* fix the watermark when we change the latency
* fix latency measurement
* move rewinding code into its own function
* make use of new function pa_alsa_recover_from_poll() were applicable
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2299 fefdeb5f-60dc-0310-8127-8f9354f1896f
- Remove "source" word from monitor source description
- Increase default tsched watermark to 20ms again
- For the first iteration after snd_pcm_start() halve the sleep time as workaround for USB devices with quick starts
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2291 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Change pa_memblockq to carry silence memchunk instead of memblock and adapt all users
* Add new call pa_sink_input_get_silence() to get the suitable silence block for a sink input
* Implement monitoring sources properly by adding a delay queue to even out rewinds
* Remove pa_{sink|source}_ping() becaused unnecessary these days and not used
* Fix naming of various rewind related functions. Downstream is now _request_rewind(), upstream is _process_rewind()
* Fix volume adjustments for a single stream in pa_sink_render()
* Properly handle prebuf-style buffer underruns in pa_sink_input
* Don't allow rewinding to more than the last underrun
* Rework default buffering metrics selection for native protocol
* New functions pa_memblockq_prebuf_active(), pa_memblockq_silence()
* add option "mixer_reset=" to module-alsa-sink
* Other cleanups
git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2283 fefdeb5f-60dc-0310-8127-8f9354f1896f
- 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
