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

[PATCH] ASoC: mediatek: mt8173-rt5650-rt5514: fix refcount leak in mt8173_rt5650_rt5514_dev_probe()
by Wang Yufen 07 Dec '22

07 Dec '22

The node returned by of_parse_phandle() with refcount incremented, of_node_put() needs be called when finish using it. So add it in the error path in mt8173_rt5650_rt5514_dev_probe(). Fixes: 0d1d7a664288 ("ASoC: mediatek: Refine mt8173 driver and change config option") Signed-off-by: Wang Yufen <wangyufen(a)huawei.com> --- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 12f40c8..f803f12 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -200,14 +200,16 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5514_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto out; } mt8173_rt5650_rt5514_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); if (!mt8173_rt5650_rt5514_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto out; } mt8173_rt5650_rt5514_codec_conf[0].dlc.of_node = mt8173_rt5650_rt5514_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node; @@ -216,6 +218,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); +out: of_node_put(platform_node); return ret; } -- 1.8.3.1

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[PATCH] ASoC: mediatek: mt8183: fix refcount leak in mt8183_mt6358_ts3a227_max98357_dev_probe()
by Wang Yufen 07 Dec '22

07 Dec '22

The node returned by of_parse_phandle() with refcount incremented, of_node_put() needs be called when finish using it. So add it in the error path in mt8183_mt6358_ts3a227_max98357_dev_probe(). Fixes: 11c0269017b2 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227") Signed-off-by: Wang Yufen <wangyufen(a)huawei.com> --- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index a860852..48c14be 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -677,8 +677,10 @@ static int mt8183_i2s2_init(struct snd_soc_pcm_runtime *rtd) } card = (struct snd_soc_card *)of_device_get_match_data(&pdev->dev); - if (!card) + if (!card) { + of_node_put(platform_node); return -EINVAL; + } card->dev = &pdev->dev; ec_codec = of_parse_phandle(pdev->dev.of_node, "mediatek,ec-codec", 0); @@ -767,8 +769,10 @@ static int mt8183_i2s2_init(struct snd_soc_pcm_runtime *rtd) } priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; + if (!priv) { + ret = -ENOMEM; + goto out; + } snd_soc_card_set_drvdata(card, priv); @@ -776,7 +780,8 @@ static int mt8183_i2s2_init(struct snd_soc_pcm_runtime *rtd) if (IS_ERR(priv->pinctrl)) { dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n", __func__); - return PTR_ERR(priv->pinctrl); + ret = PTR_ERR(priv->pinctrl); + goto out; } for (i = 0; i < PIN_STATE_MAX; i++) { @@ -809,6 +814,7 @@ static int mt8183_i2s2_init(struct snd_soc_pcm_runtime *rtd) ret = devm_snd_soc_register_card(&pdev->dev, card); +out: of_node_put(platform_node); of_node_put(ec_codec); of_node_put(hdmi_codec); -- 1.8.3.1

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[PATCH 0/3] ASoC: Intel: boards: updates for SOF boards
by Pierre-Louis Bossart 07 Dec '22

07 Dec '22

One new JasperLake configuration, core refactoring between RT1316 and RT1318 and a minor uninitialized variable corner case. Bard Liao (1): ASoC: Intel: sof_realtek_common: set ret = 0 as initial value Brent Lu (1): ASoC: Intel: sof_rt5682: add jsl_rt5682 board config Gongjun Song (1): ASoC: Intel: sof_sdw: use common helpers for all Realtek amps sound/soc/intel/boards/Makefile | 3 +- sound/soc/intel/boards/sof_realtek_common.c | 3 +- sound/soc/intel/boards/sof_rt5682.c | 6 + sound/soc/intel/boards/sof_sdw.c | 10 +- sound/soc/intel/boards/sof_sdw_common.h | 20 +- sound/soc/intel/boards/sof_sdw_rt1316.c | 239 ------------------ sound/soc/intel/boards/sof_sdw_rt1318.c | 120 --------- .../{sof_sdw_rt1308.c => sof_sdw_rt_amp.c} | 139 +++++++--- .../intel/common/soc-acpi-intel-jsl-match.c | 5 + 9 files changed, 133 insertions(+), 412 deletions(-) delete mode 100644 sound/soc/intel/boards/sof_sdw_rt1316.c delete mode 100644 sound/soc/intel/boards/sof_sdw_rt1318.c rename sound/soc/intel/boards/{sof_sdw_rt1308.c => sof_sdw_rt_amp.c} (59%) -- 2.34.1

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[PATCH] ASoC: audio-graph-card: fix refcount leak of cpu_ep in __graph_for_each_link()
by Wang Yufen 07 Dec '22

07 Dec '22

The of_get_next_child() returns a node with refcount incremented, and decrements the refcount of prev. So in the error path of the while loop, of_node_put() needs be called for cpu_ep. Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2") Signed-off-by: Wang Yufen <wangyufen(a)huawei.com> --- sound/soc/generic/audio-graph-card.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index fe7cf97..5daa824 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -485,8 +485,10 @@ static int __graph_for_each_link(struct asoc_simple_priv *priv, of_node_put(codec_ep); of_node_put(codec_port); - if (ret < 0) + if (ret < 0) { + of_node_put(cpu_ep); return ret; + } codec_port_old = codec_port; } -- 1.8.3.1

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[bug report] ALSA: hda: Interrupt servicing and BDL setup for compress streams
by Dan Carpenter 07 Dec '22

07 Dec '22

Hello Cezary Rojewski, The patch 3e9582267e3a: "ALSA: hda: Interrupt servicing and BDL setup for compress streams" from Dec 2, 2022, leads to the following Smatch static checker warning: sound/hda/hdac_stream.c:544 snd_hdac_stream_setup_periods() error: uninitialized symbol 'dmab'. sound/hda/hdac_stream.c 487 int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev) 488 { 489 struct hdac_bus *bus = azx_dev->bus; 490 struct snd_pcm_substream *substream = azx_dev->substream; 491 struct snd_compr_stream *cstream = azx_dev->cstream; 492 struct snd_pcm_runtime *runtime = NULL; 493 struct snd_dma_buffer *dmab; 494 __le32 *bdl; 495 int i, ofs, periods, period_bytes; 496 int pos_adj, pos_align; 497 498 if (substream) { 499 runtime = substream->runtime; 500 dmab = snd_pcm_get_dma_buf(substream); 501 } else if (cstream) { 502 dmab = snd_pcm_get_dma_buf(cstream); 503 } dmab is not initialized on else path. 504 505 /* reset BDL address */ 506 snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0); 507 snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0); 508 509 period_bytes = azx_dev->period_bytes; 510 periods = azx_dev->bufsize / period_bytes; 511 512 /* program the initial BDL entries */ 513 bdl = (__le32 *)azx_dev->bdl.area; 514 ofs = 0; 515 azx_dev->frags = 0; 516 517 pos_adj = bus->bdl_pos_adj; 518 if (runtime && !azx_dev->no_period_wakeup && pos_adj > 0) { 519 pos_align = pos_adj; 520 pos_adj = DIV_ROUND_UP(pos_adj * runtime->rate, 48000); 521 if (!pos_adj) 522 pos_adj = pos_align; 523 else 524 pos_adj = roundup(pos_adj, pos_align); 525 pos_adj = frames_to_bytes(runtime, pos_adj); 526 if (pos_adj >= period_bytes) { 527 dev_warn(bus->dev, "Too big adjustment %d\n", 528 pos_adj); 529 pos_adj = 0; 530 } else { 531 ofs = setup_bdle(bus, dmab, azx_dev, 532 &bdl, ofs, pos_adj, true); This is okay because runtime is set. 533 if (ofs < 0) 534 goto error; 535 } 536 } else 537 pos_adj = 0; 538 539 for (i = 0; i < periods; i++) { 540 if (i == periods - 1 && pos_adj) 541 ofs = setup_bdle(bus, dmab, azx_dev, This is okay because pos_adj is checked. 542 &bdl, ofs, period_bytes - pos_adj, 0); 543 else 544 ofs = setup_bdle(bus, dmab, azx_dev, ^^^^ Potentially uninitialized? 545 &bdl, ofs, period_bytes, 546 !azx_dev->no_period_wakeup); 547 if (ofs < 0) 548 goto error; 549 } 550 return 0; 551 552 error: 553 dev_err(bus->dev, "Too many BDL entries: buffer=%d, period=%d\n", 554 azx_dev->bufsize, period_bytes); 555 return -EINVAL; 556 } regards, dan carpenter

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[PATCH] ASoc: SOF: Fix sof-audio-pci-intel-tgl shutdown timeout during hibernation
by Zhen Ni 07 Dec '22

07 Dec '22

On Dell Latitude 3420 Notebook, sof-audio-pci-intel-tgl may fail to shutdown sporadically during hibernation as following log: [ 43.281110] PM: Image saving done [ 43.281699] PM: hibernation: Wrote 2828852 kbytes in 2.78 seconds(1017.57 MB/s) [ 43.282359] PM: SI [ 43.345156] kvm: exiting hardware virtualization [ 43.345865] auxiliary snd_sof.hda-probes.0: shutdown [ 43.346359] skl_hda_dsp_generic skl_hda_dsp_generic: shutdown [ 43.346849] skl_hda_codec hdmi ehdaudio0D2: shutdown [ 43.398204] snd_hda_codec_realtek ehdaudio0DO: shutdown [ 43.419621] dmic-codec dmic-codec: shutdown [ 43.420194] sof-audio-pci-intel-tgl 0000:00:1f.3: shutdown Call wait_xxx_timeout() to process the timeout. Signed-off-by: Zhen Ni <nizhen(a)uniontech.com> --- sound/core/init.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 5377f94eb211..9bd674d7a0fd 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -574,9 +574,10 @@ void snd_card_disconnect_sync(struct snd_card *card) } spin_lock_irq(&card->files_lock); - wait_event_lock_irq(card->remove_sleep, + wait_event_lock_irq_timeout(card->remove_sleep, list_empty(&card->files_list), - card->files_lock); + card->files_lock, + msecs_to_jiffies(2000)); spin_unlock_irq(&card->files_lock); } EXPORT_SYMBOL_GPL(snd_card_disconnect_sync); @@ -659,7 +660,7 @@ int snd_card_free(struct snd_card *card) if (ret) return ret; /* wait, until all devices are ready for the free operation */ - wait_for_completion(&released); + wait_for_completion_timeout(&released, msecs_to_jiffies(2000)) return 0; } -- 2.20.1

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[PATCH 0/4] Make QMI message rules const
by Jeff Johnson 07 Dec '22

07 Dec '22

Change ff6d365898d ("soc: qcom: qmi: use const for struct qmi_elem_info") allows QMI message encoding/decoding rules to be const. So now update the definitions in the various client to take advantage of this. Patches for ath10k and ath11k were perviously sent separately. This series depends upon: https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux.git/commit/?h=fo… This is in the for-next banch of: git.kernel.org/pub/scm/linux/kernel/git/qcom/linux.git Hence this series is also based upon that tree/branch. Jeff Johnson (4): net: ipa: Make QMI message rules const remoteproc: sysmon: Make QMI message rules const slimbus: qcom-ngd-ctrl: Make QMI message rules const soc: qcom: pdr: Make QMI message rules const drivers/net/ipa/ipa_qmi_msg.c | 20 ++++++++++---------- drivers/net/ipa/ipa_qmi_msg.h | 20 ++++++++++---------- drivers/remoteproc/qcom_sysmon.c | 8 ++++---- drivers/slimbus/qcom-ngd-ctrl.c | 8 ++++---- drivers/soc/qcom/pdr_internal.h | 20 ++++++++++---------- 5 files changed, 38 insertions(+), 38 deletions(-) -- 2.37.0

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[PATCH] ALSA: hda/realtek: Add quirk for Lenovo TianYi510Pro-14IOB
by edward-p 07 Dec '22

07 Dec '22

Lenovo TianYi510Pro-14IOB (17aa:3742) require quirk for enabling headset-mic Signed-off-by: edward-p <edward(a)edward-p.xyz> Link: https://bugzilla.kernel.org/show_bug.cgi?id=216756 --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5c036385666..a2e1bdc06468 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10960,6 +10960,17 @@ static void alc897_fixup_lenovo_headset_mic(struct hda_codec *codec, } } +static void alc897_fixup_lenovo_headset_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_automute_hook = alc897_hp_automute_hook; + } +} + static const struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -11043,6 +11054,8 @@ enum { ALC897_FIXUP_LENOVO_HEADSET_MIC, ALC897_FIXUP_HEADSET_MIC_PIN, ALC897_FIXUP_HP_HSMIC_VERB, + ALC897_FIXUP_LENOVO_HEADSET_MODE, + ALC897_FIXUP_HEADSET_MIC_PIN2, }; static const struct hda_fixup alc662_fixups[] = { @@ -11469,6 +11482,19 @@ static const struct hda_fixup alc662_fixups[] = { { } }, }, + [ALC897_FIXUP_LENOVO_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc897_fixup_lenovo_headset_mode, + }, + [ALC897_FIXUP_HEADSET_MIC_PIN2] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a11140 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -11521,6 +11547,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x3742, "Lenovo TianYi510Pro-14IOB", ALC897_FIXUP_HEADSET_MIC_PIN2), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO), -- 2.38.1

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topology: plugins: nhlt: fix ssp debug
by GitHub pull_request - opened 07 Dec '22

07 Dec '22

alsa-project/alsa-utils pull request #180 was opened from juimonen: Nhlt ssp plugin debug feature was broken due to recent change in ssp indexing, thus fix it. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/180 Patch URL : https://github.com/alsa-project/alsa-utils/pull/180.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v3 -next] ALSA:usb-audio:add the information of KT0206 device driven by USB audio
by Dicheng Wang 07 Dec '22

07 Dec '22

From: wangdicheng <wangdicheng(a)kylinos.cn> Add relevant information to the quirks-table.h file. The test passes and the sound source file plays normally. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- v3:add body information v2:use USB_DEVICE_VENDOR_SPEC() suggested by Takashi Iwai sound/usb/quirks-table.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 874fcf245747..271884e35003 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -76,6 +76,8 @@ { USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f0a) }, /* E-Mu 0204 USB */ { USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f19) }, +/* Ktmicro Usb_audio device */ +{ USB_DEVICE_VENDOR_SPEC(0x31b2, 0x0011) }, /* * Creative Technology, Ltd Live! Cam Sync HD [VF0770] -- 2.25.1

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