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

[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

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[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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[alsa-devel] [PATCH] ALSA: msnd: Optimize / harden DSP and MIDI loops
by grygorii tertychnyi 08 Sep '17

08 Sep '17

From: Takashi Iwai <tiwai(a)suse.de> commit 20e2b791796bd68816fa115f12be5320de2b8021 upstream. The ISA msnd drivers have loops fetching the ring-buffer head, tail and size values inside the loops. Such codes are inefficient and fragile. This patch optimizes it, and also adds the sanity check to avoid the endless loops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196131 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196133 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Cc: xe-linux-external(a)cisco.com Signed-off-by: grygorii tertychnyi <gtertych(a)cisco.com> --- sound/isa/msnd/msnd_midi.c | 30 +++++++++++++++--------------- sound/isa/msnd/msnd_pinnacle.c | 23 ++++++++++++----------- 2 files changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index ffc67fd80c23..58e59cd3c95c 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -120,24 +120,24 @@ void snd_msndmidi_input_read(void *mpuv) unsigned long flags; struct snd_msndmidi *mpu = mpuv; void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + u16 head, tail, size; spin_lock_irqsave(&mpu->input_lock, flags); - while (readw(mpu->dev->MIDQ + JQS_wTail) != - readw(mpu->dev->MIDQ + JQS_wHead)) { - u16 wTmp, val; - val = readw(pwMIDQData + 2 * readw(mpu->dev->MIDQ + JQS_wHead)); - - if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, - &mpu->mode)) - snd_rawmidi_receive(mpu->substream_input, - (unsigned char *)&val, 1); - - wTmp = readw(mpu->dev->MIDQ + JQS_wHead) + 1; - if (wTmp > readw(mpu->dev->MIDQ + JQS_wSize)) - writew(0, mpu->dev->MIDQ + JQS_wHead); - else - writew(wTmp, mpu->dev->MIDQ + JQS_wHead); + head = readw(mpu->dev->MIDQ + JQS_wHead); + tail = readw(mpu->dev->MIDQ + JQS_wTail); + size = readw(mpu->dev->MIDQ + JQS_wSize); + if (head > size || tail > size) + goto out; + while (head != tail) { + unsigned char val = readw(pwMIDQData + 2 * head); + + if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode)) + snd_rawmidi_receive(mpu->substream_input, &val, 1); + if (++head > size) + head = 0; + writew(head, mpu->dev->MIDQ + JQS_wHead); } + out: spin_unlock_irqrestore(&mpu->input_lock, flags); } EXPORT_SYMBOL(snd_msndmidi_input_read); diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 4c072666115d..a31ea6c22d19 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -170,23 +170,24 @@ static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) { struct snd_msnd *chip = dev_id; void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + u16 head, tail, size; /* Send ack to DSP */ /* inb(chip->io + HP_RXL); */ /* Evaluate queued DSP messages */ - while (readw(chip->DSPQ + JQS_wTail) != readw(chip->DSPQ + JQS_wHead)) { - u16 wTmp; - - snd_msnd_eval_dsp_msg(chip, - readw(pwDSPQData + 2 * readw(chip->DSPQ + JQS_wHead))); - - wTmp = readw(chip->DSPQ + JQS_wHead) + 1; - if (wTmp > readw(chip->DSPQ + JQS_wSize)) - writew(0, chip->DSPQ + JQS_wHead); - else - writew(wTmp, chip->DSPQ + JQS_wHead); + head = readw(chip->DSPQ + JQS_wHead); + tail = readw(chip->DSPQ + JQS_wTail); + size = readw(chip->DSPQ + JQS_wSize); + if (head > size || tail > size) + goto out; + while (head != tail) { + snd_msnd_eval_dsp_msg(chip, readw(pwDSPQData + 2 * head)); + if (++head > size) + head = 0; + writew(head, chip->DSPQ + JQS_wHead); } + out: /* Send ack to DSP */ inb(chip->io + HP_RXL); return IRQ_HANDLED; -- 2.10.3.dirty

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[alsa-devel] [bug] Missing "front" definition for byt-max98090
by Tanu Kaskinen 08 Sep '17

08 Sep '17

PulseAudio 11 stopped using "hw:" as a fallback for analog stereo if "front:" doesn't work. This caused a regression on "Toshiba Chromebook 2 (Swanky)", because PulseAudio doesn't any more do automatic mixer adjustments when plugging in headphones. Apparently on this machine "hw:1,0" works fine as a substitute for "front:1,0", but we're not going to revert the change in PulseAudio, so could someone add the "front" definition for this machine in alsa-lib? Unfortunately I'm not proficient enough in the alsa-lib configuration system to make a patch myself. alsa-info here: https://bugs.freedesktop.org/attachment.cgi?id=133993 [1] https://bugs.freedesktop.org/show_bug.cgi?id=102560 -- Tanu https://www.patreon.com/tanuk

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[alsa-devel] [PATCH 23/25] ALSA/dummy: Replace tasklet with softirq hrtimer
by Anna-Maria Gleixner 08 Sep '17

08 Sep '17

From: Thomas Gleixner <tglx(a)linutronix.de> The tasklet is used to defer the execution of snd_pcm_period_elapsed() to the softirq context. Using the CLOCK_MONOTONIC_SOFT base invokes the timer callback in softirq context as well which renders the tasklet useless. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Anna-Maria Gleixner <anna-maria(a)linutronix.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org --- sound/drivers/dummy.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -376,17 +376,9 @@ struct dummy_hrtimer_pcm { ktime_t period_time; atomic_t running; struct hrtimer timer; - struct tasklet_struct tasklet; struct snd_pcm_substream *substream; }; -static void dummy_hrtimer_pcm_elapsed(unsigned long priv) -{ - struct dummy_hrtimer_pcm *dpcm = (struct dummy_hrtimer_pcm *)priv; - if (atomic_read(&dpcm->running)) - snd_pcm_period_elapsed(dpcm->substream); -} - static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer) { struct dummy_hrtimer_pcm *dpcm; @@ -394,7 +386,8 @@ static enum hrtimer_restart dummy_hrtime dpcm = container_of(timer, struct dummy_hrtimer_pcm, timer); if (!atomic_read(&dpcm->running)) return HRTIMER_NORESTART; - tasklet_schedule(&dpcm->tasklet); + + snd_pcm_period_elapsed(dpcm->substream); hrtimer_forward_now(timer, dpcm->period_time); return HRTIMER_RESTART; } @@ -421,7 +414,6 @@ static int dummy_hrtimer_stop(struct snd static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm) { hrtimer_cancel(&dpcm->timer); - tasklet_kill(&dpcm->tasklet); } static snd_pcm_uframes_t @@ -466,12 +458,10 @@ static int dummy_hrtimer_create(struct s if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC_SOFT, HRTIMER_MODE_REL); dpcm->timer.function = dummy_hrtimer_callback; dpcm->substream = substream; atomic_set(&dpcm->running, 0); - tasklet_init(&dpcm->tasklet, dummy_hrtimer_pcm_elapsed, - (unsigned long)dpcm); return 0; }

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[alsa-devel] [PATCH] sound: soc: fsl: Do not set DAI sysclk when it is equal to system freq
by Lukasz Majewski 08 Sep '17

08 Sep '17

The problem is visible in the following setup (on the imx6q): "simple-audio-card" -> ssi2 -> I2S + I2C -> codec The function call log (simple-card probe -> CONFIG_SND_SIMPLE_CARD): asoc_simple_card_init_dai() @ sound/soc/generic/simple-card-utils.c snd_soc_dai_set_sysclk() fsl_ssi_set_dai_sysclk() @ sound/soc/fsl/fsl_ssi.c The last call is changing the bit clock (BCLK) frequency to SSI's IP block clock (ipg = 66 MHz) [1]. This is wrong, since IMX SSI block requires the I2S BCLK to be less than 1/5 of [1]. As a result the driver initialization passes without any errors, but the speaker-test test case breaks. This commit checks if the fsl_ssi_set_dai_sysclk() frequency passed is not equal to [1]. Signed-off-by: Lukasz Majewski <lukma(a)denx.de> --- sound/soc/fsl/fsl_ssi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 173cb84..1186fa9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -809,6 +809,8 @@ static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + if (clk_get_rate(ssi_private->clk) == freq) + return 0; ssi_private->bitclk_freq = freq; -- 2.1.4

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[alsa-devel] [patch 0/4] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 08 Sep '17

08 Sep '17

Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. This code change doesn't do much except just refactor snd_soc_dapm_new_pcm function. Now the part of allocating memory for kcontrol is allocated to a new function and it is called only in the case it is needed. anish kumar (4): ASoC: dapm: fix error path in snd_soc_dapm_new_pcm ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm ASoC: dapm: Avoid creating kcontrol for params ASoC: dapm: Remove the redundant check sound/soc/soc-dapm.c | 123 ++++++++++++++++++++++++++++----------------------- 1 file changed, 67 insertions(+), 56 deletions(-) -- 2.10.1 (Apple Git-78)

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[alsa-devel] bootstrap.sh
by Russell Treleaven 07 Sep '17

07 Sep '17

sorry wrong list. -- Sincerely, Russell Treleaven sip:rtreleaven@sip.bunnykick.ca

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[alsa-devel] autopoint should be required for bootstrap.sh
by Russell Treleaven 07 Sep '17

07 Sep '17

I attempted to bootstrap pulseaudio and got a non obvious error. log is here https://pastebin.com/CctNdLwD installing autopoint fixed the problem the following patch makes bootstrap.sh fail if autopoint is not available. diff --git a/bootstrap.sh b/bootstrap.sh index 57494fc..0a05a3e 100755 --- a/bootstrap.sh +++ b/bootstrap.sh @@ -32,13 +32,17 @@ fi # configure file faulty. if ! pkg-config --version &>/dev/null; then echo "pkg-config is required to bootstrap this program" - DIE=1 + exit 1 fi - # Other necessary programs -intltoolize --version >/dev/null || DIE=1 -test "$DIE" = 1 && exit 1 - +if ! autopoint --version &>/dev/null ; then + echo "autopoint is required to bootstrap this program" + exit 1 +fi +if ! intltoolize --version >/dev/null ; then + echo "intltoolize is required to bootstrap this program" + exit 1 +fi autopoint --force AUTOPOINT='intltoolize --automake --copy' autoreconf --force --install --verbose -- Sincerely, Russell Treleaven sip:rtreleaven@sip.bunnykick.ca

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