pa_stream_* functions return negative on error, despite the defines
for error codes being positive.
I only got to repro the error twice, so I'm not sure 100% sure this
fixes the issue (the negative var being uninitialized after returning
from pa_stream_get_latency).
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
PulseAudio 1.0 supports per-source-output volumes, and this exposes the
functionality via the GstStreamVolume interface.
When compiled against pre-1.0 PulseAudio, the interface is not
implemented, and the "volume" or "mute" properties are not available.
This bit of ugliness will go away when we can depend on PulseAudio 1.0
or greater.
https://bugzilla.gnome.org/show_bug.cgi?id=595055
This introduces a new bin which wraps around pulsesink and depending on
the formats supported by the sink, plugs in/out a decodebin2 as
required. This allows users to switch sinks on the stream and adapts
accordingly (for example, you could watch a movie in passthrough mode on
your receiver which supports AC3 decode, then plug out and switch to a
non-digital profile to continue uninterrupted on analog output).
The bin is required because doing the same with playbin2/playsink will
require API changes that cannot be made in 0.10. With 0.11/1.0, we
should be able to ask for upstream caps renegotiation to deal with all
this.
https://bugzilla.gnome.org/show_bug.cgi?id=657179
There's no use in splitting the incoming data down to the segsize
limit - by writing as much as possible in one chunk, we increase
performance and avoid PulseAudio unnecessary rewinds.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This adds support for various compressed formats (AC3, E-AC3, DTS and
MP3) payloaded in IEC 61937 format (used for transmission over S/PDIF,
HDMI and Bluetooth).
The acceptcaps() function allows bins to probe for what formats the sink
being connected to support. This only works after the element is set to
at least READY.
If the underlying sink changes and the format we are streaming is not
available, we emit a message that will allow upstream elements/bins to
block and renegotiate a new format.
This exposes the source output index of the record stream that we open
so that clients can use this with the introspection if they want (to
move the stream, for example).
Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the
deferred call to be run before returning. This causes a race when
READY->NULL is executed shortly after, which stops the mainloop. This
leaks the element reference which is passed as userdata for the callback
(introduced in commit 7cf996, bug #614765).
The correct fix is to wait in READY->NULL for all outstanding calls to
be fired (since libpulse doesn't provide a DestroyNotify for the
userdata). We get rid of the reference passing from 7cf996 altogether,
since finalization from the callback would anyways lead to a deadlock.
Re-fixes bug #614765.
So that pulsesrc/pulsesink get chosen over other possible PRIMARY
src/sinks by autoaudiosink. Presumably, if pulse is available, it
is always preferred over another src/sink.
Fixes: #647540.
This drops support fof PulseAudio versions prior to 0.9.16, which was
released about 1.5 years ago. Testing with very old versions is not
feasible and we don't want to maintain 2 independent code-paths.
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.
g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.
Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
GCC 4.6.x spits warnings about such usage of variables. The variables in
raw1394 were marked with G_GNUC_UNUSED as this seemed omre appropriate.
The others were removed.
Pulsesink was recently changed to defer uncorking until there is data
to write. This condition will however never occur when EOS in being
rendered (since that marks the end of data). Changing to PAUSED state
while EOS is being waited on results in a hang: pausing corks the
stream, which will never be undone since there is no more data when
going back to PLAYING. If pulsesink is the clock provider, deadlock
ensues since time doesn't continue in corked state and the clock id
for EOS wait never fires.
Fixes#645961.
Not doing so can result in a deadlock when two threads enter
gst_pulseringbuffer_open_device at the same time, as
pa_threaded_mainloop_wait releases the mainloop lock while waiting,
allowing another thread to take it, resulting in a deadlock as two
threads waits for the lock the other is holding.
https://bugzilla.gnome.org/show_bug.cgi?id=643087
By allowing larger chunks to be sent, PulseAudio will have a
lower CPU usage. This is especially important on low-end machines,
where PulseAudio can crash if packets are coming in at a higher
rate than PulseAudio can process them.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
After starting the ringbuffer, we wait for enough data to arrive before
uncorking the stream. This will cause the pipeline to stall if we get an
EOS (or otherwise need to flush the stream) before sufficient data
becomes available. This patch makes sure that the stream is uncorked
while flushing to avoid this problem.
Fixes issue with a webkit unit test testing reverse playback of
an MP4 H.264/AAC file.
https://bugzilla.gnome.org/show_bug.cgi?id=639740
This makes the call to pa_stream_cork() during ringbuffer pause()
synchronous, which makes sure that the clock does not advance after we
take a snapshot for start_time.
https://bugzilla.gnome.org/show_bug.cgi?id=639240
* ext/pulse/pulsesrc.c (gst_pulsesrc_class_init, gst_pulsesrc_init)
(gst_pulsesrc_set_property, gst_pulsesrc_get_property)
(gst_pulsesrc_open): Add a "client" property, as in pulsesink.
Fixes#634914
Don't uncork in the _start method just yet but wait until we have written some
samples to pulseaudio. This avoid underruns on pulseaudio and less crackling
noises when starting.
Make the is_dead check more clear and add an option to check for the status of
the stream in addition to the context.
We don't need a stream to get the device_description string.
Fixes#630317
