- Jul 06, 2016
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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- Jul 05, 2016
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Sebastian Dröge authored
There's a small window for a race condition otherwise.
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- Jul 04, 2016
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Sebastian Dröge authored
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Sebastian Dröge authored
If necessary, a demuxer will have to invent something here but this is only a problem with non-conformant files anyway.
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Sebastian Dröge authored
Without, raw AAC can't be handled and we have some information available in the decoder that most likely allows us to decode the stream in one way or another. This is the same code already used by matroskademux for the same reasons, and ffmpeg/vlc play such files just fine too by guesswork.
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Sebastian Dröge authored
The resulting file is not going to be playable without guesswork and raw caps should always have codec_data.
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- Jul 01, 2016
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This is to handle cases where upstream handles the fragmented streaming in TIME segments and sends us data with gaps within fragments. This would happen when dealing with trick-modes. When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples, it must obey the following rules: * The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET * The buffers containing the first sample after a gap: * MUST start at the beginning of a sample, * MUST have the DISCONT flag set, * MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment. https://bugzilla.gnome.org/show_bug.cgi?id=767354
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Tim-Philipp Müller authored
gst_v4l2_clear_error() doesn't work like g_clear_error(), it doesn't NULLify the pointer, so set freed debug string to NULL so it doesn't get freed again if gst_v4l2_clear_error() is called twice on the error. CID 1362901
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Commit 7873bede caused new caps event per moof without consideration of duplication. https://bugzilla.gnome.org/show_bug.cgi?id=768268
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- Jun 30, 2016
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10 bytes was allocated for stream_format but size of "byte-stream" is more. Use g_strdup() instead. https://bugzilla.gnome.org/show_bug.cgi?id=753760
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- Jun 29, 2016
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Sebastian Dröge authored
Otherwise set_flushing() will have undefined semantics and nowadays causes a g_critical() to warn about that.
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Thiago Sousa Santos authored
Update the blocksize depending on how much is obtained from a read of the input stream. This avoids doing too many reads in small chunks when larger amounts of data are available and also prevents using a very large memory area to read a small chunk of data. https://bugzilla.gnome.org/show_bug.cgi?id=767833
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- Jun 28, 2016
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state If we consider the RTSP state, what can happen is that it is PLAYING but the element already asynchronously tried to PAUSE and it just did not happen yet. We would then override this setting to PAUSED (while the element actually is in PAUSED) and set the RTSP state to PLAYING again. This would then cause us to produce packets while the sinks are all PAUSED, piling up thousands of packets in the rtpjitterbuffer and other elements and finally failing.
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- Jun 27, 2016
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Sebastian Dröge authored
They are however supported by ffmpeg and apparently used out there. https://bugzilla.gnome.org/show_bug.cgi?id=768006
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- Jun 24, 2016
- Jun 21, 2016
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Nicolas Dufresne authored
This is partly automated using "make update" in docs/plugins, but also required manual merge. Additionally, missing plugins and elements have been added.
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Nicolas Dufresne authored
From ac2f647 to f363b32
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Now we don't have to rely on a special value for the tile number. https://bugzilla.gnome.org/show_bug.cgi?id=767817
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Tim-Philipp Müller authored
gstrtpj2kpay.c:364:21: error: implicit truncation from 'int' to bitfield changes value from -1 to 65535 https://bugzilla.gnome.org/show_bug.cgi?id=767817
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Tim-Philipp Müller authored
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Need to call gst_check_drop_buffers() to release the buffers exchanged during the test. https://bugzilla.gnome.org/show_bug.cgi?id=766561
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Flush the bus when cleaning up so pending messages are destroyed. https://bugzilla.gnome.org/show_bug.cgi?id=766561
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- Flush the bus so messages aren't leaked - Fix pad leak https://bugzilla.gnome.org/show_bug.cgi?id=766561
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Sebastian Dröge authored
This is supposed to be either in the codec_data (avc stream format) or inside the stream, and we extract it from there. It should not be set from a property as it's stream specific. https://bugzilla.gnome.org/show_bug.cgi?id=767789
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- Jun 20, 2016
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Aleix Conchillo Flaqué authored
The Session Data Protocol doesn't allow specifying a cipher for the SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher "aes-128-icm" is the default for SRTP and SRTCP, but if we want to have an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead. https://bugzilla.gnome.org/show_bug.cgi?id=767799
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- Jun 17, 2016
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Tim-Philipp Müller authored
.. by registering the types from the plugin init function. This seems to help, but we'll see if it's enough (might need similar things elsewhere). https://bugzilla.gnome.org/show_bug.cgi?id=693911 https://bugzilla.gnome.org/show_bug.cgi?id=674885
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Sebastian Dröge authored
And the caps in the sink pad template already used variant (only).
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There is no valid use case when this property is needed since the values must be in either codec_data or buffer data. https://bugzilla.gnome.org/show_bug.cgi?id=753760
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Cosmetic change to read NALU type in gst_rtp_h265_pay_decode_nal() the same way as in other places. https://bugzilla.gnome.org/show_bug.cgi?id=753760
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