This is needed as a precursor to allowing capture of IEC61937
formats. We now also need to include the channel map while converting
format info to caps so that a correct channel mask is generated for
pulsesrc's caps.
PA_INVALID_INDEX, the default value, is unfortunately !0.
Setting the volume before the stream is created will put the ring
buffer in error state. Unfortunately, that's what spice-gtk does.
PulseAudio defines PA_RATE_MAX as the maximum sampling rate that it
supports. We were previously exposing a maximum rate of INT_MAX, which
is incorrect, but worked because nothing was really using a rate greater
than 384000 kHz.
While playing DSD data, we hit a case where there might be very high
sample rates (>1MHz), and pulsesink fails during stream creation with
such streams because it erroneously advertises that it supports such
rates.
Since PA_RATE_MAX is #define'd to (8*48000U), we can't just use it in
the caps string. Instead, we fix up the rate to what we actually support
whenever we use our macro caps.
GstAudioRingBuffer doesn't needs us to have at least 2 segments. We make
sure that if our buffer parameters are such that the maxlength is not at
least 2x fragsize, we still request the ringbuffer to keep that much
space so it continues to work.
https://bugzilla.gnome.org/show_bug.cgi?id=770446
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
This reverts commit 0dd46accf6.
With some audiosinks, starting the ringbuffer on the first commit
causes audio glitches at startup by starting to output segments
from the ringbuffer before it has been filled / fully prerolled. This
doesn't usually happen with pulsesink because we map the pulseaudio
ringbuffer directly, but we should keep things consistent with
other sinks with regards to startup latency, plus it gives more
headway to avoid glitching, should the initial 2nd segment take
more than 10ms to generate.
https://bugzilla.gnome.org/show_bug.cgi?id=657076
In pulsesink_query function, we use a switch for the query
type. In the CAPS case, there is no 'break', instead we
return right away. Use a break and return at the end of
the function instead for better code readability.
https://bugzilla.gnome.org/show_bug.cgi?id=744461
If we can not create probe stream in query_getcaps function, it will appear
memory leakage from format info.
The following patch prevent memory leakage in pulsesink.
https://bugzilla.gnome.org/show_bug.cgi?id=743178
We need a mechanism in PulseAudio to allow running code outside the
mainloop lock. Then we'd be able to post to the bus (taking the
GST_OBJECT_LOCK), without worrying about locking order with the mainloop
lock, which is the current cause of deadlocks while trying to post the
stream status messages.
https://bugzilla.gnome.org/show_bug.cgi?id=736071
This gives a quick introduction to how the pulsesink/pulsesrc code
interacts with the pa_threaded_mainloop that we start up to communicate
with the server.
The stream status messages are emitted in the PA mainloop thread, which
means the mainloop lock is taken, followed by the Gst object lock (by
gst_element_post_message()). In all other locations, the order of
locking is reversed (this is unavoidable in a bunch of cases where the
object lock is taken by GstBaseSink or GstAudioBaseSink, and then we get
control to take the mainloop lock).
The only way to guarantee that the defer callback for stream status
messages doesn't deadlock is to either stop posting those messages, or
make sure that the message emission is completed before we proceed to
any point that might take the object lock before the mainloop lock
(which is what we do after this patch).
https://bugzilla.gnome.org/show_bug.cgi?id=736071
The audio library considers them as encoded formats and does not fill in the
sample width. The audio ringbuffers identifies the format as alaw/mulaw and that
is always 8 bits.
This reverts commit 01457027e0.
We'll just depend on PulseAudio 2.0 or above instead of having the bug
partially fixed based on the installed libpulse version.
