- allow setting of the requested latency of a sink input/source output before _put() is called
- allow sinks/sources to have a "minimal" latency which applies to all requested latencies by sink inputs/source outputs
- add new client library flags PA_STREAM_ADJUST_LATENCY, PA_STREAM_START_MUTED
- allow client library to fill in 0 to buffer_attr fields
- update module-alsa-source following module-alsa-sink
- other cleanups and fixes
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2215 fefdeb5f-60dc-0310-8127-8f9354f1896f
pa_memblock is now an opaque structure. Access to its fields is now done
through various accessor functions in a thread-safe manner.
pa_memblock_acquire() and pa_memblock_release() are now used to access the
attached audio data. Why? To allow safe manipulation of the memory pointer
maintained by the memory block. Internally _acquire() and _release() maintain a
reference counter. Please do not confuse this reference counter whith the one
maintained by pa_memblock_ref()/_unref()!
As a side effect this patch removes all direct usages of AO_t and replaces it
with pa_atomic_xxx based code.
This stuff needs some serious testing love. Especially if threads are actively
used.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1404 fefdeb5f-60dc-0310-8127-8f9354f1896f
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
* alsa-source: if "Capture" is not found as mixer track name, fallback to "Mic"
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@993 fefdeb5f-60dc-0310-8127-8f9354f1896f
* if an ALSA device doesn't support the sampling freq requested, use what ALSA suggests and resample if this deviates more than 10% from what we requested
* fix segfault freeing an unitialized mixer_fdl field
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@992 fefdeb5f-60dc-0310-8127-8f9354f1896f
* add some more validity checks to pa_source_new(),pa_sink_new(),pa_sink_input_new(),pa_source_output_new()
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@888 fefdeb5f-60dc-0310-8127-8f9354f1896f
* fix fragment size calculation in module-alsa-sink
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@855 fefdeb5f-60dc-0310-8127-8f9354f1896f
but following their API properly should avoid problems in the future.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@606 fefdeb5f-60dc-0310-8127-8f9354f1896f
* daemon/ - Contains the files specific to the polypaudio daemon.
* modules/ - All loadable modules.
* polyp/ - Files that are part of the public, application interface or
are only used in libpolyp.
* polypcore/ - All other shared files.
* tests/ - Test programs.
* utils/ - Utility programs.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@487 fefdeb5f-60dc-0310-8127-8f9354f1896f
