https://bugs.freedesktop.org/show_bug.cgi?id=93637
GitLab Migration User changed:
What|Removed |Added
Resolution|--- |MOVED
Status|NEW
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #29 from Raymond ---
http://git.alsa-project.org/?p=alsa-plugins.git;a=blob;f=a52/pcm_a52.c;hb=HEAD
tatic int a52_set_hw_constraint(struct a52_ctx *rec)
{
static unsigned int accesses[] = {
https://bugs.freedesktop.org/show_bug.cgi?id=93637
5...@idlegandalf.com changed:
What|Removed |Added
CC||5...@idlegandalf.com
--- Comment
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #24 from Tom Yan ---
(In reply to Raymond from comment #21)
> How did pulseaudio create headphone1 and headphone2 ports?
>
> ports: analog-output;output-speaker:
> Analog Output / Speaker (priority 9910,
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #25 from Tom Yan ---
(In reply to Raymond from comment #22)
> aplay use buffer size as start threshold but pulseaudio use -1
>
> pcm start after three periods have written, this mean you have at least
> written
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #26 from Tom Yan ---
(In reply to Raymond from comment #23)
> What are the values of hw_ptr and appl_ptr in proc/sound?
`find /proc/asound/ -name *ptr*` returns nothing. And there's no /proc/sound.
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https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #27 from Raymond ---
http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/aliases.conf;hb=HEAD
http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/CMI8788.conf;hb=HEAD
You
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #23 from Raymond ---
What are the values of hw_ptr and appl_ptr in proc/sound?
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https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #21 from Raymond ---
How did pulseaudio create headphone1 and headphone2 ports?
ports: analog-output;output-speaker:
Analog Output / Speaker (priority 9910, latency offset 0 usec, available:
unknown)
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #22 from Raymond ---
aplay use buffer size as start threshold but pulseaudio use -1
pcm start after three periods have written, this mean you have at least
written 20ms data
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #18 from Tom Yan ---
(In reply to Raymond from comment #17)
> http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/alsa-
> sink.c
>
> if (u->first) {
> pa_log_info("Starting playback.");
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #13 from Tom Yan ---
Created attachment 120917
--> https://bugs.freedesktop.org/attachment.cgi?id=120917=edit
.asoundrc having the "type a52" pcm wrapped by a "type plug" pcm
Not sure what you mean, but I got
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #14 from Tom Yan ---
(In reply to Raymond from comment #10)
> I: [pulseaudio] alsa-sink.c: Using 3.0 fragments of size 18432 bytes
> (32.00ms), buffer size is 55296 bytes (96.00ms)
> I: [pulseaudio] alsa-sink.c:
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #15 from Tom Yan ---
(In reply to Raymond from comment #11)
> do a52 plugin support non interleaved access mode ?
>
> D: [pulseaudio] alsa-util.c: Slave: A52 Output Plugin
> D: [pulseaudio] alsa-util.c: Its setup
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #16 from Tom Yan ---
Hmm seems like PulseAudio will use a "plug" pcm to wrap it anyway because a52
plugin does not support interleaved access but pulse does not support
non-interleaved access, and pulse is aware
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #17 from Raymond ---
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/alsa-sink.c
if (u->first) {
pa_log_info("Starting playback.");
snd_pcm_start(u->pcm_handle);
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #11 from Raymond ---
do a52 plugin support non interleaved access mode ?
D: [pulseaudio] alsa-util.c: Slave: A52 Output Plugin
D: [pulseaudio] alsa-util.c: Its setup is:
D: [pulseaudio] alsa-util.c:
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #10 from Raymond ---
I: [pulseaudio] alsa-sink.c: Using 3.0 fragments of size 18432 bytes (32.00ms),
buffer size is 55296 bytes (96.00ms)
I: [pulseaudio] alsa-sink.c: Disabling rewind for device
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #12 from Raymond ---
non interleaved access mode mean the period 1536 frames 6 channels data cannot
be rewinded
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--- Comment #8 from Tom Yan ---
(In reply to Raymond from comment #7)
> http://git.alsa-project.org/?p=alsa-plugins.git;a=blob_plain;f=doc/a52.txt;
> hb=HEAD
>
> Do pulseaudio misuse a52 plugin since it only supoort fixed
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #9 from Raymond ---
Seem use plug:a52:0 instead of a52:0
D: [pulseaudio] alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
D: [pulseaudio] alsa-util.c: Managed to open a52:0
D: [pulseaudio]
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #6 from Tom Yan ---
I am using the following in Arch Linux:
libpulse 7.1-3
pulseaudio 7.1-3
alsa-lib 1.1.0-1
alsa-plugins 1.1.0-1
alsa-utils 1.1.0-1
linux 4.3.3-2
glibc 2.22-3
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Tom Yan changed:
What|Removed |Added
Attachment #120884|pacmd list without |pacmd list with .asoundrc
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #4 from Tom Yan ---
Created attachment 120884
--> https://bugs.freedesktop.org/attachment.cgi?id=120884=edit
pacmd list without .asoundrc
[tom@localhost ~]$ diff before_list after_list
421a422,424
>
https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #1 from Tom Yan ---
Created attachment 120881
--> https://bugs.freedesktop.org/attachment.cgi?id=120881=edit
pulseaudio -vvv without the .asoundrc
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https://bugs.freedesktop.org/show_bug.cgi?id=93637
--- Comment #3 from Tom Yan ---
Created attachment 120883
--> https://bugs.freedesktop.org/attachment.cgi?id=120883=edit
pacmd list without .asoundrc
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--- Comment #2 from Tom Yan ---
Created attachment 120882
--> https://bugs.freedesktop.org/attachment.cgi?id=120882=edit
pulseaudio -vvv with .asoundrc
from starting pulse to set-card-profile and core dumped
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