TFDTs with time 0 are being ignored since commit 1fc3d42f. They're
mistaken with the case of not having TFDT, but those two cases
must be distinguished in some way.
This patch passes an extra boolean flag when the TFDT is present.
This is now the condition being evaluated, instead of checking for
0 time.
https://bugzilla.gnome.org/show_bug.cgi?id=780410
If we have multiple tracks with timecodes, or it's not the first track
that has timecodes, or not the first buffer, we already started a chunk
for media data. We now need to "close" that chunk because we wrote data
for the timecode track and a new chunk has to be started for the
original track the next time it has data.
Similar to what was done in adaptivedemux, ignore seek
events we've already handled - such as when they are received
on every srcpad of files with lots of streams.
Otherwise mdatleft will have a value calculated from the initial
mdatsize minus the parts of the stream that we saw, which is not
including all the parts of the stream that might've been skipped.
This breaks gst-validate on the build server (though not locally),
and a unit test, and I can't run unit tests right now for some
unrelated reason.
This reverts commit 0747b56f8e.
This debug statement is meant to print the time since the last (early)
RTCP transmission, not the last regular RTCP transmission (which also
happens to be set a few lines above to current_time, so the debug output
is just confusing)
Take into account the atoms at the end of the 'trak' atom when
recovering it. So that its size (already computed and added in the trak
size) isn't making offsets wrong.
https://bugzilla.gnome.org/show_bug.cgi?id=771478
Fix the check for whether the start time of the segment has
been reached when playing in reverse. Otherwise, playback
stops after reaching the start of any file part, instead of
continuing until all parts within the segment have played
We parse the next moof in advance of having pushed
all samples from the previous one in some cases, and
we'll still need the crypto info from the previous
fragment so keep around any unused crypto info entries
when adding new ones
qtdemux.c: In function ‘qtdemux_parse_samples’:
qtdemux.c:8450:39: error: ‘*’ in boolean context, suggest ‘&&’ instead [-Werror=int-in-bool-context]
if (stream->samples_per_frame * stream->bytes_per_frame) {
~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_reset’:
gstmpegaudioparse.c:209:3: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size]
memset (mp3parse->xing_seek_table_inverse, 0, 256);
^~~~~~
gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_handle_first_frame’:
gstmpegaudioparse.c:951:7: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size]
memset (mp3parse->xing_seek_table_inverse, 0, 256);
^~~~~~
This prevents storing an infinite amount of e.g. comment headers if they
come without a new initialization header in front of them. There can
only be one header of each type.
If we also replace all headers when receiving any possibly following
comments header, we would throw away the config header before being able
to make use of it.
A sparse stream's ending timestamp can be considerably smaller
than the ending timestamps of the other streams, which can lead
to skipping considerable time from the next part.
https://bugzilla.gnome.org/show_bug.cgi?id=761086
In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP
timestamps is more than (3 * jbuf->clock_rate) we call
rtp_jitter_buffer_reset_skew which resets pts to 0. So components down
the pipeline (playes, mixers) just skip frames/samples until pts becomes
equal to pts before gap.
In version 1.10.2 and before this checking was bypassed for packets with
"estimated dts", and gaps were handled correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=778341
The payloader needs to reset and update the vorbis config data which is
pushed on the network if it receives new headers, or at least, it may
have to do so.
Without this, the stream configuration could change without the
payloader sending the new configuration to the other side.
This reverts commit 107902ec51.
This commit intended to ensure that keyframe seeks land at the
start timestamp of a keyframe, rather than in the middle of one,
but they cause trouble on files with sparse streams, or with
JPEG 'cover art' tracks that have only one or a few JPEG samples
with very long durations.
That's still desirable for doing seamless cutting of videos,
but needs a rethink for implementation.
https://bugzilla.gnome.org/show_bug.cgi?id=778690
Add a new boolean surround-delay property that makes
audioecho just apply a delay to certain channels to create
a surround effect, rather than an echo on all
channels. This is useful when upmixing from stereo - for example.
Add a surround-mask property to control which channels
are considered surround sound channels when adding a
delay with surround-delay = true
Original patch from Jochen Henneberg <jh@henneberg-systemdesign.com>
