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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1091>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/841>
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Warning: GstPbutils: gst_codec_utils_h264_get_profile_flags_level:
unknown parameter 'codec_data' in documentation comment, should be 'codecs_data
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1279>
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In ALSA, there is possible temporary failures that may require a retry,
though in certain situation, this may leak to the write() function
holding on a lock forever preventing the pipeline from going to pause
or stop. Fix this by shortly dropping the lock between retries.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1261>
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This moves the "written X frames" lower so that we don't trace
confusing negative values on errors and add the error code in the
"Write error" log.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1261>
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These might not exist inside the structure and then we would potentially
keep around uninitialized memory from the caller in the out parameter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/887>
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to allow subclasses to handle packet loss / missing data
Subclasses could use the new vfunc to activate packet loss concealment,
for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1274>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/886>
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We should record the ownership of the data before we reset the bitwriter.
Or we will always dup the buffer data and leak the memory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/886>
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In reset_and_get_data and reset_and_get_buffer, it fails to include
the trailing bits less than 8. So, when the bit_size is not byte
aligned, the trailing bits are lost in the return buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/886>
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The new keyframe is needed when the deadline of the buffer has exeeded
the waiting time, not while it is within it.
Also, since we look at the deadline of the frame, log that instead of PTS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1278>
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This feels like exactly like a case that should fail.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>
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Since the return value is documented to possibly be smaller than 0,
then it needs to be signed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1258>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
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Otherwise, it will drop valid buffers on a simple segment update
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
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Continue as-is on segment update.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
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Dropping it to 0 makes videorate push buffers from timestamp 0 again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
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AAC codec_data is a just collection of AAC profile, samplerate, and
channels. We can know samplerate and channels from parsed
SampleEntry data. Although the AAC profile is unknown there,
let's assume it as AAC-LC like we've been doing for the version 1
atom.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1082>
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The initial single queue srcresult is OK, it hasn't been
NOT_LINKED since 2007.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/885>
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.. when computing the high id.
After a flush for instance, sq->srcresult is reset to OK,
yet it doesn't make sense to pick a non-existing position
id as the high id when a queue doesn't contain any items
in that situation either.
It is in any case completely OK to let the not-linked stream
get consumed without throttling at this stage, as any
first packet arriving on other single queues will get assigned
a higher position id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/885>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1078>
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All of that is advertised through the codec_data itself so can change
just fine within isomp4.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1071>
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Previously only demuxing when stored via the RIFF/AVI mapping was
supported.
See https://github.com/FFmpeg/FFV1/blob/master/ffv1.md#matroska-file-format
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/923
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1080>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1081>
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The PrivateStream should keep track of stream tags only. Likewise, the
GstDiscovererInfo should keep track of global tags only.
This patch fixes the issue where the discoverer would report duplicated tag
titles, especially for Matroska media files. The Matroska demuxer emits
correctly-scoped tags, but downstream was making no distinction of them.
Fixes #598, #836, https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/827
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1275>
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The extension version doesn't have the ARB suffix.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1273>
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After sending or retrieving data, gstrtspconnection resets the socket's
timeout to 0 (infinite). This could cause problems if sending and
receiving at the same time. For example, if RTCP data is sent from the
streaming thread while gstrtspsrc is already retrieving data.
With this patch, timeout is only reset to 0 if there is no other
thread using the socket.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1260>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/884>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1270>
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In case of interlaced JPEG file, we are doubling stride.
The scratch scan line should take account of it as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1042>
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Use gap events to update the next_time of a queue the same
as buffers or segment events. Fixes problems where a group
consisting only of sparse streams primarily driven by
gap events would stall with a full multiqueue because
unlinked streams in the group were not being woken to
push data.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/879>
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Initial gap events should not be discarded on the input streams,
but instead cause unblocking just as buffers do.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1239>
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The subtitles in ogg/kate are identified using subtitle/ caps names.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1213>
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This works around some AVI files storing byte-stream data in the
codec_data. The previous workaround was only checking for
0x00000001 (4 bytes) instead of 0x000001 (3 bytes).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1072>
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The QQuickItem::size() method was introduced in 5.10, so use direct width() and
height() access instead.
Fixes #908
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1069>
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Matches the current list at
https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-5
as of 2021-September.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1267>
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This makes it clearer how to use the plugin in an API driven application.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
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Otherwise, it just fails to add the extension, which makes no
sense. And our level element produces levels higher than 127 in some
cases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
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This makes it a bit easier to understand how to use it in an application.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
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Tests adding the extension based on the caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1265>
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