December 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ASoC: rsnd: don't call clk_disable_unprepare() if can't use
by Kuninori Morimoto 18 Dec '20

18 Dec '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We need to care clock accessibility, because we might can't use clock for some reasons. It sets clk_rate for each clocks when enabled. This means it doesn't have clk_rate if we can't use. We can avoid to call clk_disable_unprepare() in such case. Link: https://lore.kernel.org/r/CAMuHMdWvB+p=2JqTsO7bR8uJqKqO5A2XgXFXsVAjHk3hcxgc… Reported-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - reset adg->clk_rate[i] = 0 sound/soc/sh/rcar/adg.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index b9aacf3d3b29..abdfd9cf91e2 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -366,25 +366,27 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; - int i, ret; + int i; for_each_rsnd_clk(clk, adg, i) { - ret = 0; if (enable) { - ret = clk_prepare_enable(clk); + int ret = clk_prepare_enable(clk); /* * We shouldn't use clk_get_rate() under * atomic context. Let's keep it when * rsnd_adg_clk_enable() was called */ - adg->clk_rate[i] = clk_get_rate(adg->clk[i]); + adg->clk_rate[i] = 0; + if (ret < 0) + dev_warn(dev, "can't use clk %d\n", i); + else + adg->clk_rate[i] = clk_get_rate(clk); } else { - clk_disable_unprepare(clk); + if (adg->clk_rate[i]) + clk_disable_unprepare(clk); + adg->clk_rate[i] = 0; } - - if (ret < 0) - dev_warn(dev, "can't use clk %d\n", i); } } -- 2.25.1

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[PATCH v2] ASoC: max98373: don't access volatile registers in bias level off
by Bard Liao 18 Dec '20

18 Dec '20

We will set regcache_cache_only true in suspend. As a result, regmap_read will return error when we try to read volatile registers in suspend. Besides, it doesn't make sense to read feedback data when codec is not active. To make userspace happy, this patch returns a cached value shich should be a valid value. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - Return cached values instead of 0. --- sound/soc/codecs/max98373-i2c.c | 8 ++++++++ sound/soc/codecs/max98373-sdw.c | 8 ++++++++ sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++--- sound/soc/codecs/max98373.h | 14 ++++++++++++++ 4 files changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 92921e34f948..09890478b120 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -472,6 +472,11 @@ static struct snd_soc_dai_driver max98373_dai[] = { static int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); regcache_mark_dirty(max98373->regmap); @@ -534,6 +539,9 @@ static int max98373_i2c_probe(struct i2c_client *i2c, return ret; } + max98373->cache = max98373_cache; + max98373->cache_num = ARRAY_SIZE(max98373_cache); + /* voltage/current slot & gpio configuration */ max98373_slot_config(&i2c->dev, max98373); diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index ec2e79c57357..0575d323df8a 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -245,6 +245,11 @@ static const struct regmap_config max98373_sdw_regmap = { static __maybe_unused int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); @@ -768,6 +773,9 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) max98373->regmap = regmap; max98373->slave = slave; + max98373->cache = max98373_cache; + max98373->cache_num = ARRAY_SIZE(max98373_cache); + /* Read voltage and slot configuration */ max98373_slot_config(dev, max98373); diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..31d571d4fac1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -168,6 +168,31 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum, MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0, max98373_ADC_samplerate_text); +static int max98373_feedback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); + int i; + + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* + * Register values will be cached before suspend. The cached value + * will be a valid value and userspace will happy with that. + */ + for (i = 0; i < max98373->cache_num; i++) { + if (mc->reg == max98373->cache[i].reg) { + ucontrol->value.integer.value[0] = max98373->cache[i].val; + return 0; + } + } + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new max98373_snd_controls[] = { SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_VOL_SEL_SHIFT, 1, 0), @@ -209,8 +234,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), -SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0), -SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0), +SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), +SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0, 0x3, 0), SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, @@ -226,7 +253,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0), -SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0), +SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0, + max98373_feedback_get, NULL), SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0), SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index 4ab29b9d51c7..6206c6c3a773 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -203,6 +203,17 @@ /* MAX98373_R2000_SW_RESET */ #define MAX98373_SOFT_RESET (0x1 << 0) +struct max98373_cache { + u32 reg; + u32 val; +}; + +struct max98373_cache max98373_cache[] = { + {MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0}, + {MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0}, + {MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0}, +}; + struct max98373_priv { struct regmap *regmap; int reset_gpio; @@ -212,6 +223,9 @@ struct max98373_priv { bool interleave_mode; unsigned int ch_size; bool tdm_mode; + /* cache for reading a valid fake feedback value */ + struct max98373_cache *cache; + int cache_num; /* variables to support soundwire */ struct sdw_slave *slave; bool hw_init; -- 2.17.1

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[PATCH v5 0/2] Platform driver update to support playback recover after resume
by Srinivasa Rao Mandadapu 18 Dec '20

18 Dec '20

This patch set is to add support for playback recover after hard suspend and resume. It includes: 1. Reverting part of previous commit, which is for handling registers invalid state after hard suspend. 2. Adding pm ops in component driver and do regcache sync. Changes Since v1 and v2: -- Subject lines changed Changes Since v3: -- Patch is splitted into 2 patches Changes Since v4: -- Subject lines changed Srinivasa Rao Mandadapu (2): ASoC: qcom: Fix incorrect volatile registers ASoC: qcom: Add support for playback recover after resume sound/soc/qcom/lpass-cpu.c | 20 ++---------------- sound/soc/qcom/lpass-platform.c | 46 +++++++++++++++++++++++++++++++---------- 2 files changed, 37 insertions(+), 29 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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UCM fails as CardDriver may not be defined
by GitHub issues - opened 18 Dec '20

18 Dec '20

alsa-project/alsa-ucm-conf issue #69 was opened from coolstar: UCM doesn't load as CardDriver isn't set on some platforms Changing https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/ucm.conf#L69 to `Directory "${CardLongName}"` resolves the issue ``` I: [pulseaudio] source.c: Created source 1 "alsa_input.platform-sound.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: [pulseaudio] source.c: alsa.resolution_bits = "16" I: [pulseaudio] source.c: device.api = "alsa" I: [pulseaudio] source.c: device.class = "sound" I: [pulseaudio] source.c: alsa.class = "generic" I: [pulseaudio] source.c: alsa.subclass = "generic-mix" I: [pulseaudio] source.c: alsa.name = "" I: [pulseaudio] source.c: alsa.id = "ADMAIF1 CIF ADMAIF1-0" I: [pulseaudio] source.c: alsa.subdevice = "0" I: [pulseaudio] source.c: alsa.subdevice_name = "subdevice #0" I: [pulseaudio] source.c: alsa.device = "0" I: [pulseaudio] source.c: alsa.card = "1" I: [pulseaudio] source.c: alsa.card_name = "tegra-snd-t210ref-mobile-rt565x" I: [pulseaudio] source.c: alsa.long_card_name = "tegra-snd-t210ref-mobile-rt565x" I: [pulseaudio] source.c: device.bus_path = "platform-sound" I: [pulseaudio] source.c: sysfs.path = "/devices/sound/sound/card1" I: [pulseaudio] source.c: device.form_factor = "internal" I: [pulseaudio] source.c: device.string = "front:1" I: [pulseaudio] source.c: device.buffering.buffer_size = "32768" I: [pulseaudio] source.c: device.buffering.fragment_size = "16384" I: [pulseaudio] source.c: device.access_mode = "mmap+timer" I: [pulseaudio] source.c: device.profile.name = "analog-stereo" I: [pulseaudio] source.c: device.profile.description = "Analog Stereo" I: [pulseaudio] source.c: device.description = "Built-in Audio Analog Stereo" I: [pulseaudio] source.c: module-udev-detect.discovered = "1" I: [pulseaudio] source.c: device.icon_name = "audio-card" ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/69 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v1 ] ALSA: core: memalloc: add page alignment for iram
by Robin Gong 17 Dec '20

17 Dec '20

Since mmap for userspace is based on page alignment, add page alignment for iram alloc from pool, otherwise, some good data located in the same page of dmab->area maybe touched wrongly by userspace like pulseaudio. Signed-off-by: Robin Gong <yibin.gong(a)nxp.com> --- sound/core/memalloc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 0aeeb62..0f33516 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -77,7 +77,8 @@ static void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) /* Assign the pool into private_data field */ dmab->private_data = pool; - dmab->area = gen_pool_dma_alloc(pool, size, &dmab->addr); + dmab->area = gen_pool_dma_alloc_align(pool, size, &dmab->addr, + PAGE_SIZE); } /** -- 2.7.4

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wallclock time for midi events without audio port
by Michael Kirchner 17 Dec '20

17 Dec '20

Hi, i try to get an absolute wallclock time for midi events using event->time.time.tv_sec. As I understand I have to add trigger time to get wall clock time. Therefor I use snd_pcm_status_get_trigger_htstamp(). If I use alsa audio via snd_pcm_open there is no problem. But: how can I get trigger time if I use only a midi port without audio? (usecase is: soft synth with alsa midi in -> jack audio out) Sincerely, Michael

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utils/alsa.m4: update for newer autoconf.
by GitHub pull_request - opened 17 Dec '20

17 Dec '20

alsa-project/alsa-lib pull request #106 was opened from sezero: fixes autoconf-2.70 warnings. should be compatible down to autoconf-2.59. Request URL : https://github.com/alsa-project/alsa-lib/pull/106 Patch URL : https://github.com/alsa-project/alsa-lib/pull/106.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH alsa-lib 1/2] pcm: Add snd_pcm_audio_tstamp_type_t constants
by Takashi Iwai 17 Dec '20

17 Dec '20

From: David Henningsson <diwic(a)ubuntu.com> These are mostly a copy-paste from the kernel headers. But since functions snd_pcm_audio_tstamp_config make use of these they should be added to the public API as well. Reported-by: Alex Moon <alex.r.moon(a)gmail.com> Signed-off-by: David Henningsson <diwic(a)ubuntu.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/pcm.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/include/pcm.h b/include/pcm.h index 5b0782315585..cf1eea8bbc9b 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -350,6 +350,20 @@ typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +typedef enum _snd_pcm_audio_tstamp_type { + /** + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /**< DMA time, reported as per hw_ptr */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /**< link time reported by sample or wallclock counter, reset on startup */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +} snd_pcm_audio_tstamp_type_t; + typedef struct _snd_pcm_audio_tstamp_config { /* 5 of max 16 bits used */ unsigned int type_requested:4; -- 2.26.2

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Re: [PATCH 01/01] Add VID to support native DSD reproduction on FiiO devices.
by Amadej Kastelic 17 Dec '20

17 Dec '20

Message-Id: <c759bc50134ea434bfed0a183e0ce72984f5b609.camel(a)gmail.com> Add VID to support native DSD reproduction on FiiO devices. Tested-by: Amadej Kastelic <amadejkastelic7(a)gmail.com> Signed-off-by: Emilio Moretti <emilio.moretti(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 349e1e52996d..f5fc65aef628 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ case 0x292b: /* Gustard/Ess based devices */ + case 0x2972: /* FiiO devices */ case 0x2ab6: /* T+A devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ -- 2.20.1

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Dell fixed type headset platform
by Kailang 17 Dec '20

17 Dec '20

Hi Takashi, Dell has fixed type headset platform for desktop. Attach patch was add pin verb. BIOS verb table was setting for fixed type coef. Many Thanks. BR, Kailang

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