My question is how did you check that the 32bit format is used. And, don't do top-posting.
Takashi
Sorry, i am unsure how to not top post when replying via email.
With capture and playback .formats set to SNDRV_PCM_FMTBIT_S24_3LE, jack log shows:
ALSA: final selected sample format for playback: 32bit integer little-endian
and
/proc/asound/R16/stream0 shows:
Playback: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 44100, 48000, 88200, 96000 Data packet interval: 125 us
Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 8 Endpoint: 4 IN (SYNC) Rates: 44100, 48000, 88200, 96000 Data packet interval: 125 us
On Mar 11, 2014 3:09 AM, "Takashi Iwai" tiwai@suse.de wrote:
At Mon, 10 Mar 2014 16:44:19 -0400, Jason Mancine wrote:
Yes, I have tried those .formats with no luck...it still
initializes at
32
How did you check it?
Takashi
On Mar 10, 2014 4:29 PM, "Alan Horstmann" gineera@aspect135.co.uk
wrote:
On Monday 10 March 2014 14:36, Jason Mancine wrote:
I am still working on trying to get the R16 to work for
playback.
So, the main question is how do I force ALSA to initialize this
device at
24 bit integer?
Isn't the answer in Takashi's original reply...?
> On Dec 6, 2013 11:25 AM, "Takashi Iwai" tiwai@suse.de
wrote:
>> Do you mean the 24bit physical size, i.e. each frame is
packed in
3
>> bytes? If so, you used a wrong format.
SNDRV_PCM_FMTBIT_S24_LE
is
>> for 24bit format packed in 32bit frame. If you need a
3-bytes
frame,
>> use SNDRV_PCM_FMTBIT_S24_3LE instead.
Many USB devices have this...
.type =
QUIRK_AUDIO_FIXED_ENDPOINT,
.data = & (const struct
audioformat)
{
.formats =
SNDRV_PCM_FMTBIT_S24_LE,
.formats =
SNDRV_PCM_FMTBIT_S24_3LE,
instead. That would apply for capture also. Or is that what you
have
been
trying?
Regards
Alan
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
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