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Plumb through clear-ssrc signal to rtp session, remove the corresponding rtp source #25
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Basically, don't set a DISCONT flag when detecting a seqnum jump. This is useful for ULPFEC, as a FEC packet might be hiding inside the original stream, and this stream is actually completely valid, even if the FEC packet is removed.
Also introducing a test-suite for speex. https://bugzilla.gnome.org/show_bug.cgi?id=754226
Don't send a segment event if we haven't produced data and set the output caps yet. Doing so causes a g_warning about sticky event misordering. https://bugzilla.gnome.org/show_bug.cgi?id=773510
Allows it to be configured differently for different use-cases, specifically the live scenario would probably never want this enabled. https://bugzilla.gnome.org/show_bug.cgi?id=773511
This is a Pexip specific change since we are only interested in using gstflvdemux live.
This happens all the time for us, so no point warning about it.
Multiple SIMD-ified vpx plugins will be built if available. These plugins will be put in a separate plugin directory which can be linked to if configuring with --enable-simd-symlink=avx etc.
Hard code limit to 1500 ms between each generated key frame. This should ideally be a property and set to a value derived from RTT. This avoids generating bursts of keyframes when receiving many PLIs/NACKs in close succession. 1500ms is quite large, but with 500ms and lots of packetloss, the frequent keyframe generation will more then double the bitrate from the libvpx encoder (configured at 1mbps, producing over 2mbps), potentially making an already bad situation much worse. By adding another second we only see about 1,4mbps, which is more acceptable.
Adds property that can be used to disable direct rendering.
When buffer is pushed downstream, we should not hold the buffer mapped with write access. Doing so would often lead to unneccesary memcpy later. For instance, gst_buffer_make_writable() in gst_video_decoder_finish_frame() will cause a memcpy because of _memory_get_exclusive_reference().
Change so that if the intersection of upstream and template caps does not contain a specific annex it will be not added to the caps. Previous behavior was that it was set explictily to false. This fixes an issue if an encoder (avenc_h263p) is linked late with the payloader it could end up as not negotiated.
These can spam a lot if it actually happens.
The caps for Opus is generated via a utility function that creates a new GstCaps instance after validating the codec arguments. There are possible ways to solve this, we chose the one requiring the least number of modifications. It would be also possible to simply set the caps with the specified arguments; however, this will required to move all the validation to qtdemux or require a change to gstreamer codec-utils.
…t to upstream default of 4096.
…mpl chunks Thanks to Antonio Morales for finding and reporting the issue. Fixes GHSL-2024-259 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3887 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8042>
The size does not include the 8 bytes tag and length, so an additional 8 bytes must be removed here. 8 bytes are always available at this point because otherwise the parsing of the tag and length right above would've failed. Thanks to Antonio Morales for finding and reporting the issue. Fixes GHSL-2024-260 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3888 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8042>
Thanks to Antonio Morales for finding and reporting the issue. Fixes GHSL-2024-261 Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3889 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8042>
Incorrect parsing of these bits meant that we were incorrectly parsing the VP9 uncompressed bitstream header for some profiles, as the header is of variable length and format depending on the profile. Amongst various unintended effects, this caused the width and height from the SS to be incorrectly parsed and set in the caps.
This does not work on windows and is undefined on other platforms. Use g_ascii_strtoll instead. Add test for this scenario. (fails on windows without fix!) Co-authored-by: Misha Baranov <[email protected]>
When you unplug a device you might not get it reported in device_removed() but on device_state_changed() with a new state. That is why we need to validate the state individually even after passing DEVICE_STATE_ACTIVE to IMMDeviceEnumerator_EnumAudioEndpoints()
Turns out that both can happen if you push on a pad that is being ripped away, so ignore them both. Also fixup test to compare a flow-return using int compare, since they are mostly negative!
prevent rtxsend to refer to the buffer after it was pushed without increfing it
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…lock() by calling Flush() on the reader This change allows us to unblock the source thread that could be locked for a long time waiting for ReadSample() to finish.
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Found out that ReadSample() could block until we release the samples we have buffered, so clearing the queue will force this when we want to flush effectively unblocking the pushing thread.
…w us disabling the mediafoundation codecs
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gst_mf_capture_winrt_thread_func()
winrt fails to start capture
- Clear the queue *before* calling Flush() - Call Flush() with MF_SOURCE_READER_ALL_STREAMS, specifying the stream index is not necessary
…rtp checkpoint later nasty things can happent
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