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.
The getcaps function we added uses some pa_format_info_get_prop...
accessor functions that were only added in 2.0, so we only have our
getcaps implementation exist if we're compiling against libpulse 2.0 or
above.
Eventually, we could bump the minimum requirement to 2.0 or above.
https://bugzilla.gnome.org/show_bug.cgi?id=686459
getcaps is called frequently during stream setup, and creating a new
stream each time is very inefficient. There's some more room for
optimisation by caching the queried sink formats as well, but this needs
some more changes to listen for format changes on the sink (for when
supported formats change between probe stream creation and sink
querying).
https://bugzilla.gnome.org/show_bug.cgi?id=686459
This allows us to have more fine-tuned caps in READY or above. However,
this is _really_ inefficient since we create a new stream and query sink
for every getcaps in READY, which on a simple gst-launch line happens
about 35 times. The next step is to cache getcaps results.
https://bugzilla.gnome.org/show_bug.cgi?id=686459
This makes sure that we update segdone based on the read index received
during latency updates. As the comment notes, we make some compromises
to deal with the fact that segdone is a segment multiple, while the read
index offers finer granularity. The updates are also not very often
(100ms since that is how often automatic timing updates are provided).
All this is required for the baseaudiosink sample alignment code to work
at all.
https://bugzilla.gnome.org/show_bug.cgi?id=694257
A bug in PulseAudio causes PA_STREAM_START_MUTED to be rejected on
record streams. Until this is fixed upstream, we mute the stream
manually at startup. Based on a patch by Alban Browaeys
<prahal@yahoo.com>.
https://bugzilla.gnome.org/show_bug.cgi?id=684469
When we start and renegotiate, there is a moment where the stream is created but
not yet connected. Make sure all functions deal with this situation correctly
instead of erroring out.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681247
When get_time is called but we are not yet negotiated, return 0 instead of
posting an error. It's possible that the base class is still negotiating when
our get_time is called.
This makes sure that we:
a) Destroy an existing stream if a negotiate() request comes in: this is
required when receiving a downstream renegotiation request after a
stream has been created.
b) Create a new stream on prepare(): this is required since we do a
setcaps() in negotiate(), which causes the stream to be dropped by a
ringbuffer release() call (this does not happen during first negotiation
since the release is only done on a running ringbuffer). The subsequent
call to ringbuffer acquire() fails because the stream was lost on
release().
https://bugzilla.gnome.org/show_bug.cgi?id=681247
If converting a PA channel map to gst channel positions results in a
valid set of channel positions, we clear the unpositioned flag from the
ringbuffer spec.
This itereates over the GstAudioInfo to set invalid channel positions
rather than use memset() which works right now because it assumes that
GST_AUDIO_CHANNEL_POSITION_INVALID is -1.
Post the notify outside of the pa_lock to avoid a deadlock caused by basesrc
calling get_time with the object lock.
Reset the clock on connect.
Post clock-lost and clock-provide messages.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673977
Conflicts:
ext/pulse/pulsesrc.c
When we explicitely set the mute property to FALSE, connect to pulseaudio with
the PA_STREAM_START_UNMUTED flag set, otherwise pulseaudio will use its
previously used value (which might start the stream muted).
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=672401
pulseaudiosink breaks visualisations in its current form, so let's
prevent it from being autoplugged for the time being.
The best we can hope to do in the 0.10 series is query the list of
available sinks and their formats, and expose these as the bin's sinkpad
caps. While this is not a comprehensive solution, it will make sure that
we're only trying to support compressed formats if we're certain that
one exists.
The long-term fix for this will be in the form of proper upstream
renegotiation support in the 0.11/1.0 series.
https://bugzilla.gnome.org/show_bug.cgi?id=666361
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).
This is causing us lots of headaches in 0.10 and needs to be done
differently and properly in 0.11. playbin or decodebin should
reconfigure themselves based on reconfigure events, for example.
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.
Conflicts:
ext/pulse/pulseaudiosink.c
ext/pulse/pulsesrc.c
gst/audioparsers/gstaacparse.c
gst/audioparsers/gstamrparse.c
gst/audioparsers/gstdcaparse.c
gst/audioparsers/gstflacparse.c
gst/effectv/gstradioac.c
gst/effectv/gstradioac.h
gst/effectv/gstripple.c
Some possible FIXMEs remaining in the audio parser getcaps functions.
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
