Previously we advanced the in_data pointer by bps for every channel, and then
later again for block_size*bps. This caused us to be one sample further than
expected if an input buffer covered two analysis frames. And in the end lead
to completely bogus values reported by level.
https://bugzilla.gnome.org/show_bug.cgi?id=746065
These are outside the expected range of sequence numbers and should be
clipped, especially for RTSP they might belong to packets from before a seek
or a previous stream in general.
For output device, we should not update the buffer with flags and
timestamp when we dequeue. The information in the v4l2_buffer is not
meaningful and it breaks the case where the buffer is rendered at
multiple places.
https://bugzilla.gnome.org/show_bug.cgi?id=745438
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.
