- Jul 14, 2017
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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 13, 2017
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Sebastian Dröge authored
Regression introduced by 86b427dc https://bugzilla.gnome.org/show_bug.cgi?id=784812
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- Jul 12, 2017
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Josep Torra authored
Fixes a negotiation error seen when trying to playback of a .MOV file with a mono AAC audio stream decoded by avcdec_aac that doesn't set channel-mask field but sink was requiring channel-mask=0x3.
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- Jul 09, 2017
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Yasushi SHOJI authored
Do not allocate payload size outbuf if appending payload buffer. The commit 137672ff attached payload to the output buffer but forgot to remove payload allocation. That effectively doubled payload size and add zero'ed or random bytes. Makes the following pipeline work again: gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay ! gsmdec ! autoaudiosink https://bugzilla.gnome.org/show_bug.cgi?id=784616
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Nicolas Dufresne authored
This element was not checking if mapping the RTP buffer and the payload worked, and was not checking if the RTX payload was large enough. https://bugzilla.gnome.org/show_bug.cgi?id=784484
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- Jul 05, 2017
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Tim-Philipp Müller authored
Get seqnum before unreffing the seek event. https://bugzilla.gnome.org/show_bug.cgi?id=784486
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- Jul 04, 2017
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Sebastian Dröge authored
Also fixes a crash caused by freeing an uninitialized mutex in an error case. https://bugzilla.gnome.org//show_bug.cgi?id=784282
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Sebastian Dröge authored
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- Jun 20, 2017
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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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- Jun 17, 2017
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Vivia Nikolaidou authored
They can cause us to deadlock, while we're waiting for a new frame and upstream is waiting for the allocation query to be answered before sending a frame https://bugzilla.gnome.org/show_bug.cgi?id=783753
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Sebastian Dröge authored
We currently send data to the RTSP connection from multiple threads: whenever a command is to be handled and whenever RTCP is generated. This can cause data corruption or worse if both happen at the same time. As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive() calls with a mutex. While this means that we hold a mutex during the IO operation, this is not actually a problem as the IO operation can be interrupted (gst_rtsp_connection_flush()) at any time and is blocking by itself anyway.
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- Jun 15, 2017
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Sebastian Dröge authored
The last entry will most likely get new samples added to it in "robust" muxing mode, changing the samples_per_chunk and thus making it wrong to keep the last two entries merged. It will run into an assertion later when adding a new sample to the chunk. Thanks to gdiener@cardinalpeak.com for the analysis of the bug and proposal for a solution.
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- Jun 13, 2017
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Sebastian Dröge authored
There might be other chunks after the data chunk, so clipping the chunk size with the data size can lead to a negative number and all following calculations go wrong and cause crashes or worse. This was introduced in 3ac119bb. https://bugzilla.gnome.org/show_bug.cgi?id=783760
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- Jun 12, 2017
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Juan Navarro authored
This adds printing the actual value of any unknown RTCP PT to the already existing WARNING log message. https://bugzilla.gnome.org/show_bug.cgi?id=783248
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- Jun 02, 2017
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Tim-Philipp Müller authored
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vijay authored
While reusing aacparse caps were not set.This fix enables aacparse to reuse in same pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=783027
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- May 29, 2017
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Vivia Nikolaidou authored
Timecode trak is only supported for mov right now, not for mp4. That code would otherwise create an invalid trak if the muxed video contained timecode metadata. https://bugzilla.gnome.org/show_bug.cgi?id=782684
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- May 12, 2017
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Sebastian Dröge authored
Print the right one in debug output to get meaningful numbers.
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- May 09, 2017
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- May 08, 2017
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Dustin Spicuzza authored
It's more accurate and allows cancellation. https://bugzilla.gnome.org/show_bug.cgi?id=773681
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Tim-Philipp Müller authored
Such as 1.3.0 as on raspbian.
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Nirbheek Chauhan authored
We were unnecessarily looping/goto-ing repeatedly when we had exactly the amount of data as the free space, and also when the free space was too small. This, as it turns out, is a very common scenario with Directsound on Windows. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=773681 We have to do polling here because the event notification API that Directsound exposes cannot be used with live playback since all events must be registered in advance with the capture buffer, you cannot add/remove them once playback has begun. Directsoundsrc had the same problem. See also: https://bugzilla.gnome.org/show_bug.cgi?id=781249
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- May 04, 2017
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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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- May 02, 2017
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Seungha Yang authored
Since mss has no moov, default stsd entry should be created with media-caps. https://bugzilla.gnome.org/show_bug.cgi?id=782042
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- Apr 27, 2017
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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
That case is correctly handled below but not in the debug output. https://bugzilla.gnome.org/show_bug.cgi?id=781270
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- Apr 25, 2017
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Sebastian Dröge authored
If gst_segment_do_seek() fails, we shouldn't try seeking on that resulting segment but just error out. Crashes further down the line otherwise.
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- Apr 24, 2017
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Tim-Philipp Müller authored
From 60aeef6 to 48a5d85
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Tim-Philipp Müller authored
Make sure avc output doesn't contain SPS/PPS inline, but byte-stream output does.
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Tim-Philipp Müller authored
SPS/PPS are in the caps in this case and shouldn't be in the stream data.
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- Apr 21, 2017
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Sebastian Dröge authored
If no clock was provided directly by rtspsrc. This behaviour was removed by f8013487 and results in rtspsrc not providing the system clock via the rtpjitterbuffer. As a result, if another element like an audio sink, provides a clock, the pipeline would select that (when going to PAUSED/PLAYING again later). Audio clocks usually don't progress in PAUSED, and thus our live source won't be able to use the clock to produce data, making the sink never preroll and everything is stuck.
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Jürgen Sachs authored
Fixes stream where sample_description_id is specified in the tfhd https://bugzilla.gnome.org/show_bug.cgi?id=778337
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