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

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[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

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[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

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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)

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[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

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[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

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[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

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[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

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[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

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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

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[alsa-devel] ALSA: snd-usb-usx2y: remove bogus frame checks
by Guido Aulisi 08 Oct '13

08 Oct '13

Hi, I did some more testing on my US122: I used my patch (see below) and it worked (with default nrpack=4), but I got "should not be here with counts=42" errors, then after some time the kernel crashed, but I couldn't copy the trace. Then I set nrpack=1 and I got no "should not be here..." and it was more stable, no crash. I only checked playback, not recording, nor MIDI, with aplay I played some songs for about one hour... I hope this can be useful. But I think there are other bugs in this driver. Ciao Guido Aulisi The patch is: diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..cd16fcc 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -328,7 +328,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) + if (likely((urb->start_frame & 0x03FF) == (usX2Y->wait_iso_frame & 0x03FF))) subs->completed_urb = urb; else { usX2Y_error_sequence(usX2Y, subs, urb);

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[alsa-devel] Can't get capturing from RME Hammerfall MADI card to work: sound loops & silcence
by Niels Laukens 08 Oct '13

08 Oct '13

Hi, I'm trying to capture audio from an RME Hammerfall MADI card (PCI ID 10ee:3fc6). The setup previously worked fine on Windows, so I'm currently assuming that the card and the input signal is OK. I wrote a very simple ALSA-application that simply opens the device, sets the access format to NONINTERLEAVED (which seems to be the only option the card likes), selects 32bit sampling (although the audio is 24bit, ALSA rejects this option), 48kHz sampling and 64 channels (also, the only option the card accepts). Next, it loops with snd_pcm_readn() to get 100ms worth of audio, and dumps channel 10 to file. (Full source in attachement) The result is strange for various reasons: * The second half of every buffer returned by snd_pcm_readn() is always zero's. * The call seems to return way to fast: I would expect to get 48000 frames/second, but judging from the timing of the snd_pcm_readn() call, it's more like 1536000 frames/second (i.e. 32 fold faster). * When playing back the resulting file, I do recognize some music, but it sounds like it's the same part over and over again. Also, restarting the capture program minutes later, returns the same (at least as far as my ears & brain can recall) fragment of audio. What am I doing wrong here? (Samples of the PCM-file are available upon request) I've also tried a related test with `arecord`, with similar but different results: * The audio was also interspersed with silence * The capture duration did not match the requested '-d' option: the recording was a factor of 2 shorter than requested. * There was more audio in the recording: if I manually cut out the silent parts, the audio sounded reasonable. What am I missing here? Is this the right place to ask? Or should this go to the RME forum? Thanks in advance, Niels

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[alsa-devel] Audio Miniconference 2013 schedule - Edinburgh 21st October
by Mark Brown 07 Oct '13

07 Oct '13

The audio mini-summit in Edinburgh is now less than a month away so it seems like a good time to start pulling together a schedule. Looking at the topics people mentioned when filling out the registration form there were three major topics that people wanted to cover: - Compressed audio review: tinycompress, integration into userspace, driver upstreaming. - UCM and other use case management status & review. - Effects offload to DSPs with dynamically instantiatable algorithms, and general dynamic DSP firmware handling. Plus a few things that only one or two people brought up: - Process stuff (the list, bug tracking and so on). - Some relatively specific hardware support topics: - HDMI hotplug handling throughout the stack - Mic status LED handling throughout the stack. - Code cleanups: - Moving away from the custom ALSA device registration infrastructure. - General code modernisation. - General routing control in userspace - UIs, automation. The first question is if there is anything important that is missing from the above list - please speak up soon if there is! It'd also be good to have volunteers to lead the discussions in each area. In terms of format I'd propose that we go through the list above in roughly that order, having someone lead discussion on the topic. Does this sound good to people? As previously mentioned it'd be good if anyone planning to attend could fill in the form at: https://docs.google.com/forms/d/105uav9tkwwkDU4P1T3IOWjfsa3kgtDbFAJOlHgBQR5… to help with planning & coordination.

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[alsa-devel] [PATCH v2] ALSA: snd-usb-usx2y: remove bogus frame checks
by Daniel Mack 07 Oct '13

07 Oct '13

The frame check in i_usX2Y_urb_complete() and i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This patch removes the check code entirely. Cc: fzu(a)wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 22 +++------------------- sound/usb/usx2y/usx2yhwdeppcm.c | 7 +------ 2 files changed, 4 insertions(+), 25 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..6234a51 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index f2a1acd..814d0e8 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + subs->completed_urb = urb; capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]; capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Connect 'Left Line1R Mux' and 'Right Line1L Mux'
by Peter Ujfalusi 07 Oct '13

07 Oct '13

The two paths were not connected in the DAPM route causing the associated routes to be non working and the following warnings printed in the logs: tlv320aic3x-codec 0-001b: ASoC: mux Right Line1L Mux has no paths tlv320aic3x-codec 0-001b: ASoC: mux Left Line1R Mux has no paths Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 892c108..a160b7d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -678,6 +678,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, {"Left Line1L Mux", "differential", "LINE1L"}, + {"Left Line1R Mux", "single-ended", "LINE1R"}, + {"Left Line1R Mux", "differential", "LINE1R"}, {"Left Line2L Mux", "single-ended", "LINE2L"}, {"Left Line2L Mux", "differential", "LINE2L"}, @@ -694,6 +696,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, {"Right Line1R Mux", "differential", "LINE1R"}, + {"Right Line1L Mux", "single-ended", "LINE1L"}, + {"Right Line1L Mux", "differential", "LINE1L"}, {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, -- 1.8.3.2

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Re: [alsa-devel] [PATCH] sound: usb: support wireless devices in snd_usb_parse_datainterval
by Takashi Iwai 07 Oct '13

07 Oct '13

At Tue, 1 Oct 2013 14:32:57 -0500, Thomas Pugliese wrote: > > Hi, > This patch adds support for dev speed USB_SPEED_WIRELESS in > snd_usb_parse_datainterval which allows the usb sound core to create > ISO urbs with the correct number and size of buffers. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Thanks, applied this one, too. Takashi > --- > sound/usb/helper.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/helper.c b/sound/usb/helper.c > index 6209024..51ed1ac 100644 > --- a/sound/usb/helper.c > +++ b/sound/usb/helper.c > @@ -118,6 +118,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, > { > switch (snd_usb_get_speed(chip->dev)) { > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > if (get_endpoint(alts, 0)->bInterval >= 1 && > get_endpoint(alts, 0)->bInterval <= 4) > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-usb" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

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Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Takashi Iwai 07 Oct '13

07 Oct '13

At Tue, 1 Oct 2013 14:32:15 -0500, Thomas Pugliese wrote: > > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Thanks, applied now. (BTW, please add the maintainers explicitly to Cc at the next time.) Takashi > --- > sound/usb/card.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 64952e2..ef76aa0 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -349,6 +349,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, > case USB_SPEED_LOW: > case USB_SPEED_FULL: > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > break; > default: > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-usb" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

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[alsa-devel] [PATCH 1/3] ASoC: dapm: Add support for virtual mixer controls
by Lars-Peter Clausen 07 Oct '13

07 Oct '13

This patch adds support for virtual DAPM mixer controls. They are similar to virtual DAPM enums. There is no hardware register backing the control, so changing the control's value wont have any direct effect on the hardware. But it still influences the DAPM graph by causing the path it sits on to be connected or disconnected. This in turn can cause power changes for some of the widgets on the DAPM graph, which will then modify the hardware state. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 4 ++++ include/sound/soc.h | 3 ++- sound/soc/soc-dapm.c | 45 +++++++++++++++++++++++++++------------------ 3 files changed, 33 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 27a72d5..2037c45 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -286,6 +286,8 @@ struct device; .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } +#define SOC_DAPM_SINGLE_VIRT(xname, max) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0) #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -300,6 +302,8 @@ struct device; .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/include/sound/soc.h b/include/sound/soc.h index 6dd329a..af1678c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1049,7 +1049,8 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift; + int reg, rreg; + unsigned int shift, rshift; unsigned int invert:1; unsigned int autodisable:1; }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c..25b9554 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -499,18 +499,22 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrol_news[i].private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = soc_widget_read(w, reg); - val = (val >> shift) & mask; - if (invert) - val = max - val; + if (reg != SND_SOC_NOPM) { + val = soc_widget_read(w, reg); + val = (val >> shift) & mask; + if (invert) + val = max - val; + p->connect = !!val; + } else { + p->connect = 0; + } - p->connect = !!val; } break; case snd_soc_dapm_mux: { @@ -2791,7 +2795,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2804,7 +2808,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, kcontrol->id.name); mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (dapm_kcontrol_is_powered(kcontrol)) + if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) val = (snd_soc_read(codec, reg) >> shift) & mask; else val = dapm_kcontrol_get_value(kcontrol); @@ -2835,7 +2839,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2857,19 +2861,24 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm_kcontrol_set_value(kcontrol, val); + change = dapm_kcontrol_set_value(kcontrol, val); - mask = mask << shift; - val = val << shift; + if (reg != SND_SOC_NOPM) { + mask = mask << shift; + val = val << shift; + + change = snd_soc_test_bits(codec, reg, mask, val); + } - change = snd_soc_test_bits(codec, reg, mask, val); if (change) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + if (reg != SND_SOC_NOPM) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - card->update = &update; + card->update = &update; + } soc_dapm_mixer_update_power(card, kcontrol, connect); -- 1.8.0

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[alsa-devel] [PATCH 0/7] ALSA: hda - hdmi: Various channel mapping fixes
by Anssi Hannula 07 Oct '13

07 Oct '13

Hi! Apparently there were several more bugs in the HDMI channel mapping code, so here you go. I haven't marked any for stable since these don't really affect the regular layouts that much and the patches aren't that trivial. If you want to pick some for stable, my suggestions are 1, 5, 6, maybe 3 (those I initially marked for stable until I changed my mind). Anssi Hannula (7): ALSA: hda - hdmi: Fix reported channel map on common default layouts ALSA: hda - hdmi: Fix incorrect default channel mapping for unusual CAs ALSA: hda - hdmi: Fix programmed active channel count ALSA: hda - hdmi: Fix unused slots being enabled in manual and non-PCM mappings ALSA: hda - hdmi: Fix channel maps with less common speakers ALSA: hda - hdmi: Fix available channel maps missing from TLV ALSA: hda - hdmi: Tweak debug messages to be more useful sound/pci/hda/patch_hdmi.c | 146 +++++++++++++++++++++-------------- 1 file changed, 90 insertions(+), 56 deletions(-) -- Anssi Hannula

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[alsa-devel] [PATCH] ASoC: dapm: Ignore VMID widgets for target bias
by Lars-Peter Clausen 07 Oct '13

07 Oct '13

VMID widgets behave very similar to signal generator widgets. Both are always considered to be powered up. This means that we need to ignore the VMID widgets in the same way as signal generator widgets when calculating the DAPM context's target bias level. Otherwise the presence of a VMID widget, regardless whether it is on an active path or not, will cause the DAPM context to be powered up. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c..177f8a1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1840,6 +1840,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) */ switch (w->id) { case snd_soc_dapm_siggen: + case snd_soc_dapm_vmid: break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: -- 1.8.0

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[alsa-devel] [PATCH 0/2] ASoC: fix sound on mx31moboard
by Philippe Rétornaz 07 Oct '13

07 Oct '13

Hello Theses two patches are necessary to get back the sound working on mx31moboards. The first one fix imx-ssi to work even if only the dma is available. The second one fix the audmux settings on mx31moboard boards. Thanks, Philippe Philippe Rétornaz (2): ASoC: fsl: imx-ssi: fix probe on imx31 ASoC: fsl: Fix sound on mx31moboard sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-ssi.c | 23 ++++++++++++----------- sound/soc/fsl/imx-ssi.h | 2 ++ 3 files changed, 15 insertions(+), 12 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH 3/4] ESI W192M: Add sampling rate control.
by klem.dev＠gmail.com 07 Oct '13

07 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add sampling rate control for adc/dac in Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 83 insertions(+) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index d559916..cf2b6a5 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <linux/slab.h> #include <sound/tlv.h> #include "ice1712.h" @@ -69,6 +70,37 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned char *change) +{ + unsigned char new, old; + int idx; + + /*stac9460 1*/ + for (idx = 0; idx < 7; idx++) { + if (change[idx]) { + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx] = (new != old); + if (change[idx]) + stac9460_put(ice, idx, new); + } + } + + /*stac9460 2*/ + for (idx = 0; idx < 3; idx++) { + if (change[idx + 7]) { + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx + 7] = (new != old); + if (change[idx + 7]) + stac9460_2_put(ice, idx, new); + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, @@ -97,10 +129,13 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct mutex *mute_mutex = ice->spec; unsigned char new, old; int id, idx; int change; + mutex_lock(mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; old = stac9460_get(ice, idx); @@ -128,6 +163,9 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, stac9460_2_put(ice, idx - 6, new); } } + + mutex_unlock(mute_mutex); + return change; } @@ -381,6 +419,41 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned char changed[10]; + struct mutex *mute_mutex = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(mute_mutex); + /* we have to remember current mute status for each DAC */ + stac9460_dac_mute_all(ice, 0, changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, changed); + mutex_unlock(mute_mutex); +} + + static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -479,9 +552,11 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct mutex *mute_mutex; /*WTM 192M*/ ice->num_total_dacs = 8; @@ -489,12 +564,20 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ + mute_mutex = kzalloc(sizeof(*mute_mutex), GFP_KERNEL); + if (!mute_mutex) + return -ENOMEM; + ice->spec = mute_mutex; + mutex_init(mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/4] ESI W192M: Update ESI W192M driver.
by klem.dev＠gmail.com 07 Oct '13

07 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Update Waveterminal 192M driver to support alsa changes that happened since last driver commit. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 37 ++++++++++++++++++++++--------------- 1 file changed, 22 insertions(+), 15 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..b6b650c 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -473,9 +473,10 @@ static int wtm_init(struct snd_ice1712 *ice) ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); @@ -485,19 +486,25 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x67, /* 49MHz crystal, mpu401, + * spdif-in+ 1 stereo ADC, + * 4 stereo DACs + */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /* out-en, out-int, spdif-in */ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, /* GPIO20: 0 = CD drive dig. input + * passthrough, + * 1 = SPDIF-OUT from ice1724 + */ }; -- 1.8.4.rc3

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[alsa-devel] [PATCH] ALSA: hda - Fix mono speakers and headset mic on Dell Vostro 5470
by David Henningsson 07 Oct '13

07 Oct '13

On this machine, DAC on node 0x03 seems to give mono output. Also, it needs additional patches for headset mic support. It supports CTIA style headsets only. Alsa-info available at the bug link below. Cc: stable(a)kernel.org (v3.10+) BugLink: https://bugs.launchpad.net/bugs/1236228 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- 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 0e303b9..52c26d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3496,6 +3496,15 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, } } +static void alc290_fixup_mono_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + /* Remove DAC node 0x03, as it seems to be + giving mono output */ + snd_hda_override_wcaps(codec, 0x03, 0); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3522,6 +3531,7 @@ enum { ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -3535,6 +3545,7 @@ enum { ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, + ALC290_FIXUP_MONO_SPEAKERS, }; static const struct hda_fixup alc269_fixups[] = { @@ -3712,6 +3723,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC269_FIXUP_DELL3_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -3804,6 +3824,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC290_FIXUP_MONO_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc290_fixup_mono_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3845,6 +3871,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.7.9.5

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[alsa-devel] [PATCH]: remove unnecessary/obsolete compat/hsearch_r.c
by Waldemar Brodkorb 07 Oct '13

07 Oct '13

The usage of hsearch functions where removed long time ago in commit 273d115de05574251bdd661747ecb68449a5cf1d. This patch highly increases portability for non-glibc systems. Remove the complete compat directory as requested by Takashi Iwai. Signed-off-by: Waldemar Brodkorb <wbx(a)openadk.org> --- configure.in | 4 +- src/Makefile.am | 7 +- src/compat/Makefile.am | 8 -- src/compat/hsearch_r.c | 236 ------------------------------------------------ 4 files changed, 3 insertions(+), 252 deletions(-) delete mode 100644 src/compat/Makefile.am delete mode 100644 src/compat/empty.c delete mode 100644 src/compat/hsearch_r.c diff --git a/configure.in b/configure.in index 00fff2b..35fd89f 100644 --- a/configure.in +++ b/configure.in @@ -64,8 +64,6 @@ AC_HEADER_TIME dnl Checks for library functions. AC_PROG_GCC_TRADITIONAL -AC_CHECK_FUNC([hsearch_r], [HAVE_HSEARCH_R=yes]) -AM_CONDITIONAL([ALSA_HSEARCH_R], [test "x$HAVE_HSEARCH_R" != xyes]) AC_CHECK_FUNCS([uselocale]) SAVE_LIBRARY_VERSION @@ -659,7 +657,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/pcm/Makefile src/pcm/scopes/Makefile \ src/rawmidi/Makefile src/timer/Makefile \ src/hwdep/Makefile src/seq/Makefile src/ucm/Makefile \ - src/compat/Makefile src/alisp/Makefile \ + src/alisp/Makefile \ src/conf/Makefile src/conf/alsa.conf.d/Makefile \ src/conf/cards/Makefile \ src/conf/pcm/Makefile \ diff --git a/src/Makefile.am b/src/Makefile.am index 08f482a..8f789fe 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -46,8 +46,8 @@ if BUILD_ALISP SUBDIRS += alisp libasound_la_LIBADD += alisp/libalisp.la endif -SUBDIRS += compat conf -libasound_la_LIBADD += compat/libcompat.la @ALSA_DEPLIBS@ +SUBDIRS += conf +libasound_la_LIBADD += @ALSA_DEPLIBS@ libasound_la_LDFLAGS = -version-info $(COMPATNUM) $(VSYMS) $(SYMFUNCS) $(LDFLAGS_NOUNDEFINED) @@ -84,9 +84,6 @@ ucm/libucm.la: instr/libinstr.la: $(MAKE) -C instr libinstr.la -compat/libcompat.la: - $(MAKE) -C compat libcompat.la - alisp/libalisp.la: $(MAKE) -C alisp libalisp.la diff --git a/src/compat/Makefile.am b/src/compat/Makefile.am deleted file mode 100644 index 01f54fc..0000000 --- a/src/compat/Makefile.am +++ /dev/null @@ -1,8 +0,0 @@ -noinst_LTLIBRARIES = libcompat.la -EXTRA_libcompat_la_SOURCES = hsearch_r.c - -if ALSA_HSEARCH_R -libcompat_la_SOURCES = empty.c hsearch_r.c -else -libcompat_la_SOURCES = empty.c -endif diff --git a/src/compat/empty.c b/src/compat/empty.c deleted file mode 100644 index e69de29..0000000 diff --git a/src/compat/hsearch_r.c b/src/compat/hsearch_r.c deleted file mode 100644 index 96ceac1..0000000 --- a/src/compat/hsearch_r.c +++ /dev/null @@ -1,236 +0,0 @@ -/* Copyright (C) 1993, 1995, 1996, 1997 Free Software Foundation, Inc. - This file is part of the GNU C Library. - Contributed by Ulrich Drepper <drepper(a)gnu.ai.mit.edu>, 1993. - - The GNU C Library is free software; you can redistribute it and/or - modify it under the terms of the GNU Lesser General Public License as - published by the Free Software Foundation; either version 2.1 of the - License, or (at your option) any later version. - - The GNU C Library is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - Lesser General Public License for more details. - - You should have received a copy of the GNU Lesser General Public - License along with the GNU C Library; see the file COPYING.LIB. If not, - write to the Free Software Foundation, Inc., 59 Temple Place - Suite 330, - Boston, MA 02111-1307, USA. */ - -#include <errno.h> -#include <malloc.h> -#include <string.h> - -#define __USE_GNU -#ifndef __set_errno -#define __set_errno(e) errno = (e) -#endif -#include <search.h> - -/* [Aho,Sethi,Ullman] Compilers: Principles, Techniques and Tools, 1986 - [Knuth] The Art of Computer Programming, part 3 (6.4) */ - - -/* The reentrant version has no static variables to maintain the state. - Instead the interface of all functions is extended to take an argument - which describes the current status. */ -typedef struct _ENTRY -{ - unsigned int used; - ENTRY entry; -} -_ENTRY; - - -/* For the used double hash method the table size has to be a prime. To - correct the user given table size we need a prime test. This trivial - algorithm is adequate because - a) the code is (most probably) called a few times per program run and - b) the number is small because the table must fit in the core */ -static int -isprime (unsigned int number) -{ - /* no even number will be passed */ - unsigned int div = 3; - - while (div * div < number && number % div != 0) - div += 2; - - return number % div != 0; -} - - -/* Before using the hash table we must allocate memory for it. - Test for an existing table are done. We allocate one element - more as the found prime number says. This is done for more effective - indexing as explained in the comment for the hsearch function. - The contents of the table is zeroed, especially the field used - becomes zero. */ -int -hcreate_r (nel, htab) - size_t nel; - struct hsearch_data *htab; -{ - /* Test for correct arguments. */ - if (htab == NULL) - { - __set_errno (EINVAL); - return 0; - } - - /* There is still another table active. Return with error. */ - if (htab->table != NULL) - return 0; - - /* Change nel to the first prime number not smaller as nel. */ - nel |= 1; /* make odd */ - while (!isprime (nel)) - nel += 2; - - htab->size = nel; - htab->filled = 0; - - /* allocate memory and zero out */ - htab->table = (_ENTRY *) calloc (htab->size + 1, sizeof (_ENTRY)); - if (htab->table == NULL) - return 0; - - /* everything went alright */ - return 1; -} - - -/* After using the hash table it has to be destroyed. The used memory can - be freed and the local static variable can be marked as not used. */ -void -hdestroy_r (htab) - struct hsearch_data *htab; -{ - /* Test for correct arguments. */ - if (htab == NULL) - { - __set_errno (EINVAL); - return; - } - - if (htab->table != NULL) - /* free used memory */ - free (htab->table); - - /* the sign for an existing table is an value != NULL in htable */ - htab->table = NULL; -} - - -/* This is the search function. It uses double hashing with open addressing. - The argument item.key has to be a pointer to an zero terminated, most - probably strings of chars. The function for generating a number of the - strings is simple but fast. It can be replaced by a more complex function - like ajw (see [Aho,Sethi,Ullman]) if the needs are shown. - - We use an trick to speed up the lookup. The table is created by hcreate - with one more element available. This enables us to use the index zero - special. This index will never be used because we store the first hash - index in the field used where zero means not used. Every other value - means used. The used field can be used as a first fast comparison for - equality of the stored and the parameter value. This helps to prevent - unnecessary expensive calls of strcmp. */ -int -hsearch_r (item, action, retval, htab) - ENTRY item; - ACTION action; - ENTRY **retval; - struct hsearch_data *htab; -{ - unsigned int hval; - unsigned int count; - unsigned int len = strlen (item.key); - unsigned int idx; - - /* Compute an value for the given string. Perhaps use a better method. */ - hval = len; - count = len; - while (count-- > 0) - { - hval <<= 4; - hval += item.key[count]; - } - - /* First hash function: simply take the modulo but prevent zero. */ - hval %= htab->size; - if (hval == 0) - ++hval; - - /* The first index tried. */ - idx = hval; - - if (htab->table[idx].used) - { - /* Further action might be required according to the action value. */ - unsigned hval2; - - if (htab->table[idx].used == hval - && strcmp (item.key, htab->table[idx].entry.key) == 0) - { - if (action == ENTER) - htab->table[idx].entry.data = item.data; - - *retval = &htab->table[idx].entry; - return 1; - } - - /* Second hash function, as suggested in [Knuth] */ - hval2 = 1 + hval % (htab->size - 2); - - do - { - /* Because SIZE is prime this guarantees to step through all - available indexes. */ - if (idx <= hval2) - idx = htab->size + idx - hval2; - else - idx -= hval2; - - /* If we visited all entries leave the loop unsuccessfully. */ - if (idx == hval) - break; - - /* If entry is found use it. */ - if (htab->table[idx].used == hval - && strcmp (item.key, htab->table[idx].entry.key) == 0) - { - if (action == ENTER) - htab->table[idx].entry.data = item.data; - - *retval = &htab->table[idx].entry; - return 1; - } - } - while (htab->table[idx].used); - } - - /* An empty bucket has been found. */ - if (action == ENTER) - { - /* If table is full and another entry should be entered return - with error. */ - if (action == ENTER && htab->filled == htab->size) - { - __set_errno (ENOMEM); - *retval = NULL; - return 0; - } - - htab->table[idx].used = hval; - htab->table[idx].entry = item; - - ++htab->filled; - - *retval = &htab->table[idx].entry; - return 1; - } - - __set_errno (ESRCH); - *retval = NULL; - return 0; -} -- 1.7.9.5

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