PulseAudio defines PA_RATE_MAX as the maximum sampling rate that it
supports. We were previously exposing a maximum rate of INT_MAX, which
is incorrect, but worked because nothing was really using a rate greater
than 384000 kHz.
While playing DSD data, we hit a case where there might be very high
sample rates (>1MHz), and pulsesink fails during stream creation with
such streams because it erroneously advertises that it supports such
rates.
Since PA_RATE_MAX is #define'd to (8*48000U), we can't just use it in
the caps string. Instead, we fix up the rate to what we actually support
whenever we use our macro caps.
This patch enables matroskademux to receive seeks before it reaches
GST_MATROSKA_READ_STATE_DATA.
Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/514
This also enables receiving seeks in the element READY state.
When such a seek is received, it is stored to be later handled when
GST_MATROSKA_READ_STATE_DATA is reached.
Reset RTPSession when rtpsession changes state from PAUSED to READY.
Without this change, a stored last_rtptime in RTPSource could interfere
with RTP timestamp generation in RTCP Sender Report.
Fixes#510
This commit adds a .gitlab-ci.yml file, which uses a feature
to fetch the config from a centralized repository. The intent is
to have all the gstreamer modules use the same configuration.
The configuration is currently hosted at the gst-ci repository
under the gitlab/ci_template.yml path.
Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
If there was no interlace-mode field in the caps. Read back the value
selected by the driver. This way, if the driver does not support
progressive, then it will automatically negotiate the returned mode
unless this mode is not supported by GStreamer.
This method was already used for colorimetry. Just like colorimetry, the
interlace mode is not longer probed by v4l2src dues to performance
issues.
Fixes#511
If ctts (CompositionOffsetBox) has larger sample_offset
(offset between PTS and DTS) than (2 * duration) of the stream,
assume the ctts box to be corrupted and ignore the box.
https://bugzilla.gnome.org/show_bug.cgi?id=797262
This fixes a bug where in some files mehd.fragment_duration is one unit
less than the actual duration of the fragmented movie, as explained below:
mehd.fragment_duration is computed by scaling the end timestamp of
the last frame of the movie in (in nanoseconds) by the movie timescale.
In some situations, the end timestamp is innacurate due to lossy conversion to
fixed point required by GstBuffer upstream.
Take for instance a movie with 3 frames at exactly 3 fps.
$ gst-launch-1.0 -v videotestsrc num-buffers=3 \
! video/x-raw, framerate="(fraction)3/1" \
! x264enc \
! fakesink silent=false
dts: 999:59:59.333333334, pts: 1000:00:00.000000000, duration: 0:00:00.333333333
dts: 999:59:59.666666667, pts: 1000:00:00.666666666, duration: 0:00:00.333333334
dts: 1000:00:00.000000000, pts: 1000:00:00.333333333, duration: 0:00:00.333333333
The end timestamp is calculated by qtmux in this way:
end timestamp = last frame DTS + last frame DUR - first frame DTS =
= 1000:00:00.000000000 + 0:00:00.333333333 - 999:59:59.333333334 =
= 0:00:00.999999999
qtmux needs to round this timestamp to the declared movie timescale, which can
ameliorate this distortion, but it's important that round-neareast is used;
otherwise it would backfire badly.
Take for example a movie with a timescale of 30 units/s.
0.999999999 s * 30 units/s = 29.999999970 units
A round-floor (as it was done before this patch) would set fragment_duration to
29 units, amplifying the original distorsion from 1 nanosecond up to 33
milliseconds less than the correct value. The greatest distortion would occur
in the case where timescale = framerate, where an entire frame duration would
be subtracted.
Also, rounding is added to tkhd duration computation too, which
potentially has the same problem.
https://bugzilla.gnome.org/show_bug.cgi?id=793959
There is no specific needs to duplicate the FD. Unlike the exportation,
we don't depend on code that will call close. This will make debugging
easyer since the traced FD will match the exporter.
... before the old streams is not exposed yet for MSS stream.
In case of DASH, newly configured streams will be exposed
whenever demux got moov without delay.
Meanwhile, since there is no moov box in MSS stream,
the caps will act like moov. Then, there is delay for exposing new pads
until demux got the first moof.
So, following scenario is possible only for MSS but not for DASH,
STREAM-START -> CAPS -> (configure stream but NOT EXPOSED YET)
-> STREAM-START-> CAPS (configure stream again).
In above scenario, we can reuse old stream without any stream reconfigure.
https://bugzilla.gnome.org/show_bug.cgi?id=797239
Apart from the obvious drawbacks of hardcoding, the drawback here was
that, if we subtracted 2 frames (instead of 2.6) from the target running
time, we'd request the next keyframe a bit too far into the future,
which would make our files split at the wrong position.
https://bugzilla.gnome.org/show_bug.cgi?id=797293
Flv does not support various channels in AAC stream format, for example
flvdemux detect an audio channels of 2(stereo) when the AAC really is 1(mono).
https://bugzilla.gnome.org/show_bug.cgi?id=797275
Flv does support changing the stream type and stream properties
after the headers were started to be written, and for example H264
codec_data changes can be supported.
https://bugzilla.gnome.org/show_bug.cgi?id=797256
