August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: simple-scu-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-scu-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-scu-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 0bb3da7..48606c6 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -300,6 +300,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) static struct platform_driver asoc_simple_card = { .driver = { .name = "simple-scu-audio-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: audio-graph-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using audio-graph-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2f7c8cf..488c52f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -325,6 +325,7 @@ static int asoc_graph_card_remove(struct platform_device *pdev) static struct platform_driver asoc_graph_card = { .driver = { .name = "asoc-audio-graph-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_graph_of_match, }, .probe = asoc_graph_card_probe, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using audio-graph-scu-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-scu-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 06b6fcf..a967aa1 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -401,6 +401,7 @@ static int asoc_graph_card_remove(struct platform_device *pdev) static struct platform_driver asoc_graph_card = { .driver = { .name = "asoc-audio-graph-scu-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_graph_of_match, }, .probe = asoc_graph_card_probe, -- 1.9.1

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[alsa-devel] [PATCH 0/4] ASoC: Intel: Skylake: Fixes to free resources on init failure
by Subhransu S. Prusty 23 Aug '17

23 Aug '17

Some resources are not freed for ipc and dsp init failures. This results in random failures in a stress test environment. So fixed to free the required resources and also fixed the DSP core reference counting for firmware load failure. Subhransu S. Prusty (4): ASoC: Intel: Skylake: Fix to free dsp resource on ipc_init failure ASoC: Intel: Skylake: Fix to free resources for dsp_init failure ASoC: Intel: Skylake: Fix to free correct dev id in free_irq ASoC: Intel: Skylake: Fix DSP core ref count for init failure sound/soc/intel/skylake/bxt-sst.c | 4 +++- sound/soc/intel/skylake/cnl-sst.c | 4 +++- sound/soc/intel/skylake/skl-messages.c | 26 +++++++++++++++++++------- sound/soc/intel/skylake/skl-sst-dsp.c | 4 ++-- sound/soc/intel/skylake/skl-sst.c | 4 +++- sound/soc/intel/skylake/skl.c | 2 +- 6 files changed, 31 insertions(+), 13 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH][next] ASoC: tegra: remove unused variable machine, fixes build warning
by Colin King 23 Aug '17

23 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> Remove unused variable 'machine', fixes build warning: sound/soc/tegra/tegra_wm8903.c: In function 'tegra_wm8903_remove': sound/soc/tegra/tegra_wm8903.c:206:23: warning: unused variable 'machine' [-Wunused-variable] struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); ^~~~~~~ Fixes: 2a8bd83af30e ("ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/tegra/tegra_wm8903.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2384cb9cb95c..18bdae59a4df 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -203,7 +203,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); wm8903_mic_detect(codec, NULL, 0, 0); -- 2.14.1

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[alsa-devel] [PATCH] ALSA: USB-MIDI: Use common error handling code in __snd_usbmidi_create()
by SF Markus Elfring 23 Aug '17

23 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 23 Aug 2017 12:20:07 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/usb/midi.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index a35f41467237..bd9d02268724 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2435,10 +2435,8 @@ int __snd_usbmidi_create(struct snd_card *card, err = -ENXIO; break; } - if (err < 0) { - kfree(umidi); - return err; - } + if (err < 0) + goto free_midi; /* create rawmidi device */ out_ports = 0; @@ -2448,23 +2446,25 @@ int __snd_usbmidi_create(struct snd_card *card, in_ports += hweight16(endpoints[i].in_cables); } err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports); - if (err < 0) { - kfree(umidi); - return err; - } + if (err < 0) + goto free_midi; /* create endpoint/port structures */ if (quirk && quirk->type == QUIRK_MIDI_MIDIMAN) err = snd_usbmidi_create_endpoints_midiman(umidi, &endpoints[0]); else err = snd_usbmidi_create_endpoints(umidi, endpoints); - if (err < 0) { - return err; - } + if (err < 0) + goto exit; usb_autopm_get_interface_no_resume(umidi->iface); list_add_tail(&umidi->list, midi_list); return 0; + +free_midi: + kfree(umidi); +exit: + return err; } EXPORT_SYMBOL(__snd_usbmidi_create); -- 2.14.0

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[alsa-devel] [PATCH] ASoC: Add delay quirk for H650e/Jabra 550a USB headsets
by Joakim Tjernlund 23 Aug '17

23 Aug '17

These headsets reports a lot of: cannot set freq 44100 to ep 0x81 and need a small delay between sample rate settings, just like Zoom R16/24. Add both headsets to the Zoom R16/24 quirk for a 1 ms delay between control msgs. Signed-off-by: Joakim Tjernlund <joakim.tjernlund(a)infinera.com> --- sound/usb/quirks.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index eb4b9f7a571e..6f395150243f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1308,10 +1308,13 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); - /* Zoom R16/24 needs a tiny delay here, otherwise requests like - * get/set frequency return as failed despite actually succeeding. + /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, + * otherwise requests like get/set frequency return as failed despite + * actually succeeding. */ - if (chip->usb_id == USB_ID(0x1686, 0x00dd) && + if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || + chip->usb_id == USB_ID(0x046d, 0x0a46) || + chip->usb_id == USB_ID(0x0b0e, 0x0349)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(1); } -- 2.13.5

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[alsa-devel] [PATCH] ALSA: core: Use common error handling code in two functions
by SF Markus Elfring 23 Aug '17

23 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 21:01:01 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/core/pcm.c | 30 ++++++++++++++++++------------ sound/core/timer.c | 10 ++++++---- 2 files changed, 24 insertions(+), 16 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 89c7485519cb..048df9658f50 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -783,21 +783,27 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, INIT_LIST_HEAD(&pcm->list); if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); - if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { - snd_pcm_free(pcm); - return err; - } - if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count)) < 0) { - snd_pcm_free(pcm); - return err; - } - if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { - snd_pcm_free(pcm); - return err; - } + + err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, + playback_count); + if (err < 0) + goto free_pcm; + + err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count); + if (err < 0) + goto free_pcm; + + err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops); + if (err < 0) + goto free_pcm; + if (rpcm) *rpcm = pcm; return 0; + +free_pcm: + snd_pcm_free(pcm); + return err; } /** diff --git a/sound/core/timer.c b/sound/core/timer.c index a9b9a277e00c..6d73a63f6e2b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -2096,8 +2096,7 @@ static int __init alsa_timer_init(void) err = snd_timer_register_system(); if (err < 0) { pr_err("ALSA: unable to register system timer (%i)\n", err); - put_device(&timer_dev); - return err; + goto put_timer; } err = snd_register_device(SNDRV_DEVICE_TYPE_TIMER, NULL, 0, @@ -2105,12 +2104,15 @@ static int __init alsa_timer_init(void) if (err < 0) { pr_err("ALSA: unable to register timer device (%i)\n", err); snd_timer_free_all(); - put_device(&timer_dev); - return err; + goto put_timer; } snd_timer_proc_init(); return 0; + +put_timer: + put_device(&timer_dev); + return err; } static void __exit alsa_timer_exit(void) -- 2.14.0

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[alsa-devel] [sound:topic/asus-e100h-4.13 13/17] arch/x86/platform/atom/punit_atom_debug.c:230:41: error: 'punit_freeze_nb' undeclared
by kbuild test robot 23 Aug '17

23 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asus-e100h-4.13 head: b50dc59e3e31709349588be7a1ac8991e9c5f8a5 commit: 7cfa6efe32e12013183d68f991dd586814b740da [13/17] Intel Atom suspend: add debug check for S0ix blockers config: x86_64-randconfig-x008-201734 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 7cfa6efe32e12013183d68f991dd586814b740da # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): arch/x86/platform/atom/punit_atom_debug.c: In function 'punit_atom_debug_exit': >> arch/x86/platform/atom/punit_atom_debug.c:230:41: error: 'punit_freeze_nb' undeclared (first use in this function) intel_idle_freeze_notifier_unregister(&punit_freeze_nb.nb); ^~~~~~~~~~~~~~~ arch/x86/platform/atom/punit_atom_debug.c:230:41: note: each undeclared identifier is reported only once for each function it appears in vim +/punit_freeze_nb +230 arch/x86/platform/atom/punit_atom_debug.c 226 227 static void __exit punit_atom_debug_exit(void) 228 { 229 #ifdef CONFIG_PM_DEBUG > 230 intel_idle_freeze_notifier_unregister(&punit_freeze_nb.nb); 231 #endif 232 punit_dbgfs_unregister(); 233 } 234 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ALSA: hda - Add stereo mic quirk for Lenovo G50-70 (17aa:3978)
by Takashi Iwai 23 Aug '17

23 Aug '17

Lenovo G50-70 (17aa:3978) with Conexant codec chip requires the similar workaround for the inverted stereo dmic like other Lenovo models. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1020657 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8c1289963c80..a81aacf684b2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -947,6 +947,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), -- 2.14.0

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