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.
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
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
We need different symbol names, because these symbols are also present
in the fragmented plugin ... which will cause conflicts when doing
static linking
The number of FFTs is calculated with the following formula:
guint nfft = 2 * bands - 2;
nfft is passed to gst_fft_f32_new() as the len argument and is of type
unsigned integer. This method required that len is at leas 1, then
maximum G_MAXINT, as other values would be negative. If we extrapolate
from the formula above it means we need "bands" to be between 2 and
((guint)G_MAXINT + 2) / 2).
https://bugzilla.gnome.org/show_bug.cgi?id=744213
Using the sparse streams can make the push-based seeking return
too far in the stream. It also can lead to issues as the
sparse streams will be ignored when restarting playback and,
if the sparse stream is the one that has the earliest sample,
it will confuse qtdemux's offsets as one stream will have
an earlier offset than the demuxer's one which might lead to
early EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=742661
Parse the 'sidx' atom and update the total duration according to the
parser result. The isoff parser code is imported from
gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data()
function was factored out of the gst_isoff_sidx_parser_add_buffer()
function.
https://bugzilla.gnome.org/show_bug.cgi?id=743578
According to RFC 4585 section 3.5.3 step 1 we are not allowed to send
an early RTCP packet for the very first one. It must be a regular one.
Also make sure to not use last_rtcp_send_time in any calculations until
we actually sent an RTCP packet already. In specific this means that we
must not use it for forward reconsideration of the current RTCP send time.
Instead we don't do any forward reconsideration for the first RTCP packet.
