMIXR", "BST3 Switch", "BST3" }, + { "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 L", NULL, "ADC clock" }, + { "ADC R", NULL, "RECMIXR" }, + { "ADC R", NULL, "ADC R power" }, + { "ADC R", NULL, "ADC clock" }, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC Power"}, + + { "Sto1 ADC MIXL", "ADC1 Switch", "ADC L" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "DMIC L1" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "ADC R" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "DMIC R1" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "IF1 ADC", NULL, "Stereo1 ADC MIXL" }, + { "IF1 ADC", NULL, "Stereo1 ADC MIXR" }, + { "IF1 ADC", NULL, "I2S1" }, + + { "IF1 ADC Swap Mux", "L/R", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "R/L", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "L/L", "IF1 ADC" }, + { "IF1 ADC Swap Mux", "R/R", "IF1 ADC" }, + { "AIF1TX", NULL, "IF1 ADC Swap Mux" }, + + { "IF1 DAC", NULL, "AIF1RX" }, + { "IF1 DAC", NULL, "I2S1" }, + + { "IF1 DAC Swap Mux", "L/R", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "R/L", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "L/L", "IF1 DAC" }, + { "IF1 DAC Swap Mux", "R/R", "IF1 DAC" }, + + { "IF1 DAC L", NULL, "IF1 DAC Swap Mux" }, + { "IF1 DAC R", NULL, "IF1 DAC Swap Mux" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "IF1 DAC L" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "IF1 DAC R" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + { "dac stereo1 filter", NULL, "PLL1", rt5660_is_sys_clk_from_pll }, + + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + + { "SPK MIX", "BST3 Switch", "BST3" }, + { "SPK MIX", "BST1 Switch", "BST1" }, + { "SPK MIX", "DACL Switch", "DAC L1" }, + { "SPK MIX", "DACR Switch", "DAC R1" }, + { "SPK MIX", "OUTMIXL Switch", "OUT MIXL" }, + + { "OUT MIXL", "BST3 Switch", "BST3" }, + { "OUT MIXL", "BST2 Switch", "BST2" }, + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "RECMIXL Switch", "RECMIXL" }, + { "OUT MIXL", "DACR Switch", "DAC R1" }, + { "OUT MIXL", "DACL Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "BST1 Switch", "BST1" }, + { "OUT MIXR", "RECMIXR Switch", "RECMIXR" }, + { "OUT MIXR", "DACR Switch", "DAC R1" }, + { "OUT MIXR", "DACL Switch", "DAC L1" }, + + { "SPO MIX", "DACR Switch", "DAC R1" }, + { "SPO MIX", "DACL Switch", "DAC L1" }, + { "SPO MIX", "SPKVOL Switch", "SPKVOL" }, + { "SPO MIX", "BST1 Switch", "BST1" }, + + { "SPKVOL", "Switch", "SPK MIX" }, + { "LOUTVOL L", "Switch", "OUT MIXL" }, + { "LOUTVOL R", "Switch", "OUT MIXR" }, + + { "LOUTVOL", NULL, "LOUTVOL L" }, + { "LOUTVOL", NULL, "LOUTVOL R" }, + + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + + { "LOUT MIX", "DAC Switch", "DAC 1" }, + { "LOUT MIX", "OUTMIX Switch", "LOUTVOL" }, + + { "LOUT amp", NULL, "LOUT MIX" }, + { "LOUT amp", NULL, "VREF HP" }, + { "LOUTL", NULL, "LOUT amp" }, + { "LOUTR", NULL, "LOUT amp" }, + + { "SPK amp", NULL, "SPO MIX" }, + { "SPO", NULL, "SPK amp" }, +}; + +static int rt5660_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 rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5660->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5660->sysclk, rt5660->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5660->lrck[dai->id], dai->id); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return frame_size; + } + + if (frame_size > 32) + bclk_ms = 1; + else + bclk_ms = 0; + + rt5660->bclk[dai->id] = rt5660->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5660->bclk[dai->id], rt5660->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 |= RT5660_I2S_DL_20; + break; + case 24: + val_len |= RT5660_I2S_DL_24; + break; + case 8: + val_len |= RT5660_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5660_AIF1: + mask_clk = RT5660_I2S_BCLK_MS1_MASK | RT5660_I2S_PD1_MASK; + val_clk = bclk_ms << RT5660_I2S_BCLK_MS1_SFT | + pre_div << RT5660_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5660_I2S1_SDP, RT5660_I2S_DL_MASK, + val_len); + snd_soc_update_bits(codec, RT5660_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 rt5660_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5660->master[dai->id] = 1; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5660_I2S_MS_S; + rt5660->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 |= RT5660_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 |= RT5660_I2S_DF_LEFT; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5660_I2S_DF_PCM_A; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5660_I2S_DF_PCM_B; + break; + + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5660_AIF1: + snd_soc_update_bits(codec, RT5660_I2S1_SDP, + RT5660_I2S_MS_MASK | RT5660_I2S_BP_MASK | + RT5660_I2S_DF_MASK, reg_val); + break; + + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5660_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5660->sysclk && clk_id == rt5660->sysclk_src) + return 0; + + switch (clk_id) { + case RT5660_SCLK_S_MCLK: + reg_val |= RT5660_SCLK_SRC_MCLK; + break; + + case RT5660_SCLK_S_PLL1: + reg_val |= RT5660_SCLK_SRC_PLL1; + break; + + case RT5660_SCLK_S_RCCLK: + reg_val |= RT5660_SCLK_SRC_RCCLK; + break; + + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5660_GLB_CLK, RT5660_SCLK_SRC_MASK, + reg_val); + + rt5660->sysclk = freq; + rt5660->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5660_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 rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5660->pll_src && freq_in == rt5660->pll_in && + freq_out == rt5660->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5660->pll_in = 0; + rt5660->pll_out = 0; + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_SCLK_SRC_MASK, RT5660_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5660_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_PLL1_SRC_MASK, RT5660_PLL1_SRC_MCLK); + break; + + case RT5660_PLL1_S_BCLK: + snd_soc_update_bits(codec, RT5660_GLB_CLK, + RT5660_PLL1_SRC_MASK, RT5660_PLL1_SRC_BCLK1); + break; + + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5660_PLL_CTRL1, + pll_code.n_code << RT5660_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5660_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5660_PLL_M_SFT | + pll_code.m_bp << RT5660_PLL_M_BP_SFT); + + rt5660->pll_in = freq_in; + rt5660->pll_out = freq_out; + rt5660->pll_src = source; + + return 0; +} + +static int rt5660_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5660_priv *rt5660 = 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, RT5660_GEN_CTRL1, + RT5660_DIG_GATE_CTRL, RT5660_DIG_GATE_CTRL); + + if (IS_ERR(rt5660->mclk)) + break; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { + clk_disable_unprepare(rt5660->mclk); + } else { + ret = clk_prepare_enable(rt5660->mclk); + if (ret) + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, RT5660_PWR_ANLG1, + RT5660_PWR_VREF1 | RT5660_PWR_MB | + RT5660_PWR_BG | RT5660_PWR_VREF2, + RT5660_PWR_VREF1 | RT5660_PWR_MB | + RT5660_PWR_BG | RT5660_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5660_PWR_ANLG1, + RT5660_PWR_FV1 | RT5660_PWR_FV2, + RT5660_PWR_FV1 | RT5660_PWR_FV2); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5660_GEN_CTRL1, + RT5660_DIG_GATE_CTRL, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5660_probe(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + /* Check if MCLK provided */ + rt5660->mclk = devm_clk_get(codec->dev, "mclk"); + if (PTR_ERR(rt5660->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + rt5660->codec = codec; + + return 0; +} + +static int rt5660_remove(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5660_RESET, 0); +} + +#ifdef CONFIG_PM +static int rt5660_suspend(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5660->regmap, true); + regcache_mark_dirty(rt5660->regmap); + + return 0; +} + +static int rt5660_resume(struct snd_soc_codec *codec) +{ + struct rt5660_priv *rt5660 = snd_soc_codec_get_drvdata(codec); + + if (rt5660->pdata.poweroff_codec_in_suspend) + usleep_range(350000, 400000); + + regcache_cache_only(rt5660->regmap, false); + regcache_sync(rt5660->regmap); + + return 0; +} +#else +#define rt5660_suspend NULL +#define rt5660_resume NULL +#endif + +#define RT5660_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5660_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5660_aif_dai_ops = { + .hw_params = rt5660_hw_params, + .set_fmt = rt5660_set_dai_fmt, + .set_sysclk = rt5660_set_dai_sysclk, + .set_pll = rt5660_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5660_dai[] = { + { + .name = "rt5660-aif1", + .id = RT5660_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5660_STEREO_RATES, + .formats = RT5660_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5660_STEREO_RATES, + .formats = RT5660_FORMATS, + }, + .ops = &rt5660_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5660 = { + .probe = rt5660_probe, + .remove = rt5660_remove, + .suspend = rt5660_suspend, + .resume = rt5660_resume, + .set_bias_level = rt5660_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt5660_snd_controls, + .num_controls = ARRAY_SIZE(rt5660_snd_controls), + .dapm_widgets = rt5660_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5660_dapm_widgets), + .dapm_routes = rt5660_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5660_dapm_routes), + }, +}; + +static const struct regmap_config rt5660_regmap = { + .reg_bits = 8, + .val_bits = 16, + .use_single_rw = true, + + .max_register = RT5660_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5660_ranges) * + RT5660_PR_SPACING), + .volatile_reg = rt5660_volatile_register, + .readable_reg = rt5660_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5660_reg, + .num_reg_defaults = ARRAY_SIZE(rt5660_reg), + .ranges = rt5660_ranges, + .num_ranges = ARRAY_SIZE(rt5660_ranges), +}; + +static const struct i2c_device_id rt5660_i2c_id[] = { + { "rt5660", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5660_i2c_id); + +static const struct of_device_id rt5660_of_match[] = { + { .compatible = "realtek,rt5660", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5660_of_match); + +static const struct acpi_device_id rt5660_acpi_match[] = { + { "10EC5660", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); + +static int rt5660_parse_dt(struct rt5660_priv *rt5660, struct device *dev) +{ + rt5660->pdata.in1_diff = device_property_read_bool(dev, + "realtek,in1-differential"); + rt5660->pdata.in3_diff = device_property_read_bool(dev, + "realtek,in3-differential"); + rt5660->pdata.use_ldo2 = device_property_read_bool(dev, + "realtek,use-ldo2"); + rt5660->pdata.poweroff_codec_in_suspend = device_property_read_bool(dev, + "realtek,poweroff-in-suspend"); + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5660->pdata.dmic1_data_pin); + + return 0; +} + +static int rt5660_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5660_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5660_priv *rt5660; + int ret; + unsigned int val; + + rt5660 = devm_kzalloc(&i2c->dev, sizeof(struct rt5660_priv), + GFP_KERNEL); + + if (rt5660 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5660); + + if (pdata) + rt5660->pdata = *pdata; + else if (i2c->dev.of_node) + rt5660_parse_dt(rt5660, &i2c->dev); + + rt5660->regmap = devm_regmap_init_i2c(i2c, &rt5660_regmap); + if (IS_ERR(rt5660->regmap)) { + ret = PTR_ERR(rt5660->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5660->regmap, RT5660_VENDOR_ID2, &val); + if (val != RT5660_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5660/39\n", val); + return -ENODEV; + } + + regmap_write(rt5660->regmap, RT5660_RESET, 0); + + ret = regmap_register_patch(rt5660->regmap, rt5660_patch, + ARRAY_SIZE(rt5660_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5660->pdata.dmic1_data_pin) { + regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, + RT5660_GP1_PIN_MASK, RT5660_GP1_PIN_DMIC1_SCL); + + if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_GPIO2) + regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, + RT5660_SEL_DMIC_DATA_MASK, + RT5660_SEL_DMIC_DATA_GPIO2); + else if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_IN1P) + regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, + RT5660_SEL_DMIC_DATA_MASK, + RT5660_SEL_DMIC_DATA_IN1P); + } + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5660, + rt5660_dai, ARRAY_SIZE(rt5660_dai)); +} + +static int rt5660_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5660_i2c_driver = { + .driver = { + .name = "rt5660", + .acpi_match_table = ACPI_PTR(rt5660_acpi_match), + .of_match_table = of_match_ptr(rt5660_of_match), + }, + .probe = rt5660_i2c_probe, + .remove = rt5660_i2c_remove, + .id_table = rt5660_i2c_id, +}; +module_i2c_driver(rt5660_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5660 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5660.h b/sound/soc/codecs/rt5660.h new file mode 100644 index 0000000..6cdb926 --- /dev/null +++ b/sound/soc/codecs/rt5660.h @@ -0,0 +1,847 @@ +/* + * rt5660.h -- RT5660 ALSA SoC audio driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(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 _RT5660_H +#define _RT5660_H + +#include <linux/clk.h> +#include <sound/rt5660.h> + +/* Info */ +#define RT5660_RESET 0x00 +#define RT5660_VENDOR_ID 0xfd +#define RT5660_VENDOR_ID1 0xfe +#define RT5660_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5660_SPK_VOL 0x01 +#define RT5660_LOUT_VOL 0x02 +/* I/O - Input */ +#define RT5660_IN1_IN2 0x0d +#define RT5660_IN3_IN4 0x0e +/* I/O - ADC/DAC/DMIC */ +#define RT5660_DAC1_DIG_VOL 0x19 +#define RT5660_STO1_ADC_DIG_VOL 0x1c +#define RT5660_ADC_BST_VOL1 0x1e +/* Mixer - D-D */ +#define RT5660_STO1_ADC_MIXER 0x27 +#define RT5660_AD_DA_MIXER 0x29 +#define RT5660_STO_DAC_MIXER 0x2a +#define RT5660_DIG_INF1_DATA 0x2f +/* Mixer - ADC */ +#define RT5660_REC_L1_MIXER 0x3b +#define RT5660_REC_L2_MIXER 0x3c +#define RT5660_REC_R1_MIXER 0x3d +#define RT5660_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5660_LOUT_MIXER 0x45 +#define RT5660_SPK_MIXER 0x46 +#define RT5660_SPO_MIXER 0x48 +#define RT5660_SPO_CLSD_RATIO 0x4a +#define RT5660_OUT_L_GAIN1 0x4d +#define RT5660_OUT_L_GAIN2 0x4e +#define RT5660_OUT_L1_MIXER 0x4f +#define RT5660_OUT_R_GAIN1 0x50 +#define RT5660_OUT_R_GAIN2 0x51 +#define RT5660_OUT_R1_MIXER 0x52 +/* Power */ +#define RT5660_PWR_DIG1 0x61 +#define RT5660_PWR_DIG2 0x62 +#define RT5660_PWR_ANLG1 0x63 +#define RT5660_PWR_ANLG2 0x64 +#define RT5660_PWR_MIXER 0x65 +#define RT5660_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5660_PRIV_INDEX 0x6a +#define RT5660_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5660_I2S1_SDP 0x70 +#define RT5660_ADDA_CLK1 0x73 +#define RT5660_ADDA_CLK2 0x74 +#define RT5660_DMIC_CTRL1 0x75 +/* Function - Analog */ +#define RT5660_GLB_CLK 0x80 +#define RT5660_PLL_CTRL1 0x81 +#define RT5660_PLL_CTRL2 0x82 +#define RT5660_CLSD_AMP_OC_CTRL 0x8c +#define RT5660_CLSD_AMP_CTRL 0x8d +#define RT5660_LOUT_AMP_CTRL 0x8e +#define RT5660_SPK_AMP_SPKVDD 0x92 +#define RT5660_MICBIAS 0x93 +#define RT5660_CLSD_OUT_CTRL1 0xa1 +#define RT5660_CLSD_OUT_CTRL2 0xa2 +#define RT5660_DIPOLE_MIC_CTRL1 0xa3 +#define RT5660_DIPOLE_MIC_CTRL2 0xa4 +#define RT5660_DIPOLE_MIC_CTRL3 0xa5 +#define RT5660_DIPOLE_MIC_CTRL4 0xa6 +#define RT5660_DIPOLE_MIC_CTRL5 0xa7 +#define RT5660_DIPOLE_MIC_CTRL6 0xa8 +#define RT5660_DIPOLE_MIC_CTRL7 0xa9 +#define RT5660_DIPOLE_MIC_CTRL8 0xaa +#define RT5660_DIPOLE_MIC_CTRL9 0xab +#define RT5660_DIPOLE_MIC_CTRL10 0xac +#define RT5660_DIPOLE_MIC_CTRL11 0xad +#define RT5660_DIPOLE_MIC_CTRL12 0xae +/* Function - Digital */ +#define RT5660_EQ_CTRL1 0xb0 +#define RT5660_EQ_CTRL2 0xb1 +#define RT5660_DRC_AGC_CTRL1 0xb3 +#define RT5660_DRC_AGC_CTRL2 0xb4 +#define RT5660_DRC_AGC_CTRL3 0xb5 +#define RT5660_DRC_AGC_CTRL4 0xb6 +#define RT5660_DRC_AGC_CTRL5 0xb7 +#define RT5660_JD_CTRL 0xbb +#define RT5660_IRQ_CTRL1 0xbd +#define RT5660_IRQ_CTRL2 0xbe +#define RT5660_INT_IRQ_ST 0xbf +#define RT5660_GPIO_CTRL1 0xc0 +#define RT5660_GPIO_CTRL2 0xc2 +#define RT5660_WIND_FILTER_CTRL1 0xd3 +#define RT5660_SV_ZCD1 0xd9 +#define RT5660_SV_ZCD2 0xda +#define RT5660_DRC1_LM_CTRL1 0xe0 +#define RT5660_DRC1_LM_CTRL2 0xe1 +#define RT5660_DRC2_LM_CTRL1 0xe2 +#define RT5660_DRC2_LM_CTRL2 0xe3 +#define RT5660_MULTI_DRC_CTRL 0xe4 +#define RT5660_DRC2_CTRL1 0xe5 +#define RT5660_DRC2_CTRL2 0xe6 +#define RT5660_DRC2_CTRL3 0xe7 +#define RT5660_DRC2_CTRL4 0xe8 +#define RT5660_DRC2_CTRL5 0xe9 +#define RT5660_ALC_PGA_CTRL1 0xea +#define RT5660_ALC_PGA_CTRL2 0xeb +#define RT5660_ALC_PGA_CTRL3 0xec +#define RT5660_ALC_PGA_CTRL4 0xed +#define RT5660_ALC_PGA_CTRL5 0xee +#define RT5660_ALC_PGA_CTRL6 0xef +#define RT5660_ALC_PGA_CTRL7 0xf0 + +/* General Control */ +#define RT5660_GEN_CTRL1 0xfa +#define RT5660_GEN_CTRL2 0xfb +#define RT5660_GEN_CTRL3 0xfc + +/* Index of Codec Private Register definition */ +#define RT5660_CHOP_DAC_ADC 0x3d + +/* Global Definition */ +#define RT5660_L_MUTE (0x1 << 15) +#define RT5660_L_MUTE_SFT 15 +#define RT5660_VOL_L_MUTE (0x1 << 14) +#define RT5660_VOL_L_SFT 14 +#define RT5660_R_MUTE (0x1 << 7) +#define RT5660_R_MUTE_SFT 7 +#define RT5660_VOL_R_MUTE (0x1 << 6) +#define RT5660_VOL_R_SFT 6 +#define RT5660_L_VOL_MASK (0x3f << 8) +#define RT5660_L_VOL_SFT 8 +#define RT5660_R_VOL_MASK (0x3f) +#define RT5660_R_VOL_SFT 0 + +/* IN1 and IN2 Control (0x0d) */ +#define RT5660_IN_DF1 (0x1 << 15) +#define RT5660_IN_SFT1 15 +#define RT5660_BST_MASK1 (0x7f << 8) +#define RT5660_BST_SFT1 8 +#define RT5660_IN_DF2 (0x1 << 7) +#define RT5660_IN_SFT2 7 +#define RT5660_BST_MASK2 (0x7f << 0) +#define RT5660_BST_SFT2 0 + +/* IN3 and IN4 Control (0x0e) */ +#define RT5660_IN_DF3 (0x1 << 15) +#define RT5660_IN_SFT3 15 +#define RT5660_BST_MASK3 (0x7f << 8) +#define RT5660_BST_SFT3 8 +#define RT5660_IN_DF4 (0x1 << 7) +#define RT5660_IN_SFT4 7 +#define RT5660_BST_MASK4 (0x7f << 0) +#define RT5660_BST_SFT4 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5660_DAC_L1_VOL_MASK (0x7f << 9) +#define RT5660_DAC_L1_VOL_SFT 9 +#define RT5660_DAC_R1_VOL_MASK (0x7f << 1) +#define RT5660_DAC_R1_VOL_SFT 1 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5660_ADC_L_VOL_MASK (0x3f << 9) +#define RT5660_ADC_L_VOL_SFT 9 +#define RT5660_ADC_R_VOL_MASK (0x3f << 1) +#define RT5660_ADC_R_VOL_SFT 1 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5660_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5660_STO1_ADC_L_BST_SFT 14 +#define RT5660_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5660_STO1_ADC_R_BST_SFT 12 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5660_M_ADC_L1 (0x1 << 14) +#define RT5660_M_ADC_L1_SFT 14 +#define RT5660_M_ADC_L2 (0x1 << 13) +#define RT5660_M_ADC_L2_SFT 13 +#define RT5660_M_ADC_R1 (0x1 << 6) +#define RT5660_M_ADC_R1_SFT 6 +#define RT5660_M_ADC_R2 (0x1 << 5) +#define RT5660_M_ADC_R2_SFT 5 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5660_M_ADCMIX_L (0x1 << 15) +#define RT5660_M_ADCMIX_L_SFT 15 +#define RT5660_M_DAC1_L (0x1 << 14) +#define RT5660_M_DAC1_L_SFT 14 +#define RT5660_M_ADCMIX_R (0x1 << 7) +#define RT5660_M_ADCMIX_R_SFT 7 +#define RT5660_M_DAC1_R (0x1 << 6) +#define RT5660_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5660_M_DAC_L1 (0x1 << 14) +#define RT5660_M_DAC_L1_SFT 14 +#define RT5660_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5660_DAC_L1_STO_L_VOL_SFT 13 +#define RT5660_M_DAC_R1_STO_L (0x1 << 9) +#define RT5660_M_DAC_R1_STO_L_SFT 9 +#define RT5660_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5660_DAC_R1_STO_L_VOL_SFT 8 +#define RT5660_M_DAC_R1 (0x1 << 6) +#define RT5660_M_DAC_R1_SFT 6 +#define RT5660_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5660_DAC_R1_STO_R_VOL_SFT 5 +#define RT5660_M_DAC_L1_STO_R (0x1 << 1) +#define RT5660_M_DAC_L1_STO_R_SFT 1 +#define RT5660_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5660_DAC_L1_STO_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5660_IF1_DAC_IN_SEL (0x3 << 14) +#define RT5660_IF1_DAC_IN_SFT 14 +#define RT5660_IF1_ADC_IN_SEL (0x3 << 12) +#define RT5660_IF1_ADC_IN_SFT 12 + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5660_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5660_G_BST3_RM_L_SFT 4 +#define RT5660_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5660_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5660_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5660_G_BST1_RM_L_SFT 13 +#define RT5660_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5660_G_OM_L_RM_L_SFT 10 +#define RT5660_M_BST3_RM_L (0x1 << 3) +#define RT5660_M_BST3_RM_L_SFT 3 +#define RT5660_M_BST2_RM_L (0x1 << 2) +#define RT5660_M_BST2_RM_L_SFT 2 +#define RT5660_M_BST1_RM_L (0x1 << 1) +#define RT5660_M_BST1_RM_L_SFT 1 +#define RT5660_M_OM_L_RM_L (0x1) +#define RT5660_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5660_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5660_G_BST3_RM_R_SFT 4 +#define RT5660_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5660_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5660_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5660_G_BST1_RM_R_SFT 13 +#define RT5660_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5660_G_OM_R_RM_R_SFT 10 +#define RT5660_M_BST3_RM_R (0x1 << 3) +#define RT5660_M_BST3_RM_R_SFT 3 +#define RT5660_M_BST2_RM_R (0x1 << 2) +#define RT5660_M_BST2_RM_R_SFT 2 +#define RT5660_M_BST1_RM_R (0x1 << 1) +#define RT5660_M_BST1_RM_R_SFT 1 +#define RT5660_M_OM_R_RM_R (0x1) +#define RT5660_M_OM_R_RM_R_SFT 0 + +/* LOUTMIX Control (0x45) */ +#define RT5660_M_DAC1_LM (0x1 << 14) +#define RT5660_M_DAC1_LM_SFT 14 +#define RT5660_M_LOVOL_M (0x1 << 13) +#define RT5660_M_LOVOL_LM_SFT 13 + +/* SPK Mixer Control (0x46) */ +#define RT5660_G_BST3_SM_MASK (0x3 << 14) +#define RT5660_G_BST3_SM_SFT 14 +#define RT5660_G_BST1_SM_MASK (0x3 << 12) +#define RT5660_G_BST1_SM_SFT 12 +#define RT5660_G_DACl_SM_MASK (0x3 << 10) +#define RT5660_G_DACl_SM_SFT 10 +#define RT5660_G_DACR_SM_MASK (0x3 << 8) +#define RT5660_G_DACR_SM_SFT 8 +#define RT5660_G_OM_L_SM_MASK (0x3 << 6) +#define RT5660_G_OM_L_SM_SFT 6 +#define RT5660_M_DACR_SM (0x1 << 5) +#define RT5660_M_DACR_SM_SFT 5 +#define RT5660_M_BST1_SM (0x1 << 4) +#define RT5660_M_BST1_SM_SFT 4 +#define RT5660_M_BST3_SM (0x1 << 3) +#define RT5660_M_BST3_SM_SFT 3 +#define RT5660_M_DACL_SM (0x1 << 2) +#define RT5660_M_DACL_SM_SFT 2 +#define RT5660_M_OM_L_SM (0x1 << 1) +#define RT5660_M_OM_L_SM_SFT 1 + +/* SPOMIX Control (0x48) */ +#define RT5660_M_DAC_R_SPM (0x1 << 14) +#define RT5660_M_DAC_R_SPM_SFT 14 +#define RT5660_M_DAC_L_SPM (0x1 << 13) +#define RT5660_M_DAC_L_SPM_SFT 13 +#define RT5660_M_SV_SPM (0x1 << 12) +#define RT5660_M_SV_SPM_SFT 12 +#define RT5660_M_BST1_SPM (0x1 << 11) +#define RT5660_M_BST1_SPM_SFT 11 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5660_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5660_G_BST3_OM_L_SFT 13 +#define RT5660_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5660_G_BST2_OM_L_SFT 10 +#define RT5660_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5660_G_BST1_OM_L_SFT 7 +#define RT5660_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5660_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5660_G_DAC_R1_OM_L_MASK (0x7 << 10) +#define RT5660_G_DAC_R1_OM_L_SFT 10 +#define RT5660_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5660_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5660_M_BST3_OM_L (0x1 << 5) +#define RT5660_M_BST3_OM_L_SFT 5 +#define RT5660_M_BST2_OM_L (0x1 << 4) +#define RT5660_M_BST2_OM_L_SFT 4 +#define RT5660_M_BST1_OM_L (0x1 << 3) +#define RT5660_M_BST1_OM_L_SFT 3 +#define RT5660_M_RM_L_OM_L (0x1 << 2) +#define RT5660_M_RM_L_OM_L_SFT 2 +#define RT5660_M_DAC_R_OM_L (0x1 << 1) +#define RT5660_M_DAC_R_OM_L_SFT 1 +#define RT5660_M_DAC_L_OM_L (0x1) +#define RT5660_M_DAC_L_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5660_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5660_G_BST2_OM_R_SFT 10 +#define RT5660_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5660_G_BST1_OM_R_SFT 7 +#define RT5660_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5660_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5660_G_DAC_L_OM_R_MASK (0x7 << 10) +#define RT5660_G_DAC_L_OM_R_SFT 10 +#define RT5660_G_DAC_R_OM_R_MASK (0x7 << 7) +#define RT5660_G_DAC_R_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5660_M_BST2_OM_R (0x1 << 4) +#define RT5660_M_BST2_OM_R_SFT 4 +#define RT5660_M_BST1_OM_R (0x1 << 3) +#define RT5660_M_BST1_OM_R_SFT 3 +#define RT5660_M_RM_R_OM_R (0x1 << 2) +#define RT5660_M_RM_R_OM_R_SFT 2 +#define RT5660_M_DAC_L_OM_R (0x1 << 1) +#define RT5660_M_DAC_L_OM_R_SFT 1 +#define RT5660_M_DAC_R_OM_R (0x1) +#define RT5660_M_DAC_R_OM_R_SFT 0 + +/* Power Management for Digital 1 (0x61) */ +#define RT5660_PWR_I2S1 (0x1 << 15) +#define RT5660_PWR_I2S1_BIT 15 +#define RT5660_PWR_DAC_L1 (0x1 << 12) +#define RT5660_PWR_DAC_L1_BIT 12 +#define RT5660_PWR_DAC_R1 (0x1 << 11) +#define RT5660_PWR_DAC_R1_BIT 11 +#define RT5660_PWR_ADC_L (0x1 << 2) +#define RT5660_PWR_ADC_L_BIT 2 +#define RT5660_PWR_ADC_R (0x1 << 1) +#define RT5660_PWR_ADC_R_BIT 1 +#define RT5660_PWR_CLS_D (0x1) +#define RT5660_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5660_PWR_ADC_S1F (0x1 << 15) +#define RT5660_PWR_ADC_S1F_BIT 15 +#define RT5660_PWR_DAC_S1F (0x1 << 11) +#define RT5660_PWR_DAC_S1F_BIT 11 + +/* Power Management for Analog 1 (0x63) */ +#define RT5660_PWR_VREF1 (0x1 << 15) +#define RT5660_PWR_VREF1_BIT 15 +#define RT5660_PWR_FV1 (0x1 << 14) +#define RT5660_PWR_FV1_BIT 14 +#define RT5660_PWR_MB (0x1 << 13) +#define RT5660_PWR_MB_BIT 13 +#define RT5660_PWR_BG (0x1 << 11) +#define RT5660_PWR_BG_BIT 11 +#define RT5660_PWR_HP_L (0x1 << 7) +#define RT5660_PWR_HP_L_BIT 7 +#define RT5660_PWR_HP_R (0x1 << 6) +#define RT5660_PWR_HP_R_BIT 6 +#define RT5660_PWR_HA (0x1 << 5) +#define RT5660_PWR_HA_BIT 5 +#define RT5660_PWR_VREF2 (0x1 << 4) +#define RT5660_PWR_VREF2_BIT 4 +#define RT5660_PWR_FV2 (0x1 << 3) +#define RT5660_PWR_FV2_BIT 3 +#define RT5660_PWR_LDO2 (0x1 << 2) +#define RT5660_PWR_LDO2_BIT 2 + +/* Power Management for Analog 2 (0x64) */ +#define RT5660_PWR_BST1 (0x1 << 15) +#define RT5660_PWR_BST1_BIT 15 +#define RT5660_PWR_BST2 (0x1 << 14) +#define RT5660_PWR_BST2_BIT 14 +#define RT5660_PWR_BST3 (0x1 << 13) +#define RT5660_PWR_BST3_BIT 13 +#define RT5660_PWR_MB1 (0x1 << 11) +#define RT5660_PWR_MB1_BIT 11 +#define RT5660_PWR_MB2 (0x1 << 10) +#define RT5660_PWR_MB2_BIT 10 +#define RT5660_PWR_PLL (0x1 << 9) +#define RT5660_PWR_PLL_BIT 9 + +/* Power Management for Mixer (0x65) */ +#define RT5660_PWR_OM_L (0x1 << 15) +#define RT5660_PWR_OM_L_BIT 15 +#define RT5660_PWR_OM_R (0x1 << 14) +#define RT5660_PWR_OM_R_BIT 14 +#define RT5660_PWR_SM (0x1 << 13) +#define RT5660_PWR_SM_BIT 13 +#define RT5660_PWR_RM_L (0x1 << 11) +#define RT5660_PWR_RM_L_BIT 11 +#define RT5660_PWR_RM_R (0x1 << 10) +#define RT5660_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5660_PWR_SV (0x1 << 15) +#define RT5660_PWR_SV_BIT 15 +#define RT5660_PWR_LV_L (0x1 << 11) +#define RT5660_PWR_LV_L_BIT 11 +#define RT5660_PWR_LV_R (0x1 << 10) +#define RT5660_PWR_LV_R_BIT 10 + +/* I2S1 Audio Serial Data Port Control (0x70) */ +#define RT5660_I2S_MS_MASK (0x1 << 15) +#define RT5660_I2S_MS_SFT 15 +#define RT5660_I2S_MS_M (0x0 << 15) +#define RT5660_I2S_MS_S (0x1 << 15) +#define RT5660_I2S_O_CP_MASK (0x3 << 10) +#define RT5660_I2S_O_CP_SFT 10 +#define RT5660_I2S_O_CP_OFF (0x0 << 10) +#define RT5660_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5660_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5660_I2S_I_CP_MASK (0x3 << 8) +#define RT5660_I2S_I_CP_SFT 8 +#define RT5660_I2S_I_CP_OFF (0x0 << 8) +#define RT5660_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5660_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5660_I2S_BP_MASK (0x1 << 7) +#define RT5660_I2S_BP_SFT 7 +#define RT5660_I2S_BP_NOR (0x0 << 7) +#define RT5660_I2S_BP_INV (0x1 << 7) +#define RT5660_I2S_DL_MASK (0x3 << 2) +#define RT5660_I2S_DL_SFT 2 +#define RT5660_I2S_DL_16 (0x0 << 2) +#define RT5660_I2S_DL_20 (0x1 << 2) +#define RT5660_I2S_DL_24 (0x2 << 2) +#define RT5660_I2S_DL_8 (0x3 << 2) +#define RT5660_I2S_DF_MASK (0x3) +#define RT5660_I2S_DF_SFT 0 +#define RT5660_I2S_DF_I2S (0x0) +#define RT5660_I2S_DF_LEFT (0x1) +#define RT5660_I2S_DF_PCM_A (0x2) +#define RT5660_I2S_DF_PCM_B (0x3) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5660_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5660_I2S_BCLK_MS1_SFT 15 +#define RT5660_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5660_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5660_I2S_PD1_MASK (0x7 << 12) +#define RT5660_I2S_PD1_SFT 12 +#define RT5660_I2S_PD1_1 (0x0 << 12) +#define RT5660_I2S_PD1_2 (0x1 << 12) +#define RT5660_I2S_PD1_3 (0x2 << 12) +#define RT5660_I2S_PD1_4 (0x3 << 12) +#define RT5660_I2S_PD1_6 (0x4 << 12) +#define RT5660_I2S_PD1_8 (0x5 << 12) +#define RT5660_I2S_PD1_12 (0x6 << 12) +#define RT5660_I2S_PD1_16 (0x7 << 12) +#define RT5660_DAC_OSR_MASK (0x3 << 2) +#define RT5660_DAC_OSR_SFT 2 +#define RT5660_DAC_OSR_128 (0x0 << 2) +#define RT5660_DAC_OSR_64 (0x1 << 2) +#define RT5660_DAC_OSR_32 (0x2 << 2) +#define RT5660_DAC_OSR_16 (0x3 << 2) +#define RT5660_ADC_OSR_MASK (0x3) +#define RT5660_ADC_OSR_SFT 0 +#define RT5660_ADC_OSR_128 (0x0) +#define RT5660_ADC_OSR_64 (0x1) +#define RT5660_ADC_OSR_32 (0x2) +#define RT5660_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5660_RESET_ADF (0x1 << 13) +#define RT5660_RESET_ADF_SFT 13 +#define RT5660_RESET_DAF (0x1 << 12) +#define RT5660_RESET_DAF_SFT 12 +#define RT5660_DAHPF_EN (0x1 << 11) +#define RT5660_DAHPF_EN_SFT 11 +#define RT5660_ADHPF_EN (0x1 << 10) +#define RT5660_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5660_DMIC_1_EN_MASK (0x1 << 15) +#define RT5660_DMIC_1_EN_SFT 15 +#define RT5660_DMIC_1_DIS (0x0 << 15) +#define RT5660_DMIC_1_EN (0x1 << 15) +#define RT5660_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5660_DMIC_1L_LH_SFT 13 +#define RT5660_DMIC_1L_LH_RISING (0x0 << 13) +#define RT5660_DMIC_1L_LH_FALLING (0x1 << 13) +#define RT5660_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5660_DMIC_1R_LH_SFT 12 +#define RT5660_DMIC_1R_LH_RISING (0x0 << 12) +#define RT5660_DMIC_1R_LH_FALLING (0x1 << 12) +#define RT5660_SEL_DMIC_DATA_MASK (0x1 << 11) +#define RT5660_SEL_DMIC_DATA_SFT 11 +#define RT5660_SEL_DMIC_DATA_GPIO2 (0x0 << 11) +#define RT5660_SEL_DMIC_DATA_IN1P (0x1 << 11) +#define RT5660_DMIC_CLK_MASK (0x7 << 5) +#define RT5660_DMIC_CLK_SFT 5 + +/* Global Clock Control (0x80) */ +#define RT5660_SCLK_SRC_MASK (0x3 << 14) +#define RT5660_SCLK_SRC_SFT 14 +#define RT5660_SCLK_SRC_MCLK (0x0 << 14) +#define RT5660_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5660_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5660_PLL1_SRC_MASK (0x3 << 12) +#define RT5660_PLL1_SRC_SFT 12 +#define RT5660_PLL1_SRC_MCLK (0x0 << 12) +#define RT5660_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5660_PLL1_SRC_RCCLK (0x2 << 12) +#define RT5660_PLL1_PD_MASK (0x1 << 3) +#define RT5660_PLL1_PD_SFT 3 +#define RT5660_PLL1_PD_1 (0x0 << 3) +#define RT5660_PLL1_PD_2 (0x1 << 3) + +#define RT5660_PLL_INP_MAX 40000000 +#define RT5660_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5660_PLL_N_MAX 0x1ff +#define RT5660_PLL_N_MASK (RT5660_PLL_N_MAX << 7) +#define RT5660_PLL_N_SFT 7 +#define RT5660_PLL_K_MAX 0x1f +#define RT5660_PLL_K_MASK (RT5660_PLL_K_MAX) +#define RT5660_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5660_PLL_M_MAX 0xf +#define RT5660_PLL_M_MASK (RT5660_PLL_M_MAX << 12) +#define RT5660_PLL_M_SFT 12 +#define RT5660_PLL_M_BP (0x1 << 11) +#define RT5660_PLL_M_BP_SFT 11 + +/* Class D Over Current Control (0x8c) */ +#define RT5660_CLSD_OC_MASK (0x1 << 9) +#define RT5660_CLSD_OC_SFT 9 +#define RT5660_CLSD_OC_PU (0x0 << 9) +#define RT5660_CLSD_OC_PD (0x1 << 9) +#define RT5660_AUTO_PD_MASK (0x1 << 8) +#define RT5660_AUTO_PD_SFT 8 +#define RT5660_AUTO_PD_DIS (0x0 << 8) +#define RT5660_AUTO_PD_EN (0x1 << 8) +#define RT5660_CLSD_OC_TH_MASK (0x3f) +#define RT5660_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5660_CLSD_RATIO_MASK (0xf << 12) +#define RT5660_CLSD_RATIO_SFT 12 + +/* Lout Amp Control 1 (0x8e) */ +#define RT5660_LOUT_CO_MASK (0x1 << 4) +#define RT5660_LOUT_CO_SFT 4 +#define RT5660_LOUT_CO_DIS (0x0 << 4) +#define RT5660_LOUT_CO_EN (0x1 << 4) +#define RT5660_LOUT_CB_MASK (0x1) +#define RT5660_LOUT_CB_SFT 0 +#define RT5660_LOUT_CB_PD (0x0) +#define RT5660_LOUT_CB_PU (0x1) + +/* SPKVDD detection control (0x92) */ +#define RT5660_SPKVDD_DET_MASK (0x1 << 15) +#define RT5660_SPKVDD_DET_SFT 15 +#define RT5660_SPKVDD_DET_DIS (0x0 << 15) +#define RT5660_SPKVDD_DET_EN (0x1 << 15) +#define RT5660_SPK_AG_MASK (0x1 << 14) +#define RT5660_SPK_AG_SFT 14 +#define RT5660_SPK_AG_DIS (0x0 << 14) +#define RT5660_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5660_MIC1_BS_MASK (0x1 << 15) +#define RT5660_MIC1_BS_SFT 15 +#define RT5660_MIC1_BS_9AV (0x0 << 15) +#define RT5660_MIC1_BS_75AV (0x1 << 15) +#define RT5660_MIC2_BS_MASK (0x1 << 14) +#define RT5660_MIC2_BS_SFT 14 +#define RT5660_MIC2_BS_9AV (0x0 << 14) +#define RT5660_MIC2_BS_75AV (0x1 << 14) +#define RT5660_MIC1_OVCD_MASK (0x1 << 11) +#define RT5660_MIC1_OVCD_SFT 11 +#define RT5660_MIC1_OVCD_DIS (0x0 << 11) +#define RT5660_MIC1_OVCD_EN (0x1 << 11) +#define RT5660_MIC1_OVTH_MASK (0x3 << 9) +#define RT5660_MIC1_OVTH_SFT 9 +#define RT5660_MIC1_OVTH_600UA (0x0 << 9) +#define RT5660_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5660_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5660_MIC2_OVCD_MASK (0x1 << 8) +#define RT5660_MIC2_OVCD_SFT 8 +#define RT5660_MIC2_OVCD_DIS (0x0 << 8) +#define RT5660_MIC2_OVCD_EN (0x1 << 8) +#define RT5660_MIC2_OVTH_MASK (0x3 << 6) +#define RT5660_MIC2_OVTH_SFT 6 +#define RT5660_MIC2_OVTH_600UA (0x0 << 6) +#define RT5660_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5660_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5660_PWR_CLK25M_MASK (0x1 << 4) +#define RT5660_PWR_CLK25M_SFT 4 +#define RT5660_PWR_CLK25M_PD (0x0 << 4) +#define RT5660_PWR_CLK25M_PU (0x1 << 4) + +/* EQ Control 1 (0xb0) */ +#define RT5660_EQ_SRC_MASK (0x1 << 15) +#define RT5660_EQ_SRC_SFT 15 +#define RT5660_EQ_SRC_DAC (0x0 << 15) +#define RT5660_EQ_SRC_ADC (0x1 << 15) +#define RT5660_EQ_UPD (0x1 << 14) +#define RT5660_EQ_UPD_BIT 14 + +/* Jack Detect Control (0xbb) */ +#define RT5660_JD_MASK (0x3 << 14) +#define RT5660_JD_SFT 14 +#define RT5660_JD_DIS (0x0 << 14) +#define RT5660_JD_GPIO1 (0x1 << 14) +#define RT5660_JD_GPIO2 (0x2 << 14) +#define RT5660_JD_LOUT_MASK (0x1 << 11) +#define RT5660_JD_LOUT_SFT 11 +#define RT5660_JD_LOUT_DIS (0x0 << 11) +#define RT5660_JD_LOUT_EN (0x1 << 11) +#define RT5660_JD_LOUT_TRG_MASK (0x1 << 10) +#define RT5660_JD_LOUT_TRG_SFT 10 +#define RT5660_JD_LOUT_TRG_LO (0x0 << 10) +#define RT5660_JD_LOUT_TRG_HI (0x1 << 10) +#define RT5660_JD_SPO_MASK (0x1 << 9) +#define RT5660_JD_SPO_SFT 9 +#define RT5660_JD_SPO_DIS (0x0 << 9) +#define RT5660_JD_SPO_EN (0x1 << 9) +#define RT5660_JD_SPO_TRG_MASK (0x1 << 8) +#define RT5660_JD_SPO_TRG_SFT 8 +#define RT5660_JD_SPO_TRG_LO (0x0 << 8) +#define RT5660_JD_SPO_TRG_HI (0x1 << 8) + +/* IRQ Control 1 (0xbd) */ +#define RT5660_IRQ_JD_MASK (0x1 << 15) +#define RT5660_IRQ_JD_SFT 15 +#define RT5660_IRQ_JD_BP (0x0 << 15) +#define RT5660_IRQ_JD_NOR (0x1 << 15) +#define RT5660_IRQ_OT_MASK (0x1 << 14) +#define RT5660_IRQ_OT_SFT 14 +#define RT5660_IRQ_OT_BP (0x0 << 14) +#define RT5660_IRQ_OT_NOR (0x1 << 14) +#define RT5660_JD_STKY_MASK (0x1 << 13) +#define RT5660_JD_STKY_SFT 13 +#define RT5660_JD_STKY_DIS (0x0 << 13) +#define RT5660_JD_STKY_EN (0x1 << 13) +#define RT5660_OT_STKY_MASK (0x1 << 12) +#define RT5660_OT_STKY_SFT 12 +#define RT5660_OT_STKY_DIS (0x0 << 12) +#define RT5660_OT_STKY_EN (0x1 << 12) +#define RT5660_JD_P_MASK (0x1 << 11) +#define RT5660_JD_P_SFT 11 +#define RT5660_JD_P_NOR (0x0 << 11) +#define RT5660_JD_P_INV (0x1 << 11) +#define RT5660_OT_P_MASK (0x1 << 10) +#define RT5660_OT_P_SFT 10 +#define RT5660_OT_P_NOR (0x0 << 10) +#define RT5660_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5660_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5660_IRQ_MB1_OC_SFT 15 +#define RT5660_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5660_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5660_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5660_IRQ_MB2_OC_SFT 14 +#define RT5660_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5660_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5660_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5660_MB1_OC_STKY_SFT 11 +#define RT5660_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5660_MB1_OC_STKY_EN (0x1 << 11) +#define RT5660_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5660_MB2_OC_STKY_SFT 10 +#define RT5660_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5660_MB2_OC_STKY_EN (0x1 << 10) +#define RT5660_MB1_OC_P_MASK (0x1 << 7) +#define RT5660_MB1_OC_P_SFT 7 +#define RT5660_MB1_OC_P_NOR (0x0 << 7) +#define RT5660_MB1_OC_P_INV (0x1 << 7) +#define RT5660_MB2_OC_P_MASK (0x1 << 6) +#define RT5660_MB2_OC_P_SFT 6 +#define RT5660_MB2_OC_P_NOR (0x0 << 6) +#define RT5660_MB2_OC_P_INV (0x1 << 6) +#define RT5660_MB1_OC_CLR (0x1 << 3) +#define RT5660_MB1_OC_CLR_SFT 3 +#define RT5660_MB2_OC_CLR (0x1 << 2) +#define RT5660_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5660_GP2_PIN_MASK (0x1 << 14) +#define RT5660_GP2_PIN_SFT 14 +#define RT5660_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5660_GP2_PIN_DMIC1_SDA (0x1 << 14) +#define RT5660_GP1_PIN_MASK (0x3 << 12) +#define RT5660_GP1_PIN_SFT 12 +#define RT5660_GP1_PIN_GPIO1 (0x0 << 12) +#define RT5660_GP1_PIN_DMIC1_SCL (0x1 << 12) +#define RT5660_GP1_PIN_IRQ (0x2 << 12) +#define RT5660_GPIO_M_MASK (0x1 << 9) +#define RT5660_GPIO_M_SFT 9 +#define RT5660_GPIO_M_FLT (0x0 << 9) +#define RT5660_GPIO_M_PH (0x1 << 9) + +/* GPIO Control 3 (0xc2) */ +#define RT5660_GP2_PF_MASK (0x1 << 5) +#define RT5660_GP2_PF_SFT 5 +#define RT5660_GP2_PF_IN (0x0 << 5) +#define RT5660_GP2_PF_OUT (0x1 << 5) +#define RT5660_GP2_OUT_MASK (0x1 << 4) +#define RT5660_GP2_OUT_SFT 4 +#define RT5660_GP2_OUT_LO (0x0 << 4) +#define RT5660_GP2_OUT_HI (0x1 << 4) +#define RT5660_GP2_P_MASK (0x1 << 3) +#define RT5660_GP2_P_SFT 3 +#define RT5660_GP2_P_NOR (0x0 << 3) +#define RT5660_GP2_P_INV (0x1 << 3) +#define RT5660_GP1_PF_MASK (0x1 << 2) +#define RT5660_GP1_PF_SFT 2 +#define RT5660_GP1_PF_IN (0x0 << 2) +#define RT5660_GP1_PF_OUT (0x1 << 2) +#define RT5660_GP1_OUT_MASK (0x1 << 1) +#define RT5660_GP1_OUT_SFT 1 +#define RT5660_GP1_OUT_LO (0x0 << 1) +#define RT5660_GP1_OUT_HI (0x1 << 1) +#define RT5660_GP1_P_MASK (0x1) +#define RT5660_GP1_P_SFT 0 +#define RT5660_GP1_P_NOR (0x0) +#define RT5660_GP1_P_INV (0x1) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5660_SV_MASK (0x1 << 15) +#define RT5660_SV_SFT 15 +#define RT5660_SV_DIS (0x0 << 15) +#define RT5660_SV_EN (0x1 << 15) +#define RT5660_SPO_SV_MASK (0x1 << 14) +#define RT5660_SPO_SV_SFT 14 +#define RT5660_SPO_SV_DIS (0x0 << 14) +#define RT5660_SPO_SV_EN (0x1 << 14) +#define RT5660_OUT_SV_MASK (0x1 << 12) +#define RT5660_OUT_SV_SFT 12 +#define RT5660_OUT_SV_DIS (0x0 << 12) +#define RT5660_OUT_SV_EN (0x1 << 12) +#define RT5660_ZCD_DIG_MASK (0x1 << 11) +#define RT5660_ZCD_DIG_SFT 11 +#define RT5660_ZCD_DIG_DIS (0x0 << 11) +#define RT5660_ZCD_DIG_EN (0x1 << 11) +#define RT5660_ZCD_MASK (0x1 << 10) +#define RT5660_ZCD_SFT 10 +#define RT5660_ZCD_PD (0x0 << 10) +#define RT5660_ZCD_PU (0x1 << 10) +#define RT5660_SV_DLY_MASK (0xf) +#define RT5660_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5660_ZCD_SPO_MASK (0x1 << 15) +#define RT5660_ZCD_SPO_SFT 15 +#define RT5660_ZCD_SPO_DIS (0x0 << 15) +#define RT5660_ZCD_SPO_EN (0x1 << 15) +#define RT5660_ZCD_OMR_MASK (0x1 << 8) +#define RT5660_ZCD_OMR_SFT 8 +#define RT5660_ZCD_OMR_DIS (0x0 << 8) +#define RT5660_ZCD_OMR_EN (0x1 << 8) +#define RT5660_ZCD_OML_MASK (0x1 << 7) +#define RT5660_ZCD_OML_SFT 7 +#define RT5660_ZCD_OML_DIS (0x0 << 7) +#define RT5660_ZCD_OML_EN (0x1 << 7) +#define RT5660_ZCD_SPM_MASK (0x1 << 6) +#define RT5660_ZCD_SPM_SFT 6 +#define RT5660_ZCD_SPM_DIS (0x0 << 6) +#define RT5660_ZCD_SPM_EN (0x1 << 6) +#define RT5660_ZCD_RMR_MASK (0x1 << 5) +#define RT5660_ZCD_RMR_SFT 5 +#define RT5660_ZCD_RMR_DIS (0x0 << 5) +#define RT5660_ZCD_RMR_EN (0x1 << 5) +#define RT5660_ZCD_RML_MASK (0x1 << 4) +#define RT5660_ZCD_RML_SFT 4 +#define RT5660_ZCD_RML_DIS (0x0 << 4) +#define RT5660_ZCD_RML_EN (0x1 << 4) + +/* General Control 1 (0xfa) */ +#define RT5660_PWR_VREF_HP (0x1 << 11) +#define RT5660_PWR_VREF_HP_SFT 11 +#define RT5660_DIG_GATE_CTRL (0x1) +#define RT5660_DIG_GATE_CTRL_SFT 0 + +/* System Clock Source */ +#define RT5660_SCLK_S_MCLK 0 +#define RT5660_SCLK_S_PLL1 1 +#define RT5660_SCLK_S_RCCLK 2 + +/* PLL1 Source */ +#define RT5660_PLL1_S_MCLK 0 +#define RT5660_PLL1_S_BCLK 1 + +enum { + RT5660_AIF1, + RT5660_AIFS, +}; + +struct rt5660_priv { + struct snd_soc_codec *codec; + struct rt5660_platform_data pdata; + struct regmap *regmap; + struct clk *mclk; + + int sysclk; + int sysclk_src; + int lrck[RT5660_AIFS]; + int bclk[RT5660_AIFS]; + int master[RT5660_AIFS]; + + int pll_src; + int pll_in; + int pll_out; +}; + +#endif -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ASoC: da7219: software reset codec at probe
by Xing Zheng 14 Sep '16

14 Sep '16

From: Hsin-Yu Chao <hychao(a)chromium.org> Da7219 does not trigger interrupt to report jack status when system boots from warm reset because its power remains on during warm reset. Doing software reset at probe to handle this. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v2: - change the commit message for more clear explain sound/soc/codecs/da7219.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 737e914..9d08c11 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1941,6 +1941,14 @@ static int da7219_i2c_probe(struct i2c_client *i2c, return ret; } + /* Software reset codec. */ + regmap_write_bits(da7219->regmap, DA7219_ACCDET_CONFIG_1, + DA7219_ACCDET_EN_MASK, 0); + regmap_write_bits(da7219->regmap, DA7219_CIF_CTRL, + DA7219_CIF_REG_SOFT_RESET_MASK, 0); + regmap_write_bits(da7219->regmap, DA7219_SYSTEM_ACTIVE, + DA7219_SYSTEM_ACTIVE_MASK, 0); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219, &da7219_dai, 1); if (ret < 0) { -- 1.9.1

1 0

[alsa-devel] [PATCH 4/4] hda: thinkpad_helper: Add support for hid-lenovo LED control
by Dennis Wassenberg 14 Sep '16

14 Sep '16

Make the thinkpad_helper able to support not only led control over acpi with thinkpad_acpi driver but also led control over hid-lenovo. The hid-lenovo driver adapted the led control api of thinkpad_acpi. Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- sound/pci/hda/thinkpad_helper.c | 149 ++++++++++++++++++++++++++++++---------- 1 file changed, 111 insertions(+), 38 deletions(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 62741a7..c24a4a9 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -2,79 +2,152 @@ * to be included from codec driver */ -#if IS_ENABLED(CONFIG_THINKPAD_ACPI) - +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) || IS_ENABLED(CONFIG_HID_LENOVO) #include <linux/acpi.h> +#include <linux/hid-lenovo.h> #include <linux/thinkpad_acpi.h> -static int (*led_set_func)(int, bool); +static int (*led_set_func_tpacpi)(int, bool); +static int (*led_set_func_hid_lenovo)(int, bool); static void (*old_vmaster_hook)(void *, int); static bool is_thinkpad(struct hda_codec *codec) { + return (codec->core.subsystem_id >> 16 == 0x17aa); +} + +static bool is_thinkpad_acpi(struct hda_codec *codec) +{ return (codec->core.subsystem_id >> 16 == 0x17aa) && (acpi_dev_found("LEN0068") || acpi_dev_found("IBM0068")); } -static void update_tpacpi_mute_led(void *private_data, int enabled) +static void update_thinkpad_mute_led(void *private_data, int enabled) { if (old_vmaster_hook) old_vmaster_hook(private_data, enabled); - if (led_set_func) - led_set_func(TPACPI_LED_MUTE, !enabled); + if (led_set_func_tpacpi) + led_set_func_tpacpi(TPACPI_LED_MUTE, !enabled); + + if (led_set_func_hid_lenovo) + led_set_func_hid_lenovo(HID_LENOVO_LED_MUTE, !enabled); } -static void update_tpacpi_micmute_led(struct hda_codec *codec, + + +static void update_thinkpad_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - if (!ucontrol || !led_set_func) + if (!ucontrol) return; if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { /* TODO: How do I verify if it's a mono or stereo here? */ bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); + if (led_set_func_tpacpi) + led_set_func_tpacpi(TPACPI_LED_MICMUTE, !val); + if (led_set_func_hid_lenovo) + led_set_func_hid_lenovo(HID_LENOVO_LED_MICMUTE, !val); } } -static void hda_fixup_thinkpad(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +static int hda_fixup_thinkpad_acpi(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; - bool removefunc = false; + int ret = -ENXIO; - if (action == HDA_FIXUP_ACT_PROBE) { - if (!is_thinkpad(codec)) - return; - if (!led_set_func) - led_set_func = symbol_request(tpacpi_led_set); - if (!led_set_func) { - codec_warn(codec, - "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); - return; - } + if (!is_thinkpad(codec)) + return -ENODEV; + if (!is_thinkpad_acpi(codec)) + return -ENODEV; + if (!led_set_func_tpacpi) + led_set_func_tpacpi = symbol_request(tpacpi_led_set); + if (!led_set_func_tpacpi) { + codec_warn(codec, + "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); + return -ENOENT; + } - removefunc = true; - if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { - old_vmaster_hook = spec->vmaster_mute.hook; - spec->vmaster_mute.hook = update_tpacpi_mute_led; - removefunc = false; - } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->num_adc_nids > 1) - codec_dbg(codec, - "Skipping micmute LED control due to several ADCs"); - else { - spec->cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } + if (led_set_func_tpacpi(TPACPI_LED_MUTE, false) >= 0) { + old_vmaster_hook = spec->vmaster_mute.hook; + spec->vmaster_mute.hook = update_thinkpad_mute_led; + ret = 0; + } + + if (led_set_func_tpacpi(TPACPI_LED_MICMUTE, false) >= 0) { + if (spec->num_adc_nids > 1) + codec_dbg(codec, + "Skipping micmute LED control due to several ADCs"); + else { + spec->cap_sync_hook = update_thinkpad_micmute_led; + ret = 0; } } - if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + return ret; +} + +static int hda_fixup_thinkpad_hid(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + int ret = 0; + + if (!is_thinkpad(codec)) + return -ENODEV; + if (!led_set_func_hid_lenovo) + led_set_func_hid_lenovo = symbol_request(hid_lenovo_led_set); + if (!led_set_func_hid_lenovo) { + codec_warn(codec, + "Failed to find hid-lenovo symbol hid_lenovo_led_set\n"); + return -ENOENT; + } + + if (update_thinkpad_mute_led != spec->vmaster_mute.hook) + old_vmaster_hook = spec->vmaster_mute.hook; + + // do not remove hook if setting delay does not work currently because + // it is a usb hid devices which is not connected right now + // maybe is will be connected later + led_set_func_hid_lenovo(HID_LENOVO_LED_MUTE, false); + spec->vmaster_mute.hook = update_thinkpad_mute_led; + + led_set_func_hid_lenovo(HID_LENOVO_LED_MICMUTE, false); + spec->cap_sync_hook = update_thinkpad_micmute_led; + + return ret; +} + +static void hda_fixup_thinkpad(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + int ret_fixup_acpi = 0; + int ret_fixup_hid = 0; + bool remove = 0; + + if (action == HDA_FIXUP_ACT_PROBE) { + ret_fixup_acpi = hda_fixup_thinkpad_acpi(codec); + ret_fixup_hid = hda_fixup_thinkpad_hid(codec); + } + + if (led_set_func_tpacpi && + (action == HDA_FIXUP_ACT_FREE || ret_fixup_acpi)) { + symbol_put(tpacpi_led_set); - led_set_func = NULL; + remove = true; + } + + if (led_set_func_hid_lenovo && + (action == HDA_FIXUP_ACT_FREE || ret_fixup_hid)) { + + symbol_put(hid_lenovo_led_set); + remove = true; + } + + + if (remove) { + led_set_func_tpacpi = NULL; + led_set_func_hid_lenovo = NULL; old_vmaster_hook = NULL; } } -- 1.9.1

2 3

[alsa-devel] [PATCH 2/4] HID: add device id for Lenovo X1 Tablet Cover
by Dennis Wassenberg 14 Sep '16

14 Sep '16

Add device id for Lenovo X1 Tablet Cover Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- drivers/hid/hid-core.c | 1 + drivers/hid/hid-ids.h | 1 + 2 files changed, 2 insertions(+) diff --git a/drivers/hid/hid-core.c b/drivers/hid/hid-core.c index 6add0b6..ba6a200 100644 --- a/drivers/hid/hid-core.c +++ b/drivers/hid/hid-core.c @@ -2111,6 +2111,7 @@ void hid_disconnect(struct hid_device *hdev) { HID_BLUETOOTH_DEVICE(USB_VENDOR_ID_NINTENDO, USB_DEVICE_ID_NINTENDO_WIIMOTE2) }, { HID_USB_DEVICE(USB_VENDOR_ID_RAZER, USB_DEVICE_ID_RAZER_BLADE_14) }, { HID_USB_DEVICE(USB_VENDOR_ID_CMEDIA, USB_DEVICE_ID_CM6533) }, + { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_X1_COVER) }, { } }; diff --git a/drivers/hid/hid-ids.h b/drivers/hid/hid-ids.h index 3466f0d..1f08fb5 100644 --- a/drivers/hid/hid-ids.h +++ b/drivers/hid/hid-ids.h @@ -615,6 +615,7 @@ #define USB_DEVICE_ID_LENOVO_CUSBKBD 0x6047 #define USB_DEVICE_ID_LENOVO_CBTKBD 0x6048 #define USB_DEVICE_ID_LENOVO_TPPRODOCK 0x6067 +#define USB_DEVICE_ID_LENOVO_X1_COVER 0x6085 #define USB_VENDOR_ID_LG 0x1fd2 #define USB_DEVICE_ID_LG_MULTITOUCH 0x0064 -- 1.9.1

2 2

[alsa-devel] [PATCH 1/4] hda: Rename fixup thinkpad_acpi to fixup_thinkpad
by Dennis Wassenberg 14 Sep '16

14 Sep '16

This is some preparatory work to not only support led control in thinkpad_helper over thinkpad_acpi. A further patch will introduce led control over hid-lenovo. This will make it possible to use the thinkpad_helper not only with lenovo devices via acpi but also with hid-lenovo driver if there is a hid-lenovo compatible device found. Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- sound/pci/hda/patch_conexant.c | 14 +++++++------- sound/pci/hda/patch_realtek.c | 24 ++++++++++++------------ sound/pci/hda/thinkpad_helper.c | 4 ++-- 3 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56fefbd..d54ffd9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -254,7 +254,7 @@ enum { CXT_FIXUP_HEADPHONE_MIC, CXT_FIXUP_GPIO1, CXT_FIXUP_ASPIRE_DMIC, - CXT_FIXUP_THINKPAD_ACPI, + CXT_FIXUP_THINKPAD, CXT_FIXUP_OLPC_XO, CXT_FIXUP_CAP_MIX_AMP, CXT_FIXUP_TOSHIBA_P105, @@ -263,7 +263,7 @@ enum { CXT_FIXUP_MUTE_LED_EAPD, }; -/* for hda_fixup_thinkpad_acpi() */ +/* for hda_fixup_thinkpad() */ #include "thinkpad_helper.c" static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -670,7 +670,7 @@ static void cxt_fixup_cap_mix_amp_5047(struct hda_codec *codec, .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lenovo_tp410, .chained = true, - .chain_id = CXT_FIXUP_THINKPAD_ACPI, + .chain_id = CXT_FIXUP_THINKPAD, }, [CXT_PINCFG_LEMOTE_A1004] = { .type = HDA_FIXUP_PINS, @@ -727,9 +727,9 @@ static void cxt_fixup_cap_mix_amp_5047(struct hda_codec *codec, .chained = true, .chain_id = CXT_FIXUP_GPIO1, }, - [CXT_FIXUP_THINKPAD_ACPI] = { + [CXT_FIXUP_THINKPAD] = { .type = HDA_FIXUP_FUNC, - .v.func = hda_fixup_thinkpad_acpi, + .v.func = hda_fixup_thinkpad, }, [CXT_FIXUP_OLPC_XO] = { .type = HDA_FIXUP_FUNC, @@ -828,7 +828,7 @@ static void cxt_fixup_cap_mix_amp_5047(struct hda_codec *codec, SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} @@ -839,7 +839,7 @@ static void cxt_fixup_cap_mix_amp_5047(struct hda_codec *codec, { .id = CXT_FIXUP_GPIO1, .name = "gpio1" }, { .id = CXT_FIXUP_HEADPHONE_MIC_PIN, .name = "headphone-mic-pin" }, { .id = CXT_PINCFG_LENOVO_TP410, .name = "tp410" }, - { .id = CXT_FIXUP_THINKPAD_ACPI, .name = "thinkpad" }, + { .id = CXT_FIXUP_THINKPAD, .name = "thinkpad" }, { .id = CXT_PINCFG_LEMOTE_A1004, .name = "lemote-a1004" }, { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1fe8750..6d78d76 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4754,7 +4754,7 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec, } } -/* for hda_fixup_thinkpad_acpi() */ +/* for hda_fixup_thinkpad() */ #include "thinkpad_helper.c" /* for dell wmi mic mute led */ @@ -4819,8 +4819,8 @@ enum { ALC290_FIXUP_MONO_SPEAKERS_HSJACK, ALC290_FIXUP_SUBWOOFER, ALC290_FIXUP_SUBWOOFER_HSJACK, - ALC269_FIXUP_THINKPAD_ACPI, - ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269_FIXUP_THINKPAD, + ALC269_FIXUP_DMIC_THINKPAD, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, @@ -5052,7 +5052,7 @@ enum { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, .chained = true, - .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + .chain_id = ALC269_FIXUP_THINKPAD, }, [ALC269_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -5175,7 +5175,7 @@ enum { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, .chained = true, - .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + .chain_id = ALC269_FIXUP_THINKPAD, }, [ALC269VB_FIXUP_ASUS_ZENBOOK] = { .type = HDA_FIXUP_FUNC, @@ -5261,17 +5261,17 @@ enum { .chained = true, .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, }, - [ALC269_FIXUP_THINKPAD_ACPI] = { + [ALC269_FIXUP_THINKPAD] = { .type = HDA_FIXUP_FUNC, - .v.func = hda_fixup_thinkpad_acpi, + .v.func = hda_fixup_thinkpad, .chained = true, .chain_id = ALC269_FIXUP_SKU_IGNORE, }, - [ALC269_FIXUP_DMIC_THINKPAD_ACPI] = { + [ALC269_FIXUP_DMIC_THINKPAD] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic, .chained = true, - .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + .chain_id = ALC269_FIXUP_THINKPAD, }, [ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -5471,7 +5471,7 @@ enum { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, .chained = true, - .chain_id = ALC269_FIXUP_THINKPAD_ACPI + .chain_id = ALC269_FIXUP_THINKPAD }, [ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY] = { .type = HDA_FIXUP_FUNC, @@ -5691,7 +5691,7 @@ enum { SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), - SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5766,7 +5766,7 @@ enum { SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), - SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD), {} }; diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index f0955fd..62741a7 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -38,7 +38,7 @@ static void update_tpacpi_micmute_led(struct hda_codec *codec, } } -static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, +static void hda_fixup_thinkpad(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct hda_gen_spec *spec = codec->spec; @@ -81,7 +81,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, #else /* CONFIG_THINKPAD_ACPI */ -static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, +static void hda_fixup_thinkpad(struct hda_codec *codec, const struct hda_fixup *fix, int action) { } -- 1.9.1

2 2

[alsa-devel] [v7 PATCH] amidi: add sysex-interval option
by Felipe F. Tonello 14 Sep '16

14 Sep '16

This patch adds a new option to amidi tool: sysex-interval. It adds a delay (in milliseconds) in between each SysEx message - it searches for a 0xF7 byte. This is very useful when sending firmware updates to a remote device via SysEx or any other use that requires this delay in between SysEx messages. `amidi' manual was updated with an example usage as well. Signed-off-by: Felipe F. Tonello <eu(a)felipetonello.com> --- amidi/amidi.1 | 16 ++++++++++- amidi/amidi.c | 88 +++++++++++++++++++++++++++++++++++++++++++++++------------ 2 files changed, 86 insertions(+), 18 deletions(-) diff --git a/amidi/amidi.1 b/amidi/amidi.1 index 86beb27a1e3b..5bc24ba3ad54 100644 --- a/amidi/amidi.1 +++ b/amidi/amidi.1 @@ -1,4 +1,4 @@ -.TH AMIDI 1 "16 Apr 2016" +.TH AMIDI 1 "30 Aug 2016" .SH NAME amidi \- read from and write to ALSA RawMIDI ports @@ -120,6 +120,12 @@ received MIDI commands. Does not ignore Clock bytes (F8h) when saving or printing received MIDI commands. +.TP +.I \-i, \-\-sysex-interval=mseconds +Adds a delay in between each SysEx message sent to a device. It is +useful when sending firmware updates via SysEx messages to a remote +device. + .SH EXAMPLES .TP @@ -130,6 +136,14 @@ to port .I hw:0. .TP +.B amidi \-p hw:1,0,0 -s firmware.syx \-i 100 +will send the MIDI commands in +.I firmware.syx +to port +.I hw:1,0,0 +with 100 milliseconds delay in between each SysEx message. + +.TP .B amidi \-S 'F0 43 10 4C 00 00 7E 00 F7' sends an XG Reset to the default port. diff --git a/amidi/amidi.c b/amidi/amidi.c index c20512cc96a7..a8264f181cf3 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -52,6 +52,7 @@ static int receive_file; static int dump; static float timeout; static int stop; +static int sysex_interval; static snd_rawmidi_t *input, **inputp; static snd_rawmidi_t *output, **outputp; @@ -70,19 +71,20 @@ static void usage(void) printf( "Usage: amidi options\n" "\n" - "-h, --help this help\n" - "-V, --version print current version\n" - "-l, --list-devices list all hardware ports\n" - "-L, --list-rawmidis list all RawMIDI definitions\n" - "-p, --port=name select port by name\n" - "-s, --send=file send the contents of a (.syx) file\n" - "-r, --receive=file write received data into a file\n" - "-S, --send-hex=\"...\" send hexadecimal bytes\n" - "-d, --dump print received data as hexadecimal bytes\n" - "-t, --timeout=seconds exits when no data has been received\n" - " for the specified duration\n" - "-a, --active-sensing include active sensing bytes\n" - "-c, --clock include clock bytes\n"); + "-h, --help this help\n" + "-V, --version print current version\n" + "-l, --list-devices list all hardware ports\n" + "-L, --list-rawmidis list all RawMIDI definitions\n" + "-p, --port=name select port by name\n" + "-s, --send=file send the contents of a (.syx) file\n" + "-r, --receive=file write received data into a file\n" + "-S, --send-hex=\"...\" send hexadecimal bytes\n" + "-d, --dump print received data as hexadecimal bytes\n" + "-t, --timeout=seconds exits when no data has been received\n" + " for the specified duration\n" + "-a, --active-sensing include active sensing bytes\n" + "-c, --clock include clock bytes\n" + "-i, --sysex-interval=mseconds delay in between each SysEx message\n"); } static void version(void) @@ -230,6 +232,47 @@ static void rawmidi_list(void) snd_output_close(output); } +static int send_midi_interleaved(void) +{ + int err; + char *data = send_data; + size_t buffer_size; + snd_rawmidi_params_t *param; + snd_rawmidi_status_t *st; + + snd_rawmidi_status_alloca(&st); + + snd_rawmidi_params_alloca(&param); + snd_rawmidi_params_current(output, param); + buffer_size = snd_rawmidi_params_get_buffer_size(param); + + while (data < (send_data + send_data_length)) { + int len = send_data + send_data_length - data; + char *temp; + + if (data > send_data) { + snd_rawmidi_status(output, st); + do { + /* 320 µs per byte as noted in Page 1 of MIDI spec */ + usleep((buffer_size - snd_rawmidi_status_get_avail(st)) * 320); + snd_rawmidi_status(output, st); + } while(snd_rawmidi_status_get_avail(st) < buffer_size); + usleep(sysex_interval * 1000); + } + + /* find end of SysEx */ + if ((temp = memchr(data, 0xf7, len)) != NULL) + len = temp - data + 1; + + if ((err = snd_rawmidi_write(output, data, len)) < 0) + return err; + + data += len; + } + + return 0; +} + static void load_file(void) { int fd; @@ -411,7 +454,7 @@ static void add_send_hex_data(const char *str) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlLp:s:r:S::dt:ac"; + static const char short_options[] = "hVlLp:s:r:S::dt:aci:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -425,6 +468,7 @@ int main(int argc, char *argv[]) {"timeout", 1, NULL, 't'}, {"active-sensing", 0, NULL, 'a'}, {"clock", 0, NULL, 'c'}, + {"sysex-interval", 1, NULL, 'i'}, { } }; int c, err, ok = 0; @@ -474,6 +518,9 @@ int main(int argc, char *argv[]) case 'c': ignore_clock = 0; break; + case 'i': + sysex_interval = atoi(optarg); + break; default: error("Try `amidi --help' for more information."); return 1; @@ -549,9 +596,16 @@ int main(int argc, char *argv[]) error("cannot set blocking mode: %s", snd_strerror(err)); goto _exit; } - if ((err = snd_rawmidi_write(output, send_data, send_data_length)) < 0) { - error("cannot send data: %s", snd_strerror(err)); - goto _exit; + if (!sysex_interval) { + if ((err = snd_rawmidi_write(output, send_data, send_data_length)) < 0) { + error("cannot send data: %s", snd_strerror(err)); + return err; + } + } else { + if ((err = send_midi_interleaved()) < 0) { + error("cannot send data: %s", snd_strerror(err)); + return err; + } } } -- 2.9.3

6 7

[alsa-devel] [PATCH] ALSA: seq: fix to copy from/to user space
by Takashi Sakamoto 13 Sep '16

13 Sep '16

When checking value of request for copy operation, current implementation compares shifted value to macros, while these macros are already shifted. As a result, it never performs to copy from/to user space. This commit fixes the bug. Fixes: 8ce8eb601c71('ALSA: seq: add an alternative way to handle ioctl requests' Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/seq/seq_clientmgr.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 811b95b..4c93520 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2122,7 +2122,7 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, * within 13 bits. We can safely pick up the size from the command. */ size = _IOC_SIZE(handler->cmd); - if (_IOC_DIR(handler->cmd) & IOC_IN) { + if (handler->cmd & IOC_IN) { if (copy_from_user(&buf, (const void __user *)arg, size)) return -EFAULT; } @@ -2132,7 +2132,7 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, /* Some commands includes a bug in 'dir' field. */ if (handler->cmd == SNDRV_SEQ_IOCTL_SET_QUEUE_CLIENT || handler->cmd == SNDRV_SEQ_IOCTL_SET_CLIENT_POOL || - (_IOC_DIR(handler->cmd) & IOC_OUT)) + (handler->cmd & IOC_OUT)) if (copy_to_user((void __user *)arg, &buf, size)) return -EFAULT; } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Tony Lindgren 13 Sep '16

13 Sep '16

We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Let's fix the issue by adding PM QoS latency requirement for McBSP. Based on my experiments a working latency value is somewhere around 35 ms with 40 ms breaking. So let's use a safer value of 30 ms. Note that this is different from snd_pcm_hw_constraint_step() as that can configure things based on the buffer and period size. However, that does not help to block idle between the fifo fills. Note that in theory some omaps are capable of playing audio while hitting deeper idle states. If somebody needs that and gets it working, we can set the latency requirements accordingly. Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- sound/soc/omap/mcbsp.c | 10 ++++++++++ sound/soc/omap/mcbsp.h | 2 ++ 2 files changed, 12 insertions(+) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -643,6 +644,10 @@ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) int enable_srg = 0; u16 w; + /* Prevent omap hardware from hitting off between fifo fills */ + pm_qos_add_request(&mcbsp->pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + 30 * 1000); + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -731,6 +736,8 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); + + pm_qos_remove_request(&mcbsp->pm_qos_req); } int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) @@ -1098,6 +1105,9 @@ err_thres: void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (dev_pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 61e93b1..e603f33 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -325,6 +325,8 @@ struct omap_mcbsp { unsigned int in_freq; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, -- 2.9.3

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