August 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Galaxy Flex Book
by Mike Pozulp 14 Aug '20

14 Aug '20

The Flex Book uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug, like my Samsung Notebook. The Flex Book owner used Early Patching to confirm that this quirk fixes the bug. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2477f3ed7237..449ea64919ec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7688,6 +7688,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), -- 2.26.2

2 1

[PATCH] soundwire: SDCA: detect sdca_cascade interrupt
by Bard Liao 14 Aug '20

14 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The SoundWire 1.2 specification defines an "SDCA cascade" bit which handles a logical OR of all SDCA interrupt sources (up to 30 defined). Due to limitations of the addressing space, this bit is located in the SDW_DP0_INT register when DP0 is used, or alternatively in the DP0_SDCA_Support_INTSTAT register when DP0 is not used. To allow for both cases to be handled, this bit will be checked in the main device-level interrupt handling code. This will result in the register being read twice if DP0 is enabled, but it's not clear how to optimize this case. It's also more logical to deal with this interrupt at the device than the port level, this bit is really not DP0 specific and its location in the DP0_INTSTAT bit is only due to the lack of free space in SCP_INTSTAT_1. The SDCA_Cascade bit cannot be masked or cleared, so the interrupt handling only forwards the detection to the Slave driver, which will deal with reading the relevant SDCA status bits and clearing them. The bus driver only signals the detection. The communication with the Slave driver is based on the same interrupt callback, with only an extension to provide the status of the sdca_cascade bit. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 14 ++++++++++++++ include/linux/soundwire/sdw.h | 4 ++++ 2 files changed, 18 insertions(+) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index e6e0fb9a81b4..17c697f5ad78 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1361,6 +1361,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) int port_num, stat, ret, count = 0; unsigned long port; bool slave_notify = false; + bool sdca_cascade = false; u8 buf, buf2[2], _buf, _buf2[2]; sdw_modify_slave_status(slave, SDW_SLAVE_ALERT); @@ -1388,6 +1389,18 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) goto io_err; } + if (slave->prop.is_sdca) { + ret = sdw_read(slave, SDW_DP0_INT); + if (ret < 0) { + dev_err(slave->bus->dev, + "SDW_DP0_INT read failed:%d\n", ret); + goto io_err; + } + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + if (sdca_cascade) + slave_notify = true; + } + do { /* * Check parity, bus clash and Slave (impl defined) @@ -1453,6 +1466,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) /* Update the Slave driver */ if (slave_notify && slave->ops && slave->ops->interrupt_callback) { + slave_intr.sdca_cascade = sdca_cascade; slave_intr.control_port = clear; memcpy(slave_intr.port, &port_status, sizeof(slave_intr.port)); diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 76052f12c9f7..a51494646a2c 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -355,6 +355,7 @@ struct sdw_dpn_prop { * @dp0_prop: Data Port 0 properties * @src_dpn_prop: Source Data Port N properties * @sink_dpn_prop: Sink Data Port N properties + * @is_sdca: the Slave supports the SDCA specification */ struct sdw_slave_prop { u32 mipi_revision; @@ -376,6 +377,7 @@ struct sdw_slave_prop { struct sdw_dp0_prop *dp0_prop; struct sdw_dpn_prop *src_dpn_prop; struct sdw_dpn_prop *sink_dpn_prop; + bool is_sdca; }; /** @@ -465,10 +467,12 @@ struct sdw_slave_id { /** * struct sdw_slave_intr_status - Slave interrupt status + * @sdca_cascade: set if the Slave device reports an SDCA interrupt * @control_port: control port status * @port: data port status */ struct sdw_slave_intr_status { + bool sdca_cascade; u8 control_port; u8 port[15]; }; -- 2.17.1

1 0

[PATCH] soundwire: SDCA: add helper macro to access controls
by Bard Liao 14 Aug '20

14 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The upcoming SDCA (SoundWire Device Class Audio) specification defines a hiearchical encoding to interface with Class-defined capabilities, based on which audio function, entity, control and channel being used. The specification is not yet accessible to the general public but this information is released with explicit permission from the MIPI Board to avoid delays with SDCA support on Linux platforms. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- include/linux/soundwire/sdw_registers.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/include/linux/soundwire/sdw_registers.h b/include/linux/soundwire/sdw_registers.h index 5d3c271af7d1..3ca09a02b9fc 100644 --- a/include/linux/soundwire/sdw_registers.h +++ b/include/linux/soundwire/sdw_registers.h @@ -305,4 +305,14 @@ #define SDW_CASC_PORT_MASK_INTSTAT3 1 #define SDW_CASC_PORT_REG_OFFSET_INTSTAT3 2 +/* v1.2 device - SDCA address mapping */ +#define SDW_SDCA_CTL(fun, ent, ctl, ch) (BIT(30) | \ + (((fun) & 0x7) << 22) | \ + (((ent) & 0x40) << 15) | \ + (((ent) & 0x3f) << 7) | \ + (((ctl) & 0x30) << 15) | \ + (((ctl) & 0x0f) << 3) | \ + (((ch) & 0x38) << 12) | \ + ((ch) & 0x07)) + #endif /* __SDW_REGISTERS_H */ -- 2.17.1

1 0

[PATCH v2] ASoC: Intel: boards: eve: Fix DMIC records zero
by Harsha Priya 13 Aug '20

13 Aug '20

This patch adds a dapm route to provide early mclk/sclk for for DMIC on SSP0(rt5514) and Headset on SSP1(rt5663). The sclk rate for both codecs is different which is taken care of in the platform_clock_control().The platform has one mclk and one sclk. Two variables sclk0 and sclk1 are used for the two codecs on differnet ssp that use different clock rate. This change ensures the DMIC PCM port will return valid data Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- v1 -> v2: - Only one mclk with same rate is used, so changed to using one variable - dropping ssp_ prefix from sclk variable names to make them sound right. - removing a return statment that was not required - fixed commit message accordingly .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 146 ++++++++++++++------- 1 file changed, 97 insertions(+), 49 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index b34cf6c..155f2b4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -54,7 +54,8 @@ struct kbl_codec_private { struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; struct clk *mclk; - struct clk *sclk; + struct clk *sclk0; + struct clk *sclk1; }; enum { @@ -77,13 +78,29 @@ static const struct snd_kcontrol_new kabylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event, int ssp_num) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + struct clk *sclk; + unsigned long sclk_rate; int ret = 0; + switch (ssp_num) { + case 0: + sclk = priv->sclk0; + sclk_rate = 6144000; + break; + case 1: + sclk = priv->sclk1; + sclk_rate = 3072000; + break; + default: + dev_err(card->dev, "Invalid ssp_num %d\n", ssp_num); + return -EINVAL; + } + /* * MCLK/SCLK need to be ON early for a successful synchronization of * codec internal clock. And the clocks are turned off during @@ -91,38 +108,48 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, */ switch (event) { case SND_SOC_DAPM_PRE_PMU: - /* Enable MCLK */ ret = clk_set_rate(priv->mclk, 24000000); if (ret < 0) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); - return ret; + dev_err(card->dev, "Can't set rate for mclk for ssp%d, err: %d\n", + ssp_num, ret); + return ret; } - ret = clk_prepare_enable(priv->mclk); - if (ret < 0) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); - return ret; + if (!__clk_is_enabled(priv->mclk)) { + /* Enable MCLK */ + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk for ssp%d, err: %d\n", + ssp_num, ret); + return ret; + } } - /* Enable SCLK */ - ret = clk_set_rate(priv->sclk, 3072000); + ret = clk_set_rate(sclk, sclk_rate); if (ret < 0) { - dev_err(card->dev, "Can't set rate for sclk, err: %d\n", - ret); + dev_err(card->dev, "Can't set rate for sclk for ssp%d, err: %d\n", + ssp_num, ret); clk_disable_unprepare(priv->mclk); return ret; } - ret = clk_prepare_enable(priv->sclk); - if (ret < 0) { - dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); - clk_disable_unprepare(priv->mclk); + if (!__clk_is_enabled(sclk)) { + /* Enable SCLK */ + ret = clk_prepare_enable(sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk for ssp%d, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(priv->mclk); + return ret; + } } break; case SND_SOC_DAPM_POST_PMD: - clk_disable_unprepare(priv->mclk); - clk_disable_unprepare(priv->sclk); + if (__clk_is_enabled(priv->mclk)) + clk_disable_unprepare(priv->mclk); + + if (__clk_is_enabled(sclk)) + clk_disable_unprepare(sclk); break; default: return 0; @@ -131,6 +158,18 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int platform_clock_control_ssp0(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 0); +} + +static int platform_clock_control_ssp1(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 1); +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -139,15 +178,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_SUPPLY("Platform Clock SSP0", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp0, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Platform Clock SSP1", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp1, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ - { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "Platform Clock SSP1" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -156,7 +197,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock SSP1" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -180,6 +221,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Rx", NULL, "Right HiFi Capture" }, /* DMIC */ + { "DMIC", NULL, "Platform Clock SSP0" }, { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, { "DMIC2L", NULL, "DMIC" }, @@ -666,7 +708,7 @@ static int kabylake_set_bias_level(struct snd_soc_card *card, if (!component || strcmp(component->name, RT5514_DEV_NAME)) return 0; - if (IS_ERR(priv->mclk)) + if (IS_ERR(priv->mclk0)) return 0; /* @@ -757,6 +799,28 @@ static struct snd_soc_card kabylake_audio_card = { .late_probe = kabylake_card_late_probe, }; +static int kabylake_audio_clk_get(struct device *dev, const char *id, + struct clk **clk) +{ + int ret = 0; + + if (!clk) + return -EINVAL; + + *clk = devm_clk_get(dev, id); + if (IS_ERR(*clk)) { + ret = PTR_ERR(*clk); + if (ret == -ENOENT) { + dev_info(dev, "Failed to get %s, defer probe\n", id); + return -EPROBE_DEFER; + } + + dev_err(dev, "Failed to get %s with err:%d\n", id, ret); + } + + return ret; +} + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; @@ -777,33 +841,17 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; - ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_sclk", &ctx->sclk0); + if (ret != 0) return ret; - } - ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_mclk", &ctx->mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_sclk", &ctx->sclk1); + if (ret != 0) return ret; - } return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

5 9

[PATCH] pcm: dmix: fix access to sum-buffer in non-interleaved mixing mode
by Vijay Palaniswamy (RBEI/ECF3) 13 Aug '20

13 Aug '20

Hi All, When dmix uses non-interleaved mixing mode the offset and step width to sum_buffer was calculated by using the dmix channels instead of the slave channels. This leads to audio distortions due to frame corruption. Issue happens while doing channel wise mixing of src and destination buffers. Attached the patch to fix the offset and step width calculation based on slave channels for sum buffer. So mixed data will be saved in respective channels of sum buffer. Steps to reproduce the issue: With below configuratio, Do aplay on both device in parallel for audio distortion pcm.dmix_2_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0 1] slave { pcm "hardware" channels 2 periods 4 period_time 40000 } } pcm.dmix_1_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0] slave { pcm "hardware" channels 1 periods 4 period_time 40000 } } pcm.hardware { type hw card 0 channels 2 rate 48000 format S16_LE } Best regards Palaniswamy Vijay

2 1

pcm: dmix: fix access to sum-buffer in non-interleaved mixing mode
by GitHub pull_request - opened 13 Aug '20

13 Aug '20

alsa-project/alsa-lib pull request #78 was opened from vijaypalaniswamy: When dmix uses non-interleaved mixing mode the offset and step width to sum_buffer was calculated by using the dmix channels instead of the slave channels. This leads to audio distortions due to frame corruption. example: - With below configuratio, Do aplay on both device in parallel for audio distortion pcm.dmix_2_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0 1] slave { pcm "hardware" channels 2 periods 4 period_time 40000 } } pcm.dmix_1_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0] slave { pcm "hardware" channels 1 periods 4 period_time 40000 } } pcm.hardware { type hw card 0 channels 2 rate 48000 format S16_LE } Signed-off-by: Vijay Palaniswamy <vijay.palaniswamy(a)in.bosch.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/78 Patch URL : https://github.com/alsa-project/alsa-lib/pull/78.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb-audio: ignore broken processing/extension unit
by tom.ty89＠gmail.com 13 Aug '20

13 Aug '20

From: Tom Yan <tom.ty89(a)gmail.com> Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). --- sound/usb/mixer.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c3..e0b7174c1043 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2367,7 +2367,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2470,6 +2470,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); -- 2.28.0

1 0

[PATCH] [v2] ALSA: echoaudio: Fix potential Oops in snd_echo_resume()
by Dinghao Liu 13 Aug '20

13 Aug '20

Freeing chip on error may lead to an Oops at the next time the system goes to resume. Fix this by removing all snd_echo_free() calls on error. Fixes: 47b5d028fdce8 ("ALSA: Echoaudio - Add suspend support #2") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- Changelog: v2: - Remove snd_echo_free() calls in all error paths. --- sound/pci/echoaudio/echoaudio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 6aeb99aa2414..868f1dcd27d6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2213,7 +2213,6 @@ static int snd_echo_resume(struct device *dev) if (err < 0) { kfree(commpage_bak); dev_err(dev, "resume init_hw err=%d\n", err); - snd_echo_free(chip); return err; } @@ -2240,7 +2239,6 @@ static int snd_echo_resume(struct device *dev) if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(chip->card->dev, "cannot grab irq\n"); - snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq; -- 2.17.1

2 1

[PATCH] ASoC: intel: Fix memleak in sst_media_open
by Dinghao Liu 13 Aug '20

13 Aug '20

When power_up_sst() fails, stream needs to be freed just like when try_module_get() fails. However, current code is returning directly and ends up leaking memory. Fixes: 0121327c1a68b ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 49b9f18472bc..79fedf9e3da1 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -330,8 +330,10 @@ static int sst_media_open(struct snd_pcm_substream *substream, runtime->private_data = stream; ret_val = power_up_sst(stream); - if (ret_val < 0) + if (ret_val < 0) { + kfree(stream); return ret_val; + } /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, -- 2.17.1

3 2

Re: [PATCH] dt-bindings: Whitespace clean-ups in schema files
by Geert Uytterhoeven 13 Aug '20

13 Aug '20

Hi Rob, On Wed, Aug 12, 2020 at 10:36 PM Rob Herring <robh(a)kernel.org> wrote: > Clean-up incorrect indentation, extra spaces, long lines, and missing > EOF newline in schema files. Most of the clean-ups are for list > indentation which should always be 2 spaces more than the preceding > keyword. > > Found with yamllint (which I plan to integrate into the checks). > Signed-off-by: Rob Herring <robh(a)kernel.org> Thanks for your patch! > --- a/Documentation/devicetree/bindings/clock/renesas,cpg-clocks.yaml > +++ b/Documentation/devicetree/bindings/clock/renesas,cpg-clocks.yaml > @@ -24,9 +24,9 @@ properties: > - const: renesas,r8a7778-cpg-clocks # R-Car M1 > - const: renesas,r8a7779-cpg-clocks # R-Car H1 > - items: > - - enum: > - - renesas,r7s72100-cpg-clocks # RZ/A1H > - - const: renesas,rz-cpg-clocks # RZ/A1 > + - enum: > + - renesas,r7s72100-cpg-clocks # RZ/A1H > + - const: renesas,rz-cpg-clocks # RZ/A1 This change breaks alignment of the comments at the end of each line. > - const: renesas,sh73a0-cpg-clocks # SH-Mobile AG5 (I only checked the files I care about) If you don't update commit e0fe7fc6f2ca0781 ("dt-bindings: Whitespace clean-ups in schema files"), I can send a patch after v5.9-rc1. Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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