Script 'mail_helper' called by obssrc
Hello community,
here is the log from the commit of package gstreamer-plugins-base for
openSUSE:Factory checked in at 2021-04-10 15:26:32
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/gstreamer-plugins-base (Old)
and /work/SRC/openSUSE:Factory/.gstreamer-plugins-base.new.2401 (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "gstreamer-plugins-base"
Sat Apr 10 15:26:32 2021 rev:76 rq:883602 version:1.18.4
Changes:
--------
---
/work/SRC/openSUSE:Factory/gstreamer-plugins-base/gstreamer-plugins-base.changes
2021-01-20 18:24:19.307348540 +0100
+++
/work/SRC/openSUSE:Factory/.gstreamer-plugins-base.new.2401/gstreamer-plugins-base.changes
2021-04-10 15:27:14.962369425 +0200
@@ -1,0 +2,16 @@
+Tue Mar 30 08:59:41 UTC 2021 - Antonio Larrosa <alarr...@suse.com>
+
+- Update to version 1.18.4:
+ + tag: id3v2: fix frame size check and potential invalid reads
+ + audio: Fix gst_audio_buffer_truncate() meta handling for non-interleaved
audio
+ + audioresample: respect buffer layout when draining
+ + audioaggregator: fix input_buffer ownership
+ + decodebin3: change stream selection message owner, so that the app sends
the stream-selection event to the right element
+ + rtspconnection: correct data_size when tunneled mode
+ + uridecodebin3: make caps property work
+ + video-converter: Don't upsample invalid lines
+ + videodecoder: Fix racy critical when pool negotiation occurs during flush
+ + video: Convert gst_video_info_to_caps() to take self as const ptr
+ + examples: added qt core dependency for qt overlay example
+
+-------------------------------------------------------------------
Old:
----
gst-plugins-base-1.18.3.tar.xz
New:
----
gst-plugins-base-1.18.4.tar.xz
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Other differences:
------------------
++++++ gstreamer-plugins-base.spec ++++++
--- /var/tmp/diff_new_pack.jy40ng/_old 2021-04-10 15:27:15.534370099 +0200
+++ /var/tmp/diff_new_pack.jy40ng/_new 2021-04-10 15:27:15.534370099 +0200
@@ -20,10 +20,10 @@
%define gst_branch 1.0
%define gstreamer_req_version %(echo %{version} | sed -e "s/+.*//")
Name: gstreamer-plugins-base
-Version: 1.18.3
+Version: 1.18.4
Release: 0
Summary: GStreamer Streaming-Media Framework Plug-Ins
-License: LGPL-2.1-or-later AND GPL-2.0-or-later
+License: GPL-2.0-or-later AND LGPL-2.1-or-later
Group: Productivity/Multimedia/Other
URL: https://gstreamer.freedesktop.org
# Disable tarball source url, use _service
++++++ _service ++++++
--- /var/tmp/diff_new_pack.jy40ng/_old 2021-04-10 15:27:15.554370122 +0200
+++ /var/tmp/diff_new_pack.jy40ng/_new 2021-04-10 15:27:15.554370122 +0200
@@ -9,7 +9,7 @@
<!--
<param name="changesgenerate">enable</param>
-->
- <param name="revision">1.18.3</param>
+ <param name="revision">1.18.4</param>
<param name="scm">git</param>
</service>
<service name="recompress" mode="disabled">
++++++ gst-plugins-base-1.18.3.tar.xz -> gst-plugins-base-1.18.4.tar.xz ++++++
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-plugins-base-1.18.3/ChangeLog
new/gst-plugins-base-1.18.4/ChangeLog
--- old/gst-plugins-base-1.18.3/ChangeLog 2021-01-13 22:07:13.000000000
+0100
+++ new/gst-plugins-base-1.18.4/ChangeLog 2021-03-15 18:48:00.000000000
+0100
@@ -1,3 +1,158 @@
+=== release 1.18.4 ===
+
+2021-03-15 17:47:59 +0000 Tim-Philipp M??ller <t...@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * gst-plugins-base.doap:
+ * meson.build:
+ Release 1.18.4
+
+2021-03-03 01:08:25 +0000 Tim-Philipp M??ller <t...@centricular.com>
+
+ * gst-libs/gst/tag/id3v2frames.c:
+ tag: id3v2: fix frame size check and potential invalid reads
+ Check the right variable when checking if there's
+ enough data left to read the frame size.
+ Closes
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/876
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1066>
+
+2021-03-10 14:26:22 +0100 Guillaume Desmottes
<guillaume.desmot...@collabora.com>
+
+ * gst-libs/gst/audio/gstaudioaggregator.c:
+ audioaggregator: fix input_buffer ownership
+ The way pad->priv->input_buffer reference was managed was pretty
+ spurious:
+ - it was overridden without unrefing it, which could potentially lead
to
+ leaks.
+ - we were unreffing it while keeping the pointer around, which could
+ potentially lead to use-after-free or double-free.
+ As priv->input_buffer is actually no longer used outside of the
+ aggregate() method, remove it from pad->priv to simplify the code and
+ prevent the issues desribed above.
+ Fix a single buffer leak when shutting down the pipeline as the buffer
+ returned from gst_aggregator_pad_drop_buffer() was never unreffed.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1062>
+
+2021-03-10 16:22:14 +0100 Guillaume Desmottes
<guillaume.desmot...@collabora.com>
+
+ * gst-libs/gst/audio/gstaudioaggregator.c:
+ audioaggregator: fix input buffer when converting
+ This code path is meant to convert the current buffer to the new
format
+ on update. It was using priv->input_buffer as input which is either
+ priv->buffer or a converted version of it.
+ Use priv->buffer instead as priv->input_buffer may no longer be a
valid
+ reference.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1062>
+
+2021-02-19 16:44:35 +0200 Vivia Nikolaidou <vi...@ahiru.eu>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Don't upsample/downsample/dither invalid lines
+ This is a fallout from the conversion to support multiple threads.
+ convert->upsample_p is never NULL now, it's always an allocated array
of
+ n_threads potentially-null pointers.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1054>
+
+2021-02-25 11:03:31 +0100 Kristofer Bj??rkstr??m <krist...@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ gstrtspconnection: correct data_size when tunneled mode
+ gst_rtsp_connection_send_messages_usec in tunneled mode does base64
+ encode messages. When calculating data_size 1 bytes is added, which
+ results in ending the base64 with a NULL.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1053>
+
+2021-02-24 19:51:40 +0200 Sebastian Dr??ge <sebast...@centricular.com>
+
+ * gst-libs/gst/audio/gstaudioaggregator.c:
+ audioaggregator: Log if the sample rate of one sinkpad is not accepted
+ Otherwise this can silently cause not-negotiated errors without any
+ direct hint about what went wrong.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1052>
+
+2021-02-22 15:36:53 +0900 Jeongki Kim <jeongki....@jeongki.kim>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Respect buffer layout when drain
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1050>
+
+2021-01-19 15:56:18 +0100 St??phane Cerveau <scerv...@collabora.com>
+
+ * gst/playback/gstdecodebin3.c:
+ decodebin3: change stream selection message owner
+ In order to select the streams on GST_MESSAGE_STREAM_COLLECTION,
+ the app needs to send the select-streams event
+ to the decodebin and not to the parsebin.
+ The message should be always owned by the decodebin.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1044>
+
+2021-02-15 16:05:30 +0000 Tim-Philipp M??ller <t...@centricular.com>
+
+ * gst/playback/gsturidecodebin3.c:
+ uridecodebin3: make caps property work
+ The caps set on uridecodebin3 via the "caps" property
+ were never passed to the internal decodebin3, so did
+ absolutely nothing.
+ Fixes
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/837
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1038>
+
+2021-02-13 00:27:04 +0100 Alicia Boya Garc??a <ntr...@gmail.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Fix racy critical when pool negotiation occurs during
flush
+ I found a rather reproducible race in a WebKit LayoutTest when a
player
+ was intantiated and a VP8/9 video was loaded, then torn down after
+ getting the video dimensions from the caps.
+ The crash occurs during the handling of the first frame by gstvpxdec.
+ The following actions happen sequentially leading to a crash.
+ (MT=Main Thread, ST=Streaming Thread)
+ MT: Sets pipeline state to NULL, which deactivates vpxdec's srcpad,
+ which in turn sets its FLUSHING flag.
+ ST: gst_vpx_dec_handle_frame() -- which is still running -- calls
+ gst_video_decoder_allocate_output_frame(); this in turn calls
+ gst_video_decoder_negotiate_unlocked() which fails because the
+ srcpad is FLUSHING. As a direct consequence of the negotiation
+ failure, a pool is NOT set.
+ gst_video_decoder_negotiate_unlocked() still assumes there is a
+ pool, crashing in a critical in gst_buffer_pool_acquire_buffer()
+ a couple statements later.
+ This patch fixes the bug by returning != GST_FLOW_OK when the
+ negotiation fails. If the srcpad is FLUSHING, GST_FLOW_FLUSHING is
+ returned, otherwise GST_FLOW_ERROR is used.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1037>
+
+2021-02-15 17:22:47 +0100 Jan Alexander Steffens (heftig)
<jan.steff...@ltnglobal.com>
+
+ * gst-libs/gst/audio/audio.c:
+ libs: audio: Fix gst_audio_buffer_truncate meta handling
+ In the non-interleaved case, it made `buffer` writable but then
changed
+ the meta of the non-writable buffer.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1036>
+
+2021-01-26 14:05:48 +0100 Knobe, Daniel <daniel.kn...@miele.com>
+
+ * tests/examples/overlay/meson.build:
+ overlay/example: added qt core dependency for qt overlay example
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1026>
+
+2021-01-12 10:36:34 +0100 Marijn Suijten <marijn...@gmail.com>
+
+ * gst-libs/gst/video/video-info.c:
+ * gst-libs/gst/video/video-info.h:
+ video: Convert info_to_caps to take self as const ptr
+ This requires a slight modification to the function itself because it
+ was overwriting a member locally.
+ However, now this side-effect cannot be observed outside the function
+ anymore.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1024>
+
+2021-01-14 02:16:57 +0000 Tim-Philipp M??ller <t...@centricular.com>
+
+ * meson.build:
+ Back to development
+
=== release 1.18.3 ===
2021-01-13 21:07:11 +0000 Tim-Philipp M??ller <t...@centricular.com>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-plugins-base-1.18.3/NEWS
new/gst-plugins-base-1.18.4/NEWS
--- old/gst-plugins-base-1.18.3/NEWS 2021-01-13 22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/NEWS 2021-03-15 18:48:00.000000000 +0100
@@ -2,13 +2,13 @@
GStreamer 1.18.0 was originally released on 8 September 2020.
-The latest bug-fix release in the 1.18 series is 1.18.3 and was released
-on 13 January 2021.
+The latest bug-fix release in the 1.18 series is 1.18.4 and was released
+on 15 March 2021.
See https://gstreamer.freedesktop.org/releases/1.18/ for the latest
version of this document.
-Last updated: Wednesday 13 January 2021, 20:00 UTC (log)
+Last updated: Monday 15 March 2021, 13:00 UTC (log)
Introduction
@@ -2717,6 +2717,168 @@
- List of Merge Requests applied in 1.18.3
- List of Issues fixed in 1.18.3
+1.18.4
+
+The fourth 1.18 bug-fix release (1.18.4) was released on 15 March 2021.
+
+This release only contains bugfixes and security fixes and it should be
+safe to update from 1.18.x.
+
+Highlighted bugfixes in 1.18.4
+
+- important security fixes for ID3 tag reading, matroska and realmedia
+ parsing, and gst-libav audio decoding
+- audiomixer, audioaggregator: input buffer handling fixes
+- decodebin3: improve stream-selection message handling
+- uridecodebin3: make ???caps??? property work
+- wavenc: fix writing of INFO chunks in some cases
+- v4l2: bt601 colorimetry, allow encoder resolution changes, fix
+ decoder frame rate negotiation
+- decklinkvideosink: fix auto format detection, and fixes for 29.97fps
+ framerate output
+- mpeg-2 video handling fixes when seeking
+- avviddec: fix bufferpool negotiation and possible memory corruption
+ when changing resolution
+- various stability, performance and reliability improvements
+- memory leak fixes
+- build fixes: rpicamsrc, qt overlay example, d3d11videosink on UWP
+
+gstreamer
+
+- info: Don???t leak log function user_data if the debug system is
+ compiled out
+- task: Use SetThreadDescription() Win32 API for setting thread names,
+ which preserves thread names in dump files.
+- buffer, memory: Mark info in map functions as caller-allocates and
+ pass allocation params as const pointers where possible
+- clock: define AUTO_CLEANUP_FREE_FUNC for GstClockID
+
+gst-plugins-base
+
+- tag: id3v2: fix frame size check and potential invalid reads
+- audio: Fix gst_audio_buffer_truncate() meta handling for
+ non-interleaved audio
+- audioresample: respect buffer layout when draining
+- audioaggregator: fix input_buffer ownership
+- decodebin3: change stream selection message owner, so that the app
+ sends the stream-selection event to the right element
+- rtspconnection: correct data_size when tunneled mode
+- uridecodebin3: make caps property work
+- video-converter: Don???t upsample invalid lines
+- videodecoder: Fix racy critical when pool negotiation occurs during
+ flush
+- video: Convert gst_video_info_to_caps() to take self as const ptr
+- examples: added qt core dependency for qt overlay example
+
+gst-plugins-good
+
+- matroskademux: header parsing fixes
+- rpicamsrc: depend on posix threads and vchiq_arm to fix build on
+ raspios again
+- wavenc: Fixed INFO chunk corruption, caused by odd sized data not
+ being padded
+- wavpackdec: Add floating point format support to fix distortions in
+ some cases
+- v4l2: recognize V4L2 bt601 colorimetry again
+- v4l2videoenc: support resolution change stream encode
+- v4l2h265codec: fix HEVC profile string issue
+- v4l2object: Need keep same transfer as input caps
+- v4l2videodec: Fix vp8 and vp9 streams can???t play on board with
+ vendor bsp
+- v4l2videodec: fix src side frame rate negotiation
+
+gst-plugins-bad
+
+- avwait: Don???t post messages with the mutex locked
+- d3d11h264dec: Reconfigure decoder object on DPB size change and keep
+ track of actually configured DPB size
+- dashsink: fix double unref of sinkpad caps
+- decklinkvideosink: Use correct numerator for 29.97fps
+- decklinkvideosink: fix auto format detection
+- decklinksrc: Use a more accurate capture time
+- d3d11videosink: Fix build error on UWP
+- interlace: negotiation and buffer leak fixes
+- mpegvideoparse: do not clip, so decoder receives data from keyframe
+ even if it???s before the segment start
+- mpegtsparse: Fix switched DTS/PTS when set-timestamps=false
+- nvh264sldec: Reopen decoder object if larger DPB size is required
+- sdpsrc: fix double free if sdp is provided as string via the
+ property
+- vulkan: Fix elements long name.
+
+gst-plugins-ugly
+
+- rmdemux: Make sure we have enough data available when parsing
+ audio/video packets
+
+gst-libav
+
+- avviddec: take the maximum of the height/coded_height
+- viddec: don???t configure an incorrect buffer pool when receiving a
+ gap event
+- audiodec: fix stack overflow in gst_ffmpeg_channel_layout_to_gst()
+
+gst-rtsp-server
+
+- rtspclientsink: fix deadlock on shutdown if no data has been
+ received yet
+- rtspclientsink: fix leaks in unit tests
+- rtsp-stream: avoid deadlock in send_func
+- rtsp-client: cleanup transports during TEARDOWN
+
+gstreamer-vaapi
+
+- h264 encoder: append encoder exposure to aud
+- postproc: Fix a problem of propose_allocation when passthrough
+- glx: Iterate over FBConfig and select 8 bit color size
+
+gstreamer-sharp
+
+- no changes
+
+gst-omx
+
+- no changes
+
+gst-python
+
+- no changes
+
+gst-editing-services
+
+- group: Use proper group constructor
+
+gst-integration-testsuites
+
+- no changes
+
+gst-build
+
+- no changes
+
+Cerbero build tool and packaging changes in 1.18.4
+
+- macOS: more BigSur fixes
+- glib: Backport patch to set thread names on Windows 10
+
+Contributors to 1.18.4
+
+Alicia Boya Garc??a, Ashley Brighthope, Bing Song, Branko Subasic, Edward
+Hervey, Guillaume Desmottes, Haihua Hu, He Junyan, Hou Qi, Jan Alexander
+Steffens (heftig), Jeongki Kim, Jordan Petridis, Knobe, Kristofer
+Bj??rkstr??m, Marijn Suijten, Matthew Waters, Paul Goulpi??, Philipp Zabel,
+Rafa?? Dzi??giel, Sebastian Dr??ge, Seungha Yang, Staz M, St??phane Cerveau,
+Thibault Saunier, Tim-Philipp M??ller, V??ctor Manuel J??quez Leal, Vivia
+Nikolaidou, Vladimir Menshakov,
+
+??? and many others who have contributed bug reports, translations, sent
+suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.18.4
+
+- List of Merge Requests applied in 1.18.4
+- List of Issues fixed in 1.18.4
+
Schedule for 1.20
Our next major feature release will be 1.20, and 1.19 will be the
@@ -2724,9 +2886,9 @@
development of 1.19/1.20 will happen in the git master branch.
The plan for the 1.20 development cycle is yet to be confirmed, but it
-is now expected that feature freeze will take place some time in
-January/February 2021, with the first 1.20 stable release hopefully
-around February/March 2021.
+is now expected that feature freeze will take place some time in April
+2021, with the first 1.20 stable release hopefully around April/May
+2021.
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-plugins-base-1.18.3/RELEASE
new/gst-plugins-base-1.18.4/RELEASE
--- old/gst-plugins-base-1.18.3/RELEASE 2021-01-13 22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/RELEASE 2021-03-15 18:48:00.000000000 +0100
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-base 1.18.3.
+This is GStreamer gst-plugins-base 1.18.4.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst/audioresample/gstaudioresample.c
new/gst-plugins-base-1.18.4/gst/audioresample/gstaudioresample.c
--- old/gst-plugins-base-1.18.3/gst/audioresample/gstaudioresample.c
2021-01-13 22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst/audioresample/gstaudioresample.c
2021-03-15 18:48:00.000000000 +0100
@@ -557,7 +557,8 @@
gst_audio_resample_dump_drain (GstAudioResample * resample, guint history_len)
{
gsize out_len, outsize;
- gpointer out[1];
+ GstBuffer *outbuf;
+ GstAudioBuffer abuf;
out_len =
gst_audio_converter_get_out_frames (resample->converter, history_len);
@@ -565,11 +566,19 @@
return;
outsize = out_len * resample->out.bpf;
+ outbuf = gst_buffer_new_and_alloc (outsize);
- out[0] = g_malloc (outsize);
+ if (GST_AUDIO_INFO_LAYOUT (&resample->out) ==
+ GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
+ gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL);
+ }
+
+ gst_audio_buffer_map (&abuf, &resample->out, outbuf, GST_MAP_WRITE);
gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
- out, out_len);
- g_free (out[0]);
+ abuf.planes, out_len);
+ gst_audio_buffer_unmap (&abuf);
+
+ gst_buffer_unref (outbuf);
}
static void
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-plugins-base-1.18.3/gst/playback/gstdecodebin3.c
new/gst-plugins-base-1.18.4/gst/playback/gstdecodebin3.c
--- old/gst-plugins-base-1.18.3/gst/playback/gstdecodebin3.c 2021-01-13
22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst/playback/gstdecodebin3.c 2021-03-15
18:48:00.000000000 +0100
@@ -1453,7 +1453,7 @@
}
SELECTION_LOCK (dbin);
- if (dbin->collection && collection != dbin->collection) {
+ if (dbin->collection) {
/* Replace collection message, we most likely aggregated it */
GstMessage *new_msg;
new_msg =
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst/playback/gsturidecodebin3.c
new/gst-plugins-base-1.18.4/gst/playback/gsturidecodebin3.c
--- old/gst-plugins-base-1.18.3/gst/playback/gsturidecodebin3.c 2021-01-13
22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst/playback/gsturidecodebin3.c 2021-03-15
18:48:00.000000000 +0100
@@ -1100,6 +1100,9 @@
GstURIDecodeBin3 *uridecodebin = (GstURIDecodeBin3 *) element;
switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ g_object_set (uridecodebin->decodebin, "caps", uridecodebin->caps, NULL);
+ break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = activate_next_play_item (uridecodebin);
if (ret == GST_STATE_CHANGE_FAILURE)
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-plugins-base-1.18.3/gst-libs/gst/audio/audio.c
new/gst-plugins-base-1.18.4/gst-libs/gst/audio/audio.c
--- old/gst-plugins-base-1.18.3/gst-libs/gst/audio/audio.c 2021-01-13
22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-libs/gst/audio/audio.c 2021-03-15
18:48:00.000000000 +0100
@@ -290,6 +290,10 @@
if (samples == orig_samples)
return buffer;
+ GST_DEBUG ("Truncating %" G_GSIZE_FORMAT " to %" G_GSIZE_FORMAT
+ " (trim start %" G_GSIZE_FORMAT ", end %" G_GSIZE_FORMAT ")",
+ orig_samples, samples, trim, orig_samples - trim - samples);
+
if (!meta || meta->info.layout == GST_AUDIO_LAYOUT_INTERLEAVED) {
/* interleaved */
ret = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, trim * bpf,
@@ -301,7 +305,7 @@
} else {
/* non-interleaved */
ret = gst_buffer_make_writable (buffer);
- meta = gst_buffer_get_audio_meta (buffer);
+ meta = gst_buffer_get_audio_meta (ret);
meta->samples = samples;
for (i = 0; i < meta->info.channels; i++) {
meta->offsets[i] += trim * bpf / meta->info.channels;
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst-libs/gst/audio/gstaudioaggregator.c
new/gst-plugins-base-1.18.4/gst-libs/gst/audio/gstaudioaggregator.c
--- old/gst-plugins-base-1.18.3/gst-libs/gst/audio/gstaudioaggregator.c
2021-01-13 22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-libs/gst/audio/gstaudioaggregator.c
2021-03-15 18:48:00.000000000 +0100
@@ -100,8 +100,6 @@
guint position, size; /* position in the input buffer and size of the
input buffer in number of samples */
- GstBuffer *input_buffer;
-
guint64 output_offset; /* Sample offset in output segment relative to
srcpad.segment.start where the current
position
of this input_buffer would be placed. */
@@ -139,7 +137,6 @@
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) object;
gst_buffer_replace (&pad->priv->buffer, NULL);
- gst_buffer_replace (&pad->priv->input_buffer, NULL);
G_OBJECT_CLASS (gst_audio_aggregator_pad_parent_class)->finalize (object);
}
@@ -162,7 +159,6 @@
gst_audio_info_init (&pad->info);
pad->priv->buffer = NULL;
- pad->priv->input_buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
@@ -182,7 +178,6 @@
pad->priv->output_offset = pad->priv->next_offset = -1;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
gst_buffer_replace (&pad->priv->buffer, NULL);
- gst_buffer_replace (&pad->priv->input_buffer, NULL);
GST_OBJECT_UNLOCK (aggpad);
return GST_FLOW_OK;
@@ -900,7 +895,8 @@
gboolean downstream_supports_rate = TRUE;
if (!gst_audio_info_from_caps (&info, caps)) {
- GST_WARNING_OBJECT (agg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps);
+ GST_WARNING_OBJECT (aaggpad, "Rejecting invalid caps: %" GST_PTR_FORMAT,
+ caps);
return FALSE;
}
@@ -933,6 +929,10 @@
if (!downstream_supports_rate || (first_configured_pad
&& info.rate != first_configured_pad->info.rate)) {
+ GST_WARNING_OBJECT (aaggpad,
+ "Sample rate %d can't be configured (downstream supported: %d,
configured rate: %d)",
+ info.rate, downstream_supports_rate,
+ first_configured_pad ? first_configured_pad->info.rate : 0);
gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ());
ret = FALSE;
} else {
@@ -1066,7 +1066,7 @@
if (aaggpad->priv->buffer) {
GstBuffer *new_converted_buffer =
gst_audio_aggregator_convert_buffer (aagg, GST_PAD (aaggpad),
- old_info, new_info, aaggpad->priv->input_buffer);
+ old_info, new_info, aaggpad->priv->buffer);
gst_buffer_replace (&aaggpad->priv->buffer, new_converted_buffer);
gst_buffer_unref (new_converted_buffer);
}
@@ -1809,7 +1809,6 @@
pad->priv->position = pad->priv->size;
gst_buffer_replace (&pad->priv->buffer, NULL);
- gst_buffer_replace (&pad->priv->input_buffer, NULL);
return FALSE;
}
@@ -1839,7 +1838,6 @@
if (pad->priv->position == pad->priv->size) {
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
- gst_buffer_replace (&pad->priv->input_buffer, NULL);
GST_LOG_OBJECT (pad, "Finished mixing buffer, waiting for next");
return FALSE;
}
@@ -2087,14 +2085,15 @@
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad);
+ GstBuffer *input_buffer;
if (!pad_eos)
is_eos = FALSE;
- pad->priv->input_buffer = gst_aggregator_pad_peek_buffer (aggpad);
+ input_buffer = gst_aggregator_pad_peek_buffer (aggpad);
GST_OBJECT_LOCK (pad);
- if (!pad->priv->input_buffer) {
+ if (!input_buffer) {
if (timeout) {
if (pad->priv->output_offset < next_offset) {
gint64 diff = next_offset - pad->priv->output_offset;
@@ -2119,24 +2118,21 @@
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
pad->priv->buffer =
gst_audio_aggregator_convert_buffer
- (aagg, GST_PAD (pad), &pad->info, &srcpad->info,
- pad->priv->input_buffer);
+ (aagg, GST_PAD (pad), &pad->info, &srcpad->info, input_buffer);
else
- pad->priv->buffer = gst_buffer_ref (pad->priv->input_buffer);
+ pad->priv->buffer = gst_buffer_ref (input_buffer);
if (!gst_audio_aggregator_fill_buffer (aagg, pad)) {
gst_buffer_replace (&pad->priv->buffer, NULL);
- gst_buffer_replace (&pad->priv->input_buffer, NULL);
- pad->priv->buffer = NULL;
+ gst_buffer_unref (input_buffer);
dropped = TRUE;
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
continue;
}
- } else {
- gst_buffer_unref (pad->priv->input_buffer);
}
+ gst_buffer_unref (input_buffer);
if (!pad->priv->buffer && !dropped && pad_eos) {
GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state");
@@ -2164,7 +2160,6 @@
GST_AUDIO_INFO_RATE (&srcpad->info))), pad->priv->buffer);
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
- gst_buffer_replace (&pad->priv->input_buffer, NULL);
dropped = TRUE;
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst-libs/gst/rtsp/gstrtspconnection.c
new/gst-plugins-base-1.18.4/gst-libs/gst/rtsp/gstrtspconnection.c
--- old/gst-plugins-base-1.18.3/gst-libs/gst/rtsp/gstrtspconnection.c
2021-01-13 22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-libs/gst/rtsp/gstrtspconnection.c
2021-03-15 18:48:00.000000000 +0100
@@ -1905,7 +1905,7 @@
memset (&serialized_messages[i], 0, sizeof (serialized_messages[i]));
serialized_messages[i].data = (guint8 *) base64_buffer;
- serialized_messages[i].data_size = (out_buffer - base64_buffer) + 1;
+ serialized_messages[i].data_size = (out_buffer - base64_buffer);
n_vectors++;
} else {
n_vectors++;
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst-libs/gst/tag/id3v2frames.c
new/gst-plugins-base-1.18.4/gst-libs/gst/tag/id3v2frames.c
--- old/gst-plugins-base-1.18.3/gst-libs/gst/tag/id3v2frames.c 2021-01-13
22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-libs/gst/tag/id3v2frames.c 2021-03-15
18:48:00.000000000 +0100
@@ -109,7 +109,7 @@
if (work->frame_flags & (ID3V2_FRAME_FORMAT_COMPRESSION |
ID3V2_FRAME_FORMAT_DATA_LENGTH_INDICATOR)) {
- if (work->hdr.frame_data_size <= 4)
+ if (frame_data_size <= 4)
return FALSE;
if (ID3V2_VER_MAJOR (work->hdr.version) == 3) {
work->parse_size = GST_READ_UINT32_BE (frame_data);
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst-libs/gst/video/gstvideodecoder.c
new/gst-plugins-base-1.18.4/gst-libs/gst/video/gstvideodecoder.c
--- old/gst-plugins-base-1.18.3/gst-libs/gst/video/gstvideodecoder.c
2021-01-13 22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-libs/gst/video/gstvideodecoder.c
2021-03-15 18:48:00.000000000 +0100
@@ -4258,8 +4258,19 @@
needs_reconfigure = gst_pad_check_reconfigure (decoder->srcpad);
if (G_UNLIKELY (decoder->priv->output_state_changed || needs_reconfigure)) {
if (!gst_video_decoder_negotiate_unlocked (decoder)) {
- GST_DEBUG_OBJECT (decoder, "Failed to negotiate, fallback allocation");
gst_pad_mark_reconfigure (decoder->srcpad);
+ if (GST_PAD_IS_FLUSHING (decoder->srcpad)) {
+ GST_DEBUG_OBJECT (decoder,
+ "Failed to negotiate a pool: pad is flushing");
+ goto flushing;
+ } else if (!decoder->priv->pool || decoder->priv->output_state_changed) {
+ GST_DEBUG_OBJECT (decoder,
+ "Failed to negotiate a pool and no previous pool to reuse");
+ goto error;
+ } else {
+ GST_DEBUG_OBJECT (decoder,
+ "Failed to negotiate a pool, falling back to the previous pool");
+ }
}
}
@@ -4272,6 +4283,10 @@
return flow_ret;
+flushing:
+ GST_VIDEO_DECODER_STREAM_UNLOCK (decoder);
+ return GST_FLOW_FLUSHING;
+
error:
GST_VIDEO_DECODER_STREAM_UNLOCK (decoder);
return GST_FLOW_ERROR;
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst-libs/gst/video/video-converter.c
new/gst-plugins-base-1.18.4/gst-libs/gst/video/video-converter.c
--- old/gst-plugins-base-1.18.3/gst-libs/gst/video/video-converter.c
2021-01-13 22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-libs/gst/video/video-converter.c
2021-03-15 18:48:00.000000000 +0100
@@ -2906,7 +2906,7 @@
gst_line_cache_get_lines (cache->prev, idx, out_line, start_line,
n_lines);
- if (convert->upsample) {
+ if (convert->upsample[idx]) {
GST_DEBUG ("doing upsample %d-%d %p", start_line, start_line + n_lines - 1,
lines[0]);
gst_video_chroma_resample (convert->upsample[idx], lines,
@@ -3107,7 +3107,7 @@
gst_line_cache_get_lines (cache->prev, idx, out_line, start_line,
n_lines);
- if (convert->downsample) {
+ if (convert->downsample[idx]) {
GST_DEBUG ("downsample line %d %d-%d %p", in_line, start_line,
start_line + n_lines - 1, lines[0]);
gst_video_chroma_resample (convert->downsample[idx], lines,
@@ -3130,7 +3130,7 @@
lines = gst_line_cache_get_lines (cache->prev, idx, out_line, in_line, 1);
destline = lines[0];
- if (convert->dither) {
+ if (convert->dither[idx]) {
GST_DEBUG ("Dither line %d %p", in_line, destline);
gst_video_dither_line (convert->dither[idx], destline, 0, out_line,
convert->out_width);
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst-libs/gst/video/video-info.c
new/gst-plugins-base-1.18.4/gst-libs/gst/video/video-info.c
--- old/gst-plugins-base-1.18.3/gst-libs/gst/video/video-info.c 2021-01-13
22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-libs/gst/video/video-info.c 2021-03-15
18:48:00.000000000 +0100
@@ -641,7 +641,7 @@
* Returns: a new #GstCaps containing the info of @info.
*/
GstCaps *
-gst_video_info_to_caps (GstVideoInfo * info)
+gst_video_info_to_caps (const GstVideoInfo * info)
{
GstCaps *caps;
const gchar *format;
@@ -686,14 +686,14 @@
if (GST_VIDEO_INFO_MULTIVIEW_MODE (info) != GST_VIDEO_MULTIVIEW_MODE_NONE) {
const gchar *caps_str = NULL;
+ GstVideoMultiviewFlags multiview_flags =
+ GST_VIDEO_INFO_MULTIVIEW_FLAGS (info);
/* If the half-aspect flag is set, applying it into the PAR of the
* resulting caps now seems safe, and helps with automatic behaviour
* in elements that aren't explicitly multiview aware */
- if (GST_VIDEO_INFO_MULTIVIEW_FLAGS (info) &
- GST_VIDEO_MULTIVIEW_FLAGS_HALF_ASPECT) {
- GST_VIDEO_INFO_MULTIVIEW_FLAGS (info) &=
- ~GST_VIDEO_MULTIVIEW_FLAGS_HALF_ASPECT;
+ if (multiview_flags & GST_VIDEO_MULTIVIEW_FLAGS_HALF_ASPECT) {
+ multiview_flags &= ~GST_VIDEO_MULTIVIEW_FLAGS_HALF_ASPECT;
switch (GST_VIDEO_INFO_MULTIVIEW_MODE (info)) {
case GST_VIDEO_MULTIVIEW_MODE_SIDE_BY_SIDE:
case GST_VIDEO_MULTIVIEW_MODE_SIDE_BY_SIDE_QUINCUNX:
@@ -716,7 +716,7 @@
if (caps_str != NULL) {
gst_caps_set_simple (caps, "multiview-mode", G_TYPE_STRING,
caps_str, "multiview-flags", GST_TYPE_VIDEO_MULTIVIEW_FLAGSET,
- GST_VIDEO_INFO_MULTIVIEW_FLAGS (info), GST_FLAG_SET_MASK_EXACT,
NULL);
+ multiview_flags, GST_FLAG_SET_MASK_EXACT, NULL);
}
}
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/gst-libs/gst/video/video-info.h
new/gst-plugins-base-1.18.4/gst-libs/gst/video/video-info.h
--- old/gst-plugins-base-1.18.3/gst-libs/gst/video/video-info.h 2021-01-13
22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-libs/gst/video/video-info.h 2021-03-15
18:48:00.000000000 +0100
@@ -452,7 +452,7 @@
gboolean gst_video_info_from_caps (GstVideoInfo *info, const GstCaps
* caps);
GST_VIDEO_API
-GstCaps * gst_video_info_to_caps (GstVideoInfo *info);
+GstCaps * gst_video_info_to_caps (const GstVideoInfo *info);
GST_VIDEO_API
gboolean gst_video_info_convert (GstVideoInfo *info,
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-plugins-base-1.18.3/gst-plugins-base.doap
new/gst-plugins-base-1.18.4/gst-plugins-base.doap
--- old/gst-plugins-base-1.18.3/gst-plugins-base.doap 2021-01-13
22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/gst-plugins-base.doap 2021-03-15
18:48:00.000000000 +0100
@@ -36,6 +36,16 @@
<release>
<Version>
+ <revision>1.18.4</revision>
+ <branch>1.18</branch>
+ <name></name>
+ <created>2021-03-15</created>
+ <file-release
rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.18.4.tar.xz";
/>
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.18.3</revision>
<branch>1.18</branch>
<name></name>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-plugins-base-1.18.3/meson.build
new/gst-plugins-base-1.18.4/meson.build
--- old/gst-plugins-base-1.18.3/meson.build 2021-01-13 22:07:13.000000000
+0100
+++ new/gst-plugins-base-1.18.4/meson.build 2021-03-15 18:48:00.000000000
+0100
@@ -1,5 +1,5 @@
project('gst-plugins-base', 'c',
- version : '1.18.3',
+ version : '1.18.4',
meson_version : '>= 0.48',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-plugins-base-1.18.3/tests/examples/overlay/meson.build
new/gst-plugins-base-1.18.4/tests/examples/overlay/meson.build
--- old/gst-plugins-base-1.18.3/tests/examples/overlay/meson.build
2021-01-13 22:07:13.000000000 +0100
+++ new/gst-plugins-base-1.18.4/tests/examples/overlay/meson.build
2021-03-15 18:48:00.000000000 +0100
@@ -9,7 +9,7 @@
if have_cxx # check for C++ support
qt5_mod = import('qt5')
- qt5widgets_dep = dependency('qt5', modules : ['Gui', 'Widgets'],
+ qt5widgets_dep = dependency('qt5', modules : ['Core', 'Gui', 'Widgets'],
required: get_option('examples'))
# FIXME Add a way to get that information out of the qt5 module
