September 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] hda-emu: Add support for new pm_notify callback format
by David Henningsson 05 Sep '12

05 Sep '12

The format of the pm_notify callback has recently changed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 24 ++++++++++++++++++++++++ hda-emu.c | 12 ++++++++++++ 2 files changed, 36 insertions(+) diff --git a/configure.ac b/configure.ac index 18118f3..43db88f 100644 --- a/configure.ac +++ b/configure.ac @@ -222,6 +222,30 @@ if test "$OLD_HDA_CMD" = "1"; then AC_DEFINE(OLD_HDA_CMD) fi +AC_MSG_CHECKING(for new pm_notify callback) +ac_save_CFLAGS="$CFLAGS" +CFLAGS="$CFLAGS -Iinclude" +HAVE_NEW_PM_NOTIFY= +AC_TRY_COMPILE([ +#include "hda-types.h" +#include "hda-log.h" +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include "kernel/hda_codec.h" +],[ + extern struct hda_bus_ops *ops; + ops->pm_notify(NULL, 0); +], + [AC_MSG_RESULT(yes); HAVE_NEW_PM_NOTIFY="1"], + [AC_MSG_RESULT(no)], + [AC_MSG_RESULT(unknown)] +) +CFLAGS=$ac_save_CFLAGS +if test "$HAVE_NEW_PM_NOTIFY" = "1"; then + AC_DEFINE(HAVE_NEW_PM_NOTIFY) +fi + AC_MSG_CHECKING(for attach_pcm bus ops) ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS -Iinclude" diff --git a/hda-emu.c b/hda-emu.c index b4bd8a9..2a2b3d1 100644 --- a/hda-emu.c +++ b/hda-emu.c @@ -776,6 +776,13 @@ static int azx_pcm_create(struct hda_codec *codec) * power management */ +#ifdef HAVE_NEW_PM_NOTIFY +static void new_pm_notify(struct hda_bus *bus, bool power_up) +{ + hda_log(HDA_LOG_INFO, "PM-Notified, power_up = %d\n", power_up); +} +#else + static void pm_notify(struct hda_bus *bus) { hda_log(HDA_LOG_INFO, "PM-Notified\n"); @@ -787,6 +794,7 @@ static void old_pm_notify(struct hda_codec *codec) pm_notify(codec->bus); } #endif +#endif /* HAVE_NEW_PM_NOTIFY */ /* @@ -1112,7 +1120,11 @@ int main(int argc, char **argv) #ifdef HAVE_HDA_ATTACH_PCM temp.ops.attach_pcm = attach_pcm; #endif +#ifdef HAVE_NEW_PM_NOTIFY + temp.ops.pm_notify = new_pm_notify; +#else temp.ops.pm_notify = pm_notify; +#endif #endif /* OLD_HDA_CMD */ gather_codec_hooks(); -- 1.7.9.5

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Re: [alsa-devel] ALSA: core: add support for compress_offload
by Dan Carpenter 05 Sep '12

05 Sep '12

Hello Vinod Koul, The patch b21c60a4edd2: "ALSA: core: add support for compress_offload" from Dec 23, 2011, leads to the following warning: sound/core/compress_offload.c:85 snd_compr_open() warn: bitwise AND condition is false here 83 if (f->f_flags & O_WRONLY) 84 dirn = SND_COMPRESS_PLAYBACK; 85 else if (f->f_flags & O_RDONLY) 86 dirn = SND_COMPRESS_CAPTURE; 87 else { 88 pr_err("invalid direction\n"); 89 return -EINVAL; 90 } O_RDONLY is 0 so we never do compressed capture. Perhaps the tests should be: if ((file->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; else if (file->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; else return -EINVAL; Also it seems like a bad thing to print an error here, because can the user flood dmesg? regards, dan carpenter

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[alsa-devel] [patch] ASoC: wm0010: unlock on error path
by Dan Carpenter 05 Sep '12

05 Sep '12

We're holding the wm0010->lock mutex when we goto err_core. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This driver is only in linux-next. diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 5f99148..1c052d6 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -663,7 +663,9 @@ abort: wm0010_halt(codec); mutex_unlock(&wm0010->lock); return ret; + err_core: + mutex_unlock(&wm0010->lock); regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), wm0010->core_supplies); err:

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Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Takashi Iwai 05 Sep '12

05 Sep '12

At Wed, 5 Sep 2012 15:00:15 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Applied this one, too. Thanks. Takashi > --- > sound/pci/emu10k1/memory.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c > index 0a43662..1dce0a39 100644 > --- a/sound/pci/emu10k1/memory.c > +++ b/sound/pci/emu10k1/memory.c > @@ -263,8 +263,8 @@ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *b > spin_lock_irqsave(&emu->memblk_lock, flags); > if (blk->mapped_page >= 0) { > /* update order link */ > - list_del(&blk->mapped_order_link); > - list_add_tail(&blk->mapped_order_link, &emu->mapped_order_link_head); > + list_move_tail(&blk->mapped_order_link, > + &emu->mapped_order_link_head); > spin_unlock_irqrestore(&emu->memblk_lock, flags); > return 0; > } > >

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Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Takashi Iwai 05 Sep '12

05 Sep '12

At Wed, 5 Sep 2012 14:33:21 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Thanks, applied. Takashi > --- > sound/drivers/opl4/opl4_synth.c | 9 +++------ > 1 file changed, 3 insertions(+), 6 deletions(-) > > diff --git a/sound/drivers/opl4/opl4_synth.c b/sound/drivers/opl4/opl4_synth.c > index 49b9e24..4b91adc 100644 > --- a/sound/drivers/opl4/opl4_synth.c > +++ b/sound/drivers/opl4/opl4_synth.c > @@ -504,8 +504,7 @@ void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_cha > spin_lock_irqsave(&opl4->reg_lock, flags); > for (i = 0; i < voices; i++) { > voice[i] = snd_opl4_get_voice(opl4); > - list_del(&voice[i]->list); > - list_add_tail(&voice[i]->list, &opl4->on_voices); > + list_move_tail(&voice[i]->list, &opl4->on_voices); > voice[i]->chan = chan; > voice[i]->note = note; > voice[i]->velocity = vel & 0x7f; > @@ -555,8 +554,7 @@ void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_cha > > static void snd_opl4_voice_off(struct snd_opl4 *opl4, struct opl4_voice *voice) > { > - list_del(&voice->list); > - list_add_tail(&voice->list, &opl4->off_voices); > + list_move_tail(&voice->list, &opl4->off_voices); > > voice->reg_misc &= ~OPL4_KEY_ON_BIT; > snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc); > @@ -571,8 +569,7 @@ void snd_opl4_note_off(void *private_data, int note, int vel, struct snd_midi_ch > > static void snd_opl4_terminate_voice(struct snd_opl4 *opl4, struct opl4_voice *voice) > { > - list_del(&voice->list); > - list_add_tail(&voice->list, &opl4->off_voices); > + list_move_tail(&voice->list, &opl4->off_voices); > > voice->reg_misc = (voice->reg_misc & ~OPL4_KEY_ON_BIT) | OPL4_DAMP_BIT; > snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc); >

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Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Clemens Ladisch 05 Sep '12

05 Sep '12

Wei Yongjun wrote: > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Acked-by: Clemens Ladisch <clemens(a)ladisch.de>

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[alsa-devel] [PATCH 0/3] HDMI HBR(High Bit Rate) Feature bug fix for Intel's Chips
by Wang Xingchao 05 Sep '12

05 Sep '12

Please Note This patchset has not been tested yet on any Hardware platform. i could only confirm there's no compile error. :) I test previous single patch on the Yamaha A/V receiver which works well, the test results was also confirmed by some guys from XBMC forum. The idea of this patch comes from Anssi, Big credit to him at first! Any comments are appreciated! Changelog: * do not touch channel_allocation priority order, use new api to get correct channels number * use spdif->status to check audio type, pcm or non-pcm * create new non_pcm_mapping[] for non_pcm audio in seperate block * add non_pcm hdmi_spec_per_cvt to remember previous audio type, used to determine whether need a channel mapping update during the pcm/non-pcm stream switch with same channel number. Wang Xingchao (3): ALSA: hdmi - fix channel_allocation array wrong order ALSA: HDMI - Enable HBR feature on Intel chips ALSA: HDMI - Setup channel mapping for non_pcm audio sound/pci/hda/patch_hdmi.c | 56 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 51 insertions(+), 5 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: HDMI - Enable HBR feature on Intel chips
by Wang Xingchao 05 Sep '12

05 Sep '12

HDMI channel remapping apparently effects HBR packets on Intel's chips. For compressed non-PCM audio, use "straight-through" channel mapping. For uncompressed multi-channel pcm audio, use normal mapping. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/patch_hdmi.c | 50 +++++++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d9439c5..fda89e3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -231,6 +231,11 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, /* Dolby Surround */ { .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, /* surround40 */ { .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, /* surround41 */ @@ -239,22 +244,15 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, /* surround51 */ { .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, - /* surround71 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, - -{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, { .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, { .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, { .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, @@ -537,22 +535,36 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, + hda_nid_t cvt_nid, int ca) { int i; int err; - - if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); + unsigned short val; + int tmp_mapping[8]; + bool non_pcm = false; + + val = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + non_pcm = !!(val & AC_DIG1_NONAUDIO); + + if (hdmi_channel_mapping[ca][1] == 0 || non_pcm) { + for (i = 0; i < channel_allocations[ca].channels; i++) { + if (non_pcm) + tmp_mapping[i] = i | (i << 4); + else + hdmi_channel_mapping[ca][i] = i | (i << 4); + } for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + if (non_pcm) + tmp_mapping[i] = 0xf | (i << 4); + else + hdmi_channel_mapping[ca][i] = 0xf | (i << 4); } for (i = 0; i < 8; i++) { err = snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); + non_pcm ? tmp_mapping[i] : hdmi_channel_mapping[ca][i]); if (err) { snd_printdd(KERN_NOTICE "HDMI: channel mapping failed\n"); @@ -686,7 +698,7 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, } static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - struct snd_pcm_substream *substream) + hda_nid_t cvt_nid, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; @@ -731,13 +743,13 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, * sizeof(*dp_ai) to avoid partial match/update problems when * the user switches between HDMI/DP monitors. */ + hdmi_setup_channel_mapping(codec, pin_nid, cvt_nid, ca); if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, sizeof(ai))) { snd_printdd("hdmi_setup_audio_infoframe: " "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -1151,7 +1163,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, substream); + hdmi_setup_audio_infoframe(codec, pin_idx, cvt_nid, substream); pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); -- 1.7.9.5

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Re: [alsa-devel] How to define card specific pcm devices?
by Raymond Yau 05 Sep '12

05 Sep '12

2012-8-17 下午10:51 於 "Tanu Kaskinen" <tanu.kaskinen(a)digia.com> 寫道： > > Ping? > > If card specific pcm definitions are not possible, would it be > acceptable to have a generic "front_left:CARD" or "first_channel:CARD" > pcm, which would transform "front:CARD" or "hw:CARD" into a mono device? As pulseaudio server use SND_PCM_NO_AUTO_CHANNEL which diable channel conversion, this left channel device only work with other alsa applications except pulseaudio when your sound card does not support mono. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ff4af902cf4ac0… the solution is to change the mixing in pulseaudio by using left channel for this mic and those stereo mic with inverted phase , and average for mono playback device . stereo mic and line-in how about the 4 channel mic? > > -- Tanu > > > On Fri, 2012-08-10 at 14:25 +0300, Tanu Kaskinen wrote: > > Hi, > > > > I'm writing UCM configuration for RX-51 (Nokia N900), and I have a > > problem: "hw:RX51" can't be opened in mono mode, but there's only one > > integrated mono microphone, and its audio is routed to the left input > > channel. Applications will get only silence in the right channel when > > they use "hw:RX51". > > > > The best solution that I have come up is to define a pcm that converts > > "hw:RX51" into a mono device. The problem is that I don't know how > > exactly to do that. I guess the route plugin can be used for that: > > > > pcm.rx51_mono_from_left { > > type route > > slave.pcm hw:RX51 > > slave.channels 2 > > ttable.0.0 1 > > } > > > > I tried putting that definition in /usr/share/alsa/cards/RX-51.conf (and > > also RX51.conf, since it's not entirely clear to me when the dash should > > be omitted). I have this in the UCM configuration: > > > > Value { > > CapturePCM "rx51_mono_from_left" > > CaptureChannels "1" > > } > > > > This is printed to the PulseAudio log: > > > > (alsa-lib)pcm.c: Unknown PCM rx51_mono_from_left > > > > So the rx51_mono_from_left definition is not loaded. I'm not terribly > > surprised: I guess the card configuration files are loaded only when > > necessary, and when PulseAudio tries to open "rx51_mono_from_left", > > alsa-lib can't really know that it's somehow related to the RX-51 card. > > > > How should I solve this? > > > >

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[alsa-devel] [PATCH] ASoC: tegra: move platform data header
by Stephen Warren 05 Sep '12

05 Sep '12

From: Stephen Warren <swarren(a)nvidia.com> Move the Tegra+WM8903 ASoC platform data header out of arch/arm/mach-tegra, as a pre-requisite of single zImage. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- .../sound/tegra_wm8903.h | 0 sound/soc/tegra/tegra_wm8903.c | 3 +-- 2 files changed, 1 insertions(+), 2 deletions(-) rename arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h => include/sound/tegra_wm8903.h (100%) diff --git a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h b/include/sound/tegra_wm8903.h similarity index 100% rename from arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h rename to include/sound/tegra_wm8903.h diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index d4f14e4..cee13b7 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -34,13 +34,12 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <mach/tegra_wm8903_pdata.h> - #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tegra_wm8903.h> #include "../codecs/wm8903.h" -- 1.7.0.4

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