Erich Boleyn <erich@uruk.org> wrote:
> Using RTP for multi-room music streaming, updated to Pulse 0.9.14 from
> 0.9.9, RTP reception new crashes with a segfault on all machines at
> the first "Updating sample rate" log message.
>
> Source of the segfault appears to be null pointer for
> "impl_update_rates" function in resampler routine, perhaps
> uninitialized resamplers in general?
A fresh look after work made the resampler initialization code pop out.
The problem is in the sink connection being made from
"module-rtp-recv.c", the "PA_SINK_INPUT_VARIABLE_RATE" flag should be
passed into "pa_sink_input_new", but is not there. Made the change and
tested it, fixes the problem. Checked and head-of- tree off of the
pulseaudio.org source browsing link does not have this fix either.
One-liner patch attached.
pa_logXXX(__FILE__":
and replace them by
pa_logXXX("
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1272 fefdeb5f-60dc-0310-8127-8f9354f1896f
is to allocate all audio memory blocks from a per-process memory pool which is
available as read-only SHM segment to other local processes. Then, instead of
writing the actual audio data to the socket just write references to this
shared memory pool.
To work optimally all memory blocks should now be of type PA_MEMBLOCK_POOL or
PA_MEMBLOCK_POOL_EXTERNAL. The function pa_memblock_new() now generates memory
blocks of this type by default.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1266 fefdeb5f-60dc-0310-8127-8f9354f1896f
* fix building of padsp
* remove a warning when compiling padsp.c
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@972 fefdeb5f-60dc-0310-8127-8f9354f1896f
* modify pa_sink_input_new() to take initial volume settings as argument
* call pa_sink_input_set_volume() when changing stream volume in protocol-esound.c to make sure that subscribe events are issued properly
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@858 fefdeb5f-60dc-0310-8127-8f9354f1896f
* use server cookie as RTP SSRC
* enable SVN keywords
* add new option "loop" for RTP sender module
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@716 fefdeb5f-60dc-0310-8127-8f9354f1896f
