October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH -next] ASoC: SOF: select SND_INTEL_DSP_CONFIG in SND_SOC_SOF_PCI
by Mao Wenan 28 Oct '19

28 Oct '19

When SND_SOC_SOF_PCI=y, and SND_INTEL_DSP_CONFIG=m, below errors can be seen: sound/soc/sof/sof-pci-dev.o: In function `sof_pci_probe': sof-pci-dev.c:(.text+0xb9): undefined reference to `snd_intel_dsp_driver_probe' After commit 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code"), sof_pci_probe() will call snd_intel_dsp_driver_probe(), so it should select SND_INTEL_DSP_CONFIG in Kconfig SND_SOC_SOF_PCI. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Mao Wenan <maowenan(a)huawei.com> --- sound/soc/sof/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 56a3ab6..a9b2be2 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_SOF_PCI select SND_SOC_ACPI if ACPI select SND_SOC_SOF_OPTIONS select SND_SOC_SOF_INTEL_PCI if SND_SOC_SOF_INTEL_TOPLEVEL + select SND_INTEL_DSP_CONFIG help This adds support for PCI enumeration. This option is required to enable Intel Skylake+ devices -- 2.7.4

2 1

[alsa-devel] [PATCH] Fix JACK plugin buffer sizes
by John McKay 27 Oct '19

27 Oct '19

I was having a conversation on the Archlinux forums[1] about using the alsa-plugins JACK plugin. While one of the bugs that we found has already been fixed in the git alsa-lib, there is another bug relating to buffer sizes. Currently snd_pcm_hw_params_set_buffer_size_near() doesn't really work when you are using this plugin unless you happen to request a buffer size that almost exactly matches a whole multiple of periods that the JACK is asking for. After some debugging, I discovered the reason is that some plugins, notably the 'plug' plugin that most everybody is going to use to change frequency/format directly attempt to manipulate the buffer size. This causes the code in pcm_ioplug.c to fail because it cannot find a number of periods/period time that can fit the buffer. The included patch stops this by specifying the list of acceptable buffer sizes as well, so that when other plugins try to change the buffer it always picks one that will work. It does have some problems in the JACK plugin now offers a number of periods/sizes that don't exist but in practice asking for huge buffers just seems to give you a more sane value. [1] https://bbs.archlinux.org/viewtopic.php?id=250116 --- jack/pcm_jack.c 2019-05-10 06:57:24.000000000 +0000 +++ jack/pcm_jack.c 2019-10-27 03:43:36.053734268 +0000 @@ -428,6 +428,7 @@ 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 bsize_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->num_ports); @@ -441,6 +442,9 @@ for (i = 1; i <= ARRAY_SIZE(psize_list); i++) psize_list[i-1] = jack_buffer_bytes * i; + for (i = 1; i <= ARRAY_SIZE(bsize_list); i++) + bsize_list[i-1] = jack_buffer_bytes * (i + 1); + 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 || @@ -452,6 +456,8 @@ rate, rate)) < 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_list(&jack->io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, + ARRAY_SIZE(bsize_list), bsize_list)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_PERIODS, 2, 64)) < 0) return err;

1 0

[alsa-devel] [PATCH] ALSA: bebob: Fix prototype of helper function to return negative value
by Takashi Sakamoto 26 Oct '19

26 Oct '19

A helper function of ALSA bebob driver returns negative value in a function which has a prototype to return unsigned value. This commit fixes it by changing the prototype. Fixes: eb7b3a056cd8 ("ALSA: bebob: Add commands and connections/streams management" Cc: <stable(a)vger.kernel.org> # v3.16+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 7ac0d9f495c4..f7f0db5aa811 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -252,8 +252,7 @@ int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob, return err; } -static unsigned int -map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) +static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) { unsigned int sec, sections, ch, channels; unsigned int pcm, midi, location; -- 2.20.1

2 1

[alsa-devel] [PATCH v2 00/12] Add Samus Hotwording for RT5677
by Curtis Malainey 26 Oct '19

26 Oct '19

This patch series adds the hotwording implementation used in the Pixelbook on the RT5677 driver. Known Issues: There is a known issue where the system will fail to detect a hotword if suspended while the stream is open. This is due to the fact that the haswell-dsp suspends its I2S MCLK before the RT5677 suspends which causes the writes and reads to become corrupted as a result. Any recommendations to correct this behaviour would be appreciated. Changes since v1: * Addressed Mark's and Kuninori's comments * Adjusted DAI Links * Squashed related patches * Removed all kcontrol writes * Addressed race condition with dapm * Added firmware boot check * Moved all clocking to driver API * Moved DAPM routes that spanned drivers to machine driver Cc: Dylan Reid <dgreid(a)chromium.org> Cc: Ben Zhang <benzh(a)chromium.org> Ben Zhang (8): ASoC: rt5677: Add a PCM device for streaming hotword via SPI ASoC: rt5677: Load firmware via SPI using delayed work ASoC: bdw-rt5677: Add a DAI link for rt5677 SPI PCM device ASoC: rt5677: Enable jack detect while DSP is running ASoC: rt5677: Add DAPM audio path for hotword stream ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile ASoC: rt5677: Stop and restart DSP over suspend/resume ASoC: rt5677: Disable irq at suspend Curtis Malainey (4): ASoC: rt5677: Allow VAD to be shut on/off at all times ASoC: bdw-rt5677: Turn on MCLK1 for DSP via DAPM ASoC: rt5677: Set ADC clock to use PLL and enable ASRC ASoC: rt5677: Wait for DSP to boot before loading firmware sound/soc/codecs/rt5677-spi.c | 397 ++++++++++++++++++++++++ sound/soc/codecs/rt5677-spi.h | 1 + sound/soc/codecs/rt5677.c | 447 +++++++++++++++++++++++----- sound/soc/codecs/rt5677.h | 11 +- sound/soc/intel/boards/bdw-rt5677.c | 51 ++++ 5 files changed, 834 insertions(+), 73 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

4 19

[alsa-devel] [PATCH V2] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 25 Oct '19

25 Oct '19

The output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitaion in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Change in v2 - remove p2p/m2m word - use use_ideal_rate sound/soc/fsl/fsl_asrc.c | 37 +++++++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..89cf333154c7 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -259,8 +259,11 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * It configures those ASRC registers according to a configuration instance * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. + * + * Note: + * use_ideal_rate = true is need by some case which need higher performance. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +271,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,8 +355,10 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; - if (div[IN] == 0) { + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; + if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; @@ -360,18 +366,29 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) clk = asrc_priv->asrck_clk[clk_index[OUT]]; - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + /* + * Output rate should be align with the out samplerate. If set too + * high output rate, there will be lots of Overload. + * But some case need higher performance, then we can use + * IDEAL_RATIO_RATE specifically for such case. + */ + clk_rate = clk_get_rate(clk); + if (ideal && use_ideal_rate) + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); else - div[OUT] = clk_get_rate(clk) / outrate; + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; - if (div[OUT] == 0) { + if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; } + /* Divider range is [1, 1024] */ + div[IN] = min_t(u32, 1024, div[IN]); + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +577,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH V2] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Shengjiu Wang 25 Oct '19

25 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Change in v2 -add lock for fsl_esai_trigger_start. sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..9b28e2af26e4 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock to handle reset and stop behavior * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect reset and stop */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH v2] ASoC: ti: omap-mcpdm: Add support for pdmclk clock handling
by Peter Ujfalusi 25 Oct '19

25 Oct '19

McPDM module receives it's functional clock from external source. This clock is the pdmclk provided by the twl6040 audio IC. If the clock is not available all register accesses to McPDM fails and the module is not operational. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Hi, this has been lurking in my next-wip branch for some time... It might not needed anymore as I have not heard about issues with McPDM for a while. It was dropped back then because some core parts were not picked in time and this caused some issues, but can not find the exact reason Regards, Peter sound/soc/ti/omap-mcpdm.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index b8c8290265c7..2c536947b69f 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -17,6 +17,7 @@ #include <linux/err.h> #include <linux/io.h> #include <linux/irq.h> +#include <linux/clk.h> #include <linux/slab.h> #include <linux/pm_runtime.h> #include <linux/of_device.h> @@ -42,6 +43,7 @@ struct omap_mcpdm { int irq; struct pm_qos_request pm_qos_req; int latency[2]; + struct clk *pdmclk; struct mutex mutex; @@ -416,6 +418,10 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int ret; + ret = clk_prepare_enable(mcpdm->pdmclk); + if (ret) + return ret; + pm_runtime_enable(mcpdm->dev); /* Disable lines while request is ongoing */ @@ -454,6 +460,7 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) if (pm_qos_request_active(&mcpdm->pm_qos_req)) pm_qos_remove_request(&mcpdm->pm_qos_req); + clk_disable_unprepare(mcpdm->pdmclk); return 0; } @@ -473,12 +480,19 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai) mcpdm->pm_active_count++; } + clk_disable_unprepare(mcpdm->pdmclk); + return 0; } static int omap_mcpdm_resume(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(mcpdm->pdmclk); + if (ret) + return ret; if (mcpdm->pm_active_count) { while (mcpdm->pm_active_count--) @@ -573,6 +587,15 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; + mcpdm->pdmclk = devm_clk_get(&pdev->dev, "pdmclk"); + if (IS_ERR(mcpdm->pdmclk)) { + if (PTR_ERR(mcpdm->pdmclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + dev_warn(&pdev->dev, "Error getting pdmclk (%ld)!\n", + PTR_ERR(mcpdm->pdmclk)); + mcpdm->pdmclk = NULL; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, &omap_mcpdm_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 2

[alsa-devel] sof-skl_hda: move hdmi as separate use case
by GitHub pull_request - opened 25 Oct '19

25 Oct '19

alsa-project/alsa-ucm-conf pull request #6 was opened from juimonen: Move hdmi definitions as separate use case and conf file. This will enable Pulseaudio to create separate profile for hdmi and make routing much more fluent than with single profile. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/6 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/6.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] sof-skl_hda: fix headphone jack name
by GitHub pull_request - opened 25 Oct '19

25 Oct '19

alsa-project/alsa-ucm-conf pull request #5 was opened from juimonen: Change "Headphone Jack" to "Headphone Mic Jack". Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/5 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/5.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [ANNOUNCE] 2019 Linux Audio miniconference
by Mark Brown 25 Oct '19

25 Oct '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue) and will be free of charge to attend, generously sponsored by Intel. Thus far for the agenda we have: - Use case management enhancements (Curtis) - DSP framework integration (Liam?) - Soundwire status (Pierre?) - Componentisation status/plans (Me) As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

8 19