September 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: msnd: Optimize / harden DSP and MIDI loops
by grygorii tertychnyi 12 Sep '17

12 Sep '17

Hi Greg, Could you please apply it for 4.4-stable. This fixes https://nvd.nist.gov/vuln/detail/CVE-2017-9985 Takashi Iwai (1): ALSA: msnd: Optimize / harden DSP and MIDI loops sound/isa/msnd/msnd_midi.c | 30 +++++++++++++++--------------- sound/isa/msnd/msnd_pinnacle.c | 23 ++++++++++++----------- 2 files changed, 27 insertions(+), 26 deletions(-) -- 2.10.3.dirty

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[alsa-devel] Question: alsa-lib snd_pcm_htimestamp inner working and precision
by gianluca a 12 Sep '17

12 Sep '17

Hi all, I am trying to understand the precision of htimestamp for sample captured and I am not sure if I understood well, because I have found old informations togheter with new informations and I've the doubt that something changed. I am interested in this because of a project for my degree thesis. snd_pcm_hw_params_supports_audio_wallclock_ts returns 0 on capture, so my hardware doesn't support wallclock timestamps on capture. I've understood that SND_PCM_TSTAMP_ENABLE is equivalent to SND_PCM_TSTAMP_MMAP, so it updates timestamp at every position update on DMA from the driver. I've read elsewhere that this timestamp is updated every period-size ( equal interrupt), is it correct? I've digged in the code to understand well inner working of htimestamp, but I've found that snd_pcm_htimestamp in pcm.c calls: err = pcm->fast_ops->htimestamp(pcm->fast_op_arg, avail, tstamp); And I have not found where fast_ops is initialized, maybe in the driver? Anyway I thougth that maybe in my case the called function is snd_pcm_hw_htimestamp in pcm_hw.c : "while (1) { avail1 = snd_pcm_hw_avail_update(pcm); if (avail1 < 0) return avail1; if (ok && (snd_pcm_uframes_t)avail1 == *avail) break; *avail = avail1; *tstamp = snd_pcm_hw_fast_tstamp(pcm); ok = 1; }" that calls snd_pcm_hw_fast_tstamp, that calls finally: "#define FAST_PCM_TSTAMP(hw) \ ((hw)->mmap_status->tstamp) " And finally is this the timestamp that is updated every period-size? Am I correct that in the case that hardware doesn't support wallclock timestamp is snd_pcm_hw_htimestamp in pcm_hw.c the called function? Thanks in advance

1 0

[alsa-devel] [PATCH v5 00/17] Add support for Cirrus Logic CS47L35/L85/L90/L91 codecs
by Richard Fitzgerald 12 Sep '17

12 Sep '17

The Cirrus Logic CS47L35, CS47L85, CS47L90/91 codecs are complex audio SoC devices. In addition to the core audio capability they have onboard GPIO, regulators, DSPs and interrupt controller and a large register map space accessed over SPI or I2C. This family of codecs is based around common IP blocks and they are managed by a set of common drivers referred to as "Madera". Summary of changes since V4: - Fixed review comments on MFD code - Fixed dependency error in core codec driver - minor bugfix to core codec driver Richard Fitzgerald (17): mfd: madera: Add register definitions for Cirrus Logic Madera codecs mfd: madera: Add DT bindings for Cirrus Logic Madera codecs mfd: madera: Add common support for Cirrus Logic Madera codecs mfd: madera: Register map tables for Cirrus Logic CS47L35 mfd: madera: Register map tables for Cirrus Logic CS47L85 mfd: madera: Register map tables for Cirrus Logic CS47L90/91 regulator: arizona-micsupp: Add support for Cirrus Logic Madera codecs regulator: arizona-ldo1: Add support for Cirrus Logic Madera codecs irqchip: Add driver for Cirrus Logic Madera codecs pinctrl: madera: Add DT bindings for Cirrus Logic Madera codecs pinctrl: madera: Add driver for Cirrus Logic Madera codecs gpio: madera: Support Cirrus Logic Madera class codecs ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 Documentation/devicetree/bindings/mfd/madera.txt | 102 + .../bindings/pinctrl/cirrus,madera-pinctrl.txt | 99 + .../bindings/regulator/arizona-regulator.txt | 3 +- Documentation/devicetree/bindings/sound/madera.txt | 67 + MAINTAINERS | 23 + drivers/gpio/Kconfig | 6 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-madera.c | 196 + drivers/irqchip/Kconfig | 3 + drivers/irqchip/Makefile | 1 + drivers/irqchip/irq-madera.c | 301 ++ drivers/mfd/Kconfig | 48 + drivers/mfd/Makefile | 13 + drivers/mfd/cs47l35-tables.c | 1610 +++++++ drivers/mfd/cs47l85-tables.c | 3010 +++++++++++++ drivers/mfd/cs47l90-tables.c | 2675 ++++++++++++ drivers/mfd/madera-core.c | 599 +++ drivers/mfd/madera-i2c.c | 139 + drivers/mfd/madera-spi.c | 138 + drivers/mfd/madera.h | 44 + drivers/pinctrl/Kconfig | 1 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/cirrus/Kconfig | 15 + drivers/pinctrl/cirrus/Makefile | 11 + drivers/pinctrl/cirrus/pinctrl-cs47l35.c | 46 + drivers/pinctrl/cirrus/pinctrl-cs47l85.c | 60 + drivers/pinctrl/cirrus/pinctrl-cs47l90.c | 58 + drivers/pinctrl/cirrus/pinctrl-madera.c | 1074 +++++ drivers/pinctrl/cirrus/pinctrl-madera.h | 40 + drivers/regulator/Kconfig | 15 +- drivers/regulator/arizona-ldo1.c | 82 +- drivers/regulator/arizona-micsupp.c | 71 +- include/dt-bindings/sound/madera.h | 27 + include/linux/irqchip/irq-madera-pdata.h | 23 + include/linux/irqchip/irq-madera.h | 98 + include/linux/mfd/madera/core.h | 196 + include/linux/mfd/madera/pdata.h | 61 + include/linux/mfd/madera/registers.h | 3987 ++++++++++++++++++ include/sound/madera-pdata.h | 61 + sound/soc/codecs/Kconfig | 23 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs47l35.c | 1748 ++++++++ sound/soc/codecs/cs47l85.c | 2722 ++++++++++++ sound/soc/codecs/cs47l90.c | 2661 ++++++++++++ sound/soc/codecs/madera.c | 4439 ++++++++++++++++++++ sound/soc/codecs/madera.h | 470 +++ 46 files changed, 27066 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/madera.txt create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 drivers/gpio/gpio-madera.c create mode 100644 drivers/irqchip/irq-madera.c create mode 100644 drivers/mfd/cs47l35-tables.c create mode 100644 drivers/mfd/cs47l85-tables.c create mode 100644 drivers/mfd/cs47l90-tables.c create mode 100644 drivers/mfd/madera-core.c create mode 100644 drivers/mfd/madera-i2c.c create mode 100644 drivers/mfd/madera-spi.c create mode 100644 drivers/mfd/madera.h create mode 100644 drivers/pinctrl/cirrus/Kconfig create mode 100644 drivers/pinctrl/cirrus/Makefile create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l35.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l85.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs47l90.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.c create mode 100644 drivers/pinctrl/cirrus/pinctrl-madera.h create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/linux/irqchip/irq-madera-pdata.h create mode 100644 include/linux/irqchip/irq-madera.h create mode 100644 include/linux/mfd/madera/core.h create mode 100644 include/linux/mfd/madera/pdata.h create mode 100644 include/linux/mfd/madera/registers.h create mode 100644 include/sound/madera-pdata.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h -- 1.9.1

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[alsa-devel] [PATCH] ALSA: FireWire-MOTU: Use common error handling code in snd_motu_stream_start_duplex()
by SF Markus Elfring 12 Sep '17

12 Sep '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 6 Sep 2017 13:30:14 +0200 Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/firewire/motu/motu-stream.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index dc5541c8b359..73e7a5e527fc 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -253,24 +253,21 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu, unsigned int rate) if (err < 0) { dev_err(&motu->unit->device, "fail to start isochronous comm: %d\n", err); - stop_both_streams(motu); - return err; + goto stop_streams; } err = start_isoc_ctx(motu, &motu->rx_stream); if (err < 0) { dev_err(&motu->unit->device, "fail to start IT context: %d\n", err); - stop_both_streams(motu); - return err; + goto stop_streams; } err = protocol->switch_fetching_mode(motu, true); if (err < 0) { dev_err(&motu->unit->device, "fail to enable frame fetching: %d\n", err); - stop_both_streams(motu); - return err; + goto stop_streams; } } @@ -281,12 +278,15 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu, unsigned int rate) dev_err(&motu->unit->device, "fail to start IR context: %d", err); amdtp_stream_stop(&motu->rx_stream); - stop_both_streams(motu); - return err; + goto stop_streams; } } return 0; + +stop_streams: + stop_both_streams(motu); + return err; } void snd_motu_stream_stop_duplex(struct snd_motu *motu) -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: rt5670: add set_bclk_ratio in dai ops
by Bard Liao 12 Sep '17

12 Sep '17

We need to set a specific bit for 50 bclk rate. So add set_bclk_ratio function to set the bit. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9545764..dbdd01c 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2582,6 +2582,22 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5670_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (dai->id != RT5670_AIF1) + return 0; + + if (ratio == 50) + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x0800, 0x0800); + else + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x0800, 0x0); + + return 0; +} + static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2712,6 +2728,7 @@ static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_fmt = rt5670_set_dai_fmt, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, + .set_bclk_ratio = rt5670_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5670_dai[] = { -- 2.7.4

2 1

[alsa-devel] [PATCH 3/3] ASoC: max98927: Removed obsolete variables
by Ryan Lee 11 Sep '17

11 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changelog: Removed obsolete variables that have never been referred from driver before. sound/soc/codecs/max98927.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index 98b5f17..d215f60 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -263,17 +263,13 @@ struct max98927_priv { struct regmap *regmap; struct snd_soc_codec *codec; - struct max98927_pdata *pdata; - unsigned int spk_gain; unsigned int sysclk; unsigned int v_l_slot; unsigned int i_l_slot; bool interleave_mode; unsigned int ch_size; - unsigned int rate; unsigned int iface; unsigned int master; - unsigned int digital_gain; bool tdm_mode; }; #endif -- 1.9.1

1 0

[alsa-devel] [PATCH v3 00/11] Add STM32 DFSDM support
by Arnaud Pouliquen 11 Sep '17

11 Sep '17

Hello, I kept it as RFC as i reworked the global design (but also because not full validated...). Regarding Mark and Jonathan's comments, I decided to try an implementation with DMA support in IIO instead of handling it in ASOC. Solution seems more straightforward but as DMA cyclic mode and associated buffer block API is not part of IIO today, I still need some hacks... To be honest, I tried to integrate DMA cyclic management and associated buffer block Consumer API. but no enough expertise on IIO to success... :( Advantage vs V2: - No more data processing need in ASoC , done in IIO - ASoC uses IIO consumer interface ( except for buffer callback) Drawback : - Need to create a "SPI bus" trigger to allo to use Buffer consumer API => Need to find a solution to set this trigger by default in IIO audio driver - No standard buffer management - Implement DMA cyclic in DFSDM IIO driver. - Define specific API to handle buffer callback per period instead of callback per samples generated by iio_push_to_buffers_with_timestamp. V2-V3 delta: - New patches to support ASoC DMA codec in DT - ASoC: Add Bindins for DMIC codec driver. - ASoC: codec: add DT support in dmic codec. - New patches to allow in-kernel set of IIO buffer size and watermark - IIO: consumer: allow to set buffer sizes. - IIO DFSDM drivers - Split audio and ADC support in 2 drivers. - Implement DMA cyclic mode. - Add SPI bus Trigger for buffer management. - IIO sigma delta adc drivers - Suppress "simple and rename driver. - ASoC driver - Suppress DMA engine. - Suppress copy ops. - Use IIO consmumer interface to enable/Disable DFSDM. Regards, Arnaud Arnaud Pouliquen (11): iio: Add hardware consumer support IIO: Add DT bindings for sigma delta adc modulator IIO: ADC: add sigma delta modulator support IIO: add DT bindings for stm32 DFSDM filter IIO: ADC: add stm32 DFSDM support for Sigma delta ADC IIO: ADC: add stm32 DFSDM support for PDM microphone IIO: consumer: allow to set buffer sizes ASoC: Add Bindins for DMIC codec driver ASoC: codec: add DT support in dmic codec ASoC: add bindings for stm32 DFSDM filter ASoC: stm32: add DFSDM DAI support .../bindings/iio/adc/sigma-delta-modulator.txt | 13 + .../bindings/iio/adc/st,stm32-dfsdm-adc.txt | 120 ++++ Documentation/devicetree/bindings/sound/dmic.txt | 11 + .../devicetree/bindings/sound/st,stm32-adfsdm.txt | 41 ++ drivers/iio/Kconfig | 6 + drivers/iio/Makefile | 1 + drivers/iio/adc/Kconfig | 50 ++ drivers/iio/adc/Makefile | 4 + drivers/iio/adc/sd_adc_modulator.c | 98 +++ drivers/iio/adc/stm32-dfsdm-adc.c | 419 ++++++++++++ drivers/iio/adc/stm32-dfsdm-audio.c | 720 +++++++++++++++++++++ drivers/iio/adc/stm32-dfsdm-core.c | 658 +++++++++++++++++++ drivers/iio/adc/stm32-dfsdm.h | 372 +++++++++++ drivers/iio/buffer/industrialio-buffer-cb.c | 12 + drivers/iio/hw_consumer.c | 156 +++++ include/linux/iio/adc/stm32-dfsdm-audio.h | 41 ++ include/linux/iio/consumer.h | 13 + include/linux/iio/hw_consumer.h | 12 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/dmic.c | 8 + sound/soc/stm/Kconfig | 10 + sound/soc/stm/Makefile | 2 + sound/soc/stm/stm32_adfsdm.c | 362 +++++++++++ 24 files changed, 3131 insertions(+) create mode 100644 Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.txt create mode 100644 Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt create mode 100644 Documentation/devicetree/bindings/sound/dmic.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt create mode 100644 drivers/iio/adc/sd_adc_modulator.c create mode 100644 drivers/iio/adc/stm32-dfsdm-adc.c create mode 100644 drivers/iio/adc/stm32-dfsdm-audio.c create mode 100644 drivers/iio/adc/stm32-dfsdm-core.c create mode 100644 drivers/iio/adc/stm32-dfsdm.h create mode 100644 drivers/iio/hw_consumer.c create mode 100644 include/linux/iio/adc/stm32-dfsdm-audio.h create mode 100644 include/linux/iio/hw_consumer.h create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_adfsdm.c V2: Patch-set associated to this RFC proposes an implementation of the DFSDM features shared between ASoC and IIO frameworks. Patch-set is only a Skeleton of the drivers, so a base to discuss and validate a design. It contains minimum code to allow probing (with DT) and to expose the ASoC and IIO ABI. Hope that is sufficent in a first step to allow to focus on APIs. In this patch-set there are two new APIs used: - IIO in-kern API: based on hw_customer API proposed by Lars - ASOC <-> IIO API inspired by API defined for hdmi-codec for ASoC/DRM interconnect. API is dedicated to DFSDM only. Notice also that this design is based on following assumption: - Audio stream ABI interface is ASoC, no access to data through IIO ABI for PDM. - ASoC DMA should be used for audio transfer as designed for real time stream. - Need some runtime parameters exchange between ASoC and IIO due to the correlation between the sample frequency, the DFSDM decimation factor and the associated scaling. - "ASoC: dmaengine_pcm: add copy support" patch: I added a patch in ASoC that allows to implement a copy function to process data after DMA transfer. Requested, as DFSDM samples captured contain channel ID on 8-LSB bits and need also a potential rescale to present DAT on 24-bits. - "IIO: ADC: add sigma delta modulator support" patch: Simple dummy driver created to support external Sigma delta modulator. It is binded to DFSDM driver through hw_customer API. - 1.9.1

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[alsa-devel] Different audio sample rate between in and out(44100 Hz resp. 44100 Hz)
by Joakim Tjernlund 11 Sep '17

11 Sep '17

When starting a audio chat I often have problems with my microphon and recenly I found that the sample rate is different for in and out: # > cat /proc/asound/card1/stream0 Logitech Inc Logitech Stereo H650e at usb-0000:00:1d.0-1.7, full speed : USB Audio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 192 Momentary freq = 48000 Hz (0x30.0000) Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 1 OUT (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Capture: Status: Running Interface = 1 Altset = 1 Packet Size = 200 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 8000, 16000, 24000, 32000, 44100, 48000 This seems odd to me. Adding some debug to kernel I see: [ 129.858365] usb 1-2: 1:1: setting freq 44100 to ep 0x81 [ 129.860514] usb 1-2: 1:1: setting freq 44100 to ep 0x81 [ 131.884314] usb 1-2: 2:1: setting freq 44100 to ep 0x1 [ 131.886567] usb 1-2: 2:1: setting freq 44100 to ep 0x1 [ 131.898714] usb 1-2: 2:1: setting freq 48000 to ep 0x1 [ 131.900847] usb 1-2: 2:1: setting freq 48000 to ep 0x1 Is this expected or should I try make both the same? Jocke

1 1

[alsa-devel] [PATCH] ASoC: cs43130: Add break keyword to switch case
by Li Xu 10 Sep '17

10 Sep '17

Add break keyword to all switch case unless the case is returning. Also remove gpio check for consistency. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- This patch addresses stylistic issues in driver. sound/soc/codecs/cs43130.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 643e37f..5ba0edc 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -909,6 +909,7 @@ static int cs43130_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_2, CS43130_DSD_SRC_MASK, CS43130_DSD_SRC_XSP << CS43130_DSD_SRC_SHIFT); + break; } if (!sclk && cs43130->dais[dai->id].dai_mode == SND_SOC_DAIFMT_CBM_CFM) @@ -1039,6 +1040,7 @@ static int cs43130_pcm_ch_put(struct snd_kcontrol *kcontrol, else regmap_multi_reg_write(cs43130->regmap, pcm_ch_dis_seq, ARRAY_SIZE(pcm_ch_dis_seq)); + break; } return snd_soc_put_enum_double(kcontrol, ucontrol); @@ -1152,6 +1154,7 @@ static int cs43130_dsd_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, dsd_seq, ARRAY_SIZE(dsd_seq)); + break; } break; case SND_SOC_DAPM_POST_PMU: @@ -1162,6 +1165,7 @@ static int cs43130_dsd_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, unmute_seq, ARRAY_SIZE(unmute_seq)); + break; } break; case SND_SOC_DAPM_PRE_PMD: @@ -1184,6 +1188,7 @@ static int cs43130_dsd_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs43130->regmap, CS43130_DSD_PATH_CTL_1, CS43130_MUTE_MASK, CS43130_MUTE_EN); + break; } break; default: @@ -1206,6 +1211,7 @@ static int cs43130_pcm_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, pcm_seq, ARRAY_SIZE(pcm_seq)); + break; } break; case SND_SOC_DAPM_POST_PMU: @@ -1216,6 +1222,7 @@ static int cs43130_pcm_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, unmute_seq, ARRAY_SIZE(unmute_seq)); + break; } break; case SND_SOC_DAPM_PRE_PMD: @@ -1238,6 +1245,7 @@ static int cs43130_pcm_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs43130->regmap, CS43130_PCM_PATH_CTL_1, CS43130_MUTE_MASK, CS43130_MUTE_EN); + break; } break; default: @@ -1277,6 +1285,7 @@ static int cs43130_dac_event(struct snd_soc_dapm_widget *w, case CS43198_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, pop_free_seq2, ARRAY_SIZE(pop_free_seq2)); + break; } break; case SND_SOC_DAPM_POST_PMU: @@ -1301,6 +1310,7 @@ static int cs43130_dac_event(struct snd_soc_dapm_widget *w, case CS43198_CHIP_ID: usleep_range(12000, 12010); regmap_write(cs43130->regmap, CS43130_DXD13, 0); + break; } regmap_write(cs43130->regmap, CS43130_DXD1, 0); @@ -1311,6 +1321,7 @@ static int cs43130_dac_event(struct snd_soc_dapm_widget *w, case CS4399_CHIP_ID: regmap_multi_reg_write(cs43130->regmap, dac_postpmd_seq, ARRAY_SIZE(dac_postpmd_seq)); + break; } break; default: @@ -2133,6 +2144,7 @@ static void cs43130_imp_meas(struct work_struct *wk) cs43130_hpload_proc(cs43130, hp_dis_cal_seq2, ARRAY_SIZE(hp_dis_cal_seq2), CS43130_HPLOAD_OFF_INT, ac_idx); + break; } regmap_multi_reg_write(cs43130->regmap, hp_cln_seq, @@ -2543,6 +2555,7 @@ static int cs43130_i2c_probe(struct i2c_client *client, digital_hp_routes; soc_codec_dev_cs43130.component_driver.num_dapm_routes = ARRAY_SIZE(digital_hp_routes); + break; } ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_cs43130, @@ -2586,8 +2599,7 @@ static int cs43130_i2c_remove(struct i2c_client *client) device_remove_file(&client->dev, &dev_attr_hpload_ac_r); } - if (cs43130->reset_gpio) - gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); pm_runtime_disable(&client->dev); regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: adau17x1: Workaround for noise bug in ADC
by Ricard Wanderlof 10 Sep '17

10 Sep '17

The ADC in the ADAU1361 (and possibly other Analog Devices codecs) exhibits a cyclic variation in the noise floor (in our test setup between -87 and -93 dB), a new value being attained within this range whenever a new capture stream is started. The cycle repeats after about 10 or 11 restarts. The workaround recommended by the manufacturer is to toggle the ADOSR bit in the Converter Control 0 register each time a new capture stream is started. I have verified that the patch fixes this problem on the ADAU1361, and according to the manufacturer toggling the bit in question in this manner will at least have no detrimental effect on other chips served by this driver. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 2c1bd27..6758f78 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -90,6 +90,27 @@ static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, return 0; } +static int adau17x1_adc_fixup(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 adau *adau = snd_soc_codec_get_drvdata(codec); + + /* + * If we are capturing, toggle the ADOSR bit in Converter Control 0 to + * avoid losing SNR (workaround from ADI). This must be done after + * the ADC(s) have been enabled. According to the data sheet, it is + * normally illegal to set this bit when the sampling rate is 96 kHz, + * but according to ADI it is acceptable for this workaround. + */ + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_ADOSR, ADAU17X1_CONVERTER0_ADOSR); + regmap_update_bits(adau->regmap, ADAU17X1_CONVERTER0, + ADAU17X1_CONVERTER0_ADOSR, 0); + + return 0; +} + static const char * const adau17x1_mono_stereo_text[] = { "Stereo", "Mono Left Channel (L+R)", @@ -121,7 +142,8 @@ static const struct snd_soc_dapm_widget adau17x1_dapm_widgets[] = { SND_SOC_DAPM_MUX("Right DAC Mode Mux", SND_SOC_NOPM, 0, 0, &adau17x1_dac_mode_mux), - SND_SOC_DAPM_ADC("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0), + SND_SOC_DAPM_ADC_E("Left Decimator", NULL, ADAU17X1_ADC_CONTROL, 0, 0, + adau17x1_adc_fixup, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_ADC("Right Decimator", NULL, ADAU17X1_ADC_CONTROL, 1, 0), SND_SOC_DAPM_DAC("Left DAC", NULL, ADAU17X1_DAC_CONTROL0, 0, 0), SND_SOC_DAPM_DAC("Right DAC", NULL, ADAU17X1_DAC_CONTROL0, 1, 0), diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index bf04b7ef..db35003 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -129,5 +129,7 @@ bool adau17x1_has_dsp(struct adau *adau); #define ADAU17X1_CONVERTER0_CONVSR_MASK 0x7 +#define ADAU17X1_CONVERTER0_ADOSR BIT(3) + #endif -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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