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

[PATCH] ASoC: Intel: boards: Select SND_SOC_HDAC_HDMI for Soundwire machine driver
by Richard Fitzgerald 25 May '21

25 May '21

Add select of SND_SOC_HDAC_HDMI for SND_SOC_INTEL_SOUNDWIRE_SOF_MACH. Without this the build results in missing references to hdac_hdmi_jack_port_init and hdac_hdmi_jack_init. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/intel/boards/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index eef5f4ac87c5..6a1639b14b67 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -601,6 +601,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_DMIC select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_INTEL_SOF_MAXIM_COMMON + select SND_SOC_HDAC_HDMI help Add support for Intel SoundWire-based platforms connected to MAX98373, RT700, RT711, RT1308 and RT715 -- 2.20.1

3 3

[PATCH 00/11] drm/vc4: hdmi: Enable Channel Mapping, IEC958, HBR Passthrough using hdmi-codec
by Maxime Ripard 25 May '21

25 May '21

Hi, hdmi-codec allows to have a lot of HDMI-audio related infrastructure in place, it's missing a few controls to be able to provide HBR passthrough. This series adds more infrastructure for the drivers, and leverages it in the vc4 HDMI controller driver. One thing that felt a bit weird is that even though https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-dri… mentions that the iec958 mask control should be a mixer control and the default control should be a PCM one, it feels a bit weird to have two different control type for two controls so similar, and other drivers are pretty inconsistent with this. Should we update the documentation? Thanks! Maxime Dom Cobley (5): drm/vc4: hdmi: Set HD_CTL_WHOLSMP and HD_CTL_CHALIGN_SET drm/vc4: hdmi: Set HDMI_MAI_FMT drm/vc4: hdmi: Set VC4_HDMI_MAI_CONFIG_FORMAT_REVERSE drm/vc4: hdmi: Remove firmware logic for MAI threshold setting ARM: dts: bcm2711: Tune DMA parameters for HDMI audio Maxime Ripard (6): snd: iec958: split status creation and fill ASoC: hdmi-codec: Rework to support more controls ASoC: hdmi-codec: Add iec958 controls ASoC: hdmi-codec: Add a prepare hook drm/vc4: hdmi: Register HDMI codec drm/vc4: hdmi: Remove redundant variables arch/arm/boot/dts/bcm2711.dtsi | 4 +- drivers/gpu/drm/vc4/Kconfig | 1 + drivers/gpu/drm/vc4/vc4_hdmi.c | 322 ++++++++++++++------------------- drivers/gpu/drm/vc4/vc4_hdmi.h | 5 +- drivers/gpu/drm/vc4/vc4_regs.h | 30 +++ include/sound/hdmi-codec.h | 12 +- include/sound/pcm_iec958.h | 8 + sound/core/pcm_iec958.c | 131 +++++++++----- sound/soc/codecs/hdmi-codec.c | 219 +++++++++++++++++----- 9 files changed, 456 insertions(+), 276 deletions(-) -- 2.31.1

2 17

[PATCH -next] ALSA: control_led - use DEVICE_ATTR_*() macro
by YueHaibing 25 May '21

25 May '21

Use DEVICE_ATTR_*() 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/control_led.c | 34 +++++++++++++++++++--------------- 1 file changed, 19 insertions(+), 15 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 25f57c14f294..a5e751f26d46 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -375,7 +375,7 @@ static void snd_ctl_led_disconnect(struct snd_card *card) * sysfs */ -static ssize_t show_mode(struct device *dev, +static ssize_t mode_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -390,7 +390,8 @@ static ssize_t show_mode(struct device *dev, return sprintf(buf, "%s\n", str); } -static ssize_t store_mode(struct device *dev, struct device_attribute *attr, +static ssize_t mode_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -419,7 +420,7 @@ static ssize_t store_mode(struct device *dev, struct device_attribute *attr, return count; } -static ssize_t show_brightness(struct device *dev, +static ssize_t brightness_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -427,8 +428,8 @@ static ssize_t show_brightness(struct device *dev, return sprintf(buf, "%u\n", ledtrig_audio_get(led->trigger_type)); } -static DEVICE_ATTR(mode, 0644, show_mode, store_mode); -static DEVICE_ATTR(brightness, 0444, show_brightness, NULL); +static DEVICE_ATTR_RW(mode); +static DEVICE_ATTR_RO(brightness); static struct attribute *snd_ctl_led_dev_attrs[] = { &dev_attr_mode.attr, @@ -562,22 +563,25 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si return count; } -static ssize_t parse_attach(struct device *dev, struct device_attribute *attr, +static ssize_t attach_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); return set_led_id(led_card, buf, count, true); } -static ssize_t parse_detach(struct device *dev, struct device_attribute *attr, +static ssize_t detach_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); return set_led_id(led_card, buf, count, false); } -static ssize_t ctl_reset(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t reset_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); int err; @@ -590,8 +594,8 @@ static ssize_t ctl_reset(struct device *dev, struct device_attribute *attr, return count; } -static ssize_t ctl_list(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t list_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); struct snd_card *card; @@ -624,10 +628,10 @@ static ssize_t ctl_list(struct device *dev, return buf2 - buf; } -static DEVICE_ATTR(attach, 0200, NULL, parse_attach); -static DEVICE_ATTR(detach, 0200, NULL, parse_detach); -static DEVICE_ATTR(reset, 0200, NULL, ctl_reset); -static DEVICE_ATTR(list, 0444, ctl_list, NULL); +static DEVICE_ATTR_WO(attach); +static DEVICE_ATTR_WO(detach); +static DEVICE_ATTR_WO(reset); +static DEVICE_ATTR_RO(list); static struct attribute *snd_ctl_led_card_attrs[] = { &dev_attr_attach.attr, -- 2.17.1

3 2

[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

2 5