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[PATCH v6] ASoC: tas2781: Fix wrong loading calibrated data sequence
by Shenghao Ding 04 Jun '24

04 Jun '24

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[PATCH] ASoC: dmaengine_pcm: terminate dmaengine before synchronize
by Shengjiu Wang 04 Jun '24

04 Jun '24

When dmaengine supports pause function, in suspend state, dmaengine_pause() is called instead of dmaengine_terminate_async(), In end of playback stream, the runtime->state will go to SNDRV_PCM_STATE_DRAINING, if system suspend & resume happen at this time, application will not resume playback stream, the stream will be closed directly, the dmaengine_terminate_async() will not be called before the dmaengine_synchronize(), which violates the call sequence for dmaengine_synchronize(). This behavior also happens for capture streams, but there is no SNDRV_PCM_STATE_DRAINING state for capture. So use dmaengine_tx_status() to check the DMA status if the status is DMA_PAUSED, then call dmaengine_terminate_async() to terminate dmaengine before dmaengine_synchronize(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/core/pcm_dmaengine.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 12aa1cef11a1..7346c993a067 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -358,6 +358,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan); int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_tx_state state; + enum dma_status status; + + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status == DMA_PAUSED) + dmaengine_terminate_async(prtd->dma_chan); dmaengine_synchronize(prtd->dma_chan); kfree(prtd); @@ -378,6 +384,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_tx_state state; + enum dma_status status; + + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status == DMA_PAUSED) + dmaengine_terminate_async(prtd->dma_chan); dmaengine_synchronize(prtd->dma_chan); dma_release_channel(prtd->dma_chan); -- 2.34.1

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[PATCH] soundwire: bus: suppress probe deferral errors
by Johan Hovold 04 Jun '24

04 Jun '24

Use dev_err_probe() to avoid logging errors on probe deferral: wsa884x-codec sdw:4:0:0217:0204:00:0: Probe of wsa884x-codec failed: -517 Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- drivers/soundwire/bus_type.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus_type.c b/drivers/soundwire/bus_type.c index c32faace618f..8c3746f0d527 100644 --- a/drivers/soundwire/bus_type.c +++ b/drivers/soundwire/bus_type.c @@ -112,9 +112,9 @@ static int sdw_drv_probe(struct device *dev) if (!name) name = drv->driver.name; - dev_err(dev, "Probe of %s failed: %d\n", name, ret); dev_pm_domain_detach(dev, false); - return ret; + + return dev_err_probe(dev, ret, "Probe of %s failed\n", name); } mutex_lock(&slave->sdw_dev_lock); -- 2.44.1

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[Question] how to get underlying file descriptors
by GitHub issues - opened 03 Jun '24

03 Jun '24

alsa-project/alsa-lib issue #399 was opened from pfeatherstone: I would like to use Alsa with the asio c++ library. I think all I need is the underlying file descriptor(s) for the device which I can then pass to Asio for polling and do all my asyncio. At the moment you're forced to use pollfd using snd_pcm_poll_descriptors(). Can I just read the fds or are there subtleties? Cheers Issue URL : https://github.com/alsa-project/alsa-lib/issues/399 Repository URL: https://github.com/alsa-project/alsa-lib

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hcontrol,mixer: improve callback GIL state handling
by GitHub pull_request - opened 03 Jun '24

03 Jun '24

alsa-project/alsa-python pull request #12 was opened from perexg: It's an attempt to fix warning seen in #11 . Request URL : https://github.com/alsa-project/alsa-python/pull/12 Patch URL : https://github.com/alsa-project/alsa-python/pull/12.patch Repository URL: https://github.com/alsa-project/alsa-python

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[PATCH 0/3] Dead structs in sound/soc/codecs
by linux＠treblig.org 02 Jun '24

02 Jun '24

From: "Dr. David Alan Gilbert" <linux(a)treblig.org> Clean out a bunch of old structs in sound/soc/codecs. Build tested only. Signed-off-by: Dr. David Alan Gilbert <linux(a)treblig.org> Dr. David Alan Gilbert (3): ASoC: codecs: lpass-rx-macro: remove unused struct 'rx_macro_reg_mask_val' ASoC: codecs: wm0010: remove unused struct 'wm0010_spi_msg' ASoC: codecs: cx2072x: remove unused struct 'cx2072x_eq_ctrl' sound/soc/codecs/cx2072x.c | 5 ----- sound/soc/codecs/lpass-rx-macro.c | 6 ------ sound/soc/codecs/wm0010.c | 8 -------- 3 files changed, 19 deletions(-) -- 2.45.1

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USB-Audio: Add support for MOTU Ultralite mk5
by GitHub pull_request - opened 01 Jun '24

01 Jun '24

alsa-project/alsa-ucm-conf pull request #422 was opened from clement-dufour: Adds support for the [MOTU Ultralite mk5](https://motu.com/en-us/products/gen5/ultralite-mk5/) soundcard. I named each device after their names on the MOTU Cuemix Control application. Successfully tested on my hardware, see the [alsa-info.sh output](http://alsa-project.org/db/?f=ca07253279e53fa354820f356f6beb5d2026d…. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/422 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/422.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/7] ALSA: hda: Improvements to hda_component
by Simon Trimmer 01 Jun '24

01 Jun '24

This series of patches makes sure that the existing consumers of the infrastructure unbind their interface before they begin destroying the driver, moves duplicated members from the instanced component structures into a new parent structure and introduces locking so that consumers of the interface do not use stale data. Simon Trimmer (7): ALSA: hda: cs35l56: Component should be unbound before deconstruction ALSA: hda: cs35l41: Component should be unbound before deconstruction ALSA: hda/tas2781: Component should be unbound before deconstruction ALSA: hda: hda_component: Introduce component parent structure ALSA: hda: hda_component: Change codecs to use component parent structure ALSA: hda: hda_component: Move codec field into the parent ALSA: hda: hda_component: Protect shared data with a mutex sound/pci/hda/cs35l41_hda.c | 47 +++++++++++--------- sound/pci/hda/cs35l56_hda.c | 29 +++++++------ sound/pci/hda/hda_component.c | 76 ++++++++++++++++++++------------- sound/pci/hda/hda_component.h | 48 ++++++++++++++------- sound/pci/hda/patch_realtek.c | 17 ++++---- sound/pci/hda/tas2781_hda_i2c.c | 37 ++++++++-------- 6 files changed, 148 insertions(+), 106 deletions(-) -- 2.34.1

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[PATCH v2 0/6] ASoC: Drop or replace of_gpio.h
by Andy Shevchenko 31 May '24

31 May '24

Replace or drop the legacy header that is subject to remove. Not all of them were compile-tested, the series might have hidden compilation errors. In v2: - added tags (Kuninori, Charles) - ripped out TAS2781 (it's a mess from GPIO handling perspective) Andy Shevchenko (6): ASoC: codecs: Remove unused of_gpio.h ASoC: fsl: Remove unused of_gpio.h ASoC: rockchip: Remove unused of_gpio.h ASoC: codecs: Replace of_gpio.h by proper one ASoC: generic: Replace of_gpio.h by proper one ASoC: samsung: Replace of_gpio.h by proper one sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4458.c | 1 - sound/soc/codecs/aw88395/aw88395.c | 2 +- sound/soc/codecs/aw88399.c | 1 - sound/soc/codecs/cs53l30.c | 1 - sound/soc/codecs/max98390.c | 1 - sound/soc/codecs/pcm3168a.c | 1 - sound/soc/codecs/rk817_codec.c | 1 - sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tlv320adc3xxx.c | 1 - sound/soc/codecs/tlv320adcx140.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/ts3a227e.c | 1 - sound/soc/codecs/wsa883x.c | 1 - sound/soc/fsl/imx-es8328.c | 1 - sound/soc/fsl/imx-rpmsg.c | 2 -- sound/soc/generic/audio-graph-card2-custom-sample.c | 3 ++- sound/soc/rockchip/rockchip_i2s.c | 1 - sound/soc/rockchip/rockchip_spdif.c | 1 - sound/soc/samsung/aries_wm8994.c | 2 +- 23 files changed, 4 insertions(+), 24 deletions(-) -- 2.43.0.rc1.1336.g36b5255a03ac

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[PATCHv4 7/9] ASoC: fsl-asoc-card: add DT clock "cpu_sysclk" with generic codec
by Elinor Montmasson 31 May '24

31 May '24

Add an optional DT clock "cpu_sysclk" to get the CPU DAI system-clock frequency when using the generic codec. It is set for both Tx and Rx. The way the frequency value is used is up to the CPU DAI driver implementation. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 9aca8ad15372..c7fc9c16f761 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -754,6 +754,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) snd_soc_of_parse_tdm_slot(np, NULL, NULL, &priv->cpu_priv.slot_num, &priv->cpu_priv.slot_width); + struct clk *cpu_sysclk = clk_get(&pdev->dev, "cpu_sysclk"); + if (!IS_ERR(cpu_sysclk)) { + priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(cpu_sysclk); + priv->cpu_priv.sysclk_freq[RX] = priv->cpu_priv.sysclk_freq[TX]; + clk_put(cpu_sysclk); + } } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; -- 2.34.1

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