October 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC] ASoC: pcm: Trigger all commands on error
by Markus Pargmann 10 Oct '13

10 Oct '13

If a start command is triggered and an error occures in the first called function, e.g. DMA, all other functions are not executed and the error value is returned immediately. In case of a start command, the calling function will call undo_start, calling this trigger function with the stop command. It will call stop for each function. But only the first function was started previously. The other functions may fail in the assumption that a stop command always comes after a start command. As the API does not specify the behaviour of trigger functionpointers, I think this should be fixed in the function calling the trigger functionpointers. This patch changes the behaviour. The trigger function calls all functionpointers independent of their returncodes. The first error-code is returned. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- sound/soc/soc-pcm.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..23fc25b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -616,26 +616,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; + int ret = 0; + int err = 0; if (codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } if (platform->driver->ops && platform->driver->ops->trigger) { ret = platform->driver->ops->trigger(substream, cmd); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } if (cpu_dai->driver->ops->trigger) { ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } - return 0; + + return err; } static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, -- 1.8.4.rc3

2 13

[alsa-devel] [PATCH] ASoC: mc13783: add mixer controls
by Steffen Trumtrar 10 Oct '13

10 Oct '13

Add more kcontrols for the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> --- sound/soc/codecs/mc13783.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05..fafe999 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -534,6 +534,30 @@ static struct snd_soc_dapm_route mc13783_routes[] = { static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", "Mono", "Mono Mix"}; +static const char * const mc13783_alsp[] = {"Off", "Codec", "Right"}; + +static const char * const mc13783_ahs[] = {"Codec", "Mixer"}; + +static const struct soc_enum mc13783_enum_asp = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 4, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + +static const struct soc_enum mc13783_enum_alsp = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 7, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + +static const struct soc_enum mc13783_enum_ahs = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 11, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_arxout = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 17, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_codec = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 4, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + static const struct soc_enum mc13783_enum_3d_mixer = SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), mc13783_3d_mixer); @@ -541,8 +565,32 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + SOC_ENUM("Earpiece Switch", mc13783_enum_asp), + SOC_ENUM("Loudspeaker Switch", mc13783_enum_alsp), + + SOC_ENUM("Headset Switch", mc13783_enum_ahs), + SOC_SINGLE("Headset Amp Right Switch", MC13783_AUDIO_RX0, 9, 1, 0), + SOC_SINGLE("Headset Amp Left Switch", MC13783_AUDIO_RX0, 10, 1, 0), + + SOC_ENUM("Line out Amp Switch", mc13783_enum_arxout), + SOC_SINGLE("Line out Amp Right Switch", MC13783_AUDIO_RX0, 15, 1, 0), + SOC_SINGLE("Line out Amp Left Switch", MC13783_AUDIO_RX0, 16, 1, 0), + + SOC_SINGLE("Codec Capture Mix Switch", MC13783_AUDIO_RX0, 21, 1, 0), + SOC_SINGLE("PCM Capture Mix Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mix Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("Codec Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("Codec Capture Switch", MC13783_AUDIO_RX1, 0, 1, 0), + + SOC_SINGLE("PGA Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("PGA Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- 1.8.4.rc3

3 5

[alsa-devel] [PATCH] ALSA: hda - Enable surround speakers (when line out is also present)
by David Henningsson 10 Oct '13

10 Oct '13

In the case where we have both line out and more than stereo speakers, the speaker DACs will end up in extra_out_nid. In fact, AFAIU, speakers are the only ones that can end up in extra_out_nid, and if we have several of those, they should be surround outputs rather than copy front. BugLink: https://bugs.launchpad.net/bugs/1236965 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) Notes: * Thanks to Raymond Yau for the pointer * Not tested (yet), except in hda-emu * Not sure if this should be sent to stable or not, given that it is a change of behaviour * Alsainfo available here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1236965/+attachm… diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5b6c4e3..68801ba 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5395,11 +5395,6 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, mout->hp_out_nid[i], stream_tag, 0, format); - for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) - if (!mout->no_share_stream && mout->extra_out_nid[i]) - snd_hda_codec_setup_stream(codec, - mout->extra_out_nid[i], - stream_tag, 0, format); /* surrounds */ for (i = 1; i < mout->num_dacs; i++) { @@ -5410,6 +5405,20 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format); } + + /* extra surrounds */ + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) { + int ch = 0; + if (!mout->extra_out_nid[i]) + break; + if (chs >= (i + 1) * 2) + ch = i * 2; + else if (!mout->no_share_stream) + break; + snd_hda_codec_setup_stream(codec, mout->extra_out_nid[i], + stream_tag, ch, format); + } + return 0; } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare); -- 1.7.9.5

2 1

Re: [alsa-devel] Cannot load snd-usb-audio on 3.12-rc2
by Takashi Iwai 10 Oct '13

10 Oct '13

At Mon, 23 Sep 2013 16:41:07 -0700, Sarah Sharp wrote: > > The snd-usb-audio driver no longer loads properly on 3.12-rc2 when I > plug in my USB headset. It worked fine on 3.11-rc4 (not sure about > vanilla 3.11). > > Trying to manually load the driver fails: > > sarah@xanatos:~$ sudo modprobe snd-usb-audio > FATAL: Error inserting snd_usb_audio > (/lib/modules/3.12.0-rc2/kernel/sound/usb/snd-usb-audio.ko): Invalid argument > > dmesg shows: > > [ 44.231668] snd_usb_audio: `-2' invalid for parameter `index' There is no corresponding code change in the USB audio driver side, so I guess it must be in the module parameter parser that got broken. Does the patch below fix? thanks, Takashi --- diff --git a/kernel/params.c b/kernel/params.c index 81c4e78..c00d5b5 100644 --- a/kernel/params.c +++ b/kernel/params.c @@ -254,11 +254,11 @@ int parse_args(const char *doing, STANDARD_PARAM_DEF(byte, unsigned char, "%hhu", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(short, short, "%hi", long, kstrtoul); +STANDARD_PARAM_DEF(short, short, "%hi", long, kstrtol); STANDARD_PARAM_DEF(ushort, unsigned short, "%hu", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(int, int, "%i", long, kstrtoul); +STANDARD_PARAM_DEF(int, int, "%i", long, kstrtol); STANDARD_PARAM_DEF(uint, unsigned int, "%u", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(long, long, "%li", long, kstrtoul); +STANDARD_PARAM_DEF(long, long, "%li", long, kstrtol); STANDARD_PARAM_DEF(ulong, unsigned long, "%lu", unsigned long, kstrtoul); int param_set_charp(const char *val, const struct kernel_param *kp)

2 2

[alsa-devel] firewire amdtp details
by Uli Franke 09 Oct '13

09 Oct '13

Hi Clemens One of the reasons why we are working on snd_dice is because we experienced (rare but unacceptable nevertheless) audio dropouts with the original driver. The DICE PLL lost lock in such cases. Investigations showed that this varies with the hardware/temperature/sun spots/snowfall. This led me to the conclusion that the device buffering/SYT-timing must be somehow close to its limits (late/early). In such a case just little node clock differences (induced by temperature and alike) can cause the system to drop out of the PLL's locking range. During capture mode SYT implementation/review I re-read the SYT code and found the following: from amdtp_stream_set_parameters: /* default buffering in the device */ s->transfer_delay = TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; if (s->flags & CIP_BLOCKING) /* additional buffering needed to adjust for no-data packets */ s->transfer_delay += TICKS_PER_SECOND * amdtp_syt_intervals[sfc] / rate; which sets a global transfer delay. But later in calculate_syt I get the impression that (not exactly) the same operations are performed again accidentally: if (syt_offset < TICKS_PER_CYCLE) { syt_offset += TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; if (s->flags & CIP_BLOCKING) syt_offset += s->transfer_delay; syt = (cycle + syt_offset / TICKS_PER_CYCLE) << 12; syt += syt_offset % TICKS_PER_CYCLE; return syt & 0xffff; } When I interpret the code from set_parameters as some sort of precomputation, the code in calculate_syt should be something like this: if (syt_offset < TICKS_PER_CYCLE) { syt_offset += s->transfer_delay; syt = (cycle + syt_offset / TICKS_PER_CYCLE) << 12; syt += syt_offset % TICKS_PER_CYCLE; return syt & 0xffff; } This could explain that the stream gets delayed too much - resulting in late timing (to be verified though). Therefore I'm a bit curious if you could enlighten me about this as s->transfer_delay isn't used in any other place. Additionally I appreciate any justifications for the subtraction of TICKS_PER_CYCLE from the global transfer delay. This would help me to get a better understanding of snd_dice's SYT mechanics. Regards Uli

2 1

[alsa-devel] [PATCH] ASoC: dapm: Use async I/O for DAPM sequences
by Mark Brown 09 Oct '13

09 Oct '13

From: Mark Brown <broonie(a)linaro.org> Within a DAPM sequence we normally don't care about when exactly a register write has completed so long as they happen in the order we requested. This means that we can issue most of the writes we do asynchronously which should maximise the ability of the underlying frameworks to keep the hardware busy, providing a small performance improvement on some systems. We currently ensure that all writes are completed both when changing to a different device and when calling into the regulator and clock frameworks. This should ensure that the previous ordering is maintained. We also ensure that writes are completed prior to calling into widget event functions since some event functions implement delays. This should be improved in future so that widgets can disable this sync in order to add extra writes. Signed-off-by: Mark Brown <broonie(a)linaro.org> xvdfvsd --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 25b9554..905e4c5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -409,6 +409,12 @@ static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) mutex_unlock(&w->platform->mutex); } +static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec && dapm->codec->using_regmap) + regmap_async_complete(dapm->codec->control_data); +} + static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { @@ -417,8 +423,9 @@ static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, int ret; if (w->codec && w->codec->using_regmap) { - ret = regmap_update_bits_check(w->codec->control_data, - reg, mask, value, &change); + ret = regmap_update_bits_check_async(w->codec->control_data, + reg, mask, value, + &change); if (ret != 0) return ret; } else { @@ -1201,6 +1208,8 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, { int ret; + soc_dapm_async_complete(w->dapm); + if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, false); @@ -1234,6 +1243,8 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, if (!w->clk) return -EIO; + soc_dapm_async_complete(w->dapm); + #ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_prepare_enable(w->clk); @@ -1426,6 +1437,7 @@ static void dapm_seq_check_event(struct snd_soc_card *card, if (w->event && (w->event_flags & event)) { pop_dbg(w->dapm->dev, card->pop_time, "pop test : %s %s\n", w->name, ev_name); + soc_dapm_async_complete(w->dapm); trace_snd_soc_dapm_widget_event_start(w, event); ret = w->event(w, NULL, event); trace_snd_soc_dapm_widget_event_done(w, event); @@ -1498,6 +1510,7 @@ static void dapm_seq_run(struct snd_soc_card *card, struct list_head *list, int event, bool power_up) { struct snd_soc_dapm_widget *w, *n; + struct snd_soc_dapm_context *d; LIST_HEAD(pending); int cur_sort = -1; int cur_subseq = -1; @@ -1528,6 +1541,9 @@ static void dapm_seq_run(struct snd_soc_card *card, cur_subseq); } + if (cur_dapm && w->dapm != cur_dapm) + soc_dapm_async_complete(cur_dapm); + INIT_LIST_HEAD(&pending); cur_sort = -1; cur_subseq = INT_MIN; @@ -1586,6 +1602,10 @@ static void dapm_seq_run(struct snd_soc_card *card, cur_dapm->seq_notifier(cur_dapm, i, cur_subseq); } + + list_for_each_entry(d, &card->dapm_list, list) { + soc_dapm_async_complete(d); + } } static void dapm_widget_update(struct snd_soc_card *card) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ASoC: dapm: Run clock and regulator events separately to other supplies
by Mark Brown 09 Oct '13

09 Oct '13

From: Mark Brown <broonie(a)linaro.org> In order to avoid trying to use an external clock or supply for an on-chip supply prior to it being enabled move the clock and regulator supply events to a separate step in DAPM sequencing from normal supply events. This should have minimal practical impact since these widgets are sorted using SND_SOC_NOPM which is a negative value and hence sorted separately to any real register writes, though it may be relevant if supplies have event callbacks only. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 50 +++++++++++++++++++++++++------------------------- 1 file changed, 25 insertions(+), 25 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 905e4c5..eb26d5a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -59,31 +59,31 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_clock_supply] = 1, - [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_supply] = 2, + [snd_soc_dapm_micbias] = 3, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai_in] = 3, - [snd_soc_dapm_dai_out] = 3, - [snd_soc_dapm_aif_in] = 3, - [snd_soc_dapm_aif_out] = 3, - [snd_soc_dapm_mic] = 4, - [snd_soc_dapm_mux] = 5, - [snd_soc_dapm_virt_mux] = 5, - [snd_soc_dapm_value_mux] = 5, - [snd_soc_dapm_dac] = 6, - [snd_soc_dapm_switch] = 7, - [snd_soc_dapm_mixer] = 7, - [snd_soc_dapm_mixer_named_ctl] = 7, - [snd_soc_dapm_pga] = 8, - [snd_soc_dapm_adc] = 9, - [snd_soc_dapm_out_drv] = 10, - [snd_soc_dapm_hp] = 10, - [snd_soc_dapm_spk] = 10, - [snd_soc_dapm_line] = 10, - [snd_soc_dapm_kcontrol] = 11, - [snd_soc_dapm_post] = 12, + [snd_soc_dapm_dai_in] = 4, + [snd_soc_dapm_dai_out] = 4, + [snd_soc_dapm_aif_in] = 4, + [snd_soc_dapm_aif_out] = 4, + [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_mux] = 6, + [snd_soc_dapm_virt_mux] = 6, + [snd_soc_dapm_value_mux] = 6, + [snd_soc_dapm_dac] = 7, + [snd_soc_dapm_switch] = 8, + [snd_soc_dapm_mixer] = 8, + [snd_soc_dapm_mixer_named_ctl] = 8, + [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_adc] = 10, + [snd_soc_dapm_out_drv] = 11, + [snd_soc_dapm_hp] = 11, + [snd_soc_dapm_spk] = 11, + [snd_soc_dapm_line] = 11, + [snd_soc_dapm_kcontrol] = 12, + [snd_soc_dapm_post] = 13, }; static int dapm_down_seq[] = { @@ -109,10 +109,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_dai_in] = 10, [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, - [snd_soc_dapm_clock_supply] = 12, - [snd_soc_dapm_regulator_supply] = 12, [snd_soc_dapm_supply] = 12, - [snd_soc_dapm_post] = 13, + [snd_soc_dapm_clock_supply] = 13, + [snd_soc_dapm_regulator_supply] = 13, + [snd_soc_dapm_post] = 14, }; static void pop_wait(u32 pop_time) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Use module_usb_driver
by Sachin Kamat 09 Oct '13

09 Oct '13

module_usb_driver makes code simpler by removing the boilerplate. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/usb/card.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9d9de8d..d979050 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -751,15 +751,4 @@ static struct usb_driver usb_audio_driver = { .supports_autosuspend = 1, }; -static int __init snd_usb_audio_init(void) -{ - return usb_register(&usb_audio_driver); -} - -static void __exit snd_usb_audio_cleanup(void) -{ - usb_deregister(&usb_audio_driver); -} - -module_init(snd_usb_audio_init); -module_exit(snd_usb_audio_cleanup); +module_usb_driver(usb_audio_driver); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add fixup for ASUS N56VZ
by Takashi Iwai 08 Oct '13

08 Oct '13

ASUS N56VZ needs a fixup for the bass speaker pin, which was already provided via model=asus-mode4. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=841645 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 52c26d3..ed9deb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- 1.8.4

1 0

Re: [alsa-devel] [Alsa-user] hdmi sound rerouted?
by Clemens Ladisch 08 Oct '13

08 Oct '13

Dagg Stompler wrote: >> daggs(a)gmx.com wrote: >>> card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] >>> card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] >>> card 2: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] >> >> The second one is called "Generic" because your kernel does not (yet) >> know the hardware. >> >>> 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Cedar HDMI Audio [Radeon HD 5400/6300 Series] >>> 06:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV635 HDMI Audio [Radeon HD 3600 Series] >> >> These are the HDMI outputs of *two* graphics cards. The first one's >> model is so new that the driver doesn't know it. > > the 3600 doesn't have hdmi socket so I think it is irrelevant for my issue. The RV635 chip does support HDMI outputs (even if none are actually connected), so the corresponding PCI device shows up here. >>> If you update the kernel, you will get two devices named "HDA ATI HDMI". > > upgrade to where? Oops. Even the latest driver lacks the PCI IDs of many HDMI HDA devices (aa50/aa58/aa60/aa68/aa80/aa88/aa90/aa98: Cypress/Juniper/Redwood/Cedar/ Cayman+Antilles/Barts/Turks/Caicos). Alex, are these devices that require disabling snooping? Regards, Clemens

4 5