June 2015 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] ALSA: hdac_ext: add hdac extended controller
by Dan Carpenter 26 Jun '15

26 Jun '15

Hello Jeeja KP, The patch 0b00a5615dc4: "ALSA: hdac_ext: add hdac extended controller" from Jun 11, 2015, leads to the following static checker warning: sound/hda/ext/hdac_ext_controller.c:47 snd_hdac_ext_bus_parse_capabilities() warn: unsigned 'offset' is never less than zero. sound/hda/ext/hdac_ext_controller.c:54 snd_hdac_ext_bus_parse_capabilities() warn: unsigned 'cur_cap' is never less than zero. sound/hda/ext/hdac_ext_controller.c 38 int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *ebus) 39 { 40 unsigned int cur_cap; 41 unsigned int offset; 42 struct hdac_bus *bus = &ebus->bus; 43 unsigned int counter = 0; 44 45 offset = snd_hdac_chip_readl(bus, LLCH); 46 47 if (offset < 0) ^^^^^^^^^^ 48 return -EIO; 49 50 /* Lets walk the linked capabilities list */ 51 do { 52 cur_cap = _snd_hdac_chip_read(l, bus, offset); 53 54 if (cur_cap < 0) ^^^^^^^^^^^ 55 return -EIO; regards, dan carpenter

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Re: [alsa-devel] Need help with Behringer Firepower FCA610 profile
by Raymond Yau 26 Jun '15

26 Jun '15

>> > >> > I'd prefer a multi-purpose configuration with multiple >> sinks/sources: >> > >> > - Source 'DAW': aux1,aux2,aux3,aux4,aux5,aux6 >> > - Source 'Mic/Instrument': channel 0/1 = left/right >> > - Source 'Line': channel 2/3 = left/right >> > - Source 'S/PDIF' channel 4/5 = left/right >> > - Source 'TV' channel 4/5 = left/right >> (=S/PDIF) >> > >> > - Sink 'DAW': >> aux1,aux2,aux3,aux4,aux5,aux6,aux7,aux8,aux9,aux10 >> > - Sink 'Stereo1/Headphone A': channel 0/1 = left/right >> > - Sink 'Stereo2/Headphone B': channel 2/3 = left/right >> > - Sink 'Stereo3': channel 4/5 = left/right >> > - Sink 'Stereo4': channel 6/7 = left/right >> > - Sink 'Stereo5/SPDIF': channel 8/9 = left/right >> > - Sink 'Surround 7.1': front-left, front-right, >> rear-left, rear-right, center, lfe, side-left, side-right >> > - Sink 'Surround 7.1 -> 5.0': front-left, front-right, >> rear-left, rear-right, center, lfe -> front-left + front-right, >> side-left -> front-left + rear-left, side-right -> front-right + >> rear-right >> >> http://cgit.freedesktop.org/pulseaudio/pulseaudio/plain/src/modules/alsa/mi… >> [1] >> >> >> Pulseaudio only support different device >> >> If your driver only have simgle device for 10 channel playback and 6 >> channel capture > > > ALSA provides the following hardware-devices/plugs: > > - sysdefault:CARD=FCA610 > - front:CARD=FCA610,DEV=0 > - surround21:CARD=FCA610,DEV=0 > - surround40:CARD=FCA610,DEV=0 > - surround41:CARD=FCA610,DEV=0 > - surround50:CARD=FCA610,DEV=0 > - surround51:CARD=FCA610,DEV=0 > - surround71:CARD=FCA610,DEV=0 > - iec958:CARD=FCA610,DEV=0 > - dmix:CARD=FCA610,DEV=0 > - dsnoop:CARD=FCA610,DEV=0 > - hw:CARD=FCA610,DEV=0 > - plughw:CARD=FCA610,DEV=0 Are you sure that those devices really work ?

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[alsa-devel] [PATCH] ALSA: hda - restore the MIC FIXUP for some Dell machines
by Hui Wang 26 Jun '15

26 Jun '15

Those FIXUPs were applied to the machines through pin quirks, but recently the PCI_QUIRK makes them can't apply to the machines. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99851 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8e02cdf..78dfec9d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4522,6 +4522,8 @@ enum { ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13_GPIO6, + ALC288_FIXUP_DELL_XPS_13, + ALC288_FIXUP_DISABLE_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -5053,9 +5055,23 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC288_FIXUP_DISABLE_AAMIX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC288_FIXUP_DELL_XPS_13_GPIO6 + }, + [ALC288_FIXUP_DELL_XPS_13] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_dell_xps13, + .chained = true, + .chain_id = ALC288_FIXUP_DISABLE_AAMIX + }, [ALC292_FIXUP_DISABLE_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE }, [ALC292_FIXUP_DELL_E7X] = { .type = HDA_FIXUP_FUNC, @@ -5096,7 +5112,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC292_FIXUP_DELL_E7X), + SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.1

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[alsa-devel] [PATCH] ALSA: jack: Fix endless loop at unique index detection
by Takashi Iwai 26 Jun '15

26 Jun '15

While the commit [d0a601c278de: ALSA: jack: Fix the id uniqueness check] fixes the wrong string check, it leads to a worse result -- the loop in get_available_index() goes into an endless loop. The cause is that snd_ctl_find_id() returns the object assigned to the numid if it's set. Thus it points to the previous entry again. This patch clears the numid field for the next call properly. Reported-and-tested-by: Tomáš Pružina <pruzinat(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/ctljack.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/ctljack.c b/sound/core/ctljack.c index 9149a4aefa95..84a3cd683068 100644 --- a/sound/core/ctljack.c +++ b/sound/core/ctljack.c @@ -41,8 +41,11 @@ static int get_available_index(struct snd_card *card, const char *name) sid.iface = SNDRV_CTL_ELEM_IFACE_CARD; strlcpy(sid.name, name, sizeof(sid.name)); - while (snd_ctl_find_id(card, &sid)) + while (snd_ctl_find_id(card, &sid)) { sid.index++; + /* reset numid; otherwise snd_ctl_find_id() hits this again */ + sid.numid = 0; + } return sid.index; } -- 2.4.4

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[alsa-devel] DPCM: BE to BE cpu DAIs question‏
by srv_kaichieh.chaung 26 Jun '15

26 Jun '15

Hi, Our platform use ASoC structure. Currently, we want to refactor our driver to use DPCM structure. I followed the documentation of DPCM, but have some trouble implementing hostless BE to BE dai links. My question is: 1. Is there any code example for this(hostless BE to BE)? Since I didn't see any existing code on linux kernel. 2. From documentation, i have to assgin a snd_soc_pcm_stream for BE dais. It seems that the BE<->BE dai links can only support one spec (format/rates/ch). Or am i getting it wrong, does it support dynamically change of the spec? Thanks in advanced! -------------------------------------------------------------------------------------- I have tried some implementations, but failed. My linux version is 3.18. Illustration, PCM -----------------> BE1 ↑ -------< BE2 Code snippet, static const struct snd_soc_dapm_route machine_routes[] = { {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, FE_STNAME}, {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, BE2_CAPTURE_CPU_DAI_STNAME}, }; static const struct snd_soc_pcm_stream dai_params = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, }; static struct snd_soc_dai_link mt_soc_dai_common[] = { { .name = FE_PCM, .stream_name = FE_STNAME, .cpu_dai_name = FE_CPUDAINAME, // cannot use dummy!!!! .platform_name = FE_PCMNAME, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .init = mt_soc_audio_init3, .ops = &kc_snd_soc_ops, .dynamic = 1, .dpcm_playback = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, }, /* BE DAIs */ { .name = "BE2", .cpu_dai_name = BE2_PLAYBACK_CPU_DAI, //"snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, { .name = BE1_PLAYBACK, .cpu_dai_name = BE1_PLAYBACK_CPU_DAI, .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, 1. when only routes PCM-->BE1, If i add .params in BE1 soc dai link, when i start PCM, it will crash. and get "Unable to handle kernel NULL pointer dereference at virtual address 000000d0" Seems to crash in dpcm_be_dai_startup() If i removed .params, it will not crash, and play normally. 2. when only with BE2--> BE1, BE2 is never trigger, where did i get wrong? Thanks. ************* Email Confidentiality Notice ******************** The information contained in this e-mail message (including any attachments) may be confidential, proprietary, privileged, or otherwise exempt from disclosure under applicable laws. It is intended to be conveyed only to the designated recipient(s). Any use, dissemination, distribution, printing, retaining or copying of this e-mail (including its attachments) by unintended recipient(s) is strictly prohibited and may be unlawful. If you are not an intended recipient of this e-mail, or believe that you have received this e-mail in error, please notify the sender immediately (by replying to this e-mail), delete any and all copies of this e-mail (including any attachments) from your system, and do not disclose the content of this e-mail to any other person. Thank you!

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[alsa-devel] DPCM: BE to BE cpu DAIs question
by Chuang Eric 26 Jun '15

26 Jun '15

Hi, Our platform use ASoC structure. Currently, we want to refactor our driver to use DPCM structure.I followed the documentation of DPCM, but have some trouble implementing hostless BE to BE dai links. My question is:1. Is there any code example for this(hostless BE to BE)?Since I didn't see any existing code on linux kernel.2. From documentation, i have to assgin a snd_soc_pcm_stream for BE dais. It seems that the BE<->BE dai links can only support one spec (format/rates/ch). Or am i getting it wrong, does it support dynamically change of the spec? Thanks in advanced! -------------------------------------------------------------------------------------- I have tried some implementations, but failed. My linux version is 3.18. Illustration,PCM -----------------> BE1 ↑ -------< BE2 Code snippet, static const struct snd_soc_dapm_route machine_routes[] = { {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, FE_STNAME}, {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, BE2_CAPTURE_CPU_DAI_STNAME},}; static const struct snd_soc_pcm_stream dai_params = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2,}; static struct snd_soc_dai_link mt_soc_dai_common[] = { { .name = FE_PCM, .stream_name = FE_STNAME, .cpu_dai_name = FE_CPUDAINAME, // cannot use dummy!!!! .platform_name = FE_PCMNAME, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .init = mt_soc_audio_init3, .ops = &kc_snd_soc_ops, .dynamic = 1, .dpcm_playback = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, }, /* BE DAIs */ { .name = "BE2", .cpu_dai_name = BE2_PLAYBACK_CPU_DAI, //"snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, { .name = BE1_PLAYBACK, .cpu_dai_name = BE1_PLAYBACK_CPU_DAI, .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, 1. when only routes PCM-->BE1, If i add .params in BE1 soc dai link, when i start PCM, it will crash. and get "Unable to handle kernel NULL pointer dereference at virtual address 000000d0" Seems to crash in dpcm_be_dai_startup() If i removed .params, it will not crash, and play normally. 2. when only with BE2--> BE1, BE2 is never trigger, where did i get wrong? Thanks.

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[alsa-devel] [PATCH 1/2] ASoC: gtm601: Fix modalias
by Axel Lin 25 Jun '15

25 Jun '15

The MODULE_ALIAS needs to match the driver name to make module auto-loading work. Also move MODULE_ALIAS close to other MODULE_* macro. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/gtm601.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/gtm601.c b/sound/soc/codecs/gtm601.c index 1b77ff2..a696e3e 100644 --- a/sound/soc/codecs/gtm601.c +++ b/sound/soc/codecs/gtm601.c @@ -70,8 +70,6 @@ static int gtm601_platform_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:gtm601_codec_audio"); - #if defined(CONFIG_OF) static const struct of_device_id gtm601_codec_of_match[] = { { .compatible = "option,gtm601", }, @@ -96,3 +94,4 @@ module_platform_driver(gtm601_codec_driver); MODULE_DESCRIPTION("ASoC gtm601 driver"); MODULE_AUTHOR("Marek Belisko <marek(a)goldelico.com>"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:gtm601"); -- 2.1.0

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[alsa-devel] [PATCH] surround41/50.conf: Use chmap syntax for better flexibility
by David Henningsson 25 Jun '15

25 Jun '15

In case the hardware only supports a specific channel map, this change would allow surround41/50 to select the correct channel map and channel count in this situation. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/pcm/surround41.conf | 11 +++++------ src/conf/pcm/surround50.conf | 11 +++++------ 2 files changed, 10 insertions(+), 12 deletions(-) diff --git a/src/conf/pcm/surround41.conf b/src/conf/pcm/surround41.conf index 10e486e..2f82381 100644 --- a/src/conf/pcm/surround41.conf +++ b/src/conf/pcm/surround41.conf @@ -53,12 +53,11 @@ pcm.!surround41 { ] } } - slave.channels 6 - ttable.0.0 1 - ttable.1.1 1 - ttable.2.2 1 - ttable.3.3 1 - ttable.4.5 1 + ttable.0.FL 1 + ttable.1.FR 1 + ttable.2.RL 1 + ttable.3.RR 1 + ttable.4.LFE 1 hint { description "4.1 Surround output to Front, Rear and Subwoofer speakers" device $DEV diff --git a/src/conf/pcm/surround50.conf b/src/conf/pcm/surround50.conf index 7b7b17e..dc95c17 100644 --- a/src/conf/pcm/surround50.conf +++ b/src/conf/pcm/surround50.conf @@ -53,12 +53,11 @@ pcm.!surround50 { ] } } - slave.channels 6 - ttable.0.0 1 - ttable.1.1 1 - ttable.2.2 1 - ttable.3.3 1 - ttable.4.4 1 + ttable.0.FL 1 + ttable.1.FR 1 + ttable.2.RL 1 + ttable.3.RR 1 + ttable.4.FC 1 hint { description "5.0 Surround output to Front, Center and Rear speakers" device $DEV -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Fix Dock Headphone on Thinkpad X250 seen as a Line Out
by David Henningsson 25 Jun '15

25 Jun '15

Thinkpad X250, when attached to a dock, has two headphone outs but no line out. Make sure we don't try to turn this into one headphone and one line out (since that disables the headphone amp on the dock). Alsa-info at http://www.alsa-project.org/db/?f=36f8764e1d782397928feec715d0ef90dfddd4c1 Cc: stable(a)vger.kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 431a20b..8e02cdf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4511,6 +4511,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPT440_DOCK2, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -4960,6 +4961,12 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_HEADSET_MODE }, [ALC292_FIXUP_TPT440_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_pincfg_no_hp_to_lineout, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK2 + }, + [ALC292_FIXUP_TPT440_DOCK2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x16, 0x21211010 }, /* dock headphone */ -- 1.9.1

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[alsa-devel] [PATCH] hda: set proper caps for newer AMD hda audio in KB/KV
by Alex Deucher 25 Jun '15

25 Jun '15

Fixes audio problems on newer asics. Noticed by: Kelly Anderson <kelly(a)xilka.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a8a1e14..479ee25 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2056,6 +2056,8 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* ATI HDMI */ + { PCI_DEVICE(0x1002, 0x1308), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x7919), @@ -2064,6 +2066,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0x9840), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0xaa08), -- 1.8.3.1

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