July 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: mt6359: Fix refcount leak bug
by Liang He 13 Jul '22

13 Jul '22

In mt6359_parse_dt() and mt6359_accdet_parse_dt(), we should call of_node_put() for the reference returned by of_get_child_by_name() which has increased the refcount. Fixes: 683530285316 ("ASoC: mt6359: fix failed to parse DT properties") Fixes: eef07b9e0925 ("ASoC: mediatek: mt6359: add MT6359 accdet jack driver") Signed-off-by: Liang He <windhl(a)126.com> --- sound/soc/codecs/mt6359-accdet.c | 1 + sound/soc/codecs/mt6359.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index 6d3d170144a0..c190628e2905 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -675,6 +675,7 @@ static int mt6359_accdet_parse_dt(struct mt6359_accdet *priv) sizeof(struct three_key_threshold)); } + of_node_put(node); dev_warn(priv->dev, "accdet caps=%x\n", priv->caps); return 0; diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 23709b180409..c9a453ce8a2a 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -2778,6 +2778,7 @@ static int mt6359_parse_dt(struct mt6359_priv *priv) ret = of_property_read_u32(np, "mediatek,mic-type-2", &priv->mux_select[MUX_MIC_TYPE_2]); + of_node_put(np); if (ret) { dev_info(priv->dev, "%s() failed to read mic-type-2, use default (%d)\n", -- 2.25.1

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[PATCH 1/2] soundwire: bus: remove use of __func__ in dev_dbg
by Bard Liao 13 Jul '22

13 Jul '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH <gregkh(a)linuxfoundation.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 8970f8560766..2772973eebb1 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -827,13 +827,13 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, mutex_lock(&bus->bus_lock); dev_vdbg(bus->dev, - "%s: changing status slave %d status %d new status %d\n", - __func__, slave->dev_num, slave->status, status); + "changing status slave %d status %d new status %d\n", + slave->dev_num, slave->status, status); if (status == SDW_SLAVE_UNATTACHED) { dev_dbg(&slave->dev, - "%s: initializing enumeration and init completion for Slave %d\n", - __func__, slave->dev_num); + "initializing enumeration and init completion for Slave %d\n", + slave->dev_num); init_completion(&slave->enumeration_complete); init_completion(&slave->initialization_complete); @@ -841,8 +841,8 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, } else if ((status == SDW_SLAVE_ATTACHED) && (slave->status == SDW_SLAVE_UNATTACHED)) { dev_dbg(&slave->dev, - "%s: signaling enumeration completion for Slave %d\n", - __func__, slave->dev_num); + "signaling enumeration completion for Slave %d\n", + slave->dev_num); complete(&slave->enumeration_complete); } @@ -1849,8 +1849,8 @@ int sdw_handle_slave_status(struct sdw_bus *bus, "Update Slave status failed:%d\n", ret); if (attached_initializing) { dev_dbg(&slave->dev, - "%s: signaling initialization completion for Slave %d\n", - __func__, slave->dev_num); + "signaling initialization completion for Slave %d\n", + slave->dev_num); complete(&slave->initialization_complete); -- 2.25.1

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[PATCH 0/3] soundwire: allocate device_number with IDA
by Bard Liao 13 Jul '22

13 Jul '22

The device_number is currently allocated in the scope of each bus and does not need to be unique at the system level. This leads e.g. on Dell devices with three or four Device1 on different bus segments. To make the device_number unique at the system level, and unified with the HDaudio/iDISP SDI values, this PRs allocates the dev_number with an IDA restricted between 4 and 11 (inclusive). Pierre-Louis Bossart (3): soundwire: bus: rename sdw_ida as sdw_bus_ida soundwire: bus: allow device number to be unique at system level soundwire: intel: set dev_num_ida_min drivers/soundwire/bus.c | 29 ++++++++++++++++++++--------- drivers/soundwire/intel.c | 4 ++++ include/linux/soundwire/sdw.h | 4 ++++ 3 files changed, 28 insertions(+), 9 deletions(-) -- 2.25.1

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[bug report] ASoC: amd: acp: ACP code generic to support newer platforms
by Dan Carpenter 13 Jul '22

13 Jul '22

Hello V sujith kumar Reddy, This is a semi-automatic email about new static checker warnings. The patch b24484c18b10: "ASoC: amd: acp: ACP code generic to support newer platforms" from Jul 7, 2022, leads to the following Smatch complaint: sound/soc/amd/acp/acp-platform.c:99 i2s_irq_handler() warn: variable dereferenced before check 'adata' (see line 94) sound/soc/amd/acp/acp-platform.c 93 struct acp_dev_data *adata = data; 94 struct acp_resource *rsrc = adata->rsrc; ^^^^^^^^^^^ The patch adds a new dereference 95 struct acp_stream *stream; 96 u16 i2s_flag = 0; 97 u32 val, val1, i; 98 99 if (!adata) ^^^^^^ The old code assumed that "adata" could be NULL but hopefully that check can be removed? 100 return IRQ_NONE; 101 regards, dan carpenter

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snd-aloop working no more on 1.2.7.1
by GitHub issues - edited 13 Jul '22

13 Jul '22

alsa-project/alsa-lib issue #251 was edited from tramchamploo: ``` pcm.er4s { # Direct hardware access type hw # Loopback card name # # Has to match "id" in the options of the snd-aloop module card "Loopback" # Loopback device ID device 0 # Number of audio channels # # Has to match the number of channels in music player app # and in the CamillaDSP input configuration channels 2 # Format of audio stream # # Has to match the format defined in the # of the CamillaDSP input configuration format "FLOAT_LE" # Sampling-rate of audio stream # # Has to match the sampling-rate defined in the # CamillaDSP configuration rate 48000 } ctl.er4s { type hw card "Loopback" } ``` This is the .asoundrc I'm using, it worked before this update. Now aplay -L doesn't show this device. Using mpd to play to the Loopback gets `snd_pcm_hw_params() failed: Invalid argument`. Here's my mpd config: ``` audio_output { type "alsa" name "Etymotic ER4S" device "er4s" format "48000:f:2" mixer_type "none" replay_gain_handler "none" auto_resample "no" auto_format "no" } resampler { plugin "soxr" quality "custom" precision "28" phase_response "50" passband_end "91" stopband_begin "100" attenu ``` Btw, alsa should really have a detailed specification on asoundrc file. The value of the format option I specify is from a dsp project, otherwise I have no idea what to write here. And according to [https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gaa14b7f2… it's in a totally different format. Issue URL : https://github.com/alsa-project/alsa-lib/issues/251 Repository URL: https://github.com/alsa-project/alsa-lib

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snd-aloop working no more on 1.2.7.1
by GitHub issues - edited 13 Jul '22

13 Jul '22

alsa-project/alsa-lib issue #251 was edited from tramchamploo: ``` pcm.er4s { # Direct hardware access type hw # Loopback card name # # Has to match "id" in the options of the snd-aloop module card "Loopback" # Loopback device ID device 0 # Number of audio channels # # Has to match the number of channels in music player app # and in the CamillaDSP input configuration channels 2 # Format of audio stream # # Has to match the format defined in the # of the CamillaDSP input configuration format "FLOAT_LE" # Sampling-rate of audio stream # # Has to match the sampling-rate defined in the # CamillaDSP configuration rate 48000 } ctl.er4s { type hw card "Loopback" } ``` This is the .asoundrc I'm using, it worked before this update. Now aplay -L doesn't show this device. Using mpd to play to the Loopback gets `snd_pcm_hw_params() failed: Invalid argument`. Btw, alsa should really have a detailed specification on asoundrc file. The value of the format option I specify is from a dsp project, otherwise I have no idea what to write here. And according to [https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gaa14b7f2… it's in a totally different format. Issue URL : https://github.com/alsa-project/alsa-lib/issues/251 Repository URL: https://github.com/alsa-project/alsa-lib

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snd-aloop working no more on 1.2.7.1
by GitHub issues - opened 13 Jul '22

13 Jul '22

alsa-project/alsa-lib issue #251 was opened from tramchamploo: `# # Place your global alsa-lib configuration here... # # pcm.er4s { # Direct hardware access type hw # Loopback card name # # Has to match "id" in the options of the snd-aloop module card "Loopback" # Loopback device ID device 0 # Number of audio channels # # Has to match the number of channels in music player app # and in the CamillaDSP input configuration channels 2 # Format of audio stream # # Has to match the format defined in the # of the CamillaDSP input configuration format "FLOAT_LE" # Sampling-rate of audio stream # # Has to match the sampling-rate defined in the # CamillaDSP configuration rate 48000 } ctl.er4s { type hw card "Loopback" } ` This is the .asoundrc I'm using, it worked before this update. Now aplay -L doesn't show this device. Using mpd to play to the Loopback gets `snd_pcm_hw_params() failed: Invalid argument`. Btw, alsa should really have a detailed specification on asoundrc file. The value of the format option I specify is from a dsp project, otherwise I have no idea what to write here. And according to [https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gaa14b7f2… it's in a totally different format. Issue URL : https://github.com/alsa-project/alsa-lib/issues/251 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/7] ALSA: kernel doc fixes
by Takashi Iwai 13 Jul '22

13 Jul '22

Hi, this is a patch series to address minor warnings at processing kernel docs. Takashi === Takashi Iwai (7): ALSA: compress: Enable kernel doc markers for some functions ALSA: pcm: Fix missing return value comments for kernel docs ALSA: dmaengine: Fix missing return value comments for kernel docs ALSA: compress: Fix kernel doc warnings ALSA: control: Fix missing return value comments for kernel docs ALSA: memalloc: Fix missing return value comments for kernel docs ALSA: core: Fix missing return value comments for kernel docs include/sound/dmaengine_pcm.h | 2 ++ include/sound/pcm.h | 50 +++++++++++++++++++++++++-- include/uapi/sound/compress_offload.h | 2 +- include/uapi/sound/compress_params.h | 6 ++-- sound/core/compress_offload.c | 9 +++-- sound/core/control.c | 16 +++++++-- sound/core/device.c | 2 ++ sound/core/info.c | 2 ++ sound/core/init.c | 10 ++++++ sound/core/isadma.c | 3 +- sound/core/memalloc.c | 10 +++++- sound/core/pcm.c | 4 +++ sound/core/pcm_dmaengine.c | 30 ++++++++++------ sound/core/pcm_memory.c | 4 +++ sound/core/pcm_native.c | 6 ++++ sound/core/vmaster.c | 3 +- 16 files changed, 136 insertions(+), 23 deletions(-) -- 2.35.3

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[PATCH] ALSA: hda/realtek - Enable the headset-mic on a Xiaomi's laptop
by Meng Tang 13 Jul '22

13 Jul '22

The headset on this machine is not defined, after applying the quirk ALC256_FIXUP_ASUS_HEADSET_MIC, the headset-mic works well Signed-off-by: Meng Tang <tangmeng(a)uniontech.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 91cd58849ee8..383a814b8539 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9369,6 +9369,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), -- 2.20.1

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[PATCH] ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc221
by Meng Tang 13 Jul '22

13 Jul '22

On a HP 288 Pro G2 MT (X9W02AV), the front mic could not be detected. In order to get it working, the pin configuration needs to be set correctly, and the ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE fixup needs to be applied. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34139c26795f..91cd58849ee8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6901,6 +6901,7 @@ enum { ALC298_FIXUP_LENOVO_SPK_VOLUME, ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, ALC269_FIXUP_ATIV_BOOK_8, + ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE, ALC221_FIXUP_HP_MIC_NO_PRESENCE, ALC256_FIXUP_ASUS_HEADSET_MODE, ALC256_FIXUP_ASUS_MIC, @@ -7837,6 +7838,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_NO_SHUTUP }, + [ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01813030 }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC221_FIXUP_HP_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9012,6 +9023,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2b5e, "HP 288 Pro G2 MT", ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC), -- 2.20.1

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