+2020-09-02 10:49:40 +0100 Justin Chadwell <me@jedevc.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: use unsigned int types to store result of QT_UINT32
+ In a few cases throughout qtdemux, the results of QT_UINT32 were being
+ stored in a signed integer, which could cause subtle bugs in the case of
+ an integer overflow, even allowing the the result to equal a negative
+ number!
+ This patch prevents this by simply storing the results of this function
+ call properly in an unsigned integer type. Additionally, we fix up the
+ length checking with stsd parsing to prevent cases of child atoms
+ exceeding their parent atom sizes.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3344>
+
+2022-11-04 17:48:01 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/qt/gstqtglutility.cc:
+ qt: initialize GError properly in gst_qt_get_gl_wrapcontext()
+ Spotted by Claus Stovgaard.
+ Fixes #1545
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3327>
+
+2022-11-04 11:10:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Add durations to raw audio buffers from the raw audio adapter in prefill mode
+ This ensures that a duration can also be calculated and stored for the
+ last buffer at EOS.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3321>
+
+2022-11-04 10:49:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Release object lock before posting an error message
+ GST_ELEMENT_ERROR() also takes the object lock and this would then
+ deadlock.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3321>
+
+2022-11-03 14:08:57 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/multifile/gstimagesequencesrc.c:
+ imagesequencesrc; Fix leaks
+ * The path was leaked
+ * The custom buffer was never freed
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
+
+2022-11-03 14:08:02 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix cenc-related leaks
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
+
+2022-11-03 14:06:58 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Don't leak metas
+ There is no correlation between the frame being NULL and the metas not being
+ present.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
+
+2022-10-31 16:08:23 +0100 Edward Hervey <edward@centricular.com>
+
+ * ext/adaptivedemux2/gstadaptivedemux-period.c:
+ * ext/adaptivedemux2/gstadaptivedemux.c:
+ adaptivedemux2: Fix collection leaks
+ * The collection on the period was never unreffed
+ * The collection in the message handler was never unreffed
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
+
+2022-11-05 03:23:43 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/mss/gstmssdemux.c:
+ mssdemux2: Update for adaptivedemux2 refactoring
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-11-03 01:48:08 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/dash/gstdashdemux.c:
+ * ext/adaptivedemux2/gstadaptivedemux-private.h:
+ * ext/adaptivedemux2/gstadaptivedemux-stream.c:
+ * ext/adaptivedemux2/gstadaptivedemux-stream.h:
+ * ext/adaptivedemux2/gstadaptivedemux.c:
+ * ext/adaptivedemux2/gstadaptivedemux.h:
+ * ext/adaptivedemux2/hls/gsthlsdemux.c:
+ adaptivedemux2: Move stream_seek() to the Stream class
+ Move the last stream specific vfunc from the demux
+ class to the stream class.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-08-21 04:31:53 +1000 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/dash/gstdashdemux.c:
+ * ext/adaptivedemux2/gstadaptivedemux-private.h:
+ * ext/adaptivedemux2/gstadaptivedemux-stream.c:
+ * ext/adaptivedemux2/gstadaptivedemux-stream.h:
+ * ext/adaptivedemux2/gstadaptivedemux-types.h:
+ * ext/adaptivedemux2/gstadaptivedemux.c:
+ * ext/adaptivedemux2/gstadaptivedemux.h:
+ * ext/adaptivedemux2/hls/gsthlsdemux.c:
+ adaptivedemux2: Refactor stream methods into the stream
+ Unlike the legacy elements, GstAdaptiveDemuxStream is a GObject now,
+ so a bunch of things that were actually stream methods on the
+ parent demux object can directly become stream methods now.
+ Move the stream class out to a header of its own.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-06-07 14:36:24 +1000 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/hls/m3u8.c:
+ * ext/adaptivedemux2/hls/m3u8.h:
+ hlsdemux2/m3u8: Implement EXT-X-GAP parsing
+ Read the EXT-X-GAP tag and set is_gap on the segment.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-06-07 14:13:39 +1000 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/hls/m3u8.c:
+ hlsdemux2/m3u8: Refactor parsing for readability
+ Small readability improvements in the parsing code
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-10-14 06:21:41 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/downloadrequest.c:
+ * ext/adaptivedemux2/downloadrequest.h:
+ adaptivedemux2/downloadhelper: Remove return val for download_request_add_buffer()
+ The function can't actually fail, and the only caller
+ was ignoring the result anyway, so remove the return value.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-10-14 06:20:06 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/downloadhelper.c:
+ adaptivedemux2/downloadhelper: Add debug output of response headers
+ Dump the HTTP response headers at TRACE level
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-10-14 06:19:11 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/downloadhelper.c:
+ adaptivedemux2/downloadhelper: Don't mark transfer as complete/error if cancelled.
+ If the state of the download request was reset to UNSENT,
+ it was cancelled. Don't update the state to COMPLETE or ERRORED
+ in on_read_ready().
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-10-14 06:17:00 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/downloadhelper.c:
+ adaptivedemux2/downloadhelper: Ignore spurious read failure
+ Sometimes g_input_stream_read_all_finish() can return
+ 0 bytes, but still succeed (return TRUE) and have more
+ data available later. Only finish the transfer
+ if it returns 0 bytes *and* FALSE with no error.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-10-14 06:15:45 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/downloadhelper.c:
+ * ext/adaptivedemux2/downloadrequest.c:
+ * ext/adaptivedemux2/downloadrequest.h:
+ adaptivedemux2/downloadhelper: Fix function name
+ Fix a typo in the name of function download_request_despatch_progress()
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-10-12 02:14:32 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/gstadaptivedemux-private.h:
+ * ext/adaptivedemux2/gstadaptivedemux.c:
+ adaptivedemux2: Remove scheduler_lock mutex
+ Remove the old unused scheduler_lock
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-10-11 03:20:11 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/gstadaptivedemux.c:
+ adaptivedemux2: Hold tracks lock accessing input_period
+ The input_period is protected by the TRACKS_LOCK,
+ so make sure to hold that when accessing it.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-08-16 23:01:46 +1000 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/gstadaptivedemux-stream.c:
+ * ext/adaptivedemux2/gstadaptivedemux.h:
+ adaptivedemux2: Add state checks and clean up obsolete variables
+ The cancelled flag was only set in the stream finalize()
+ method, after all activity on the stream has stopped anyway.
+ Replace uses of cancelled with checks on the stream state.
+ Remove the replaced flag, which was checked but never set
+ to TRUE anywhere any more.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
+
+2022-10-30 20:28:25 +0900 Seungha Yang <seungha@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvpxenc.c:
+ vpx: Complete high bitdepth vp9 en/decoding support
+ Adding 12bits variant formats to en/decoder, and high bitdepth
+ 4:4:4 (except for GBR) encoding support
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3298>
+
+2022-10-30 20:03:10 +0900 Seungha Yang <seungha@centricular.com>
+
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvpxcompat.h:
+ * ext/vpx/gstvpxdec.h:
+ * ext/vpx/gstvpxenc.h:
+ vpx: Define formats for compatibility
+ ifdef for enum values never work. Instead, define new enum type
+ and use it
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3298>
+
+2022-10-27 23:57:58 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/hls/m3u8.c:
+ hlsdemux2: m3u8: Use PDT to offset stream time when aligning playlist
+ When matching segments across playlists with Program-Date-Times,
+ use the difference in segment PDTs to adjust the stream time
+ that's being transferred. This can fix cases where the
+ segment boundaries don't align across different streams
+ and the first download gets thrown away once the PTS
+ is seen and found not to match.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3309>
+
+2022-11-01 02:17:46 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/hls/gsthlsdemux.c:
+ * ext/adaptivedemux2/hls/gsthlsdemux.h:
+ hlsdemux2: Download new header when it changes
+ Check whether the init file / MAP data for a segment
+ is different to the current data and trigger an
+ update if so. Previously, the header would only
+ be checked in HLS after switching bitrate or
+ after a seek / first download.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3307>
+
+2022-11-01 01:41:35 +1100 Jan Schmidt <jan@centricular.com>
+
+ * ext/adaptivedemux2/hls/m3u8.c:
+ * ext/adaptivedemux2/hls/m3u8.h:
+ m3u8: Expose GstM3U8InitFile methods
+ Exposure ref/unref methods for the GstM3U8InitFile type,
+ and add a gst_m3u8_init_file_equal() comparison method.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3307>
+
+2022-10-21 17:24:41 +0200 Edward Hervey <edward@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/adaptivedemux2/gstadaptivedemux-stream.c:
+ * ext/adaptivedemux2/gstadaptivedemux.c:
+ * ext/adaptivedemux2/gstadaptivedemux.h:
+ * ext/adaptivedemux2/hls/gsthlsdemux.c:
+ * ext/adaptivedemux2/hls/m3u8.c:
+ * ext/adaptivedemux2/hls/m3u8.h:
+ adaptivedemux2: Improve minimum buffering threshold
+ Previously the minimum buffering threshold was hardcoded to a specific
+ value (10s). This is suboptimal this an actual value will depend on the actual
+ stream being played.
+ This commit sets the low watermark threshold in time to 0, which is an automatic
+ mode. Subclasses can provide a stream `recommended_buffering_threshold` when
+ update_stream_info() is called.
+ Currently implemented for HLS, where we recommended 1.5 average segment
+ duration. This will result in buffering being at 100% when the 2nd segment has
+ been downloaded (minus a bit already being consumed downstream)
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3240>
+
+2022-10-28 18:57:44 +0530 Sanchayan Maity <sanchayan@asymptotic.io>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Speed up type finding for DTS
+ In order to figure out if the "raw" audio contained within the wav
+ container is actually DTS, right now we call the typefinder helper
+ which runs all typefinders.
+ Speed up this type finding process by specifying the extension.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3294>
+
+2022-10-25 13:30:15 +1100 Matthew Waters <matthew@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/isomp4/gstqtmuxmap.c:
+ mp4mux: enable muxing VP9 streams
+ As specified in https://www.webmproject.org/vp9/mp4/
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3260>
+
+2022-10-25 13:28:26 +1100 Matthew Waters <matthew@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: add support for writing vpcC box for VP9
+ Increases compatibility for VP9 in .mov in at least VLC.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3260>
+
+2022-10-04 18:21:15 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * ext/adaptivedemux2/dash/gstdashdemux.c:
+ * ext/adaptivedemux2/gstadaptivedemux.c:
+ dashdemux2: Fix the way we determine current_position after seeks
+ Without that the current_position was off after seeks, potentially
+ leading to not properly push a last fragment when a `.stop` time was
+ set.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
+
+2022-09-20 15:32:52 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * ext/adaptivedemux2/dash/gstmpdclient.c:
+ dash: Fix computing `repeat_index` when seeking in stream with a start !=0 on the first fragment
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
+
+2022-09-22 11:20:55 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Let upstream handle seeking/duration query in time if possible
+ So proper response are given for dash streams
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
+
+2022-09-21 15:01:39 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-demux.h:
+ matroskademux: Start support for upstream segments in TIME format
+ So we can use matroskademux for dash webm dash streams.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
+
+2022-01-24 16:49:52 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: grab the server while capturing screen image
+ Makes sure screen resolution doesn't change in the middle of the
+ process.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1562>
+
+2021-12-17 14:57:57 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * sys/ximage/gstximagesrc.c:
+ * sys/ximage/gstximagesrc.h:
+ ximagesrc: change video resolution when X11 screen gets resized
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1562>
+
+2022-10-23 20:32:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/meson.build:
+ * gst/xingmux/gstxingmux.c:
+ * gst/xingmux/gstxingmux.h:
+ * gst/xingmux/meson.build:
+ * gst/xingmux/plugin.c:
+ * meson_options.txt:
+ * tests/check/elements/xingmux.c:
+ * tests/check/elements/xingmux_testdata.h:
+ * tests/check/meson.build:
+ xingmux: move from gst-plugins-ugly to gst-plugins-good
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/415
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3251>
+
+2022-10-21 16:23:08 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Only EOS on timeout if all streams are timed out/EOS
+ Otherwise a stream that is just temporarily inactive might time out and
+ then can never become active again because the EOS event was sent
+ already.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3238>
+
+2022-10-18 16:51:39 +1100 Matthew Waters <matthew@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtp/gstrtpulpfecdec.c:
+ * gst/rtp/gstrtpulpfecdec.h:
+ rtpulpfecdec: add property for passthrough
+ Support for enabling and disabling decoding of FEC data decoding on
+ packet loss events and unconditional seqnum rewriting of packets.
+ See
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/581
+ for background.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3212>
+
+2022-10-14 01:23:04 +0000 Devin Anderson <danderson@microsoft.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Avoid occasional crash due to referencing freed buffer.
+ We've seen occasional crashes in the `wavparse` module associated with
+ referencing a buffer in `gst_wavparse_chain` that's already been freed. The
+ reference is stolen when the buffer is transferred to the adapter with
+ `gst_adapter_push` and, IIUC, assuming the source doesn't hold a reference to
+ the buffer, the buffer could be freed during interaction with the adapter in
+ `gst_wavparse_stream_headers`.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3179>
+
+2022-10-13 11:21:42 -0400 Julian Bouzas <julian.bouzas@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ riff: Mark jpeg as parsed
+ This is needed so that autoplugging works with avidemux and JPEG decoders that
+ need parsed sink caps (eg rockchip 'mppjpegdec' decoder). It also works fine
+ with 'jpegdec' decoder regardless.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3175>
+
+2022-10-13 00:20:45 +0000 Devin Anderson <danderson@microsoft.com>
+
+ * gst/wavparse/gstwavparse.c:
+ * tests/check/elements/wavparse.c:
+ * tests/files/corruptheadertestsrc.wav:
+ wavparse: Fix crash that occurs in push mode when header chunks are corrupted in certain ways.
+ In the case that a test is provided for, the size of the `fmt ` chunk is
+ changed from 16 bytes to 18 bytes (bytes 17 - 20 below):
+ ```
+ $ hexdump -C corruptheadertestsrc.wav
+ 00000000 52 49 46 46 e4 fd 00 00 57 41 56 45 66 6d 74 20 |RIFF....WAVEfmt |
+ 00000010 12 00 00 00 01 00 01 00 80 3e 00 00 00 7d 00 00 |.........>...}..|
+ 00000020 02 00 10 00 64 61 74 61 |....data|
+ 00000028
+ ```
+ (Note that the original file is much larger. This was the smallest sub-file
+ I could find that would generate the crash.)
+ Note that, while the same issue doesn't cause a crash in pull mode, there's a
+ different issue in that the file is processed successfully as if it was a .wav
+ file with zero samples.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3173>
+
+2022-10-11 15:00:37 +0200 Edward Hervey <edward@centricular.com>
+
+ * sys/oss4/oss4-sink.c:
+ * sys/oss4/oss4-source.c:
+ oss4: Fix debug category initialization
+ Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1456
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3158>
+
+2022-10-08 01:03:13 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxsrc: don't queue data on unlinked pads
+ Once a pad has returned NOT_LINKED, the part reader shouldn't let its
+ corresponding data queue run full and eventually (after 20 seconds)
+ stall playback.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3145>
+
+2022-10-03 20:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpsource: Don't do probation for RTX sources
+ Disable probation for RTX sources as packets will arrive very
+ irregularly and waiting for a second packet usually exceeds the deadline
+ of the retransmission.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/181
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
+
+2022-10-03 19:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/rtp/client-rtpaux.c:
+ rtp: examples: client-rtpaux: Provide correct caps by payload type and RTX pt map by session
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
+
+2019-01-25 17:04:50 -0500 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * tests/check/elements/rtpsession.c:
+ tests/check/rtpsession: extend test_internal_sources_timeout
+ to verify that rtx SSRCs do not BYE after timeout
+ See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
+
+2022-10-03 19:12:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpsession: Remember the corresponding media SSRC for RTX sources
+ This allows timing out the RTX source and sending BYE for it when the
+ actual media source belonging to it is timed out.
+ This change only applies to sending sources from this session.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
+
+2022-10-03 19:20:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpsource: Rename rtp_source_update_caps to rtp_source_update_send_caps
+ To make it clear that this is only used for sending RTP sources.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
+
+2022-10-03 13:48:36 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Rename gst_rtp_session_sink_setcaps to gst_rtp_session_setcaps_recv_rtp
+ to make it clearer that this is for setting receiver caps and to make it
+ more consistent with gst_rtp_session_setcaps_send_rtp.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
+
+2022-10-06 15:02:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Retry SETUP with non-compliant URL resolution on "Bad Request" and "Not found"
+ Various RTSP servers/cameras assume base and control URL to be simply
+ appended instead of being resolved according to the relative URL
+ resolution algorithm as mandated by the RTSP specification.
+ To work around this, try using such a non-compliant control URL if the
+ server didn't like the URL used in the first SETUP request.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1447
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/922
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3127>
+
+2022-10-04 03:57:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3115>
+
=== release 1.21.1 ===
2022-10-04 01:14:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+ * ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
projects / platform / upstream / gstreamer.git / blobdiff