May 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: line6: Improve poor error handling in line6_init_cap_control
by Hyeonggon Yoo 25 May '21

25 May '21

line6_init_cap_control doesn't free resources properly when allocations like kmalloc, usb_alloc_urb fails. It can cause memory leak when some allocations succeed, and then an error occurs. This patch makes line6_init_cap_control to properly free the resources, freeing previously allocated resources when there's an error. Signed-off-by: Hyeonggon Yoo <42.hyeyoo(a)gmail.com> --- sound/usb/line6/driver.c | 36 +++++++++++++++++++++++++++--------- 1 file changed, 27 insertions(+), 9 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 9602929b7de9..6991cb855023 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -688,34 +688,52 @@ static int line6_init_cap_control(struct usb_line6 *line6) /* initialize USB buffers: */ line6->buffer_listen = kmalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); - if (!line6->buffer_listen) - return -ENOMEM; + if (!line6->buffer_listen) { + ret = -ENOMEM; + goto fail_ret; + } line6->urb_listen = usb_alloc_urb(0, GFP_KERNEL); - if (!line6->urb_listen) - return -ENOMEM; + if (!line6->urb_listen) { + ret = -ENOMEM; + goto fail1; + } if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) { line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); - if (!line6->buffer_message) - return -ENOMEM; + if (!line6->buffer_message) { + ret = -ENOMEM; + goto fail2; + } ret = line6_init_midi(line6); if (ret < 0) - return ret; + goto fail3; } else { ret = line6_hwdep_init(line6); if (ret < 0) - return ret; + goto fail2; } ret = line6_start_listen(line6); if (ret < 0) { dev_err(line6->ifcdev, "cannot start listening: %d\n", ret); - return ret; + if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) + goto fail3; + else + goto fail2; } return 0; + +fail3: + kfree(line6->buffer_message); +fail2: + usb_free_urb(line6->urb_listen); +fail1: + kfree(line6->buffer_listen); +fail_ret: + return ret; } static void line6_startup_work(struct work_struct *work) -- 2.25.1

2 3

[GIT PULL] ASoC fixes for v5.13-rc3
by Mark Brown 25 May '21

25 May '21

The following changes since commit 25c4a9b614f101bb9f3e687960815db7dc439c0f: ASoC: simple-card: Fix breakage on kontron-sl28-var3-ads2 (2021-04-23 18:13:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.13-rc3 for you to fetch changes up to af2702549d68519ac78228e915d9b2c199056787: ASoC: qcom: lpass-cpu: Use optional clk APIs (2021-05-21 13:12:29 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.13 A collection of fixes that have come in since the merge window, mainly device specific things. The fixes to the generic cards from Morimoto-san are handling regressions that were introduced in the merge window on at least the Kontron sl28-var3-ads2. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: fsl: fix SND_SOC_IMX_RPMSG dependency Bixuan Cui (1): ASoC: codecs: lpass-tx-macro: add missing MODULE_DEVICE_TABLE Charles Keepax (5): ASoC: cs53l30: Add missing regmap use_single config ASoC: cs42l73: Add missing regmap use_single config ASoC: cs35l34: Add missing regmap use_single config ASoC: cs35l32: Add missing regmap use_single config ASoC: cs42l52: Minor tidy up of error paths Dan Carpenter (1): ASoC: cs35l33: fix an error code in probe() Hans de Goede (2): ASoC: Intel: bytcr_rt5640: Add quirk for the Glavey TM800A550L tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix 3-830 tablet Jerome Brunet (2): ASoC: stm32: do not request a new clock consummer reference ASoC: da7219: do not request a new clock consummer reference Kuninori Morimoto (5): ASoC: simple-card: add simple_parse_node() ASoC: simple-card: add simple_link_init() ASoC: audio-graph: tidyup graph_dai_link_of_dpcm() ASoC: audio-graph: tidyup graph_parse_node() ASoC: soc-dai.h: Align the word of comment for SND_SOC_DAIFMT_CBC_CFC Marco Felsch (1): ASoC: max98088: fix ni clock divider calculation Mark Brown (2): Merge series "ASoC: Revert clk_hw_get_clk() cleanup" from Jerome Brunet <jbrunet(a)baylibre.com>: Merge series "ASoC: simple-card / audio-graph re-cleanup" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Ranjani Sridharan (1): ASoC: SOF: Intel: hda: don't send DAI_CONFIG IPC for older firmware Richard Fitzgerald (1): ASoC: cs42l42: Regmap must use_single_read/write Shengjiu Wang (1): ASoC: ak5558: Correct the dai name for ak5552 Shuming Fan (1): ASoC: rt711-sdca: fix the function number of SDCA control for feature unit 0x1E Srinivas Kandagatla (1): ASoC: codecs: lpass-rx-macro: add missing MODULE_DEVICE_TABLE Stephen Boyd (1): ASoC: qcom: lpass-cpu: Use optional clk APIs Vijendar Mukunda (1): ASoC: amd: fix for pcm_read() error Zou Wei (1): ASoC: sti-sas: add missing MODULE_DEVICE_TABLE include/sound/soc-dai.h | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 10 -- sound/soc/amd/raven/acp3x.h | 1 + sound/soc/amd/raven/pci-acp3x.c | 15 +++ sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/cs35l32.c | 3 + sound/soc/codecs/cs35l33.c | 1 + sound/soc/codecs/cs35l34.c | 3 + sound/soc/codecs/cs42l42.c | 3 + sound/soc/codecs/cs42l56.c | 7 +- sound/soc/codecs/cs42l73.c | 3 + sound/soc/codecs/cs53l30.c | 3 + sound/soc/codecs/da7219.c | 5 +- sound/soc/codecs/lpass-rx-macro.c | 1 + sound/soc/codecs/lpass-tx-macro.c | 1 + sound/soc/codecs/max98088.c | 13 ++- sound/soc/codecs/rt711-sdca.c | 4 +- sound/soc/codecs/sti-sas.c | 1 + sound/soc/fsl/Kconfig | 1 + sound/soc/generic/audio-graph-card.c | 57 ++++++------ sound/soc/generic/simple-card.c | 168 +++++++++++++++++----------------- sound/soc/intel/boards/bytcr_rt5640.c | 25 +++++ sound/soc/qcom/lpass-cpu.c | 12 +-- sound/soc/sof/intel/hda-dai.c | 5 + sound/soc/stm/stm32_sai_sub.c | 5 +- 25 files changed, 202 insertions(+), 149 deletions(-)

2 1

[PATCH] ASoC: meson: use dev_err_probe
by Jerome Brunet 25 May '21

25 May '21

Use dev_err_probe() helper function to handle probe deferral. It removes the open coded test for -EPROBE_DEFER but more importantly, it sets the deferral reason in debugfs which is great for debugging. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/aiu.c | 17 +++++++---------- sound/soc/meson/axg-fifo.c | 11 +++++------ sound/soc/meson/axg-pdm.c | 9 +++------ sound/soc/meson/axg-spdifin.c | 6 ++---- sound/soc/meson/axg-spdifout.c | 6 ++---- sound/soc/meson/axg-tdm-formatter.c | 18 ++++++------------ sound/soc/meson/axg-tdm-interface.c | 10 ++++------ sound/soc/meson/meson-card-utils.c | 4 ++-- sound/soc/meson/t9015.c | 7 +++---- 9 files changed, 34 insertions(+), 54 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index ba15d5762b0b..37036adf14ce 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -219,31 +219,29 @@ static int aiu_clk_get(struct device *dev) aiu->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(aiu->pclk)) { - if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu pclk\n"); + dev_err_probe(dev, PTR_ERR(aiu->pclk), + "Can't get the aiu pclk\n"); return PTR_ERR(aiu->pclk); } aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); if (IS_ERR(aiu->spdif_mclk)) { - if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu spdif master clock\n"); + dev_err_probe(dev, PTR_ERR(aiu->spdif_mclk), + "Can't get the aiu spdif master clock\n"); return PTR_ERR(aiu->spdif_mclk); } ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids), &aiu->i2s); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the i2s clocks\n"); + dev_err_probe(dev, ret, "Can't get the i2s clocks\n"); return ret; } ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids), &aiu->spdif); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the spdif clocks\n"); + dev_err_probe(dev, ret, "Can't get the spdif clocks\n"); return ret; } @@ -282,8 +280,7 @@ static int aiu_probe(struct platform_device *pdev) ret = device_reset(dev); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Failed to reset device\n"); + dev_err_probe(dev, ret, "Failed to reset device\n"); return ret; } diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index b9af2d513e09..d670a32d56ab 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -352,17 +352,16 @@ int axg_fifo_probe(struct platform_device *pdev) fifo->pclk = devm_clk_get(dev, NULL); if (IS_ERR(fifo->pclk)) { - if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %ld\n", - PTR_ERR(fifo->pclk)); + dev_err_probe(dev, PTR_ERR(fifo->pclk), + "failed to get pclk: %ld\n", PTR_ERR(fifo->pclk)); return PTR_ERR(fifo->pclk); } fifo->arb = devm_reset_control_get_exclusive(dev, NULL); if (IS_ERR(fifo->arb)) { - if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) - dev_err(dev, "failed to get arb reset: %ld\n", - PTR_ERR(fifo->arb)); + dev_err_probe(dev, PTR_ERR(fifo->arb), + "failed to get arb reset: %ld\n", + PTR_ERR(fifo->arb)); return PTR_ERR(fifo->arb); } diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c index bfd37d49a73e..f630311b5b29 100644 --- a/sound/soc/meson/axg-pdm.c +++ b/sound/soc/meson/axg-pdm.c @@ -613,24 +613,21 @@ static int axg_pdm_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->dclk = devm_clk_get(dev, "dclk"); if (IS_ERR(priv->dclk)) { ret = PTR_ERR(priv->dclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk: %d\n", ret); return ret; } priv->sysclk = devm_clk_get(dev, "sysclk"); if (IS_ERR(priv->sysclk)) { ret = PTR_ERR(priv->sysclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d0d09f945b48..8f587c3bb5d7 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -481,16 +481,14 @@ static int axg_spdifin_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->refclk = devm_clk_get(dev, "refclk"); if (IS_ERR(priv->refclk)) { ret = PTR_ERR(priv->refclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c index e769a5ee6e27..58e02f6898b7 100644 --- a/sound/soc/meson/axg-spdifout.c +++ b/sound/soc/meson/axg-spdifout.c @@ -424,16 +424,14 @@ static int axg_spdifout_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk)) { ret = PTR_ERR(priv->mclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index cab7fa2851aa..108f11e14006 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -284,8 +284,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(formatter->pclk)) { ret = PTR_ERR(formatter->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } @@ -293,8 +292,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(formatter->sclk)) { ret = PTR_ERR(formatter->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk: %d\n", ret); return ret; } @@ -302,8 +300,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(formatter->lrclk)) { ret = PTR_ERR(formatter->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk: %d\n", ret); return ret; } @@ -311,8 +308,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); if (IS_ERR(formatter->sclk_sel)) { ret = PTR_ERR(formatter->sclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk_sel: %d\n", ret); return ret; } @@ -320,8 +316,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); if (IS_ERR(formatter->lrclk_sel)) { ret = PTR_ERR(formatter->lrclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk_sel: %d\n", ret); return ret; } @@ -329,8 +324,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->reset = devm_reset_control_get_optional_exclusive(dev, NULL); if (IS_ERR(formatter->reset)) { ret = PTR_ERR(formatter->reset); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get reset: %d\n", ret); + dev_err_probe(dev, ret, "failed to get reset: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 87cac440b369..a2c7f658f6b4 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -519,8 +519,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(iface->sclk)) { ret = PTR_ERR(iface->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk: %d\n", ret); return ret; } @@ -528,8 +527,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(iface->lrclk)) { ret = PTR_ERR(iface->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk: %d\n", ret); return ret; } @@ -545,8 +543,8 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) if (ret == -ENOENT) { iface->mclk = NULL; } else { - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", + ret); return ret; } } diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 300ac8be46ef..ac062c03cf33 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -86,8 +86,8 @@ int meson_card_parse_dai(struct snd_soc_card *card, ret = of_parse_phandle_with_args(node, "sound-dai", "#sound-dai-cells", 0, &args); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(card->dev, "can't parse dai %d\n", ret); + dev_err_probe(card->dev, ret, + "can't parse sound-dai at %pOFn\n", node); return ret; } *dai_of_node = args.np; diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c index 4c1349dd1e06..7ff16f51dbd4 100644 --- a/sound/soc/meson/t9015.c +++ b/sound/soc/meson/t9015.c @@ -259,15 +259,14 @@ static int t9015_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { - if (PTR_ERR(priv->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get core clock\n"); + dev_err_probe(dev, PTR_ERR(priv->pclk), + "failed to get core clock\n"); return PTR_ERR(priv->pclk); } priv->avdd = devm_regulator_get(dev, "AVDD"); if (IS_ERR(priv->avdd)) { - if (PTR_ERR(priv->avdd) != -EPROBE_DEFER) - dev_err(dev, "failed to AVDD\n"); + dev_err_probe(dev, PTR_ERR(priv->avdd), "failed to AVDD\n"); return PTR_ERR(priv->avdd); } -- 2.32.0.rc0

2 2

[PATCH -next] ALSA: pcm: use DEVICE_ATTR_RO macro
by YueHaibing 25 May '21

25 May '21

Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/core/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index b163164a83ec..6fd3677685d7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1004,7 +1004,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) substream->pstr->substream_opened--; } -static ssize_t show_pcm_class(struct device *dev, +static ssize_t pcm_class_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); @@ -1024,7 +1024,7 @@ static ssize_t show_pcm_class(struct device *dev, return sprintf(buf, "%s\n", str); } -static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); +static DEVICE_ATTR_RO(pcm_class); static struct attribute *pcm_dev_attrs[] = { &dev_attr_pcm_class.attr, NULL -- 2.17.1

2 1

[PATCH] ALSA: ac97: fix PM reference leak in ac97_bus_remove()
by Yufen Yu 25 May '21

25 May '21

pm_runtime_get_sync will increment pm usage counter even it failed. Forgetting to putting operation will result in reference leak here. Fix it by replacing it with pm_runtime_resume_and_get to keep usage counter balanced. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yufen Yu <yuyufen(a)huawei.com> --- sound/ac97/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index d9077e91382b..6ddf646cda65 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -520,7 +520,7 @@ static int ac97_bus_remove(struct device *dev) struct ac97_codec_driver *adrv = to_ac97_driver(dev->driver); int ret; - ret = pm_runtime_get_sync(dev); + ret = pm_runtime_resume_and_get(dev); if (ret < 0) return ret; -- 2.25.4

2 1

[PATCH v2 0/6] ALSA: Prep work for PCI rescan support
by Takashi Iwai 25 May '21

25 May '21

Hi, this is a revised patch set as a prep work for the future-planned PCI rescan support(*). The essential part is the patch to track in-flight tasks, and the rest are cleanups and fixes in the core code. With those changes, the driver can perform a sort of software suspend on the device without the hardware intervention, which is required during the PCI BAR movement, for example. v1->v2: * Move / drop unnecessary snd_power_wait() calls * Rephrase changelogs and comments, reorder patches Takashi (*) https://lore.kernel.org/alsa-devel/e25017c6-e5e4-7a24-e793-14a2e70a434e@amd… === Takashi Iwai (6): ALSA: core: Use READ_ONCE() / WRITE_ONCE() for power state change ALSA: control: Track in-flight control read/write/tlv accesses ALSA: control: Drop superfluous snd_power_wait() calls ALSA: control: Minor optimization for SNDRV_CTL_IOCTL_POWER_STATE ALSA: Drop superfluous argument from snd_power_wait() ALSA: pcm: Block the release until the system resume finishes include/sound/core.h | 50 ++++++++++++++++++++++++++++++++++--- sound/core/control.c | 38 +++++++++++++--------------- sound/core/control_compat.c | 14 +++-------- sound/core/init.c | 46 ++++++++++++++++++++++++++++------ sound/core/pcm_native.c | 6 ++++- sound/soc/soc-core.c | 2 +- sound/usb/usx2y/us122l.c | 2 +- 7 files changed, 113 insertions(+), 45 deletions(-) -- 2.26.2

3 9

[PATCH v2 0/4] ALSA: firewire-lib: drop initial NODATA packets or empty packets
by Takashi Sakamoto 25 May '21

25 May '21

Hi, This patchset is take 2 of my previous one; * https://lore.kernel.org/alsa-devel/20210523124114.272134-1-o-takashi@sakamo… The devices based on BeBoB ASICs or the devices in Tascam FireWire series transfer a batch of NODATA packet or empty packet in initial step of streaming. To avoid processing them, current implementation uses an option to skip processing content of tx packet during some initial cycles. However, the hard-coded number is not enough useful. In 1st patch, ALSA IEC 61883-1/6 packet streaming engine becomes to drop the initial packets. As a result, The tx_init_skip_cycles argument of amdtp_domain_start() function changes its meaning. In the following patches, ALSA bebob driver is refactored. Changes from v1: * Fix -Wunused-but-set-variable warning reported by 0day-ci * Add 2nd patch to obsolete unused member of structure Takashi Sakamoto (4): ALSA: firewire-lib: drop initial NODATA or empty packet ALSA: firewire-lib: obsolete callbacked member ALSA: bebob: cancel switching connection order ALSA: bebob: distinguish M-Audio ProFire Lightbridge quirk sound/firewire/amdtp-stream.c | 145 +++++++++++++++++++--------- sound/firewire/amdtp-stream.h | 6 +- sound/firewire/bebob/bebob.c | 10 +- sound/firewire/bebob/bebob.h | 1 + sound/firewire/bebob/bebob_stream.c | 25 ++--- 5 files changed, 122 insertions(+), 65 deletions(-) -- 2.27.0

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[PATCH v2 0/2] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 25 May '21

25 May '21

During review of the RT5631 machine driver, Jon Hunter suggested that it will be better to squash all the current ASoC machine drivers into a single one. This suggestion is implemented by this patchset. The RT5631 support will come later with the ASUS Transformer changes. This series needs to be approved by Jaroslav Kysela before it can be merged. Changelog: v2: - Dropped use of of_device_compatible_match(), like it was suggested by Rob Herring in a review comment to v1. - Added patch that sets card's driver_name of as Tegra ASoC drivers. In a comment to v1 Jaroslav Kysela suggested that the Tegra drivers don't set the card name properly and he was right. I opened pull request with the new Tegra UCMs and updated lookup paths for older UCMs [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/92 Dmitry Osipenko (2): ASoC: tegra: Set driver_name=tegra for all machine drivers ASoC: tegra: Unify ASoC machine drivers sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra_alc5632.c | 259 ---------- sound/soc/tegra/tegra_asoc_machine.c | 732 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 45 ++ sound/soc/tegra/tegra_max98090.c | 276 ---------- sound/soc/tegra/tegra_rt5640.c | 222 -------- sound/soc/tegra/tegra_rt5677.c | 324 ------------ sound/soc/tegra/tegra_sgtl5000.c | 211 -------- sound/soc/tegra/tegra_wm8753.c | 185 ------- sound/soc/tegra/tegra_wm8903.c | 357 +++---------- sound/soc/tegra/tegra_wm9712.c | 166 ------ sound/soc/tegra/trimslice.c | 172 ------- 13 files changed, 862 insertions(+), 2117 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

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RE: Re: [PATCH] ALSA: pcm: Need to check whether runtime is valid or not
by EUNWOO KIM 25 May '21

25 May '21

> > > Sender : Takashi Iwai <tiwai(a)suse.de> > > > Date : 2021-05-12 16:08 (GMT+9) > > > Title : Re: [PATCH] ALSA: pcm: Need to check whether runtime is v alid or not > > > > > > On Wed, 12 May 2021 06:43:23 +0200, > > > Chanho Park wrote: > > > > > > > > From: eunwoo kim <ew.kim(a)samsung.com> > > > > > > > > Since snd_pcm_ioctl_sw_params_compat has no runtime variable ch eck, > > > > if application call the ioctl after close, it can make kernel c rash. > > > > So, snd_pcm_ioctl_sw_params_compat needs to check the runtime v ariable > > > > at the beginning of the function. > > > > > > > In principle, you cannot call ioctl for an already closed file. > > > > Or do you mean other code path? > > > > > > Yes, other code path such as application layer or alsa framwork l ayer. > > > > But how can it go over the 32bit compat ioctl layer...? > > It's always tied with a file object, so the runtime object is > > assured. > > > > > > Yes, I think so too. but in fact, some case can call 32bit comp at ioctl. > > > > > In our case, suspend call is releasing all running pcm device. ot her side some application want to start music. > > > it make critical section between suspend and pcm_open from applic ation. > > > this is error and we have to solve this problem without this patc h. > > > but I think kernel don't make kernel crash even if application or system architecture have problem. > > > > I guess you're scratching a wrong surface. > > > > > could you tell me more detail? why do you think that. > > > In 64bit native case, it didn't make kernel crash on same test ca se. because native always check runtime in ioctl. > > > Try to put WARN_ON() there. If you can catch the real case, it'd b e > > worth to merge. Otherwise, it's just a wrong place to look at. > > I already found root cause in our issue case. And I am fixing this is sue via changing sequence. > > I think that you want to know issue case because why this patch need. > So I explain more about our case. > 1. application playback sound in android. > 2. call pcm_open in tinyalsa. > 3. call file open and hw param in pcm_open > 4. go to suspend > 5. freezing all task before calling sw param in pcm_open. > 6. kernel layer try to go suspend > 7. release all running substream, because all HW IPs must to go idle for power save mode. > -> detach runtime from substream. > 8. go to resume. > 9. call sw param in pcm_open. > 10-1. native case : has runtime check in ioctl -> return error -> aud io hal layer recovery hw connect. > 10-2. 32bit compat case : has not runtime check -> runtime is not val id -> make kernel crash. > > - if AAOS can support wakelock, we can postpone suspend until finishe d pcm_open. but AAOS didn't support wakelock. > I'm afraid that the approach is wrong. It breaks the fundamental PCM > state change rule. Yes, that is wrong approach I know. So I am changing that. > I guess it's a downstream driver that does that? If so, tweaking > something superfluous in the upstream code is unacceptable. > BTW, if you want further discussions, please don't drop Cc to > alsa-devel ML. Safety is very important in automotive. so it must not to make kernel crash even if someone make wrong code. because it can connect to accident. If 32bit compat don't need checking-runtime, why do pcm_native check runtime? thanks, Takashi [update?userid=ew.kim&do=bWFpbElEPTIwMjEwNTE5MjMxNjE1ZXBjbXMycDE4Y2NlMT Q1YzJhOWMwZDk5ZmViZjMxNDBjMTY3ZDIyNiZyZWNpcGllbnRBZGRyZXNzPWFsc2EtZGV2Z WxAYWxzYS1wcm9qZWN0Lm9yZw__]

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[RFC] ALSA: hda: Add Lenovo ThinkStation P340 to power_save_denylist
by Peter Ujfalusi 25 May '21

25 May '21

On playback start there is a huge plock/pop noise via the green jack used in Line out mode. The only way I was able to fix it to set the power_save to 0 for the snd_hda_intel. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Takashi, ever since I have this workstation the plock/pop noise on playback start bugged me. So far the only solution I have found is to disable the power_save which I find acceptable on a desktop compared to the pops. The alsa-info.sh output is can be found here: http://alsa-project.org/db/?f=414b1b236fc73db386ad4f938bc3b304f6b30b05 The machine have ALC623 codec and the jacks are correctly discovered and they do what they supposed to be. When I have the headset connected to the front and audio is routed there the line out (green jack) from the back still prodices the plock/pop. Is there a known quirk for similar issue or a better way to handle it? Thanks you, Peter Ujfalusi sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..e16d93d8291f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2213,6 +2213,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* Loud pluck/pop noise via the green jack on playback start */ + SND_PCI_QUIRK(0x8086, 0x06c8, "Lenovo ThinkStation P340", 0), {} }; #endif /* CONFIG_PM */ -- 2.31.1

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