hwparams_copy needs to be reset (as it is also reset for the third and
fourth try) before the second try.
If the reset is not done and the first try fails:
D: [lt-pulseaudio] alsa-util.c: Maximum hw buffer size is 743 ms
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_period_size_near() failed: Invalid argument
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
D: [lt-pulseaudio] alsa-util.c: Set only period size (to 1102 samples).
We have three failures and finally the fourth (only period size) succeed.
With this patch:
D: [lt-pulseaudio] alsa-util.c: Maximum hw buffer size is 743 ms
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
D: [lt-pulseaudio] alsa-util.c: Set period size first (to 1102 samples), buffer size second (to 4408 samples).
We only fail with the first try, the second (period followed by buffer) is
fine.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
This fixes a case where pa__done() is called while
AVAHI_MESSAGE_PUBLISH_ALL is waiting for processing. The
pa_asyncmsgq_wait_for(AVAHI_MESSAGE_SHUTDOWN_COMPLETE) call will
process all pending messages, and processing AVAHI_MESSAGE_PUBLISH_ALL
causes publish_all_services(), and that in turn accesses u->services,
which has been already freed at this point. If we are shutting down,
we shouldn't react to any of the messages that the Avahi thread is
sending to the main thread.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=76184
When given an explicit device.description in card_properties, prefer
this information over other default prefixes (e.g. 'Built-in Audio')
when constructing sink/source descriptions.
For example, if I manually configure the card description to be
"FooBar", I then expect that the sinks and created by the card also
have "FooBar" in their description instead of generic "Built-in
Audio".
In some cases, "Analog Input" could show up as well as
"Headset Mic" (or "Headphone Mic"), because I forgot to add the
relevant "required-absent" lines when I added the headset mic path.
As a result, both "Analog Input" and "Headset Mic" showed up on the
Logitech USB 530 Headset.
Reported-by: Steve Magoun <steve.magoun@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
CC modules/module_tunnel_sink_la-module-tunnel.lo
modules/module-tunnel-source-new.c: In function 'read_new_samples':
modules/module-tunnel-source-new.c:145:16: warning: declaration of 'read' shadows a global declaration [-Wshadow]
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
If mixer_handle is not NULL, then hctl_handle won't be NULL either.
The redundant check was confusing, because it looked like we would
leak the mixer_handle if mixer_handle is non-NULL and hctl_handle is
NULL.
Currently the latency information is being updated based on the encoded
SBC data instead of the decoded PCM data. Fixing this required moving
the timing update to be after the packet has been decoded.
The Nokia E7 running Symbian Belle Refresh seems to generate invalid SBC
packets every few minutes. This causes pulseaudio to disconnect the
stream and log "SBC decoding error (-3)".
If a single packet is bad, pulseaudio should keep playing the stream.
Some people want module-rtp-send to send silence when the sink that is
monitored goes idle, and some people want module-rtp-send to pause the
RTP stream to avoid unnecessary bandwidth consumption.
If a stream is started corked and remains corked, the sink/source
remained idle without being properly suspended. This patch fixes
that issue.
BugLink: https://bugs.launchpad.net/bugs/1284415
Tested-by: Ricardo Salveti <ricardo.salveti@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Steps to reproduce:
1) Leave LFE remixing disabled (the default)
2) Start playback of stereo material on e g 5.1 surround, notice nothing in LFE
3) Now change profile to e g 4.0 surround and then back to 5.1 surround
4) Notice that LFE channel is now remixed
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This makes sure that there is no window between pa_sink/source_new()
and _put() where enumerating sinks/sources causes an assert (several
calls in sink/source_get_info need a linked sink or source).
A segfault was reported on this line:
pa_return_val_if_fail(PA_SINK_IS_LINKED(pa_sink_get_state(data->sink)), -PA_ERR_BADSTATE);
After expanding the pa_sink_get_state() macro, the line looks like
this:
pa_return_val_if_fail(PA_SINK_IS_LINKED(data->sink->state), -PA_ERR_BADSTATE);
So data->sink was apparently NULL. That could happen if we try to fall
back to the default sink, but format negotiation fails.
This bug was introduced in commit
71816ecb7f.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=74646
This fixes a build error with mingw32:
pulsecore/.libs/libpulsecommon_4.99_la-lock-autospawn.o: In function `unref':
/home/abuild/rpmbuild/BUILD/pulseaudio-4.99.2/src/pulsecore/lock-autospawn.c:123: undefined reference to `pa_thread_free_nojoin'
collect2: error: ld returned 1 exit status
pa_thread_free_nojoin() was initially only implemented for the pthread
based pa_thread backend, because it was incorrectly assumed that
autospawning (the only user of pa_thread_free_nojoin()) is not used on
Windows.
Reported-By: Michael DePaulo <mikedep333@gmail.com>
Reintroduces a cleaned-up version of commit 30ce3a14e5 which
was reverted by 1ce71cbd82; for more information see
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/17479/focus=17487
The patch intends to reduce computational load when resampling AND remapping. The PA
resampler performs the following steps:
sample format conversion -> remapping -> resampling -> sample format conversion
In case the number of output channels is higher than the number of input channels, the
resampler has to be run more often than necessary. E.g. in case of mono to 4-channel remapping,
the resampler runs on 4 channels separately.
To ímprove this, the PA resampler pipeline is made adaptive:
if out-channels <= in-channels:
sample format conversion -> remapping -> resampling -> sample format conversion
if out-channels > in-channels:
sample format conversion -> resampling -> remapping -> sample format conversion
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
