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

[PATCH AUTOSEL 4.19 27/39] ASoC: soc-core: Fix the error return code in snd_soc_of_parse_audio_routing()
by Sasha Levin 10 Jul '21

10 Jul '21

From: Zhen Lei <thunder.leizhen(a)huawei.com> [ Upstream commit 7d3865a10b9ff2669c531d5ddd60bf46b3d48f1e ] When devm_kcalloc() fails, the error code -ENOMEM should be returned instead of -EINVAL. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> Link: https://lore.kernel.org/r/20210617103729.1918-1-thunder.leizhen@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 595fe20bbc6d..8531b490f6f6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3510,7 +3510,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, if (!routes) { dev_err(card->dev, "ASoC: Could not allocate DAPM route table\n"); - return -EINVAL; + return -ENOMEM; } for (i = 0; i < num_routes; i++) { -- 2.30.2

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[PATCH AUTOSEL 4.19 24/39] ALSA: ppc: fix error return code in snd_pmac_probe()
by Sasha Levin 10 Jul '21

10 Jul '21

From: Yang Yingliang <yangyingliang(a)huawei.com> [ Upstream commit 80b9c1be567c3c6bbe0d4b290af578e630485b5d ] If snd_pmac_tumbler_init() or snd_pmac_tumbler_post_init() fails, snd_pmac_probe() need return error code. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> Link: https://lore.kernel.org/r/20210616021121.1991502-1-yangyingliang@huawei.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/ppc/powermac.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 33c6be9fb388..7c70ba5e2540 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -90,7 +90,11 @@ static int snd_pmac_probe(struct platform_device *devptr) sprintf(card->shortname, "PowerMac %s", name_ext); sprintf(card->longname, "%s (Dev %d) Sub-frame %d", card->shortname, chip->device_id, chip->subframe); - if ( snd_pmac_tumbler_init(chip) < 0 || snd_pmac_tumbler_post_init() < 0) + err = snd_pmac_tumbler_init(chip); + if (err < 0) + goto __error; + err = snd_pmac_tumbler_post_init(); + if (err < 0) goto __error; break; case PMAC_AWACS: -- 2.30.2

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[PATCH AUTOSEL 4.19 21/39] ALSA: sb: Fix potential double-free of CSP mixer elements
by Sasha Levin 10 Jul '21

10 Jul '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c305366a37441c2ac90b08711cb6f032b43672f2 ] snd_sb_qsound_destroy() contains the calls of removing the previously created mixer controls, but it doesn't clear the pointers. As snd_sb_qsound_destroy() itself may be repeatedly called via ioctl, this could lead to double-free potentially. Fix it by clearing the struct fields properly afterwards. Link: https://lore.kernel.org/r/20210608140540.17885-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb16_csp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 2e00b64ef13b..b3eecde0b612 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -1086,10 +1086,14 @@ static void snd_sb_qsound_destroy(struct snd_sb_csp * p) card = p->chip->card; down_write(&card->controls_rwsem); - if (p->qsound_switch) + if (p->qsound_switch) { snd_ctl_remove(card, p->qsound_switch); - if (p->qsound_space) + p->qsound_switch = NULL; + } + if (p->qsound_space) { snd_ctl_remove(card, p->qsound_space); + p->qsound_space = NULL; + } up_write(&card->controls_rwsem); /* cancel pending transfer of QSound parameters */ -- 2.30.2

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[PATCH AUTOSEL 4.19 09/39] ALSA: ac97: fix PM reference leak in ac97_bus_remove()
by Sasha Levin 10 Jul '21

10 Jul '21

From: Yufen Yu <yuyufen(a)huawei.com> [ Upstream commit a38e93302ee25b2ca6f4ee76c6c974cf3637985e ] 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> Link: https://lore.kernel.org/r/20210524093811.612302-1-yuyufen@huawei.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 ca50ff444796..d8b227e6d4cf 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -523,7 +523,7 @@ 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; -- 2.30.2

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[PATCH AUTOSEL 4.19 03/39] Revert "ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro"
by Sasha Levin 10 Jul '21

10 Jul '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 5d6fb80a142b5994355ce675c517baba6089d199 ] This reverts commit 0edabdfe89581669609eaac5f6a8d0ae6fe95e7f. I've explained that optional FireWire card for d.2 is also built-in to d.2 Pro, however it's wrong. The optional card uses DM1000 ASIC and has 'Mackie DJ Mixer' in its model name of configuration ROM. On the other hand, built-in FireWire card for d.2 Pro and d.4 Pro uses OXFW971 ASIC and has 'd.Pro' in its model name according to manuals and user experiences. The former card is not the card for d.2 Pro. They are similar in appearance but different internally. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210518084557.102681-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/Kconfig | 4 ++-- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/oxfw/oxfw.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index a2ed164d80b4..4e0e320b77d8 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -37,7 +37,7 @@ config SND_OXFW * Mackie(Loud) Onyx 1640i (former model) * Mackie(Loud) Onyx Satellite * Mackie(Loud) Tapco Link.Firewire - * Mackie(Loud) d.4 pro + * Mackie(Loud) d.2 pro/d.4 pro (built-in FireWire card with OXFW971 ASIC) * Mackie(Loud) U.420/U.420d * TASCAM FireOne * Stanton Controllers & Systems 1 Deck/Mixer @@ -83,7 +83,7 @@ config SND_BEBOB * PreSonus FIREBOX/FIREPOD/FP10/Inspire1394 * BridgeCo RDAudio1/Audio5 * Mackie Onyx 1220/1620/1640 (FireWire I/O Card) - * Mackie d.2 (FireWire Option) and d.2 Pro + * Mackie d.2 (optional FireWire card with DM1000 ASIC) * Stanton FinalScratch 2 (ScratchAmp) * Tascam IF-FW/DM * Behringer XENIX UFX 1204/1604 diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 2bcfeee75853..8073360581f4 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -414,7 +414,7 @@ static const struct ieee1394_device_id bebob_id_table[] = { SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal), /* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */ SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal), - // Mackie, d.2 (Firewire option card) and d.2 Pro (the card is built-in). + // Mackie, d.2 (optional Firewire card with DM1000). SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal), /* Stanton, ScratchAmp */ SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal), diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 3c9aa797747b..59c05c5dc1cb 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -400,7 +400,7 @@ static const struct ieee1394_device_id oxfw_id_table[] = { * Onyx-i series (former models): 0x081216 * Mackie Onyx Satellite: 0x00200f * Tapco LINK.firewire 4x6: 0x000460 - * d.4 pro: Unknown + * d.2 pro/d.4 pro (built-in card): Unknown * U.420: Unknown * U.420d: Unknown */ -- 2.30.2

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[PATCH AUTOSEL 5.4 58/63] ALSA: usb-audio: scarlett2: Fix 6i6 Gen 2 line out descriptions
by Sasha Levin 10 Jul '21

10 Jul '21

From: "Geoffrey D. Bennett" <g(a)b4.vu> [ Upstream commit c712c6c0ff2d60478582e337185bcdd520a7dc2e ] There are two headphone outputs, and they map to the four analogue outputs. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> Link: https://lore.kernel.org/r/205e5e5348f08ded0cc4da5446f604d4b91db5bf.16242945… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_scarlett_gen2.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 8fd1f948a892..ddab4c77a690 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -254,10 +254,10 @@ static const struct scarlett2_device_info s6i6_gen2_info = { .pad_input_count = 2, .line_out_descrs = { - "Monitor L", - "Monitor R", - "Headphones L", - "Headphones R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", }, .ports = { -- 2.30.2

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[PATCH AUTOSEL 5.4 57/63] ALSA: hda: Add IRQ check for platform_get_irq()
by Sasha Levin 10 Jul '21

10 Jul '21

From: Jiajun Cao <jjcao20(a)fudan.edu.cn> [ Upstream commit 8c13212443230d03ff25014514ec0d53498c0912 ] The function hda_tegra_first_init() neglects to check the return value after executing platform_get_irq(). hda_tegra_first_init() should check the return value (if negative error number) for errors so as to not pass a negative value to the devm_request_irq(). Fix it by adding a check for the return value irq_id. Signed-off-by: Jiajun Cao <jjcao20(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Reviewed-by: Thierry Reding <treding(a)nvidia.com> Link: https://lore.kernel.org/r/20210622131947.94346-1-jjcao20@fudan.edu.cn Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_tegra.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index e378cb33c69d..2971b34c87c1 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -292,6 +292,9 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) const char *sname, *drv_name = "tegra-hda"; struct device_node *np = pdev->dev.of_node; + if (irq_id < 0) + return irq_id; + err = hda_tegra_init_chip(chip, pdev); if (err) return err; -- 2.30.2

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[PATCH AUTOSEL 5.4 55/63] ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters
by Sasha Levin 10 Jul '21

10 Jul '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 94efd726b947f265bd313605c9f73edec5469d65 ] Sparse throws the following warnings: sound/soc/intel/boards/kbl_da7219_max98357a.c:647:25: error: too long initializer-string for array of char(no space for nul char) Fix by using the 'mx' acronym for Maxim. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Link: https://lore.kernel.org/r/20210621194057.21711-6-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 ++-- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb6..69362eae65be 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -607,7 +607,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) static const struct platform_device_id kbl_board_ids[] = { { - .name = "kbl_da7219_max98357a", + .name = "kbl_da7219_mx98357a", .driver_data = (kernel_ulong_t)&kabylake_audio_card_da7219_m98357a, }, @@ -629,4 +629,4 @@ module_platform_driver(kabylake_audio) MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:kbl_da7219_max98357a"); +MODULE_ALIAS("platform:kbl_da7219_mx98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index e200baa11011..df7f82e55a5a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -113,7 +113,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_max98373", + .drv_name = "kbl_da7219_mx98373", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98373_codecs, -- 2.30.2

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[PATCH AUTOSEL 5.4 51/63] ALSA: usb-audio: scarlett2: Fix scarlett2_*_ctl_put() return values
by Sasha Levin 10 Jul '21

10 Jul '21

From: "Geoffrey D. Bennett" <g(a)b4.vu> [ Upstream commit c5d8e008032f3cd5f266d552732973a960b0bd4b ] Mixer control put callbacks should return 1 if the value is changed. Fix the sw_hw, level, pad, and button controls accordingly. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> Link: https://lore.kernel.org/r/20210620164645.GA9221@m.b4.vu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_scarlett_gen2.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 60545eeef28f..8fd1f948a892 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1179,6 +1179,8 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, /* Send SW/HW switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1239,6 +1241,8 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1289,6 +1293,8 @@ static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1344,6 +1350,8 @@ static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); -- 2.30.2

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[PATCH AUTOSEL 5.4 50/63] ALSA: usb-audio: scarlett2: Fix data_mutex lock
by Sasha Levin 10 Jul '21

10 Jul '21

From: "Geoffrey D. Bennett" <g(a)b4.vu> [ Upstream commit 9b5ddea9ce5a68d7d2bedcb69901ac2a86c96c7b ] The private->vol_updated flag was being checked outside of the mutex_lock/unlock() of private->data_mutex leading to the volume data being fetched twice from the device unnecessarily or old volume data being returned. Update scarlett2_*_ctl_get() and include the private->vol_updated flag check inside the critical region. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> Link: https://lore.kernel.org/r/20210620164643.GA9216@m.b4.vu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_scarlett_gen2.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index a1f1ff72be4f..60545eeef28f 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1028,11 +1028,10 @@ static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_mixer_data *private = mixer->private_data; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.integer.value[0] = private->master_vol; return 0; @@ -1046,11 +1045,10 @@ static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, struct scarlett2_mixer_data *private = mixer->private_data; int index = elem->control; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.integer.value[0] = private->vol[index]; return 0; @@ -1314,11 +1312,10 @@ static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_mixer_data *private = mixer->private_data; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; return 0; -- 2.30.2

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