FSF addresses used in PA sources are no longer valid and rpmlint
generates numerous warnings during packaging because of this.
This patch changes all FSF addresses to FSF web page according to
the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html
Done automatically by sed-ing through sources.
Since the hashmap stores a pointer to the key provided at pa_hashmap_put()
time, it make sense to allow the hashmap to be given ownership of the key and
have it free it at pa_hashmap_remove/free time.
To do this cleanly, we now provide the key and value free functions at hashmap
creation time with a pa_hashmap_new_full. With this, we do away with the free
function that was provided at remove/free time for freeing the value.
The previous patch removed module-gconf's dependency on the userdata
pointer of the free callback, and that was the only place where the
userdata pointer of pa_free2_cb_t was used, so now there's no need for
pa_free2_cb_t in pa_hashmap_free(). Using pa_free_cb_t instead allows
removing a significant amount of repetitive code.
modules/rtp/module-rtp-recv.c:462:8: warning: 'r' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
These are not used for anything at this point, but this
makes it easy to add ad-hoc debug prints that show the
memblockq name and to convert between bytes and usecs.
This is the beginning of work to support compressed formats natively in
PulseAudio. This adds a pa_stream_new_extended() that takes a format
structure, sends it to the server (=> protocol extension) and has the
server negotiate with the appropropriate sink to figure out what format
it should use.
This is work in progress, and works only with PCM streams. Actual
compressed format support in some sink needs to be implemented, and
extensive testing is required.
More details on how this is supposed to work is available at:
http://pulseaudio.org/wiki/PassthroughSupport
Instead <pulsecore/poll.h> should be included. That file includes poll.h on
platform where it is appropriate. Also remove some unnecessary <ioctl.h>
includes.
The same logic is applied to the sample rate adjustments in module-rtp-recv,
module-loopback and module-combine:
- Each time an adjustment is made, the new rate can differ at most 2‰ from the
old rate. Such a step is equal to 3.5 cents (a cent is 1/100th of a
semitone) and as 5 cents is generally considered the smallest observable
difference in pitch, this results in inaudible adjustments.
- The sample rate of the stream can only differ from the rate of the
corresponding sink by 25%. As these adjustments are meant to account for
very small clock drifts, any large deviation from the base rate suggests
something is seriously wrong.
- If the calculated rate is within 20Hz of the base rate, set it to the base
rate. This saves CPU because no resampling is necessary.
Move the mainloop to monotonic based time events.
Introduces 4 helper functions:
pa_{context,core}_rttime_{new,restart}(), that fill correctly a
timeval with the rtclock flag set if the mainloop supports it.
Both mainloop-test and mainloop-test-glib works with rt and timeval
based time events. PulseAudio and clients should be fully functional.
This patch has received several iterations, and this one as been
largely untested.
Signed-off-by: Marc-André Lureau <marca-andre.lureau@nokia.com>
The primary reason for this change is to allow time graphs that do not
go through the origin and hence smoothing starting from the origin is
not desired. This change will allow passing time data into the smoother
while paused and then abruptly use that data without smoothing using the
'quick fixup' flag when resuming.
Primary use case is allowing recording time graphs where the data
recorded originates from a time before the stream was created. The
resulting graft will be shifted and should not be smoothened to go
through the origin.
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
