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

[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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[alsa-devel] SOC_DAPM_MUX implementation, routes missing and do I need more dapm definitions?
by Oleksandr Müller 07 Sep '17

07 Sep '17

Hello guys, I began with the implementation of a dapm mux and am almost done but I don't get how to configure the routes and I am not sure if I need more dapm definitions. As the capture stream is the source into the SE_IN_L/SE_IN_R I would say this should be the source for the left/right mux (right side of the route definition). Then I would need a switch in the middle and on the left side as the output should be the ADC1 (left side). { "ADC1", ADC1VINSWITCHLEFT, "SE_IN_L" }, I would think it should look something like this. But then I would need some SOC_DAPM_SINGLE, I think you can see I am clearly confused right now :) I would like to have a mux where I can switch the vin source from vin1 to vin4 in alsamixer and then its saved as a default. Because in the layout plan the vin4 is connected and needs to be selected... My code so far looks like this: static const char * const pcm1863_adc_vinl_src[] = { "VIN1l", "VIN4l" }; static const char * const pcm1863_adc_vinr_src[] = { "VIN1r", "VIN4r" }; /*Stereo ADC Source VIN1 left/right */ static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin1l_enum, PCM1863_ADC1_INPUT_SEL_L, PCM1863_SEL_L_VIN1_SFT, pcm1863_adc_vinl_src); static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin1r_enum, PCM1863_ADC1_INPUT_SEL_R, PCM1863_SEL_R_VIN1_SFT, pcm1863_adc_vinr_src); /*Stereo ADC Source VIN4 left/right */ static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin4l_enum, PCM1863_ADC1_INPUT_SEL_L, PCM1863_SEL_L_VIN4_SFT, pcm1863_adc_vinl_src); static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin4r_enum, PCM1863_ADC1_INPUT_SEL_R, PCM1863_SEL_R_VIN4_SFT, pcm1863_adc_vinr_src); static const struct snd_kcontrol_new pcm1863_adc_vin1l_mux = SOC_DAPM_ENUM("ADC VIN1l Source", pcm1863_adc_vin1l_enum); static const struct snd_kcontrol_new pcm1863_adc_vin1r_mux = SOC_DAPM_ENUM("ADC VIN1r Source", pcm1863_adc_vin1r_enum); static const struct snd_kcontrol_new pcm1863_adc_vin4l_mux = SOC_DAPM_ENUM("ADC VIN4l Source", pcm1863_adc_vin4l_enum); static const struct snd_kcontrol_new pcm1863_adc_vin4r_mux = SOC_DAPM_ENUM("ADC VIN4r Source", pcm1863_adc_vin4r_enum); static const struct snd_soc_dapm_widget pcm1863_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC1", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_INPUT("SE_IN_L"), SND_SOC_DAPM_INPUT("SE_IN_R"), SND_SOC_DAPM_MUX("ADC VIN1l Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin1l_mux), SND_SOC_DAPM_MUX("ADC VIN1r Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin1r_mux), SND_SOC_DAPM_MUX("ADC VIN1l Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin4l_mux), SND_SOC_DAPM_MUX("ADC VIN1r Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin4r_mux), }; static const struct snd_soc_dapm_route pcm1863_dapm_routes[] = { { "ADC1", NULL, "SE_IN_L" }, { "ADC1", NULL, "SE_IN_R" }, { "SE_IN_L", NULL, "Capture" }, { "SE_IN_R", NULL, "Capture" }, };

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[alsa-devel] [PATCH] ASoC: All bytes readable in 6 byte PLL register in ADAU17x1
by Ricard Wanderlof 07 Sep '17

07 Sep '17

The PLL register in the ADAU17x1 is 6 bytes long, so mark all bytes as readable. This is of no importance for existing code, but is useful when instrumenting the code for debugging purposes. 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..b324c6f 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -759,7 +759,13 @@ bool adau17x1_readable_register(struct device *dev, unsigned int reg) switch (reg) { case ADAU17X1_CLOCK_CONTROL: + /* The PLL register is 6 bytes long */ case ADAU17X1_PLL_CONTROL: + case ADAU17X1_PLL_CONTROL + 1: + case ADAU17X1_PLL_CONTROL + 2: + case ADAU17X1_PLL_CONTROL + 3: + case ADAU17X1_PLL_CONTROL + 4: + case ADAU17X1_PLL_CONTROL + 5: case ADAU17X1_REC_POWER_MGMT: case ADAU17X1_MICBIAS: case ADAU17X1_SERIAL_PORT0: -- 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] How to apply a SOC_DAPM_MUX if two there are two different input register for left and right channel?
by Oleksandr Müller 07 Sep '17

07 Sep '17

Hello guys, I have some trouble I want to implement a mux within the mixer application for the VIN source of my adc. Taking a look into the datasheet of the TI-PCM1863 adc I get the information that the VIN left register is at the 0x06 and the VIN right register is at the 0x07 address. Right now my code would look like this and here comes the problem it seems that there is only a SOC_ENUM_SINGLE_DECL or a SOC_ENUM_DOUBLE_DECL but no SOC_ENUM_DOUBLE_R_DECL because the DOUBLE_DECL only allows shiftl and shiftr as the input parameters but I would need a regl regr parameter. Is the only option to duplicate the code and define a VIN_L_MUX and a VIN_R_MUX? static const char * const pcm1863_adc_vin_src[] = { "VIN1", "VIN2", "VIN3", "VIN4" }; static SOC_ENUM_SINGLE_DECL( pcm1863_adc_vin_enum, PCM1863_ADC1_INPUT_SEL_L, PCM1863_SEL_L_SHIFT, pcm1863_vin_src); static const struct snd_kcontrol_new pcm1863_adc_vin_mux = SOC_DAPM_ENUM("ADC VIN Source", pcm1863_adc_vin_enum); static const struct snd_soc_dapm_widget pcm1863_dapm_widgets[] = { [...] SND_SOC_DAPM_MUX("ADC VIN Mux", SND_SOC_NOPM, 0, 0, &pcm1863_adc_vin_mux), }; - Sanj3k

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