[The original commit message didn't have any explanation why this
change is made, so I'll add that information here myself.
--Tanu Kaskinen]
This change is from the developers of a Haskell binding[1]. According
to them, this change isn't strictly necessary, but their code gets
significantly cleaner if they can register an operation callback that
is called when the operation is cancelled due to the context getting
disconnected.
[1] https://github.com/favonia/pulse
This allows clients to get a "fake" sample space for compressed formats
that we can support. This should make size/time conversion for things
like calculating buffer attributes simpler.
These utility functions could be handy to clients.
pa_format_info_to_sample_spec_fake() isn't made public, but the return
value is changed to keep in sync with pa_format_info_to_sample_spec().
We currently only have setters and clients need to be able to query
these values as well. The return types for these functions needed to be
changed to int since this is public API now.
This patch introduces some extra protocol information, so protocol
version is bumped. This functionality is primarily needed to solve
a long standing issue in alsa-plugins, which should ignore underruns
if and only if it is obsolete, i e, if more data has been written to
the pipe in the meantime (which will automatically end the underrun).
BugLink: http://bugs.launchpad.net/bugs/805940
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This simply exposes the formats that a device supports
via a simple protocol extension that will allow clients
to setup what a connected receiver supports format wise.
This replaces the simple string used by pa_format_info's proplist with a
JSON string (accessed via new API only). This allows us to express lists
and ranges more cleanly, and embed type information for future
extensibility.
We use json-c for JSON parsing. This is a lightweight depdency (32 KB on
my system) and avoids the hassle of having to reinvent a JSON parser.
Also included is a test which verifies functionality and is
valgrind-clean.
We frequently need to free an idxset containing pa_format_infos, so
define an internal free function that can be used directly with this
(instead of defining it once-per-file).
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
Since the stream identifiers (channels) are monotonically growing integer, it
isn't a good idea to use them as index to a dynamic array, because the array
will grow all the time. This is not a problem with client connections that
don't create many streams, but, for example, long-running clients that use
libcanberra for playing event sounds, this means that the client connection
effectively leaks memory.
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>
On Tue, 31 Mar 2009, Lennart Poettering wrote:
[snip]
>
> I have now merged your patch. I had to change a few things to make it
> apply cleanly. Since I have no access to Solaris I am unable to test
> this though, so please check if things still work for you.
>
> I also worked around the realpath() issue mostly. It should work fine on
> Solaris now, as well.
Thanks. 0.9.15-test7 seems to work fine.
The only new issue is that configure --without-dbus no longer builds. I
don't need dbus for my purposes (network audio server) and it seems that
dbus is not included with Solaris. A patch for this follows.
Finn
- Add new PA_STREAM_FIX_CHANNELS, FIX_RATE, FIX_FORMAT, DONT_MOVE, VARIABLE_RATES to pa_sream_flags_t adn implement it
- Expose those flags in pacat
- Add notifications about device suspend/resume to the protocol and expose them in libpulse
- Allow changing of buffer_attr during playback
- allow disabling for remixing globally
- hookup polkit support
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2067 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
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
