We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.
https://bugzilla.gnome.org/show_bug.cgi?id=745599
Otherwise we will get not-negotiated later from rtpbin, and will never be able
to send RTCP packets back to the server. Note that error flow returns from the
RTCP pads are ignored, that's why it didn't fail more visible before.
This reverts commit 1591adf4cd.
https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
It's the beginning of an implementation of RFC 2762, which is needed for
large multicast groups. The implementation is not yet complete but why
not leave what is there and implement RFC 2762 instead?
rtpsession declares an array of maps to store srrcs but only the
the key 0 is being used. This patch replaces the array of maps
for just one map and remove useless parameters in rtpsession
https://bugzilla.gnome.org/show_bug.cgi?id=745586
In gst_avi_demux_handle_src_query, there is not needed code.
We already check about stream is vbr or not at the upper line.
o, we don't need to check this condition becase stream is not
vbr 100% in this case.
https://bugzilla.gnome.org/show_bug.cgi?id=745276
The ringbuffer does allow renegotiation, so we do not have to report
fixed caps once it is acquired (based on a similar patch for the sink
side by Ilya Konstantinov <ilya.konstantinov@gmail.com>).
Once osxaudiosink's device is open, it fixates on the initial caps and
refuses to accept new caps. This is erroneous since the Audio Unit is
can accept a new ASBD, and GstAudioRingBuffer supports reconfiguration
as well.
https://bugzilla.gnome.org/show_bug.cgi?id=743925
Ensure gst_v4l2_buffer_pool_release_buffer() releases the associated
GstV4l2MemoryGroup. In particular, this allows for closing the DMABUF
handles prior to instantiating new ones.
https://bugzilla.gnome.org/show_bug.cgi?id=745443
... instead of just counting frames. The values are supposed to be in timebase
units, not frame units. This fixes various quality problems with VP8/VP9
encoding and in general makes the encoder behave better.
Thanks to Nirbheek Chauhan for noticing this bug.
As it's very common, handle driver not setting field in buffers
by using the field value from the format. This workaround a long time
bug in UVC driver. For even buggier driver, we simply assume
progressive as before. We also only warn once, to avoid spamming.
Unlike many other seek flags, the KEY_UNIT seek
flag is not copied over into the GstSegment,
since it's only relevant for the seek itself,
so we need to pass it explicitly to the seek
handler here.
https://bugzilla.gnome.org/show_bug.cgi?id=745339
S_CROP ioctl is write-only and the device can adjust crop rectangle so
we query back the crop configuration after each S_CROP to know what has
been done.
https://bugzilla.gnome.org/show_bug.cgi?id=736133
In the V4L2 single-planar API, when format is semi-planar/planar,
drivers expect the planes to be contiguous in memory.
So this commit change the way we handle semi-planar/planar format
(n_planes > 1) when we use the single-planar API (group->n_mem == 1).
To check that planes are contiguous and have expected size, ie: no
padding. We test the fact that plane 'i' start address + plane 'i'
expected size equals to plane 'i + 1' start address. If not, we return
in error.
Math are done in bufferpool rather than in allocator because the
former is aware of video info.
https://bugzilla.gnome.org/show_bug.cgi?id=738013
Offset are relative to the buffer and there is no guarantee substracting
them will give us the plane size. So we let bufferpool make the math as
it is more aware of video info than allocator and pass a size array to
allocator import function.
Pointed out by Nicolas Dufresne <nicolas.dufresne@collabora.com>
https://bugzilla.gnome.org/show_bug.cgi?id=738013
Fixes stuttering audio when iOS AU is resampling. To make AU resample,
one has to request a rate that differs from AVAudioSession's
sampleRate. The resampling itself is not the culprit, but rather our
API misuse.
AudioUnitRender modifies the mDataByteSize members with the
actual read bytes count. Therefore, they must be reinitialized
before each AudioUnitRender. (The buffers themselves can be
preallocated.)
The "stutter" was caused by one AudioUnitRender making the buffer
too small for other AudioUnitRender invocations, making them fail
with -50 (paramErr). By way of luck, when AU didn't resample, all
AudioUnitRender invocations read the same number of bytes.
(This patch addresses some non-interleaved audio concerns, but
at this moment the elements do not support non-interleaved audio
and non-interleaved is untested.)
https://bugzilla.gnome.org/show_bug.cgi?id=744922
