[alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4

Takashi Iwai tiwai at suse.de
Fri Jan 18 17:58:36 CET 2013


At Fri, 18 Jan 2013 17:48:23 +0100,
Manolo Díaz wrote:
> 
> El vie, 18 ene 2013 a las 16:30 horas
> Takashi Iwai escribió:
> 
> >At Fri, 18 Jan 2013 22:49:37 +0800,
> >Raymond Yau wrote:
> >> 
> >> > > It's already in the repository. Now none of the input sources work for
> >> > > me: front-mic, rear-mic nor input line. Alsa-info output is attached.
> >> > >
> >> > > commit 77ecb70ef5b022a1ee80169583753d85d7a9c396
> >> >
> >> > Hmm, through a quick glance, all look OK.
> >> >
> >> 
> >> It is strange that three input source are line but audio selector are not
> >> the same
> >> 
> >> Simple mixer control 'Input Source',0
> >>   Capabilities: cenum
> >>   Items: 'Front Mic' 'Rear Mic' 'Line' 'CD'
> >>   Item0: 'Line'
> >> Simple mixer control 'Input Source',1
> >>   Capabilities: cenum
> >>   Items: 'Front Mic' 'Rear Mic' 'Line' 'CD'
> >>   Item0: 'Line'
> >> Simple mixer control 'Input Source',2
> >>   Capabilities: cenum
> >>   Items: 'Front Mic' 'Rear Mic' 'Line' 'CD'
> >>   Item0: 'Line'
> >> 
> >> 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
> >>   Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1
> >>   Amp-Out vals:  [0x36 0x36]
> >>   Connection: 11
> >>      0x38 0x39 0x3a* 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 0x1f
> >> Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out
> >>   Control: name="Capture Volume", index=1, device=0
> >>     ControlAmp: chs=3, dir=Out, idx=0, ofs=0
> >>   Control: name="Capture Switch", index=1, device=0
> >>     ControlAmp: chs=3, dir=Out, idx=0, ofs=0
> >>   Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1
> >>   Amp-Out vals:  [0x36 0x36]
> >>   Connection: 10
> >>      0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20
> >> Node 0x0e [Audio Selector] wcaps 0x30010d: Stereo Amp-Out
> >>   Control: name="Capture Volume", index=2, device=0
> >>     ControlAmp: chs=3, dir=Out, idx=0, ofs=0
> >>   Control: name="Capture Switch", index=2, device=0
> >>     ControlAmp: chs=3, dir=Out, idx=0, ofs=0
> >>   Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1
> >>   Amp-Out vals:  [0x36 0x36]
> >>   Connection: 10
> >>      0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20
> >
> >Good catch.  It's a recent regression.
> >I fixed now with the patch below.
> >
> >test/hda-gen-parser and master branches are updated now with this and
> >other fixes.
> >
> >
> >thanks,
> >
> >Takashi
> >
> >---
> >From: Takashi Iwai <tiwai at suse.de>
> >Subject: [PATCH] ALSA: hda - Fix the wrong adc_idx for capture source
> >
> >The patch "ALSA: hda - fix wrong adc_idx in generic parser" fixed the
> >adc_idx for the capture volume and capture switch controls.  But also
> >modified the adc_idx retrieval for the capture source controls
> >wrongly.  As multiple capture source controls are created in a single
> >shot with counts > 1, the id.index doesn't contain the real value.
> >The real index has to be taken via snd_ctl_get_ioffidx() as in the
> >original code.
> >
> >This patch reverts the fixes partially to recover from the
> >regression.
> >
> >Signed-off-by: Takashi Iwai <tiwai at suse.de>
> >---
> > sound/pci/hda/hda_generic.c | 5 +++--
> > 1 file changed, 3 insertions(+), 2 deletions(-)
> >
> >diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
> >index e4e71fa..29f37c9 100644
> >--- a/sound/pci/hda/hda_generic.c
> >+++ b/sound/pci/hda/hda_generic.c
> >@@ -2675,7 +2675,8 @@ static int mux_enum_get(struct snd_kcontrol *kcontrol,
> > {
> > 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
> > 	struct hda_gen_spec *spec = codec->spec;
> >-	unsigned int adc_idx = kcontrol->id.index;
> >+	/* the ctls are created at once with multiple counts */
> >+	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
> > 
> > 	ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
> > 	return 0;
> >@@ -2685,7 +2686,7 @@ static int mux_enum_put(struct snd_kcontrol *kcontrol,
> > 			    struct snd_ctl_elem_value *ucontrol)
> > {
> > 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
> >-	unsigned int adc_idx = kcontrol->id.index;
> >+	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
> > 	return mux_select(codec, adc_idx,
> > 			  ucontrol->value.enumerated.item[0]);
> > }
> 
> The problem persists after commit d821c1ef2c8ada02f1feada071a37ced69b300fe,
> master branch
> 
> arecord -fdat -Dplughw:0 -vv foo.wav
> Recording WAVE 'foo.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
> Plug PCM: Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0
> Its setup is:
>   stream       : CAPTURE
>   access       : RW_INTERLEAVED
>   format       : S16_LE
>   subformat    : STD
>   channels     : 2
>   rate         : 48000
>   exact rate   : 48000 (48000/1)
>   msbits       : 16
>   buffer_size  : 24064
>   period_size  : 6016
>   period_time  : 125333
>   tstamp_mode  : NONE
>   period_step  : 1
>   avail_min    : 6016
>   period_event : 0
>   start_threshold  : 1
>   stop_threshold   : 24064
>   silence_threshold: 0
>   silence_size : 0
>   boundary     : 6773413839565225984
>   appl_ptr     : 0
>   hw_ptr       : 0
> ##### +                                            | 10%^C
> 
> Playing the foo.wav file I can only hear noise.

You chose "Rear Mic" only in the third capture source.
There are three "Input Source" controls, and the first one corresponds
to the primary recording stream.

> Simple mixer control 'Input Source',0
>   Capabilities: cenum
>   Items: 'Front Mic' 'Rear Mic' 'Line' 'CD'
>   Item0: 'Line'
> Simple mixer control 'Input Source',1
>   Capabilities: cenum
>   Items: 'Front Mic' 'Rear Mic' 'Line' 'CD'
>   Item0: 'Front Mic'
> Simple mixer control 'Input Source',2
>   Capabilities: cenum
>   Items: 'Front Mic' 'Rear Mic' 'Line' 'CD'
>   Item0: 'Rear Mic'

BTW, you should turn down the "Digital" capture volume to the half
(0dB).  It's an artificial gain in software, so at best keep it in
0dB.

	% amixer -c0 set "Digital" 0dB


Takashi


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