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
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
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).
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
Sine the base class now does the negotiation from the streaming thread we have
to be careful and check if the stream is ready before changing its corked state.
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).
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
* 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
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
