July 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH - ALSA JACK plugin 1/1] jack: Only allow ALSA periods multiple of JACKd period
by twischer＠de.adit-jv.com 19 Jul '18

19 Jul '18

From: Laxmi Devi <ldevi(a)de.adit-jv.com> There is a higher Xrun probability whenever the ALSA period is not a multiple of the JACKd period and the ALSA buffer is only twice the period. Allowing ALSA buffers of min. 3x period is not a good solution because this would increase the latency. As an example in case of ALSA is using a period of 11 frames and JACK a period of 12 frames and a buffer of 22 frames (2x ALSA period) is used: - The buffer is filled and contains 22 frames - JACK is reading 12 frames. Buffer is only containing 10 frames - Now, ALSA has to be scheduled exactly after the last JACK read and before the next read. Otherwise we will get an Xrun In case of ALSA is using a multiple period of JACK, JACK can always read 2 periods before an Xrun if the buffer was full. In case of ALSA is using a period of 12 frames and JACK a period of 11 frames and a buffer of 24 frames (2x ALSA period) is used: - The buffer is filled and contains 24 frames - JACK is reading 11 frames. Buffer is now containing 13 frames. But ALSA can still not write to the buffer - JACK reads 11 additional frames. Buffer contains 2 frames - Now, ALSA has to be scheduled exactly after the last JACK read and before the next read. Otherwise we will get an Xrun In case of ALSA is using a multiple period of JACK, ALSA can always write to the buffer as long as the buffer is filled with less or equal of half of buffer. Therefore this patch enforces an ALSA period which is always a multiple of the JACKd period. Signed-off-by: Laxmi Devi <ldevi(a)de.adit-jv.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index fbc9daf..b39835e 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -29,6 +29,8 @@ #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> +#define MAX_PERIODS_MULTIPLE 64 + typedef enum _jack_format { SND_PCM_JACK_FORMAT_RAW } snd_pcm_jack_format_t; @@ -364,8 +366,20 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) }; unsigned int format = SND_PCM_FORMAT_FLOAT; unsigned int rate = jack_get_sample_rate(jack->client); + unsigned int psize_list[MAX_PERIODS_MULTIPLE]; + unsigned int nframes = jack_get_buffer_size(jack->client); + unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * + jack->channels); + unsigned int i; int err; + if (!jack_buffer_bytes) { + SNDERR("Buffer size is zero"); + return -EINVAL; + } + for (i = 1; i <= ARRAY_SIZE(psize_list); i++) + psize_list[i-1] = jack_buffer_bytes * i; + jack->sample_bits = snd_pcm_format_physical_width(format); if ((err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list)) < 0 || @@ -375,8 +389,8 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) jack->channels, jack->channels)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_RATE, rate, rate)) < 0 || - (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, - 128, 64*1024)) < 0 || + (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(psize_list), psize_list)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_PERIODS, 2, 64)) < 0) return err; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Add quirk for the "Connect Tablet 9" tablet
by Hans de Goede 19 Jul '18

19 Jul '18

Add a quirk for the "Connect Tablet 9" tablet, this tablet has a mono-speaker. Otherwise it works fine with the defaults. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7456566c5648..718f022b3aa0 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -486,6 +486,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP), }, + { /* Connect Tablet 9 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."), -- 2.17.1

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[alsa-devel] [PATCH 1/5] ASoC: wm_adsp: Correct algorithm list allocation size
by Charles Keepax 19 Jul '18

19 Jul '18

Commit 6396bb221514 ("treewide: kzalloc() -> kcalloc()") was overlooked when doing some refactoring to the algorithm list handling, which lead to twice as much buffer being allocated as required for reading the algorithm list. A kcalloc is no longer appropriate since the allocation size is now in bytes not registers, as such change back to kzalloc. Fixes: 7f7cca08abf4 ("ASoC: wm_adsp: Simplify handling of alg offset and length") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- 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 07c17acc8a4f..99108d18de8d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1906,7 +1906,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, /* Convert length from DSP words to bytes */ len *= sizeof(u32); - alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA); + alg = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!alg) return ERR_PTR(-ENOMEM); -- 2.11.0

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[alsa-devel] [PATCH RESEND 00/15] ASoC: meson: add axg audio subsystem support
by Jerome Brunet 19 Jul '18

19 Jul '18

This patchset adds support for the audio subsystem found on Amlogic's axg SoC family. The first SoC of this family is the A113D/X targeted at smart speaker application. The G12a SoC family, Amlogic next-gen STB chipset, will reuse this audio subsystem with a few changes. This patchset implements the DPCM machine driver and the different components it made of. Several components are still missing, such spdif input and pdm, and will submitted later on. The fancier part of this patchset is probably around the TDM. Properly modeling the SoC architecture and its capatilities made things a little more complex than usual (details in the related commit descriptions) If necessary, this patchset could broken into smaller series. For the intial submission, I wanted to give a global view of the subsystem. Hopefully, it will make it a bit easier to understand. Jerome Brunet (15): ASoC: meson: add axg fifos DT binding documentation ASoC: meson: add axg fifo base driver ASoC: meson: add axg frddr driver ASoC: meson: add axg toddr driver ASoC: meson: add axg spdif output DT bindings documentation ASoC: meson: add axg spdif output ASoC: meson: add axg tdm formatters DT bindings documentation ASoC: meson: add axg tdm interface DT bindings documentation ASoC: meson: add tdm formatter base driver ASoC: meson: add tdm interface driver ASoC: meson: add tdm output driver ASoC: meson: add tdm input driver ASoC: export snd_soc_of_get_slot_mask ASoC: meson: add axg sound card DT bindings documentation ASoC: meson: add axg sound card support .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 23 + .../bindings/sound/amlogic,axg-sound-card.txt | 124 ++++ .../bindings/sound/amlogic,axg-spdifout.txt | 20 + .../bindings/sound/amlogic,axg-tdm-formatters.txt | 28 + .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 + include/sound/soc.h | 3 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/meson/Kconfig | 64 ++ sound/soc/meson/Makefile | 21 + sound/soc/meson/axg-card.c | 671 +++++++++++++++++++++ sound/soc/meson/axg-fifo.c | 341 +++++++++++ sound/soc/meson/axg-fifo.h | 80 +++ sound/soc/meson/axg-frddr.c | 141 +++++ sound/soc/meson/axg-spdifout.c | 456 ++++++++++++++ sound/soc/meson/axg-tdm-formatter.c | 381 ++++++++++++ sound/soc/meson/axg-tdm-formatter.h | 39 ++ sound/soc/meson/axg-tdm-interface.c | 542 +++++++++++++++++ sound/soc/meson/axg-tdm.h | 78 +++ sound/soc/meson/axg-tdmin.c | 229 +++++++ sound/soc/meson/axg-tdmout.c | 259 ++++++++ sound/soc/meson/axg-toddr.c | 199 ++++++ sound/soc/soc-core.c | 7 +- 23 files changed, 3727 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt create mode 100644 sound/soc/meson/Kconfig create mode 100644 sound/soc/meson/Makefile create mode 100644 sound/soc/meson/axg-card.c create mode 100644 sound/soc/meson/axg-fifo.c create mode 100644 sound/soc/meson/axg-fifo.h create mode 100644 sound/soc/meson/axg-frddr.c create mode 100644 sound/soc/meson/axg-spdifout.c create mode 100644 sound/soc/meson/axg-tdm-formatter.c create mode 100644 sound/soc/meson/axg-tdm-formatter.h create mode 100644 sound/soc/meson/axg-tdm-interface.c create mode 100644 sound/soc/meson/axg-tdm.h create mode 100644 sound/soc/meson/axg-tdmin.c create mode 100644 sound/soc/meson/axg-tdmout.c create mode 100644 sound/soc/meson/axg-toddr.c -- 2.14.4

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[alsa-devel] [PATCH - usb-audio v2 1/1] ALSA: usb-audio: Allow changing from a bad sample rate
by Adam Goode 19 Jul '18

19 Jul '18

If the audio device is externally clocked and set to a rate that does not match the external clock, the clock will never be valid and we cannot set the rate successfully. To fix this, allow a rate change even if the clock is initially invalid, and validate again after the rate is changed. This fixes problems with MOTU UltraLite AVB hardware over USB. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/sound/usb/clock.c b/sound/usb/clock.c index c79749613fa6..db5e39d67a90 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -513,14 +513,28 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, bool writeable; u32 bmControls; + /* First, try to find a valid clock. This may trigger + * automatic clock selection if the current clock is not + * valid. + */ clock = snd_usb_clock_find_source(chip, fmt->protocol, fmt->clock, true); - if (clock < 0) - return clock; + if (clock < 0) { + /* We did not find a valid clock, but that might be + * because the current sample rate does not match an + * external clock source. Try again without validation + * and we will do another validation after setting the + * rate. + */ + clock = snd_usb_clock_find_source(chip, fmt->protocol, + fmt->clock, false); + if (clock < 0) + return clock; + } prev_rate = get_sample_rate_v2v3(chip, iface, fmt->altsetting, clock); if (prev_rate == rate) - return 0; + goto validation; if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc; @@ -577,6 +591,10 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, snd_usb_set_interface_quirk(dev); } +validation: + /* validate clock after rate change */ + if (!uac_clock_source_is_valid(chip, fmt->protocol, clock)) + return -ENXIO; return 0; } -- 2.17.1

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[alsa-devel] [PATCH 1/1] Preventive fix in sound module
by Srikanth K H 19 Jul '18

19 Jul '18

Signed-off-by: Srikanth K H <srikanth.h(a)samsung.com> --- sound/core/timer.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index b6f076bb..c7be4f1 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1192,10 +1192,12 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, break; case SNDRV_TIMER_CLASS_CARD: snd_iprintf(buffer, "C%i-%i: ", - timer->card->number, timer->tmr_device); + timer->card ? timer->card->number : -1, + timer->tmr_device); break; case SNDRV_TIMER_CLASS_PCM: - snd_iprintf(buffer, "P%i-%i-%i: ", timer->card->number, + snd_iprintf(buffer, "P%i-%i-%i: ", + timer->card ? timer->card->number : -1, timer->tmr_device, timer->tmr_subdevice); break; default: -- 1.9.1

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Re: [alsa-devel] i.MX6Q no audio with wm8962 Question
by Fabio Estevam 18 Jul '18

18 Jul '18

Hi Adam, On Wed, Jul 18, 2018 at 1:07 PM, Adam Ford <aford173(a)gmail.com> wrote: > On Tue, Jul 17, 2018 at 2:40 PM Adam Ford <aford173(a)gmail.com> wrote: >> >> I have an iMX6Q with a wm8962 attached using AUDMUX4 and SSI2, but I >> am having difficulty getting the sound to generate from the headphone >> jack. > > sorry for the noise. It turns out to be a faulty pull-up resistor on > the I2C bus > > it's working Glad you fixed the problem :-)

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[alsa-devel] [PATCH v2 0/5] ASoC: core: add support to card re-bind using component framework
by Srinivas Kandagatla 18 Jul '18

18 Jul '18

During discussion regarding card re-binding when components unregister and register back at https://lkml.org/lkml/2018/7/9/785 it was suggested that component framework can be added into core to provide this feature. With this new changes the card will re-bind once the dependent component re-registers after unregistering. This works based on the match done from component name using component framework. I have tested this patch with qdsp start-stop and module-load-unload usecase for more than 10000 times in loop on Qcom platforms. Thanks, srini Changes since v1: - Added new flag to enable/disable auto bind at card level, It is useful if machine driver/module is really removed/unloaded. Also to help cleanup the master component. Without this we will endup with a memory leak if the machine driver and memory associated with card are removed. - removed unregister from unbind callback. - updated match function as suggested by Vinod. - Added qdsp6 cleanup patches to this series. Srinivas Kandagatla (5): ASoC: core: add support to card re-bind using component framework ASoC: qdsp6: q6afe-dai: remove component fw related code ASoC: qdsp6: q6asm-dai: remove component framework related code ASoC: qdsp6: q6routing: remove component framework related code ASoC: qcom: apq8096: remove component framework related code include/sound/soc.h | 7 ++++ sound/soc/qcom/apq8096.c | 75 ++++------------------------------------ sound/soc/qcom/qdsp6/q6afe-dai.c | 34 +++--------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 35 +++---------------- sound/soc/qcom/qdsp6/q6routing.c | 32 ++++------------- sound/soc/soc-core.c | 62 +++++++++++++++++++++++++++++++++ 6 files changed, 89 insertions(+), 156 deletions(-) -- 2.16.2

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: Add documentation for power management
by Sriram Periyasamy 18 Jul '18

18 Jul '18

Add documentation for power management of HDAC HDMI codec device for various scenarios such as S0/S3, probe and playback use case. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 69 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 69 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..30ccc902e5cf 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2103,6 +2103,75 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) } #ifdef CONFIG_PM +/* + * Power management sequences + * ========================== + * + * The following explains the PM handling of HDAC HDMI with its parent + * device SKL and display power usage + * + * Probe + * ----- + * In SKL probe, + * 1. skl_probe_work() powers up the display (refcount++ -> 1) + * 2. enumerates the codecs on the link + * 3. powers down the display (refcount-- -> 0) + * + * In HDAC HDMI probe, + * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) + * 2. probe the codec + * 3. put the HDAC HDMI device to runtime suspend + * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * + * Once children are runtime suspended, SKL device also goes to runtime + * suspend + * + * HDMI Playback + * ------------- + * Open HDMI device, + * 1. skl_runtime_resume() invoked + * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * + * Close HDMI device, + * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 2. skl_runtime_suspend() invoked + * + * S0/S3 Cycle with playback in progress + * ------------------------------------- + * When the device is opened for playback, the device is runtime active + * already and the display refcount is 1 as explained above. + * + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just + * increments the PM runtime usage count of the codec since the device + * is in use already + * 2. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just + * decrements the PM runtime usage count of the codec since the device + * is in use already + * + * Once playback is stopped, the display refcount is set to 0 as explained + * above in the HDMI playback sequence. The PM handlings are designed in + * such way that to balance the refcount of display power when the codec + * device put to S3 while playback is going on. + * + * S0/S3 Cycle without playback in progress + * ---------------------------------------- + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec + * 2. skl_runtime_resume() invoked + * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * 4. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec + * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 4. skl_runtime_suspend() invoked + */ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_ext_device *edev = to_hda_ext_device(dev); -- 2.7.4

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[alsa-devel] [PATCH 2/2] ASoC: AMD: Add an always on fixed 1.8V regulator for da7219 VDDIO
by Daniel Kurtz 18 Jul '18

18 Jul '18

From: Akshu Agrawal <akshu.agrawal(a)amd.com> DA7219's VDDIO for our platform need to be configured for 1.8V. Hence we register a fixed 1.8V voltage regulator in the machine driver. Change-Id: I65fd93e1dd37c3e0d38265b4b1492ea53b93afd4 Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf4d1a4c2..58c1dcb4d2550f 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index ccddc6650b9c79..6ea0f42f5b7692 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/module.h> +#include <linux/regulator/fixed.h> +#include <linux/regulator/machine.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -278,11 +280,23 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00") +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct platform_device *reg; + + reg = regulator_register_always_on(0, "fixed-1.8V", + acp_da7219_supplies, + ARRAY_SIZE(acp_da7219_supplies), + 1800000); + if (!reg) + return -ENOMEM; machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); -- 2.18.0.203.gfac676dfb9-goog

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