[alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
Commit identified in the subject (ALSA: HDA - Add Independent Headphone for all models of ad1988/ad1989) causes problem with audio on at least two Asus boards and is still present in current kernel. The boards on which this problem is seen are Asus M3A79-T DELUXE and Asus Rampage II Gene (actually a slightly modified version used in Asus CG5290 desktop).
The symptoms: 1. XFCE and Gnome2 audio mixers crash whenever sound is started or stopped. The error message is: simple_none.c:1350: get_enum_item_ops: Assertion `helem' failed.
2. On one of the boards (Asus M3A79-T DELUXE) the mixer control labels are misconfigured since muting the Master channel does not turn the sound off. Either Master is mislabeled or it does not work correctly.
Other than that, the sound actually works fine, but it is very annoying when mixer keeps crashing. Reverting to kernel just before the commit fixes these issues. Thus it appears that the commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4 breaks autodetect for these boards. A workaround is to force the 6stack-dig model and this fixes problems on both boards, but, of course, it would be preferred if things work out of the box.
We do not understand enough of alsa in kernel to propose a patch, but hope that we have provided enough information that it will be easy to write. We are happy to provide more information and test any proposed patches.
Alsa-info for both boards can be found below.
Best regards, Miro Hodak and Manolo Diaz
Results of alsa-info
Asus Rampage II Gene:
Before the commit: http://pastebin.com/gCZurk6D After the commit (autodetect): http://pastebin.com/3hNX2g1c After the commit (6stack_dig model): http://pastebin.com/dL0UGeTW
More recent kernel: 3.6.11-gentoo (autodetect): http://pastebin.com/0gjnhYA2 3.6.11-gentoo (6stack_dig model): http://pastebin.com/K9bsWBWW
Asus M3A79-T DELUXE
Before the commit: http://pastebin.com/hyUBmpDL After the commit (autodetect) : http://pastebin.com/gzGg60bM After the commit (6stack_dig model): http://pastebin.com/y88jCMhM
Commit identified in the subject (ALSA: HDA - Add Independent Headphone
for
all models of ad1988/ad1989) causes problem with audio on at least two
Asus
boards and is still present in current kernel. The boards on which this problem is seen are Asus M3A79-T DELUXE and Asus Rampage II Gene (actually a slightly modified version used in Asus CG5290 desktop).
The symptoms:
- XFCE and Gnome2 audio mixers crash whenever sound is started or
stopped.
The error message is: simple_none.c:1350: get_enum_item_ops: Assertion `helem' failed.
This is caused by the active access flag of independent headphone control
static void activate_ctl(struct hda_codec *codec, const char *name, int active) { struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name); if (ctl) { - ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; - ctl->vd[0].access |= active ? 0 : - SNDRV_CTL_ELEM_ACCESS_INACTIVE; ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_WRITE; ctl->vd[0].access |= active ? SNDRV_CTL_ELEM_ACCESS_WRITE : 0; snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); } }
- On one of the boards (Asus M3A79-T DELUXE) the mixer control labels are
misconfigured since muting the Master channel does not turn the sound off.
For model=auto
It is bug of hda parser which put node 0x1a in speaker out pin when the pincap only support input
Your need either
ignore this node in hda parser or
Ignore the node as speaker when pincap is INPUT in patch_analog.c
/* add playback controls for speaker and HP outputs */ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, const char *pfx) { struct ad198x_spec *spec = codec->spec; hda_nid_t nid; int i, idx, err; char name[32];
if (! pin) return 0;
+ + if ((snd_hda_query_pin_caps(codec, pin) & AC_PINCTL_OUT_EN) == 0) + return 0; + idx = ad1988_pin_idx(pin); nid = ad1988_idx_to_dac(codec, idx);
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x911711f0: [Fixed] Speaker at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
Node 0x03 [Audio Output] wcaps 0x405: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x1e 0x1e] Converter: stream=0, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0
Node 0x22 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x37 0x21
Either Master is mislabeled or it does not work correctly.
Other than that, the sound actually works fine, but it is very annoying when mixer keeps crashing. Reverting to kernel just before the commit
fixes
these issues. Thus it appears that the commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4 breaks autodetect for these boards. A workaround is to force the 6stack-dig model and this fixes problems on both boards, but, of course, it would be preferred if things work out of the box.
If possible try the proposed generic hda parser from sound-unstable.git
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058330.htm...
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058367.htm...
Alsa-info for both boards can be found below.
Model=auto does not include headphone as slave of virtual master , the volume of headphone and line out jacks are controlled independently since you can play different streams to headphone and line out jacks (multi stream playback)
May need dynamically inlcude/exclude headphone play volume and switch as slave of the virtual master controls
Do you need to include HDMI playback volume as slave of the virtual master volume ?
Can HDMI and SPDIF work independently since there are two digital audio output convertor and the number of SDO of hda controller is 4 ?
Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Control: name="IEC958 Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000010: OUT Pin Default 0x0145f1a0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Other DefAssociation = 0xa, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x02
Node 0x1d [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Control: name="HDMI Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000010: OUT Pin Default 0x1856f1b0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Other DefAssociation = 0xb, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0b
For 6stack-dig
Thus model create digital capture even when there is no spdif in pin complex
Node 0x07 [Audio Input] wcaps 0x130391: Stereo Digital Control: name="IEC958 Capture Switch", index=0, device=0 Control: name="IEC958 Capture Default", index=0, device=0 Device: name="AD198x Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Delay: 3 samples Connection: 1 0x1c
Node 0x1c [Pin Complex] wcaps 0x40020b: Stereo Digital Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] Pincap 0x00000020: IN Pin Default 0x41c5f160: [N/A] SPDIF In at Ext Rear Conn = Optical, Color = Other DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
Results of alsa-info
Asus Rampage II Gene:
Before the commit: http://pastebin.com/gCZurk6D After the commit (autodetect): http://pastebin.com/3hNX2g1c After the commit (6stack_dig model): http://pastebin.com/dL0UGeTW
More recent kernel: 3.6.11-gentoo (autodetect): http://pastebin.com/0gjnhYA2 3.6.11-gentoo (6stack_dig model): http://pastebin.com/K9bsWBWW
Asus M3A79-T DELUXE
Before the commit: http://pastebin.com/hyUBmpDL After the commit (autodetect) : http://pastebin.com/gzGg60bM After the commit (6stack_dig model): http://pastebin.com/y88jCMhM
At Mon, 14 Jan 2013 11:42:23 +0800, Raymond Yau wrote:
Either Master is mislabeled or it does not work correctly.
Other than that, the sound actually works fine, but it is very annoying when mixer keeps crashing. Reverting to kernel just before the commit
fixes
these issues. Thus it appears that the commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4 breaks autodetect for these boards. A workaround is to force the 6stack-dig model and this fixes problems on both boards, but, of course, it would be preferred if things work out of the box.
If possible try the proposed generic hda parser from sound-unstable.git
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058330.htm...
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058367.htm...
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
thanks,
Takashi
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
- On one of the boards (Asus M3A79-T DELUXE) the mixer control labels
are > misconfigured since muting the Master channel does not turn the sound off.
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x911711f0: [Fixed] Speaker at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
Do you prefer a pin fixup of node 0x1a for Asus M3A79-T DELUXE
or ignore these kind of pin during parsing
case AC_JACK_SPEAKER: + if ((snd_hda_query_pin_caps(codec, pin) & AC_PINCTL_OUT_EN) == AC_PINCTL_OUT_EN) { seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) continue; speaker_out[cfg->speaker_outs].pin = nid; speaker_out[cfg->speaker_outs].seq = (assoc << 4) | seq; cfg->speaker_outs++; + } break;
At Mon, 14 Jan 2013 21:46:57 +0800, Raymond Yau wrote:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
- On one of the boards (Asus M3A79-T DELUXE) the mixer control labels
are > misconfigured since muting the Master channel does not turn the sound off.
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x911711f0: [Fixed] Speaker at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
Do you prefer a pin fixup of node 0x1a for Asus M3A79-T DELUXE
or ignore these kind of pin during parsing
case AC_JACK_SPEAKER:
if ((snd_hda_query_pin_caps(codec, pin) &AC_PINCTL_OUT_EN) == AC_PINCTL_OUT_EN) { seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) continue; speaker_out[cfg->speaker_outs].pin = nid; speaker_out[cfg->speaker_outs].seq = (assoc << 4) | seq; cfg->speaker_outs++;
} break;
The latter, it must be filtered out in general.
If a codec has really the output capability on this pin, the pincap should be overridden beforehand by the codec parser.
Care to send a proper patch?
thanks,
Takashi
At Mon, 14 Jan 2013 15:18:11 +0100, Takashi Iwai wrote:
At Mon, 14 Jan 2013 21:46:57 +0800, Raymond Yau wrote:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
- On one of the boards (Asus M3A79-T DELUXE) the mixer control labels
are > misconfigured since muting the Master channel does not turn the sound off.
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x911711f0: [Fixed] Speaker at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
Do you prefer a pin fixup of node 0x1a for Asus M3A79-T DELUXE
or ignore these kind of pin during parsing
case AC_JACK_SPEAKER:
if ((snd_hda_query_pin_caps(codec, pin) &AC_PINCTL_OUT_EN) == AC_PINCTL_OUT_EN) { seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) continue; speaker_out[cfg->speaker_outs].pin = nid; speaker_out[cfg->speaker_outs].seq = (assoc << 4) | seq; cfg->speaker_outs++;
} break;The latter, it must be filtered out in general.
If a codec has really the output capability on this pin, the pincap should be overridden beforehand by the codec parser.
Care to send a proper patch?
Never mind, I patched by myself now. It's found in test/hda-gen-parser branch of sound-unstable git tree.
thanks,
Takashi
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the
tree
above. The other model behaviors of AD codecs are unchanged at all.
Never mind, I patched by myself now. It's found in test/hda-gen-parser branch of sound-unstable git tree.
http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob_plain;f=c...
https://launchpadlibrarian.net/73038045/Card0.Codecs.codec.0.txt
How about this node of Asus P5B [1043:81ec] and P5B DELUXE WIFI ? Two line-in in input source
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x918711f0: [Fixed] Line In at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob_plain;f=c...
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x99f301f0: [Fixed] Other at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN
At Wed, 16 Jan 2013 10:04:32 +0800, Raymond Yau wrote:
[1 <text/plain; UTF-8 (7bit)>]
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the
tree
above. The other model behaviors of AD codecs are unchanged at all.
Never mind, I patched by myself now. It's found in test/hda-gen-parser branch of sound-unstable git tree.
http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob_plain;f=c...
https://launchpadlibrarian.net/73038045/Card0.Codecs.codec.0.txt
How about this node of Asus P5B [1043:81ec] and P5B DELUXE WIFI ? Two line-in in input source
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x918711f0: [Fixed] Line In at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob_plain;f=c...
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x99f301f0: [Fixed] Other at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN
In the capture case, multiple entries are allowed. If any of them are bogus, they should be fixed via pincfg fixup.
Takashi
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules
from
alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in
the
tree
above. The other model behaviors of AD codecs are unchanged at
all.
Never mind, I patched by myself now. It's found in test/hda-gen-parser branch of sound-unstable git tree.
http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob_plain;f=c...
https://launchpadlibrarian.net/73038045/Card0.Codecs.codec.0.txt
How about this node of Asus P5B [1043:81ec] and P5B DELUXE WIFI ? Two line-in in input source
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x918711f0: [Fixed] Line In at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob_plain;f=c...
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin
Default
0x99f301f0: [Fixed] Other at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN
In the capture case, multiple entries are allowed. If any of them are bogus, they should be fixed via pincfg fixup.
But device 2 is not ad198x Headphones anymore
APLAY
**** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: AD1989B Analog [AD1989B Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: AD1989B Digital [AD1989B Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 2: AD1989B Analog [AD1989B Analog] Subdevices: 1/1 Subdevice #0: subdevice #0
At Wed, 16 Jan 2013 15:19:23 +0800, Raymond Yau wrote:
[1 <text/plain; UTF-8 (7bit)>]
> Yes, testing the latest code with real machines is highly > appreciated. Try sound-unstable git tree, either master or > test/hda-migrate branch. > git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
> > Alternatively you can build the external alsa-driver modules
from
> alsa-driver-unstable snapshot tarball, too >
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
> > Note that only the behavior of model=auto has been changed in
the
tree
> above. The other model behaviors of AD codecs are unchanged at
all.
Never mind, I patched by myself now. It's found in test/hda-gen-parser branch of sound-unstable git tree.
http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob_plain;f=c...
https://launchpadlibrarian.net/73038045/Card0.Codecs.codec.0.txt
How about this node of Asus P5B [1043:81ec] and P5B DELUXE WIFI ? Two line-in in input source
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x918711f0: [Fixed] Line In at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob_plain;f=c...
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin
Default
0x99f301f0: [Fixed] Other at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN
In the capture case, multiple entries are allowed. If any of them are bogus, they should be fixed via pincfg fixup.
But device 2 is not ad198x Headphones anymore
It's just named differently.
Takashi
APLAY
**** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: AD1989B Analog [AD1989B Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: AD1989B Digital [AD1989B Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 2: AD1989B Analog [AD1989B Analog] Subdevices: 1/1 Subdevice #0: subdevice #0
Raymond Yau wrote:
- XFCE and Gnome2 audio mixers crash whenever sound is started or stopped.
The error message is: simple_none.c:1350: get_enum_item_ops: Assertion `helem' failed.
This is caused by the active access flag of independent headphone control
... which triggers bugs in these mixers.
But as long as mixers with this bug are widespread, it might be a good idea to avoid ELEM_ACCESS_INACTIVE.
Regards, Clemens
El lun, 14 ene 2013 a las 11:42 horas Raymond Yau escribió:
Commit identified in the subject (ALSA: HDA - Add Independent Headphone
for
all models of ad1988/ad1989) causes problem with audio on at least two
Asus
boards and is still present in current kernel. The boards on which this problem is seen are Asus M3A79-T DELUXE and Asus Rampage II Gene (actually a slightly modified version used in Asus CG5290 desktop).
The symptoms:
- XFCE and Gnome2 audio mixers crash whenever sound is started or
stopped.
The error message is: simple_none.c:1350: get_enum_item_ops: Assertion `helem' failed.
This is caused by the active access flag of independent headphone control
static void activate_ctl(struct hda_codec *codec, const char *name, int active) { struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name); if (ctl) {
ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;ctl->vd[0].access |= active ? 0 :SNDRV_CTL_ELEM_ACCESS_INACTIVE; ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_WRITE; ctl->vd[0].access |= active ? SNDRV_CTL_ELEM_ACCESS_WRITE : 0; snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); }}
- On one of the boards (Asus M3A79-T DELUXE) the mixer control labels are
misconfigured since muting the Master channel does not turn the sound off.
For model=auto
It is bug of hda parser which put node 0x1a in speaker out pin when the pincap only support input
Your need either
ignore this node in hda parser or
Ignore the node as speaker when pincap is INPUT in patch_analog.c
/* add playback controls for speaker and HP outputs */ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, const char *pfx) { struct ad198x_spec *spec = codec->spec; hda_nid_t nid; int i, idx, err; char name[32];
if (! pin) return 0;
if ((snd_hda_query_pin_caps(codec, pin) & AC_PINCTL_OUT_EN) == 0)return 0;idx = ad1988_pin_idx(pin); nid = ad1988_idx_to_dac(codec, idx);
No. Doing those insertions and deletions mixer crashes are still here but I have no headphone control and headphone doesn't sound in auto model.
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x911711f0: [Fixed] Speaker at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
Node 0x03 [Audio Output] wcaps 0x405: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x1e 0x1e] Converter: stream=0, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0
Node 0x22 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x37 0x21
Either Master is mislabeled or it does not work correctly.
Other than that, the sound actually works fine, but it is very annoying when mixer keeps crashing. Reverting to kernel just before the commit
fixes
these issues. Thus it appears that the commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4 breaks autodetect for these boards. A workaround is to force the 6stack-dig model and this fixes problems on both boards, but, of course, it would be preferred if things work out of the box.
If possible try the proposed generic hda parser from sound-unstable.git
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058330.htm...
I've tried to patch it, but this is what's happened (linux stable 3.7.2):
patching file sound/pci/hda/hda_generic.c Hunk #1 FAILED at 41. Hunk #2 FAILED at 1485. Hunk #3 FAILED at 2905. Hunk #4 FAILED at 3057. Hunk #5 succeeded at 1086 with fuzz 2 (offset -1994 lines). Hunk #6 FAILED at 3192. Hunk #7 FAILED at 3209. 6 out of 7 hunks FAILED -- saving rejects to file sound/pci/hda/hda_generic.c.rej can't find file to patch at input line 187 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -------------------------- |diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h |index 85d138f..5c1569c 100644 |--- a/sound/pci/hda/hda_generic.h |+++ b/sound/pci/hda/hda_generic.h
What sources is supposed should we patch?
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058367.htm...
Alsa-info for both boards can be found below.
Model=auto does not include headphone as slave of virtual master , the volume of headphone and line out jacks are controlled independently since you can play different streams to headphone and line out jacks (multi stream playback)
Then it works as expected. Front control is slave of master, but headphone isn't (Asus M3A79-T at least). Sorry, my bad.
May need dynamically inlcude/exclude headphone play volume and switch as slave of the virtual master controls
Do you need to include HDMI playback volume as slave of the virtual master volume ?
Can HDMI and SPDIF work independently since there are two digital audio output convertor and the number of SDO of hda controller is 4 ?
Sorry, but can't answer you because I have nothing that can be connected to a digital output.
Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Control: name="IEC958 Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000010: OUT Pin Default 0x0145f1a0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Other DefAssociation = 0xa, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x02
Node 0x1d [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Control: name="HDMI Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000010: OUT Pin Default 0x1856f1b0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Other DefAssociation = 0xb, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0b
For 6stack-dig
Thus model create digital capture even when there is no spdif in pin complex
Node 0x07 [Audio Input] wcaps 0x130391: Stereo Digital Control: name="IEC958 Capture Switch", index=0, device=0 Control: name="IEC958 Capture Default", index=0, device=0 Device: name="AD198x Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Delay: 3 samples Connection: 1 0x1c
Node 0x1c [Pin Complex] wcaps 0x40020b: Stereo Digital Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] Pincap 0x00000020: IN Pin Default 0x41c5f160: [N/A] SPDIF In at Ext Rear Conn = Optical, Color = Other DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
Results of alsa-info
Asus Rampage II Gene:
Before the commit: http://pastebin.com/gCZurk6D After the commit (autodetect): http://pastebin.com/3hNX2g1c After the commit (6stack_dig model): http://pastebin.com/dL0UGeTW
More recent kernel: 3.6.11-gentoo (autodetect): http://pastebin.com/0gjnhYA2 3.6.11-gentoo (6stack_dig model): http://pastebin.com/K9bsWBWW
Asus M3A79-T DELUXE
Before the commit: http://pastebin.com/hyUBmpDL After the commit (autodetect) : http://pastebin.com/gzGg60bM After the commit (6stack_dig model): http://pastebin.com/y88jCMhM
At Mon, 14 Jan 2013 13:31:22 +0100, Manolo Díaz wrote:
El lun, 14 ene 2013 a las 11:42 horas Raymond Yau escribió:
Commit identified in the subject (ALSA: HDA - Add Independent Headphone
for
all models of ad1988/ad1989) causes problem with audio on at least two
Asus
boards and is still present in current kernel. The boards on which this problem is seen are Asus M3A79-T DELUXE and Asus Rampage II Gene (actually a slightly modified version used in Asus CG5290 desktop).
The symptoms:
- XFCE and Gnome2 audio mixers crash whenever sound is started or
stopped.
The error message is: simple_none.c:1350: get_enum_item_ops: Assertion `helem' failed.
This is caused by the active access flag of independent headphone control
static void activate_ctl(struct hda_codec *codec, const char *name, int active) { struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name); if (ctl) {
ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;ctl->vd[0].access |= active ? 0 :SNDRV_CTL_ELEM_ACCESS_INACTIVE; ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_WRITE; ctl->vd[0].access |= active ? SNDRV_CTL_ELEM_ACCESS_WRITE : 0; snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); }}
- On one of the boards (Asus M3A79-T DELUXE) the mixer control labels are
misconfigured since muting the Master channel does not turn the sound off.
For model=auto
It is bug of hda parser which put node 0x1a in speaker out pin when the pincap only support input
Your need either
ignore this node in hda parser or
Ignore the node as speaker when pincap is INPUT in patch_analog.c
/* add playback controls for speaker and HP outputs */ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, const char *pfx) { struct ad198x_spec *spec = codec->spec; hda_nid_t nid; int i, idx, err; char name[32];
if (! pin) return 0;
if ((snd_hda_query_pin_caps(codec, pin) & AC_PINCTL_OUT_EN) == 0)return 0;idx = ad1988_pin_idx(pin); nid = ad1988_idx_to_dac(codec, idx);No. Doing those insertions and deletions mixer crashes are still here but I have no headphone control and headphone doesn't sound in auto model.
Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x911711f0: [Fixed] Speaker at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN
Node 0x03 [Audio Output] wcaps 0x405: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x1e 0x1e] Converter: stream=0, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0
Node 0x22 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x37 0x21
Either Master is mislabeled or it does not work correctly.
Other than that, the sound actually works fine, but it is very annoying when mixer keeps crashing. Reverting to kernel just before the commit
fixes
these issues. Thus it appears that the commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4 breaks autodetect for these boards. A workaround is to force the 6stack-dig model and this fixes problems on both boards, but, of course, it would be preferred if things work out of the box.
If possible try the proposed generic hda parser from sound-unstable.git
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058330.htm...
I've tried to patch it, but this is what's happened (linux stable 3.7.2):
patching file sound/pci/hda/hda_generic.c Hunk #1 FAILED at 41. Hunk #2 FAILED at 1485. Hunk #3 FAILED at 2905. Hunk #4 FAILED at 3057. Hunk #5 succeeded at 1086 with fuzz 2 (offset -1994 lines). Hunk #6 FAILED at 3192. Hunk #7 FAILED at 3209. 6 out of 7 hunks FAILED -- saving rejects to file sound/pci/hda/hda_generic.c.rej can't find file to patch at input line 187 Perhaps you used the wrong -p or --strip option? The text leading up to this was:
|diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h |index 85d138f..5c1569c 100644 |--- a/sound/pci/hda/hda_generic.h |+++ b/sound/pci/hda/hda_generic.h
What sources is supposed should we patch?
You need to apply over 100 patches :)
Takashi
El lun, 14 ene 2013 a las 13:48 horas Takashi Iwai escribió:
If possible try the proposed generic hda parser from sound-unstable.git
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058330.htm...
I've tried to patch it, but this is what's happened (linux stable 3.7.2):
patching file sound/pci/hda/hda_generic.c Hunk #1 FAILED at 41. Hunk #2 FAILED at 1485. Hunk #3 FAILED at 2905. Hunk #4 FAILED at 3057. Hunk #5 succeeded at 1086 with fuzz 2 (offset -1994 lines). Hunk #6 FAILED at 3192. Hunk #7 FAILED at 3209. 6 out of 7 hunks FAILED -- saving rejects to file sound/pci/hda/hda_generic.c.rej can't find file to patch at input line 187 Perhaps you used the wrong -p or --strip option? The text leading up to this was:
|diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h |index 85d138f..5c1569c 100644 |--- a/sound/pci/hda/hda_generic.h |+++ b/sound/pci/hda/hda_generic.h
What sources is supposed should we patch?
You need to apply over 100 patches :)
Is there an easy way to apply all of them?
At Mon, 14 Jan 2013 13:59:47 +0100, Manolo Díaz wrote:
El lun, 14 ene 2013 a las 13:48 horas Takashi Iwai escribió:
If possible try the proposed generic hda parser from sound-unstable.git
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058330.htm...
I've tried to patch it, but this is what's happened (linux stable 3.7.2):
patching file sound/pci/hda/hda_generic.c Hunk #1 FAILED at 41. Hunk #2 FAILED at 1485. Hunk #3 FAILED at 2905. Hunk #4 FAILED at 3057. Hunk #5 succeeded at 1086 with fuzz 2 (offset -1994 lines). Hunk #6 FAILED at 3192. Hunk #7 FAILED at 3209. 6 out of 7 hunks FAILED -- saving rejects to file sound/pci/hda/hda_generic.c.rej can't find file to patch at input line 187 Perhaps you used the wrong -p or --strip option? The text leading up to this was:
|diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h |index 85d138f..5c1569c 100644 |--- a/sound/pci/hda/hda_generic.h |+++ b/sound/pci/hda/hda_generic.h
What sources is supposed should we patch?
You need to apply over 100 patches :)
Is there an easy way to apply all of them?
Hm, didn't you receive my previous post?
To repeat myself:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
thanks,
Takashi
El lun, 14 ene 2013 a las 14:03 horas Takashi Iwai escribió:
Hm, didn't you receive my previous post?
To repeat myself:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
thanks,
Takashi
I'm cloning the sound-unstable git repository. It will take a while. I'll keep you informed.
Regards,
El lun, 14 ene 2013 a las 14:03 horas Takashi Iwai escribió:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
Mixer crashes are gone. As I wrote I have no way to test the digital output.
I haven't found a way to test multi stream playback either. After choosing HP Independant enabled I can't hear nothing through the headphone.
git show commit 9d7b2668e0d3ddf69a3872d2f3622b2af262ed8e Merge: bcf4f30 fb85635 Author: Takashi Iwai tiwai@suse.de Date: Mon Jan 14 10:15:24 2013 +0100
Merge remote-tracking branch 'stable/master'
dmesg | grep -i hda [ 2.144482] hda_codec: AD1989B: BIOS auto-probing. [ 2.167768] input: HDA ATI SB Line as /devices/pci0000:00/0000:00:14.2/sound/card0/input4 [ 2.170191] input: HDA ATI SB Rear Mic as /devices/pci0000:00/0000:00:14.2/sound/card0/input5 [ 2.172568] input: HDA ATI SB Front Mic as /devices/pci0000:00/0000:00:14.2/sound/card0/input6 [ 2.174956] input: HDA ATI SB Front Headphone as /devices/pci0000:00/0000:00:14.2/sound/card0/input7 [ 2.177378] input: HDA ATI SB Line Out Side as /devices/pci0000:00/0000:00:14.2/sound/card0/input8 [ 2.179806] input: HDA ATI SB Line Out CLFE as /devices/pci0000:00/0000:00:14.2/sound/card0/input9 [ 2.182209] input: HDA ATI SB Line Out Surround as /devices/pci0000:00/0000:00:14.2/sound/card0/input10 [ 2.184621] input: HDA ATI SB Line Out Front as /devices/pci0000:00/0000:00:14.2/sound/card0/input11 [ 2.187370] snd_hda_intel 0000:01:00.1: irq 42 for MSI/MSI-X [ 2.191501] input: HDA ATI HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:02.0/0000:01:00.1/sound/card1/input12 [ 2.266022] #0: HDA ATI SB at 0xfe8f4000 irq 16 [ 2.268244] #1: HDA ATI HDMI at 0xfe9ec000 irq 42 [ 212.984025] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj.
Regards,
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
Mixer crashes are gone. As I wrote I have no way to test the digital output.
I haven't found a way to test multi stream playback either. After choosing HP Independant enabled I can't hear nothing through the headphone.
Can you post the output of alsa-info.sh ?
speaker-test -c2 -t wav -D hw:0,2
and another application playback using hw:0,0
Do your computer chassis have the hda front audio pannel with Jack detection ?
You need to disable auto mute when testing the multi streaming playback
El mar, 15 ene 2013 a las 8:42 horas Raymond Yau escribió:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
Mixer crashes are gone. As I wrote I have no way to test the digital output.
I haven't found a way to test multi stream playback either. After choosing HP Independant enabled I can't hear nothing through the headphone.
Can you post the output of alsa-info.sh ?
speaker-test -c2 -t wav -D hw:0,2
and another application playback using hw:0,0
Do your computer chassis have the hda front audio pannel with Jack detection ?
You need to disable auto mute when testing the multi streaming playback
All right, auto mute disabled.
When Independent HP is enabled I can hear nothing through the headphone:
speaker-test -c2 -t wav -D hw:0,2
speaker-test 1.0.25
Playback device is hw:0,2 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 524288 Period size range from 32 to 262144 Using max buffer size 524288 Periods = 4 was set period_size = 131072 was set buffer_size = 524288 0 - Front Left 1 - Front Right Time per period = 5.487935 0 - Front Left 1 - Front Right Time per period = 5.469224 0 - Front Left 1 - Front Right Time per period = 5.469168 0 - Front Left 1 - Front Right Time per period = 5.469292 0 - Front Left 1 - Front Right
and so on...
When HP Independent is disabled I hear the same through front speaker and headphone, just as expected.
alsa-info.sh output is attached.
Regards,
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
Mixer crashes are gone. As I wrote I have no way to test the digital output.
I haven't found a way to test multi stream playback either. After choosing HP Independant enabled I can't hear nothing through the headphone.
Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend / resume anymore after install Ubuntu 12.04 ?
Audio can suspend / resume in fedora 10 for different version of alsa driver
Seen many ad1988 controls are missing ?
1) rear Mic, line , front Mic playback volume and switch 2) CD playback switch 3) input mix in input source 4) analog loopback mix volume and mute switch
El mié, 16 ene 2013 a las 9:52 horas Raymond Yau escribió:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
Mixer crashes are gone. As I wrote I have no way to test the digital output.
I haven't found a way to test multi stream playback either. After choosing HP Independant enabled I can't hear nothing through the headphone.
Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend / resume anymore after install Ubuntu 12.04 ?
Audio can suspend / resume in fedora 10 for different version of alsa driver
Seen many ad1988 controls are missing ?
- rear Mic, line , front Mic playback volume and switch
- CD playback switch
- input mix in input source
- analog loopback mix volume and mute switch
I can confirm both: no sound after resume and missing controls. Attached are captures from the mixer control list using linux stable and sound unstable, both with BIOS autoprobe.
Asus M3A79-T Deluxe
At Wed, 16 Jan 2013 09:49:41 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:52 horas Raymond Yau escribió:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
Mixer crashes are gone. As I wrote I have no way to test the digital output.
I haven't found a way to test multi stream playback either. After choosing HP Independant enabled I can't hear nothing through the headphone.
Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend / resume anymore after install Ubuntu 12.04 ?
Audio can suspend / resume in fedora 10 for different version of alsa driver
Seen many ad1988 controls are missing ?
- rear Mic, line , front Mic playback volume and switch
- CD playback switch
- input mix in input source
- analog loopback mix volume and mute switch
I can confirm both: no sound after resume and missing controls. Attached are captures from the mixer control list using linux stable and sound unstable, both with BIOS autoprobe.
Give alsa-info.sh outputs before and after suspend, instead.
Takashi
Seen many ad1988 controls are missing ?
- rear Mic, line , front Mic playback volume and switch
- CD playback switch
- input mix in input source
- analog loopback mix volume and mute switch
I can confirm both: no sound after resume and missing controls. Attached are captures from the mixer control list using linux stable and sound unstable, both with BIOS autoprobe.
Give alsa-info.sh outputs before and after suspend, instead.
Is there any special reason to use alt_capture_device since the input sources of three ADC of ad1988/1989 are same ?
Ones of ADC of some realtek codec has more input source than the other ADCs
**** List of CAPTURE Hardware Devices **** card 0: SB [HDA ATI SB], device 0: AD1989B Analog [AD1989B Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 2: AD1989B Analog [AD1989B Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1
Node 0x21 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Analog Mix Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Analog Mix Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 1 0x20
The input source "mix" from node 0x20 audio mixer
Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' 'CD' 'Mix' Item0: 'Front Mic'
Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=0, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x2f 0x2f] Connection: 11 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 0x1f
Node 0x20 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=6, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=6, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x39 0x33 0x38 0x3d 0x34 0x3b 0x18 0x1a
El mié, 16 ene 2013 a las 9:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 09:49:41 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:52 horas Raymond Yau escribió:
>Yes, testing the latest code with real machines is highly >appreciated. Try sound-unstable git tree, either master or >test/hda-migrate branch. > git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
> >Alternatively you can build the external alsa-driver modules from >alsa-driver-unstable snapshot tarball, too >
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
> >Note that only the behavior of model=auto has been changed in the tree >above. The other model behaviors of AD codecs are unchanged at all.
Mixer crashes are gone. As I wrote I have no way to test the digital output.
I haven't found a way to test multi stream playback either. After choosing HP Independant enabled I can't hear nothing through the headphone.
Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend / resume anymore after install Ubuntu 12.04 ?
Audio can suspend / resume in fedora 10 for different version of alsa driver
Seen many ad1988 controls are missing ?
- rear Mic, line , front Mic playback volume and switch
- CD playback switch
- input mix in input source
- analog loopback mix volume and mute switch
I can confirm both: no sound after resume and missing controls. Attached are captures from the mixer control list using linux stable and sound unstable, both with BIOS autoprobe.
Give alsa-info.sh outputs before and after suspend, instead.
Takashi
Ok.
Outputs attached. This is the diff:
--- alsa-info-before-suspend.txt 2013-01-16 13:29:59.991633927 +0100 +++ alsa-info-after-resume.txt 2013-01-16 13:31:06.353096949 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################
-!!Script ran on: Wed Jan 16 12:29:57 UTC 2013 +!!Script ran on: Wed Jan 16 12:31:03 UTC 2013
!!Linux Distribution @@ -127,7 +127,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=2 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 - Amp-Out vals: [0x23 0x23] + Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 @@ -136,7 +136,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 - Amp-Out vals: [0x1e 0x1e] + Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 @@ -146,7 +146,7 @@ Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 - Amp-Out vals: [0x23 0x23] + Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 @@ -154,7 +154,7 @@ Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 - Amp-Out vals: [0x23 0x23] + Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 @@ -192,7 +192,7 @@ Control: name="Side Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 - Amp-Out vals: [0x23 0x23] + Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0 @@ -248,7 +248,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-Out vals: [0x00 0x00] + Amp-Out vals: [0x80 0x80] Pincap 0x0000373f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214030: [Jack] HP Out at Ext Front @@ -263,7 +263,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Front Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-Out vals: [0x00 0x00] + Amp-Out vals: [0x80 0x80] Pincap 0x0000373f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01014010: [Jack] Line Out at Ext Rear @@ -315,7 +315,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Surround Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-Out vals: [0x00 0x00] + Amp-Out vals: [0x80 0x80] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black @@ -362,7 +362,7 @@ Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Control: name="SPDIF Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 - Amp-Out vals: [0x27 0x27] + Amp-Out vals: [0xa7 0xa7] Pincap 0x00000010: OUT Pin Default 0x0145f1a0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Other @@ -383,7 +383,7 @@ Node 0x1d [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Control: name="HDMI Phantom Jack", index=1, device=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 - Amp-Out vals: [0x27 0x27] + Amp-Out vals: [0xa7 0xa7] Pincap 0x00000010: OUT Pin Default 0x1856f1b0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Other @@ -410,7 +410,7 @@ 0x20 Node 0x22 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-In vals: [0x00 0x00] [0x80 0x80] + Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x37 0x21 Node 0x23 [Vendor Defined Widget] wcaps 0xf00100: Mono @@ -423,7 +423,7 @@ ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Line Out CLFE Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-Out vals: [0x00 0x00] + Amp-Out vals: [0x80 0x80] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange @@ -437,7 +437,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Side Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-Out vals: [0x00 0x00] + Amp-Out vals: [0x80 0x80] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey @@ -453,22 +453,22 @@ 0x32 0x21 Node 0x27 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-In vals: [0x00 0x00] [0x80 0x80] + Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x05 0x21 Node 0x28 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-In vals: [0x00 0x00] [0x80 0x80] + Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x0a 0x21 Node 0x29 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-In vals: [0x00 0x00] [0x80 0x80] + Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x04 0x21 Node 0x2a [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-In vals: [0x00 0x00] [0x80 0x80] + Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x06 0x21 Node 0x2b [Audio Mixer] wcaps 0x200103: Stereo Amp-In @@ -1140,7 +1140,7 @@ control.31 { iface CARD name 'Line Out Front Jack' - value true + value false comment { access read type BOOLEAN @@ -1461,5 +1461,9 @@ [ 2.225160] #0: HDA ATI SB at 0xfe8f4000 irq 16 [ 2.227341] #1: HDA ATI HDMI at 0xfe9ec000 irq 42 [ 2.230063] EXT4-fs (sda1): couldn't mount as ext3 due to feature incompatibilities +-- +[ 170.949719] PM: early resume of devices complete after 0.107 msecs +[ 170.950031] snd_hda_intel 0000:01:00.1: irq 42 for MSI/MSI-X +[ 170.950098] pcieport 0000:00:06.0: System wakeup disabled by ACPI
At Wed, 16 Jan 2013 13:39:52 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 09:49:41 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:52 horas Raymond Yau escribió:
> > >Yes, testing the latest code with real machines is highly > >appreciated. Try sound-unstable git tree, either master or > >test/hda-migrate branch. > > git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
> > > >Alternatively you can build the external alsa-driver modules from > >alsa-driver-unstable snapshot tarball, too > >
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
> > > >Note that only the behavior of model=auto has been changed in the tree > >above. The other model behaviors of AD codecs are unchanged at all. > > Mixer crashes are gone. As I wrote I have no way to test the digital > output. > > I haven't found a way to test multi stream playback either. After > choosing HP Independant enabled I can't hear nothing through the > headphone. >
Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend / resume anymore after install Ubuntu 12.04 ?
Audio can suspend / resume in fedora 10 for different version of alsa driver
Seen many ad1988 controls are missing ?
- rear Mic, line , front Mic playback volume and switch
- CD playback switch
- input mix in input source
- analog loopback mix volume and mute switch
I can confirm both: no sound after resume and missing controls. Attached are captures from the mixer control list using linux stable and sound unstable, both with BIOS autoprobe.
Give alsa-info.sh outputs before and after suspend, instead.
Takashi
Ok.
Outputs attached. This is the diff:
--- alsa-info-before-suspend.txt 2013-01-16 13:29:59.991633927 +0100 +++ alsa-info-after-resume.txt 2013-01-16 13:31:06.353096949 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################
-!!Script ran on: Wed Jan 16 12:29:57 UTC 2013 +!!Script ran on: Wed Jan 16 12:31:03 UTC 2013
!!Linux Distribution @@ -127,7 +127,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=2 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x23 0x23]
- Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0
@@ -136,7 +136,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x1e 0x1e]
- Amp-Out vals: [0x27 0x27]
Hm, are you running the latest sound-unstable tree?
There have been a few critical bugs fixed since the last time you wrote. This might be the issue you hit.
Takashi
El mié, 16 ene 2013 a las 13:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 13:39:52 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 09:49:41 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:52 horas Raymond Yau escribió:
>> >> >Yes, testing the latest code with real machines is highly >> >appreciated. Try sound-unstable git tree, either master or >> >test/hda-migrate branch. >> > git://
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
>> > >> >Alternatively you can build the external alsa-driver modules from >> >alsa-driver-unstable snapshot tarball, too >> > >
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
>> > >> >Note that only the behavior of model=auto has been changed in the tree >> >above. The other model behaviors of AD codecs are unchanged at all. >> >> Mixer crashes are gone. As I wrote I have no way to test the digital >> output. >> >> I haven't found a way to test multi stream playback either. After >> choosing HP Independant enabled I can't hear nothing through the >> headphone. >>
Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend / resume anymore after install Ubuntu 12.04 ?
Audio can suspend / resume in fedora 10 for different version of alsa driver
Seen many ad1988 controls are missing ?
- rear Mic, line , front Mic playback volume and switch
- CD playback switch
- input mix in input source
- analog loopback mix volume and mute switch
I can confirm both: no sound after resume and missing controls. Attached are captures from the mixer control list using linux stable and sound unstable, both with BIOS autoprobe.
Give alsa-info.sh outputs before and after suspend, instead.
Takashi
Ok.
Outputs attached. This is the diff:
--- alsa-info-before-suspend.txt 2013-01-16 13:29:59.991633927 +0100 +++ alsa-info-after-resume.txt 2013-01-16 13:31:06.353096949 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################
-!!Script ran on: Wed Jan 16 12:29:57 UTC 2013 +!!Script ran on: Wed Jan 16 12:31:03 UTC 2013
!!Linux Distribution @@ -127,7 +127,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=2 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x23 0x23]
- Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0
@@ -136,7 +136,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x1e 0x1e]
- Amp-Out vals: [0x27 0x27]
Hm, are you running the latest sound-unstable tree?
There have been a few critical bugs fixed since the last time you wrote. This might be the issue you hit.
Takashi
#git show commit 9d7b2668e0d3ddf69a3872d2f3622b2af262ed8e Merge: bcf4f30 fb85635 Author: Takashi Iwai tiwai@suse.de Date: Mon Jan 14 10:15:24 2013 +0100
Merge remote-tracking branch 'stable/master'
#git rebase Current branch master is up to date.
Am I doing something wrong?
At Wed, 16 Jan 2013 14:04:12 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 13:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 13:39:52 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 09:49:41 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:52 horas Raymond Yau escribió:
> > >> > >> >Yes, testing the latest code with real machines is highly > >> >appreciated. Try sound-unstable git tree, either master or > >> >test/hda-migrate branch. > >> > git:// git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git > >> > > >> >Alternatively you can build the external alsa-driver modules from > >> >alsa-driver-unstable snapshot tarball, too > >> > > > ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz > >> > > >> >Note that only the behavior of model=auto has been changed in the tree > >> >above. The other model behaviors of AD codecs are unchanged at all. > >> > >> Mixer crashes are gone. As I wrote I have no way to test the digital > >> output. > >> > >> I haven't found a way to test multi stream playback either. After > >> choosing HP Independant enabled I can't hear nothing through the > >> headphone. > >>
Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend / resume anymore after install Ubuntu 12.04 ?
Audio can suspend / resume in fedora 10 for different version of alsa driver
Seen many ad1988 controls are missing ?
- rear Mic, line , front Mic playback volume and switch
- CD playback switch
- input mix in input source
- analog loopback mix volume and mute switch
I can confirm both: no sound after resume and missing controls. Attached are captures from the mixer control list using linux stable and sound unstable, both with BIOS autoprobe.
Give alsa-info.sh outputs before and after suspend, instead.
Takashi
Ok.
Outputs attached. This is the diff:
--- alsa-info-before-suspend.txt 2013-01-16 13:29:59.991633927 +0100 +++ alsa-info-after-resume.txt 2013-01-16 13:31:06.353096949 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################
-!!Script ran on: Wed Jan 16 12:29:57 UTC 2013 +!!Script ran on: Wed Jan 16 12:31:03 UTC 2013
!!Linux Distribution @@ -127,7 +127,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=2 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x23 0x23]
- Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0
@@ -136,7 +136,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x1e 0x1e]
- Amp-Out vals: [0x27 0x27]
Hm, are you running the latest sound-unstable tree?
There have been a few critical bugs fixed since the last time you wrote. This might be the issue you hit.
Takashi
#git show commit 9d7b2668e0d3ddf69a3872d2f3622b2af262ed8e Merge: bcf4f30 fb85635 Author: Takashi Iwai tiwai@suse.de Date: Mon Jan 14 10:15:24 2013 +0100
Merge remote-tracking branch 'stable/master'
The latest master HEAD is
commit 3c8a646ae30522bd041596282ab5e9e5bc7e1c6d Merge: 261ecc2 c0f3b21 Author: Takashi Iwai tiwai@suse.de Date: Wed Jan 16 11:57:22 2013 +0100
Merge branch 'test/hda-gen-parser'
Takashi
El mié, 16 ene 2013 a las 14:16 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 14:04:12 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 13:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 13:39:52 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 09:49:41 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:52 horas Raymond Yau escribió:
>> >> >> >> >> >Yes, testing the latest code with real machines is highly >> >> >appreciated. Try sound-unstable git tree, either master or >> >> >test/hda-migrate branch. >> >> > git:// >git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git >> >> > >> >> >Alternatively you can build the external alsa-driver modules from >> >> >alsa-driver-unstable snapshot tarball, too >> >> > >> > >ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz >> >> > >> >> >Note that only the behavior of model=auto has been changed in the tree >> >> >above. The other model behaviors of AD codecs are unchanged at all. >> >> >> >> Mixer crashes are gone. As I wrote I have no way to test the digital >> >> output. >> >> >> >> I haven't found a way to test multi stream playback either. After >> >> choosing HP Independant enabled I can't hear nothing through the >> >> headphone. >> >> > >Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend >/ resume anymore after install Ubuntu 12.04 ? > >Audio can suspend / resume in fedora 10 for different version of alsa >driver > >Seen many ad1988 controls are missing ? > >1) rear Mic, line , front Mic playback volume and switch >2) CD playback switch >3) input mix in input source >4) analog loopback mix volume and mute switch
I can confirm both: no sound after resume and missing controls. Attached are captures from the mixer control list using linux stable and sound unstable, both with BIOS autoprobe.
Give alsa-info.sh outputs before and after suspend, instead.
Takashi
Ok.
Outputs attached. This is the diff:
--- alsa-info-before-suspend.txt 2013-01-16 13:29:59.991633927 +0100 +++ alsa-info-after-resume.txt 2013-01-16 13:31:06.353096949 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################
-!!Script ran on: Wed Jan 16 12:29:57 UTC 2013 +!!Script ran on: Wed Jan 16 12:31:03 UTC 2013
!!Linux Distribution @@ -127,7 +127,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=2 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x23 0x23]
- Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0
@@ -136,7 +136,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x1e 0x1e]
- Amp-Out vals: [0x27 0x27]
Hm, are you running the latest sound-unstable tree?
There have been a few critical bugs fixed since the last time you wrote. This might be the issue you hit.
Takashi
#git show commit 9d7b2668e0d3ddf69a3872d2f3622b2af262ed8e Merge: bcf4f30 fb85635 Author: Takashi Iwai tiwai@suse.de Date: Mon Jan 14 10:15:24 2013 +0100
Merge remote-tracking branch 'stable/master'The latest master HEAD is
commit 3c8a646ae30522bd041596282ab5e9e5bc7e1c6d Merge: 261ecc2 c0f3b21 Author: Takashi Iwai tiwai@suse.de Date: Wed Jan 16 11:57:22 2013 +0100
Merge branch 'test/hda-gen-parser'
Takashi
Sorry. I'll repeat those tests.
commit 3c8a646ae30522bd041596282ab5e9e5bc7e1c6d Merge: 261ecc2 c0f3b21 Author: Takashi Iwai tiwai@suse.de Date: Wed Jan 16 11:57:22 2013 +0100
Merge branch 'test/hda-gen-parser'
You're right. Things seem so much better now:
-Multi stream play does work (tested front speakers and headphone) -Sound doesn't disappear after resume
--- alsa-info-before-suspend.txt 2013-01-16 14:38:53.398515024 +0100 +++ alsa-info-after-resume.txt 2013-01-16 14:39:48.424133569 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################
-!!Script ran on: Wed Jan 16 13:38:52 UTC 2013 +!!Script ran on: Wed Jan 16 13:39:47 UTC 2013
!!Linux Distribution @@ -1138,7 +1138,7 @@ control.31 { iface CARD name 'Line Out Front Jack' - value true + value false comment { access read type BOOLEAN @@ -1453,5 +1453,9 @@ [ 2.361493] #0: HDA ATI SB at 0xfe8f4000 irq 16 [ 2.363720] #1: HDA ATI HDMI at 0xfe9ec000 irq 42 [ 2.366474] EXT4-fs (sda1): couldn't mount as ext3 due to feature incompatibilities +-- +[ 200.736358] serial 00:08: activated +[ 200.736528] snd_hda_intel 0000:01:00.1: irq 42 for MSI/MSI-X +[ 200.736593] pcieport 0000:00:06.0: System wakeup disabled by ACPI
But the number of controls is smaller than in linux stable 3.7.2. I don't know if that's a regression.
At Wed, 16 Jan 2013 14:55:09 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 14:16 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 14:04:12 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 13:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 13:39:52 +0100, Manolo Díaz wrote:
El mié, 16 ene 2013 a las 9:54 horas Takashi Iwai escribió:
At Wed, 16 Jan 2013 09:49:41 +0100, Manolo Díaz wrote: > > El mié, 16 ene 2013 a las 9:52 horas > Raymond Yau escribió: > > >> > >> >> > >> >> >Yes, testing the latest code with real machines is highly > >> >> >appreciated. Try sound-unstable git tree, either master or > >> >> >test/hda-migrate branch. > >> >> > git:// > >git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git > >> >> > > >> >> >Alternatively you can build the external alsa-driver modules from > >> >> >alsa-driver-unstable snapshot tarball, too > >> >> > > >> > > >ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz > >> >> > > >> >> >Note that only the behavior of model=auto has been changed in the tree > >> >> >above. The other model behaviors of AD codecs are unchanged at all. > >> >> > >> >> Mixer crashes are gone. As I wrote I have no way to test the digital > >> >> output. > >> >> > >> >> I haven't found a way to test multi stream playback either. After > >> >> choosing HP Independant enabled I can't hear nothing through the > >> >> headphone. > >> >> > > > >Did audio work with suspend and resume since my p5b-v ad1988 cannot suspend > >/ resume anymore after install Ubuntu 12.04 ? > > > >Audio can suspend / resume in fedora 10 for different version of alsa > >driver > > > >Seen many ad1988 controls are missing ? > > > >1) rear Mic, line , front Mic playback volume and switch > >2) CD playback switch > >3) input mix in input source > >4) analog loopback mix volume and mute switch > > I can confirm both: no sound after resume and missing controls. > Attached are captures from the mixer control list using linux stable > and sound unstable, both with BIOS autoprobe.
Give alsa-info.sh outputs before and after suspend, instead.
Takashi
Ok.
Outputs attached. This is the diff:
--- alsa-info-before-suspend.txt 2013-01-16 13:29:59.991633927 +0100 +++ alsa-info-after-resume.txt 2013-01-16 13:31:06.353096949 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################
-!!Script ran on: Wed Jan 16 12:29:57 UTC 2013 +!!Script ran on: Wed Jan 16 12:31:03 UTC 2013
!!Linux Distribution @@ -127,7 +127,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=2 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x23 0x23]
- Amp-Out vals: [0x27 0x27] Converter: stream=8, channel=0 Power states: D0 D3 Power: setting=D0, actual=D0
@@ -136,7 +136,7 @@ ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="AD1989B Analog", type="Audio", device=0 Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0
- Amp-Out vals: [0x1e 0x1e]
- Amp-Out vals: [0x27 0x27]
Hm, are you running the latest sound-unstable tree?
There have been a few critical bugs fixed since the last time you wrote. This might be the issue you hit.
Takashi
#git show commit 9d7b2668e0d3ddf69a3872d2f3622b2af262ed8e Merge: bcf4f30 fb85635 Author: Takashi Iwai tiwai@suse.de Date: Mon Jan 14 10:15:24 2013 +0100
Merge remote-tracking branch 'stable/master'The latest master HEAD is
commit 3c8a646ae30522bd041596282ab5e9e5bc7e1c6d Merge: 261ecc2 c0f3b21 Author: Takashi Iwai tiwai@suse.de Date: Wed Jan 16 11:57:22 2013 +0100
Merge branch 'test/hda-gen-parser'
Takashi
Sorry. I'll repeat those tests.
commit 3c8a646ae30522bd041596282ab5e9e5bc7e1c6d Merge: 261ecc2 c0f3b21 Author: Takashi Iwai tiwai@suse.de Date: Wed Jan 16 11:57:22 2013 +0100
Merge branch 'test/hda-gen-parser'You're right. Things seem so much better now:
-Multi stream play does work (tested front speakers and headphone) -Sound doesn't disappear after resume
Good to hear!
--- alsa-info-before-suspend.txt 2013-01-16 14:38:53.398515024 +0100 +++ alsa-info-after-resume.txt 2013-01-16 14:39:48.424133569 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################
-!!Script ran on: Wed Jan 16 13:38:52 UTC 2013 +!!Script ran on: Wed Jan 16 13:39:47 UTC 2013
!!Linux Distribution @@ -1138,7 +1138,7 @@ control.31 { iface CARD name 'Line Out Front Jack'
value true
value false@@ -1453,5 +1453,9 @@ [ 2.361493] #0: HDA ATI SB at 0xfe8f4000 irq 16 [ 2.363720] #1: HDA ATI HDMI at 0xfe9ec000 irq 42 [ 2.366474] EXT4-fs (sda1): couldn't mount as ext3 due to feature incompatibilities +-- +[ 200.736358] serial 00:08: activated +[ 200.736528] snd_hda_intel 0000:01:00.1: irq 42 for MSI/MSI-X +[ 200.736593] pcieport 0000:00:06.0: System wakeup disabled by ACPI
But the number of controls is smaller than in linux stable 3.7.2. I don't know if that's a regression.
You should diet sometimes :)
Unless you see *functional* degradation, it's no regression but regarded rather as a cleanup.
Takashi
>> >> >> >Yes, testing the latest code with real machines is
highly
>> >> >> >appreciated. Try sound-unstable git tree, either
master or
>> >> >> >test/hda-migrate branch. >> >> >> > git:// >> >
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
>> >> >> > >> >> >> >Alternatively you can build the external alsa-driver
modules from
>> >> >> >alsa-driver-unstable snapshot tarball, too >> >> >> > >> >> > >> >
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
>> >> >> > >> >> >> >Note that only the behavior of model=auto has been
changed in the tree
>> >> >> >above. The other model behaviors of AD codecs are
unchanged at all.
>> > >> >Seen many ad1988 controls are missing ? >> > >> >1) rear Mic, line , front Mic playback volume and switch >> >2) CD playback switch >> >3) input mix in input source >> >4) analog loopback mix volume and mute switch
But the number of controls is smaller than in linux stable 3.7.2. I don't know if that's a regression.
Unless you see *functional* degradation, it's no regression but regarded rather as a cleanup
Refer to commit
http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable.git;a=commitd...
The Stereo Mix of those VIA codecs allow ADD capture DAC signal
ad1988 and those realtek codecs can select input source from the audio mixer which can mix the input pins
- snd_hda_add_imux_item(imux, "Mix", 9, NULL);
Those static models (3stack , 3stack-dig, 6stack and 6stack-dig )still have the input source "mix"
static const struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { { "Front Mic", 0x1 }, /* port-B */ { "Line", 0x2 }, /* port-C */ { "Mic", 0x4 }, /* port-E */ { "CD", 0x5 }, { "Mix", 0x9 }, }, };
Most desktop codecs can pass the blue Jack signal to green Jack by unmute line playback switch and line playback volume
-- if ((err = add_control(spec, AD_CTL_WIDGET_VOL, - "Analog Mix Playback Volume", - HDA_COMPOSE_AMP_VAL(0x21, 3, 0x0, HDA_OUTPUT))) < 0) - return err; - if ((err = add_control(spec, AD_CTL_WIDGET_MUTE, - "Analog Mix Playback Switch", - HDA_COMPOSE_AMP_VAL(0x21, 3, 0x0, HDA_OUTPUT))) < 0) - return err;
At Thu, 17 Jan 2013 11:51:48 +0800, Raymond Yau wrote:
> >> >> >> >Yes, testing the latest code with real machines is
highly
> >> >> >> >appreciated. Try sound-unstable git tree, either
master or
> >> >> >> >test/hda-migrate branch. > >> >> >> > git:// > >> >
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
> >> >> >> > > >> >> >> >Alternatively you can build the external alsa-driver
modules from
> >> >> >> >alsa-driver-unstable snapshot tarball, too > >> >> >> > > >> >> > > >> >
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
> >> >> >> > > >> >> >> >Note that only the behavior of model=auto has been
changed in the tree
> >> >> >> >above. The other model behaviors of AD codecs are
unchanged at all.
> >> > > >> >Seen many ad1988 controls are missing ? > >> > > >> >1) rear Mic, line , front Mic playback volume and switch > >> >2) CD playback switch > >> >3) input mix in input source > >> >4) analog loopback mix volume and mute switch
But the number of controls is smaller than in linux stable 3.7.2. I don't know if that's a regression.
Unless you see *functional* degradation, it's no regression but regarded rather as a cleanup
Refer to commit
http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable.git;a=commitd...
The Stereo Mix of those VIA codecs allow ADD capture DAC signal
ad1988 and those realtek codecs can select input source from the audio mixer which can mix the input pins
- snd_hda_add_imux_item(imux, "Mix", 9, NULL);
Those static models (3stack , 3stack-dig, 6stack and 6stack-dig )still have the input source "mix"
static const struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { { "Front Mic", 0x1 }, /* port-B */ { "Line", 0x2 }, /* port-C */ { "Mic", 0x4 }, /* port-E */ { "CD", 0x5 }, { "Mix", 0x9 }, }, };
This can be added now by passing a hint "add_stereo_mix_input=true". Maybe it's better to enable it unconditionally for AD1988, as this codec is mostly used for desktops, not for laptops. (On laptops with auto-mic switching, the driver doesn't want to have capture sources other than mics.)
Takashi
>> >> >> >> >Yes, testing the latest code with real machines is
highly
>> >> >> >> >appreciated. Try sound-unstable git tree, either
master or
>> >> >> >> >test/hda-migrate branch. >> >> >> >> > git:// >> >> >
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
>> >> >> >> > >> >> >> >> >Alternatively you can build the external alsa-driver
modules from
>> >> >> >> >alsa-driver-unstable snapshot tarball, too >> >> >> >> > >> >> >> > >> >> >
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
>> >> >> >> > >> >> >> >> >Note that only the behavior of model=auto has been
changed in the tree
>> >> >> >> >above. The other model behaviors of AD codecs are
unchanged at all.
>> >> > >> >> >Seen many ad1988 controls are missing ? >> >> > >> >> >1) rear Mic, line , front Mic playback volume and switch >> >> >2) CD playback switch >> >> >3) input mix in input source >> >> >4) analog loopback mix volume and mute switch
But the number of controls is smaller than in linux stable 3.7.2. I don't know if that's a regression.
Unless you see *functional* degradation, it's no regression but regarded rather as a cleanup
Refer to commit
http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable.git;a=commitd...
The Stereo Mix of those VIA codecs allow ADD capture DAC signal
ad1988 and those realtek codecs can select input source from the audio mixer which can mix the input pins
- snd_hda_add_imux_item(imux, "Mix", 9, NULL);
Those static models (3stack , 3stack-dig, 6stack and 6stack-dig )still have the input source "mix"
static const struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { { "Front Mic", 0x1 }, /* port-B */ { "Line", 0x2 }, /* port-C */ { "Mic", 0x4 }, /* port-E */ { "CD", 0x5 }, { "Mix", 0x9 }, }, };
This can be added now by passing a hint "add_stereo_mix_input=true". Maybe it's better to enable it unconditionally for AD1988, as this codec is mostly used for desktops, not for laptops. (On laptops with auto-mic switching, the driver doesn't want to have capture sources other than mics.)
When there is no DAC connected to aa-mix , it is just "Input Mix" instead of "Stereo Mix"
Those "Stereo Mix" of conexant codecs used in notebook are disabled
At Thu, 17 Jan 2013 14:40:16 +0800, Raymond Yau wrote:
> >> >> >> >> >Yes, testing the latest code with real machines is
highly
> >> >> >> >> >appreciated. Try sound-unstable git tree, either
master or
> >> >> >> >> >test/hda-migrate branch. > >> >> >> >> > git:// > >> >> >
git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
> >> >> >> >> > > >> >> >> >> >Alternatively you can build the external alsa-driver
modules from
> >> >> >> >> >alsa-driver-unstable snapshot tarball, too > >> >> >> >> > > >> >> >> > > >> >> >
ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
> >> >> >> >> > > >> >> >> >> >Note that only the behavior of model=auto has been
changed in the tree
> >> >> >> >> >above. The other model behaviors of AD codecs are
unchanged at all.
> >> >> > > >> >> >Seen many ad1988 controls are missing ? > >> >> > > >> >> >1) rear Mic, line , front Mic playback volume and switch > >> >> >2) CD playback switch > >> >> >3) input mix in input source > >> >> >4) analog loopback mix volume and mute switch
But the number of controls is smaller than in linux stable 3.7.2. I don't know if that's a regression.
Unless you see *functional* degradation, it's no regression but regarded rather as a cleanup
Refer to commit
http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable.git;a=commitd...
The Stereo Mix of those VIA codecs allow ADD capture DAC signal
ad1988 and those realtek codecs can select input source from the audio mixer which can mix the input pins
- snd_hda_add_imux_item(imux, "Mix", 9, NULL);
Those static models (3stack , 3stack-dig, 6stack and 6stack-dig )still have the input source "mix"
static const struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { { "Front Mic", 0x1 }, /* port-B */ { "Line", 0x2 }, /* port-C */ { "Mic", 0x4 }, /* port-E */ { "CD", 0x5 }, { "Mix", 0x9 }, }, };
This can be added now by passing a hint "add_stereo_mix_input=true". Maybe it's better to enable it unconditionally for AD1988, as this codec is mostly used for desktops, not for laptops. (On laptops with auto-mic switching, the driver doesn't want to have capture sources other than mics.)
When there is no DAC connected to aa-mix , it is just "Input Mix" instead of "Stereo Mix"
"Stereo" doesn't mean PCM. Or maybe we just drop "Stereo" word there.
Takashi
I was finally able to do some testing on my board (Asus Rampage II Gene). Using latest sound-unstable I can confirm that the crashes in XFCE mixer are gone. Also, labels in alsamixer make more sense now - I have microphone plugged to the back and previously I had to enable Front Mic as an input source to for the microphone to work. Now, it works over "Rear Mic" as expected. Also, multistream playback works with "Independent HP" enabled. And yes, less controls then before ... Overall, a big improvement.
My alsa-info can be found here: http://pastebin.com/RaxrnZKK
Should I try with branch 'test/hda-gen-parser' (what is the difference, BTW?)
Anything else to test?
When can we expect patches from sound-unstable to appear in main kernel?
Thanks, Miro
At Thu, 17 Jan 2013 03:10:04 -0500, Miro Hodak wrote:
I was finally able to do some testing on my board (Asus Rampage II Gene). Using latest sound-unstable I can confirm that the crashes in XFCE mixer are gone. Also, labels in alsamixer make more sense now - I have microphone plugged to the back and previously I had to enable Front Mic as an input source to for the microphone to work. Now, it works over "Rear Mic" as expected. Also, multistream playback works with "Independent HP" enabled. And yes, less controls then before ... Overall, a big improvement.
Thanks for testing. Good to know that it works for you.
My alsa-info can be found here: http://pastebin.com/RaxrnZKK
Should I try with branch 'test/hda-gen-parser' (what is the difference, BTW?)
No, it's not needed. test/hda-gen-parser branch contains the development of the generic parser and Realtek codec driver, but it doesn't include the migration patches for other codec drivers to use the generic parser. So, on this branch, AD codec still uses the old driver.
Anything else to test?
That's enough so far. Let us know if you find any regressions.
When can we expect patches from sound-unstable to appear in main kernel?
Yes, the plan is to merge the stuff to 3.9 kernel.
But currently I'm not sure whether we can merge the whole patches. It's a fairly big change, and we need a bit more test coverage for codecs like Conexant, Cirrus and VIA. But, judging from your test result, we can merge the Analog codec patch, at least.
thanks,
Takashi
On Thu, Jan 17, 2013 at 4:04 AM, Takashi Iwai tiwai@suse.de wrote:
Yes, the plan is to merge the stuff to 3.9 kernel.
But currently I'm not sure whether we can merge the whole patches. It's a fairly big change, and we need a bit more test coverage for codecs like Conexant, Cirrus and VIA. But, judging from your test result, we can merge the Analog codec patch, at least.
thanks,
Takashi
Meging the Analog codec soon will be certainly appreciated if it makes sense for you.
Out of curiosity, with Indendent HP on, how can one send sound from, say, vlc, to HP? Tried to google, but could not find the answer.
At Thu, 17 Jan 2013 04:24:44 -0500, Miro Hodak wrote:
On Thu, Jan 17, 2013 at 4:04 AM, Takashi Iwai tiwai@suse.de wrote:
Yes, the plan is to merge the stuff to 3.9 kernel.
But currently I'm not sure whether we can merge the whole patches. It's a fairly big change, and we need a bit more test coverage for codecs like Conexant, Cirrus and VIA. But, judging from your test result, we can merge the Analog codec patch, at least.
thanks,
Takashi
Meging the Analog codec soon will be certainly appreciated if it makes sense for you.
Out of curiosity, with Indendent HP on, how can one send sound from, say, vlc, to HP? Tried to google, but could not find the answer.
In the independent HP mode, it's output from the secondary analog PCM (typically hw:2):
% aplay -Dplughw:2 something.wav
Takashi
On Thu, Jan 17, 2013 at 4:29 AM, Takashi Iwai tiwai@suse.de wrote:
In the independent HP mode, it's output from the secondary analog PCM (typically hw:2):
% aplay -Dplughw:2 something.wavTakashi
hw:2 does not work, aplay complains (Invalid value for card). How to find out what its called? Should I see the secondary analog PCM in alsamixer?
At Thu, 17 Jan 2013 04:42:40 -0500, Miro Hodak wrote:
[1 <text/plain; ISO-8859-1 (7bit)>] On Thu, Jan 17, 2013 at 4:29 AM, Takashi Iwai tiwai@suse.de wrote:
In the independent HP mode, it's output from the secondary analog PCM (typically hw:2):
% aplay -Dplughw:2 something.wavTakashi
hw:2 does not work, aplay complains (Invalid value for card). How to find out what its called? Should I see the secondary analog PCM in alsamixer?
Did you try plughw:2? Also, this stream is available only when independent HP mode is enabled. When disabled, PCM open returns -EBUSY error.
Takashi
On Thu, Jan 17, 2013 at 4:55 AM, Takashi Iwai tiwai@suse.de wrote:
Did you try plughw:2? Also, this stream is available only when independent HP mode is enabled. When disabled, PCM open returns -EBUSY error.
Takashi
Do you mean plughw:0,2? Because that one works with aplayer (with Independent HP enabled and Auto-Mute disabled).
Do other players (vlc, mplayer, etc.) support choice of PCM or are they hardcoded to the primary one?
At Thu, 17 Jan 2013 05:17:07 -0500, Miro Hodak wrote:
[1 <text/plain; ISO-8859-1 (7bit)>] On Thu, Jan 17, 2013 at 4:55 AM, Takashi Iwai tiwai@suse.de wrote:
Did you try plughw:2? Also, this stream is available only when independent HP mode is enabled. When disabled, PCM open returns -EBUSY error.
Takashi
Do you mean plughw:0,2? Because that one works with aplayer (with Independent HP enabled and Auto-Mute disabled).
Damn, yes, of course, it's plughw:0,2. Sorry for confusion.
Do other players (vlc, mplayer, etc.) support choice of PCM or are they hardcoded to the primary one?
You should be able to pass via option or config.
Takashi
On Thu, Jan 17, 2013 at 5:21 AM, Takashi Iwai tiwai@suse.de wrote:
At Thu, 17 Jan 2013 05:17:07 -0500,
Miro Hodak wrote:
Do you mean plughw:0,2? Because that one works with aplayer (with Independent HP enabled and Auto-Mute disabled).
Damn, yes, of course, it's plughw:0,2. Sorry for confusion.
Do other players (vlc, mplayer, etc.) support choice of PCM or are they hardcoded to the primary one?
You should be able to pass via option or config.
Takashi
OK, now as far as Auto-Mute goes, should it do what expected when enabled - i.e. muting the speakers when headphones are connected to the front panel? I have tried that, does not seem to work.
At Thu, 17 Jan 2013 05:32:03 -0500, Miro Hodak wrote:
On Thu, Jan 17, 2013 at 5:21 AM, Takashi Iwai tiwai@suse.de wrote:
At Thu, 17 Jan 2013 05:17:07 -0500,
Miro Hodak wrote:
Do you mean plughw:0,2? Because that one works with aplayer (with Independent HP enabled and Auto-Mute disabled).
Damn, yes, of course, it's plughw:0,2. Sorry for confusion.
Do other players (vlc, mplayer, etc.) support choice of PCM or are they hardcoded to the primary one?
You should be able to pass via option or config.
Takashi
OK, now as far as Auto-Mute goes, should it do what expected when enabled - i.e. muting the speakers when headphones are connected to the front panel? I have tried that, does not seem to work.
With independent HP mode on or off? If the auto-muting the speaker doesn't work even with the indep HP off, it's likely an issue of the unsolicited events.
Takashi
OK, now as far as Auto-Mute goes, should it do what expected when enabled
- i.e. muting the speakers when headphones are connected to the front panel? I have tried that, does not seem to work.
Do your computer chassis have a hda front audio panel with Jack detection ?
The front headphone Jack should return true when headphone Jack is plugged
control.35 { iface CARD name 'Front Headphone Jack' value false comment { access read type BOOLEAN count 1 } }
All ADI codecs support impedance measurement
Try the modified hda-jack-sense-test which need hda-analyzer
http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115
On Thu, Jan 17, 2013 at 7:49 AM, Raymond Yau superquad.vortex2@gmail.comwrote:
OK, now as far as Auto-Mute goes, should it do what expected when
enabled - i.e. muting the speakers when headphones are connected to the front panel? I have tried that, does not seem to work.
Do your computer chassis have a hda front audio panel with Jack detection ?
The front headphone Jack should return true when headphone Jack is plugged
control.35 { iface CARD name 'Front Headphone Jack'
value false comment { access read type BOOLEAN count 1 } }
All ADI codecs support impedance measurement
Try the modified hda-jack-sense-test which need hda-analyzer
http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115
Hmmm, ran into some python issues. Even hda-analyzer itself is not working: ... File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile raise ValueError, "wrong proc file format (%s)" % msg ValueError: wrong proc file format (integer decode 'State of AFG node 0x01:' ('Node '))
Any ideas?
OK, now as far as Auto-Mute goes, should it do what expected when
enabled - i.e. muting the speakers when headphones are connected to the front panel? I have tried that, does not seem to work.
Do your computer chassis have a hda front audio panel with Jack
detection ?
The front headphone Jack should return true when headphone Jack is
plugged
control.35 { iface CARD name 'Front Headphone Jack'
value false comment { access read type BOOLEAN count 1 } }
All ADI codecs support impedance measurement
Try the modified hda-jack-sense-test which need hda-analyzer
http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115
Hmmm, ran into some python issues. Even hda-analyzer itself is not
working:
... File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile raise ValueError, "wrong proc file format (%s)" % msg ValueError: wrong proc file format (integer decode 'State of AFG node
0x01:' ('Node '))
Any ideas?
This patch add lines which make the parser of hda_proc.py fail
http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=f4f...
On 01/17/2013 04:29 PM, Raymond Yau wrote:
File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile raise ValueError, "wrong proc file format (%s)" % msg ValueError: wrong proc file format (integer decode 'State of AFG node
0x01:' ('Node '))
Any ideas?
This patch add lines which make the parser of hda_proc.py fail
http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=f4f...
Sigh, this is not the first time new stuff in the proc file crashes hda_analyzer.
Jaroslav, can you please fix hda_analyzer so that it ignores new stuff in the proc file instead of crashing, so we don't have this problem over and over again?
At Thu, 17 Jan 2013 09:59:20 -0500, Miro Hodak wrote:
On Thu, Jan 17, 2013 at 7:49 AM, Raymond Yau superquad.vortex2@gmail.comwrote:
OK, now as far as Auto-Mute goes, should it do what expected when
enabled - i.e. muting the speakers when headphones are connected to the front panel? I have tried that, does not seem to work.
Do your computer chassis have a hda front audio panel with Jack detection ?
The front headphone Jack should return true when headphone Jack is plugged
control.35 { iface CARD name 'Front Headphone Jack'
value false comment { access read type BOOLEAN count 1 } }
All ADI codecs support impedance measurement
Try the modified hda-jack-sense-test which need hda-analyzer
http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115
Hmmm, ran into some python issues. Even hda-analyzer itself is not working: ... File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile raise ValueError, "wrong proc file format (%s)" % msg ValueError: wrong proc file format (integer decode 'State of AFG node 0x01:' ('Node '))
Ah, it's a new proc file entry that has been added recently. hda-analyzer needs to be fixed.
In anyway, only for checking the presence unsolicited events, you can do without extra program. Suppose you built your kernel with the tracing support. See "Tracepoints" option in Documentation/sound/alsa/HD-Audio.txt.
# echo 1 > /sys/kernel/debug/tracing/events/hda/hda_unsol_event/enable
(plug and unplug your headphone jacks)
# cat /sys/kernel/debug/tracing/trace
If you see some hda_unsol_event lines there, it implies that the jack detection basically works in hardware level.
Takashi
El jue, 17 ene 2013 a las 16:32 horas Takashi Iwai escribió:
At Thu, 17 Jan 2013 09:59:20 -0500, Miro Hodak wrote:
On Thu, Jan 17, 2013 at 7:49 AM, Raymond Yau superquad.vortex2@gmail.comwrote:
OK, now as far as Auto-Mute goes, should it do what expected when
enabled - i.e. muting the speakers when headphones are connected to the front panel? I have tried that, does not seem to work.
Do your computer chassis have a hda front audio panel with Jack detection ?
The front headphone Jack should return true when headphone Jack is plugged
control.35 { iface CARD name 'Front Headphone Jack'
value false comment { access read type BOOLEAN count 1 } }
All ADI codecs support impedance measurement
Try the modified hda-jack-sense-test which need hda-analyzer
http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115
Hmmm, ran into some python issues. Even hda-analyzer itself is not working: ... File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile raise ValueError, "wrong proc file format (%s)" % msg ValueError: wrong proc file format (integer decode 'State of AFG node 0x01:' ('Node '))
Ah, it's a new proc file entry that has been added recently. hda-analyzer needs to be fixed.
In anyway, only for checking the presence unsolicited events, you can do without extra program. Suppose you built your kernel with the tracing support. See "Tracepoints" option in Documentation/sound/alsa/HD-Audio.txt.
# echo 1 > /sys/kernel/debug/tracing/events/hda/hda_unsol_event/enable
(plug and unplug your headphone jacks)
# cat /sys/kernel/debug/tracing/trace
If you see some hda_unsol_event lines there, it implies that the jack detection basically works in hardware level.
Takashi
I Suppose that when auto-mute is enabled an HP independent is disabled, inserting the HP jack should disable sound from front speakers. If so then auto-mute doesn't work. Jack insertion/extraction events are well detected for front speakers and HP.
#~ cat /sys/kernel/debug/tracing/trace # tracer: nop # # entries-in-buffer/entries-written: 4/4 #P:2 # # _-----=> irqs-off # / _----=> need-resched # | / _---=> hardirq/softirq # || / _--=> preempt-depth # ||| / delay # TASK-PID CPU# |||| TIMESTAMP FUNCTION # | | | |||| | | <idle>-0 [001] d.h. 137.881919: hda_unsol_event: [0] res=8000000, res_ex=10 <idle>-0 [001] d.h. 162.511102: hda_unsol_event: [0] res=8000000, res_ex=10 <idle>-0 [001] d.h. 246.529813: hda_unsol_event: [0] res=4000000, res_ex=10 <idle>-0 [001] d.h. 272.381719: hda_unsol_event: [0] res=4000000, res_ex=10
branch tested: test/hda-gen-parser
commit 6f4c53a1467c51966e06a1d29c6f0fee85d5c6f6 Merge: 920544e 29476558 Author: Takashi Iwai tiwai@suse.de Date: Thu Jan 17 09:56:33 2013 +0100
Regards,
El jue, 17 ene 2013 a las 16:32 horas Takashi Iwai escribió:
In anyway, only for checking the presence unsolicited events, you can do without extra program. Suppose you built your kernel with the tracing support. See "Tracepoints" option in Documentation/sound/alsa/HD-Audio.txt.
# echo 1 > /sys/kernel/debug/tracing/events/hda/hda_unsol_event/enable
(plug and unplug your headphone jacks)
# cat /sys/kernel/debug/tracing/trace
If you see some hda_unsol_event lines there, it implies that the jack detection basically works in hardware level.
Takashi
My test:
HP Independent disabled, Auto-Mute enabled. When plugging front headphones, sound is in both devices. Yet tracing shows that jack detection is being picked up (one plug and unplug event):
miro2 miro # cat /sys/kernel/debug/tracing/trace # tracer: nop # # entries-in-buffer/entries-written: 2/2 #P:8 # # _-----=> irqs-off # / _----=> need-resched # | / _---=> hardirq/softirq # || / _--=> preempt-depth # ||| / delay # TASK-PID CPU# |||| TIMESTAMP FUNCTION # | | | |||| | | <idle>-0 [000] d.h. 339.442694: hda_unsol_event: [0] res=4000000, res_ex=10 <idle>-0 [000] d.h. 345.727502: hda_unsol_event: [0] res=4000000, res_ex=10
So, somehow auto-muting appears non-functional ...
We're working to build a consistent Linux audio environment for our embedded audio product,with known versions of both ALSA and the kernel (we're specifically considering kernel 3.6.6). We are not clear, however, exactly what version of the ALSA kernel-space tools are included with the kernel, nor are we clear on exactly what each ALSA package is used for.
In order to simplify any responses, I'll give my best guess as to what's included in the kernel and each ALSA package: let me know where I went wrong.
So, am I right that...
- Kernel 3.6.6 includes exactly alsa-driver-1.0.25 (I'm actually guessing not, but let me know.)
- alsa-driver-1.0.25 is all included in the kernel release, and can be ignored
- alsa-lib is the user-space library that we complile our tools against, and that communicates with ALSA in the kernel through the device and /proc tree (required)
- alsa-utils are various generic ALSA command line tools, such as amixer, aplay, alsaconf, etc (required)
- alsa-tools are various more obscure tools and loaders (probably not required)
- alsa-firmware are binary drivers for various third-party products (probably not required)
- alsa-plugins are plugins for various ALSA needs (e.g. Jack) (probably not required)
- alsa-oss is the OSS compatibilty layer (not required)
- pyalsa is the Python bindings for ALSA (not required)
And, some final questions:
- What version of ALSA and the kernel are needed to fully support implicit feedback?
- Given our audio focus, what version kernel would you recommend?
Thanks, Dan
At Thu, 17 Jan 2013 19:12:24 +0100, Manolo Díaz wrote:
El jue, 17 ene 2013 a las 16:32 horas Takashi Iwai escribió:
At Thu, 17 Jan 2013 09:59:20 -0500, Miro Hodak wrote:
On Thu, Jan 17, 2013 at 7:49 AM, Raymond Yau superquad.vortex2@gmail.comwrote:
OK, now as far as Auto-Mute goes, should it do what expected when
enabled - i.e. muting the speakers when headphones are connected to the front panel? I have tried that, does not seem to work.
Do your computer chassis have a hda front audio panel with Jack detection ?
The front headphone Jack should return true when headphone Jack is plugged
control.35 { iface CARD name 'Front Headphone Jack'
value false comment { access read type BOOLEAN count 1 } }
All ADI codecs support impedance measurement
Try the modified hda-jack-sense-test which need hda-analyzer
http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115
Hmmm, ran into some python issues. Even hda-analyzer itself is not working: ... File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile raise ValueError, "wrong proc file format (%s)" % msg ValueError: wrong proc file format (integer decode 'State of AFG node 0x01:' ('Node '))
Ah, it's a new proc file entry that has been added recently. hda-analyzer needs to be fixed.
In anyway, only for checking the presence unsolicited events, you can do without extra program. Suppose you built your kernel with the tracing support. See "Tracepoints" option in Documentation/sound/alsa/HD-Audio.txt.
# echo 1 > /sys/kernel/debug/tracing/events/hda/hda_unsol_event/enable
(plug and unplug your headphone jacks)
# cat /sys/kernel/debug/tracing/trace
If you see some hda_unsol_event lines there, it implies that the jack detection basically works in hardware level.
Takashi
I Suppose that when auto-mute is enabled an HP independent is disabled, inserting the HP jack should disable sound from front speakers. If so then auto-mute doesn't work. Jack insertion/extraction events are well detected for front speakers and HP.
#~ cat /sys/kernel/debug/tracing/trace # tracer: nop # # entries-in-buffer/entries-written: 4/4 #P:2 # # _-----=> irqs-off # / _----=> need-resched # | / _---=> hardirq/softirq # || / _--=> preempt-depth # ||| / delay # TASK-PID CPU# |||| TIMESTAMP FUNCTION # | | | |||| | | <idle>-0 [001] d.h. 137.881919: hda_unsol_event: [0] res=8000000, res_ex=10 <idle>-0 [001] d.h. 162.511102: hda_unsol_event: [0] res=8000000, res_ex=10 <idle>-0 [001] d.h. 246.529813: hda_unsol_event: [0] res=4000000, res_ex=10 <idle>-0 [001] d.h. 272.381719: hda_unsol_event: [0] res=4000000, res_ex=10
branch tested: test/hda-gen-parser
Please test with either master or test/hda-migrate branch. test/hda-gen-parser branch doesn't contain the patch for AD codecs.
In anyway, please attach alsa-info.sh outputs before and after plugging the headphone with the latest sound-unstable tree (and a proper branch).
thanks,
Takashi
I was finally able to do some testing on my board (Asus Rampage II Gene).
Using latest sound-unstable I can confirm that the crashes in XFCE mixer are gone. Also, labels in alsamixer make more sense now - I have microphone plugged to the back and previously I had to enable Front Mic as an input source to for the microphone to work. Now, it works over "Rear Mic" as expected. Also, multistream playback works with "Independent HP" enabled. And yes, less controls then before ... Overall, a big improvement.
My alsa-info can be found here: http://pastebin.com/RaxrnZKK
Should I try with branch 'test/hda-gen-parser' (what is the difference,
BTW?)
Anything else to test?
Does the Mic really work since missing boost control ?
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
Take a diff of alsa-info the auto model and 6stack-dig
At Mon, 14 Jan 2013 17:23:54 +0100, Manolo Díaz wrote:
El lun, 14 ene 2013 a las 14:03 horas Takashi Iwai escribió:
Yes, testing the latest code with real machines is highly appreciated. Try sound-unstable git tree, either master or test/hda-migrate branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git
Alternatively you can build the external alsa-driver modules from alsa-driver-unstable snapshot tarball, too ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.tar.gz
Note that only the behavior of model=auto has been changed in the tree above. The other model behaviors of AD codecs are unchanged at all.
Mixer crashes are gone. As I wrote I have no way to test the digital output.
I haven't found a way to test multi stream playback either. After choosing HP Independant enabled I can't hear nothing through the headphone.
This is the feature. The independent HP mode means that the headphone output comes from the individual stream assigned to the secondary analog PCM.
Takashi
participants (7)
-
Clemens Ladisch -
Daniel Griscom -
David Henningsson -
Manolo Díaz -
Miro Hodak -
Raymond Yau -
Takashi Iwai