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

[alsa-devel] [PATCH v2 00/23] ASoC: use devm_xxx
by Kuninori Morimoto 03 Jul '19

03 Jul '19

Hi Mark These are v2 patch of converting for devm_xxx(). v2 is caring about related empty functions. Kuninori Morimoto (23): ASoC: atmel: atmel-pcm-dma.c: use devm_snd_dmaengine_pcm_register() ASoC: sunxi: sun4i-i2s.c: use devm_snd_dmaengine_pcm_register() ASoC: tegra: tegra_pcm.c: use devm_snd_dmaengine_pcm_register() ASoC: ux500: ux500_pcm.c: use devm_snd_dmaengine_pcm_register() ASoC: au1x: i2sc.c: use devm_snd_soc_register_component() ASoC: au1x: ac97c.c: use devm_snd_soc_register_component() ASoC: au1x: psc-ac97.c: use devm_snd_soc_register_component() ASoC: au1x: psc-i2s.c: use devm_snd_soc_register_component() ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component() ASoC: cirrus: ep93xx-ac97.c: use devm_snd_soc_register_component() ASoC: rt5682: use devm_snd_soc_register_component() ASoC: ak4118: use devm_snd_soc_register_component() ASoC: cros_ec_codec: use devm_snd_soc_register_component() ASoC: fsl: imx-ssi.c: use devm_snd_soc_register_component() ASoC: fsl: mpc5200_psc_i2s.c: use devm_snd_soc_register_component() ASoC: fsl: mpc5200_psc_ac97.c: use devm_snd_soc_register_component() ASoC: pxa: pxa2xx-ac97.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_i2s.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_spdif.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_ac97.c: use devm_snd_soc_register_component() ASoC: tegra: tegra30_i2s.c: use devm_snd_soc_register_component() ASoC: ti: davinci-i2s.c: use devm_snd_soc_register_component() ASoC: ux500: ux500_msp_dai.c: use devm_snd_soc_register_component() sound/soc/atmel/atmel-pcm-dma.c | 9 ++------- sound/soc/atmel/atmel-pcm-pdc.c | 5 ----- sound/soc/atmel/atmel-pcm.h | 8 -------- sound/soc/atmel/atmel_ssc_dai.c | 11 ----------- sound/soc/au1x/ac97c.c | 4 +--- sound/soc/au1x/i2sc.c | 4 +--- sound/soc/au1x/psc-ac97.c | 6 ++---- sound/soc/au1x/psc-i2s.c | 6 ++---- sound/soc/cirrus/ep93xx-ac97.c | 10 +++------- sound/soc/cirrus/ep93xx-i2s.c | 7 ++----- sound/soc/codecs/ak4118.c | 11 ++--------- sound/soc/codecs/cros_ec_codec.c | 2 +- sound/soc/codecs/rt5682.c | 13 +++---------- sound/soc/fsl/imx-ssi.c | 8 ++------ sound/soc/fsl/mpc5200_psc_ac97.c | 3 +-- sound/soc/fsl/mpc5200_psc_i2s.c | 3 +-- sound/soc/pxa/pxa2xx-ac97.c | 3 +-- sound/soc/sunxi/sun4i-i2s.c | 4 +--- sound/soc/tegra/tegra20_ac97.c | 11 +++-------- sound/soc/tegra/tegra20_i2s.c | 11 +++-------- sound/soc/tegra/tegra20_spdif.c | 12 ++++-------- sound/soc/tegra/tegra30_i2s.c | 11 +++-------- sound/soc/tegra/tegra_pcm.c | 11 +++-------- sound/soc/tegra/tegra_pcm.h | 1 - sound/soc/ti/davinci-i2s.c | 8 ++------ sound/soc/ux500/ux500_msp_dai.c | 12 ++---------- sound/soc/ux500/ux500_pcm.c | 9 +-------- sound/soc/ux500/ux500_pcm.h | 1 - 28 files changed, 46 insertions(+), 158 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: line6: Fix write on zero-sized buffer
by Takashi Iwai 02 Jul '19

02 Jul '19

LINE6 drivers allocate the buffers based on the value returned from usb_maxpacket() calls. The manipulated device may return zero for this, and this results in the kmalloc() with zero size (and it may succeed) while the other part of the driver code writes the packet data with the fixed size -- which eventually overwrites. This patch adds a simple sanity check for the invalid buffer size for avoiding that problem. Reported-by: syzbot+219f00fb49874dcaea17(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/pcm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 72c6f8e82a7e..78c2d6cab3b5 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -560,6 +560,11 @@ int line6_init_pcm(struct usb_line6 *line6, line6pcm->max_packet_size_out = usb_maxpacket(line6->usbdev, usb_sndisocpipe(line6->usbdev, ep_write), 1); + if (!line6pcm->max_packet_size_in || !line6pcm->max_packet_size_out) { + dev_err(line6pcm->line6->ifcdev, + "cannot get proper max packet size\n"); + return -EINVAL; + } spin_lock_init(&line6pcm->out.lock); spin_lock_init(&line6pcm->in.lock); -- 2.16.4

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[alsa-devel] [PATCH v6] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 02 Jul '19

02 Jul '19

The supported formats are S16_LE and S24_LE now. However, S24_LE is not supported when TDM is 0 and it is not in the right justified mode. We should remove 24-bit format in that situation to avoid triggering error. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- Changed the order of the conditional. Remove the snd_pcm_hw_constraint_msbits function. Use removing 24 bits format instead of fixing at 16 bits format. sound/soc/codecs/max98090.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..9fbb4c31bcf1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,24 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* + * When TDM = 0, remove 24-bit format support if it is not in right + * justified mode. + */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J && + !max98090->tdm_slots) + substream->runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_S24_LE; + + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2334,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.410.gd8fdbe21b5-goog

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[alsa-devel] [PATCH -next] ASoC: madera: Remove duplicated include from cs47l35.c
by YueHaibing 02 Jul '19

02 Jul '19

Remove duplicated include. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs47l35.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 511d0d6fa962..e3585c1dab3d 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -19,7 +19,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/tlv.h> -#include <sound/tlv.h> #include <linux/irqchip/irq-madera.h> #include <linux/mfd/madera/core.h>

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[alsa-devel] Applied "ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 02 Jul '19

02 Jul '19

The patch ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From cb9002a186329b35f32f74021551aecbfe9a696c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 28 Jun 2019 13:09:00 +0900 Subject: [PATCH] ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component() We have devm_xxx version of snd_soc_register_component, let's use it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/cirrus/ep93xx-i2s.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 6ca899ba9484..0b4355e95f84 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -473,19 +473,17 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, info); - err = snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component, + err = devm_snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component, &ep93xx_i2s_dai, 1); if (err) goto fail_put_lrclk; err = devm_ep93xx_pcm_platform_register(&pdev->dev); if (err) - goto fail_unregister; + goto fail_put_lrclk; return 0; -fail_unregister: - snd_soc_unregister_component(&pdev->dev); fail_put_lrclk: clk_put(info->lrclk); fail_put_sclk: @@ -500,7 +498,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev) { struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); clk_put(info->lrclk); clk_put(info->sclk); clk_put(info->mclk); -- 2.20.1

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[alsa-devel] [PATCH -next] ASoC: rt1308: Fix platform_no_drv_owner.cocci warnings
by YueHaibing 02 Jul '19

02 Jul '19

Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt1308.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index 619a630e982e..d673506c7c39 100755 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -884,7 +884,6 @@ static void rt1308_i2c_shutdown(struct i2c_client *client) static struct i2c_driver rt1308_i2c_driver = { .driver = { .name = "rt1308", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(rt1308_of_match), .acpi_match_table = ACPI_PTR(rt1308_acpi_match), },

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[alsa-devel] [RFC PATCH 0/5] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 02 Jul '19

02 Jul '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support and module_sdw_driver macro. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Srinivas Kandagatla (5): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices soundwire: add module_sdw_driver helper macro dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.txt | 27 + .../devicetree/bindings/soundwire/bus.txt | 48 + drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 54 +- include/linux/soundwire/sdw_type.h | 11 + sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1160 +++++++++++++++++ 9 files changed, 1312 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.txt create mode 100644 Documentation/devicetree/bindings/soundwire/bus.txt create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

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[alsa-devel] [PATCH v3] ALSA: hda: Fix widget_mutex incomplete protection
by Evan Green 01 Jul '19

01 Jul '19

The widget_mutex was introduced to serialize callers to hda_widget_sysfs_{re}init. However, its protection of the sysfs widget array is incomplete. For example, it is acquired around the call to hda_widget_sysfs_reinit(), which actually creates the new array, but isn't still acquired when codec->num_nodes and codec->start_nid is updated. So the lock ensures one thread sets up the new array at a time, but doesn't ensure which thread's value will end up in codec->num_nodes. If a larger num_nodes wins but a smaller array was set up, the next call to refresh_widgets() will touch free memory as it iterates over codec->num_nodes that aren't there. The widget_lock really protects both the tree as well as codec->num_nodes, start_nid, and end_nid, so make sure it's held across that update. It should also be held during snd_hdac_get_sub_nodes(), so that a very old read from that function doesn't end up clobbering a later update. Fixes: ed180abba7f1 ("ALSA: hda: Fix race between creating and refreshing sysfs entries") Signed-off-by: Evan Green <evgreen(a)chromium.org> --- Changes in v3: - Moved locking back into callers (Takashi) - Combined update_widgets exit flow (Takashi) Changes in v2: - Introduced widget_mutex relocation sound/hda/hdac_device.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 6907dbefd08c..3842f9d34b7c 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -400,27 +400,33 @@ static void setup_fg_nodes(struct hdac_device *codec) int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs) { hda_nid_t start_nid; - int nums, err; + int nums, err = 0; + /* + * Serialize against multiple threads trying to update the sysfs + * widgets array. + */ + mutex_lock(&codec->widget_lock); nums = snd_hdac_get_sub_nodes(codec, codec->afg, &start_nid); if (!start_nid || nums <= 0 || nums >= 0xff) { dev_err(&codec->dev, "cannot read sub nodes for FG 0x%02x\n", codec->afg); - return -EINVAL; + err = -EINVAL; + goto unlock; } if (sysfs) { - mutex_lock(&codec->widget_lock); err = hda_widget_sysfs_reinit(codec, start_nid, nums); - mutex_unlock(&codec->widget_lock); if (err < 0) - return err; + goto unlock; } codec->num_nodes = nums; codec->start_nid = start_nid; codec->end_nid = start_nid + nums; - return 0; +unlock: + mutex_unlock(&codec->widget_lock); + return err; } EXPORT_SYMBOL_GPL(snd_hdac_refresh_widgets); -- 2.20.1

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[alsa-devel] [PATCH v2 0/2] ALSA: hda: Widget memory fixes
by Evan Green 01 Jul '19

01 Jul '19

This series fixes concurrency issues with the sysfs widget array. The first function patches up the locking that was introduced recently to protect more of the data structure. The second patch fixes a race between a reinit and the initial population of the array which could result in a length and array getting out of sync. Changes in v2: - Introduced widget_mutex relocation Evan Green (2): ALSA: hda: Fix widget_mutex incomplete protection ALSA: hda: Use correct start/count for sysfs init sound/hda/hdac_device.c | 21 ++++++++++++++------- sound/hda/hdac_sysfs.c | 18 ++++++++++-------- sound/hda/local.h | 3 ++- 3 files changed, 26 insertions(+), 16 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v2] ALSA: firewire-lib/fireworks: fix miss detection of received MIDI messages
by Takashi Sakamoto 01 Jul '19

01 Jul '19

In IEC 61883-6, 8 MIDI data streams are multiplexed into single MIDI conformant data channel. The index of stream is calculated by modulo 8 of the value of data block counter. In fireworks, the value of data block counter in CIP header has a quirk with firmware version v5.0.0, v5.7.3 and v5.8.0. This brings ALSA IEC 61883-1/6 packet streaming engine to miss detection of MIDI messages. This commit fixes the miss detection to modify the value of data block counter for the modulo calculation. For maintainers, this bug exists since a commit 18f5ed365d3f ("ALSA: fireworks/firewire-lib: add support for recent firmware quirk") in Linux kernel v4.2. There're many changes since the commit. This fix can be backported to Linux kernel v4.4 or later. I tagged a base commit to the backport for your convenience. Besides, my work for Linux kernel v5.3 brings heavy code refactoring and some structure members are renamed in 'sound/firewire/amdtp-stream.h'. The content of this patch brings conflict when merging -rc tree with this patch and the latest tree. I request maintainers to solve the conflict to replace 'tx_first_dbc' with 'ctx_data.tx.first_dbc'. Fixes: df075feefbd3 ("ALSA: firewire-lib: complete AM824 data block processing layer") Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-am824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index bebddc60fde8..99654e7eb2d4 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -388,7 +388,7 @@ static void read_midi_messages(struct amdtp_stream *s, u8 *b; for (f = 0; f < frames; f++) { - port = (s->data_block_counter + f) % 8; + port = (8 - s->tx_first_dbc + s->data_block_counter + f) % 8; b = (u8 *)&buffer[p->midi_position]; len = b[0] - 0x80; -- 2.20.1

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