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Diffstat (limited to 'ChangeLog')
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@@ -1,3 +1,439 @@ +=== release 1.18.5 === + +2021-09-08 20:02:48 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ChangeLog: + * NEWS: + * RELEASE: + * gst-plugins-good.doap: + * meson.build: + Release 1.18.5 + +2021-08-03 19:12:11 +0900 Seungha Yang <seungha@centricular.com> + + * ext/jpeg/gstjpegdec.c: + jpegdec: Fix crash when interlaced field height is not DCT block size aligned + In case of interlaced JPEG file, we are doubling stride. + The scratch scan line should take account of it as well. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1042> + +2021-04-13 09:23:12 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Remove some dead code + stop is not used after this point, nor do we create a new segment + here since 84725d62b57bc74ce34abde755f35bf8f948f94d + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1076> + +2021-04-10 02:53:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Do not overwrite the known duration after a seek + This breaks the duration query and also the seeking query. + Broke in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1076> + +2021-04-10 04:40:46 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Just assign the segment instead of memcpy + Assignments copy by value, we don't need to memcpy... + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1076> + +2021-04-10 03:09:44 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: De-dup seek event seqnums to avoid multiple seeks + Seek events are sent upstream on each sink, so if we receive multiple + seeks with the same seqnum, we must only perform one seek, not N seeks + where N = the number of sinks in the pipeline connected to rtspsrc. + This is the same thing done by demuxers like qtdemux or matrsokademux. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1075> + +2021-07-08 02:22:20 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/multifile/gstsplitmuxsink.c: + splitmuxsink: always use factory property when set + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1074> + +2021-09-02 08:38:54 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/avi/gstavidemux.c: + avidemux: Also detect 0x000001 as H264 byte-stream start code in codec_data + This works around some AVI files storing byte-stream data in the + codec_data. The previous workaround was only checking for + 0x00000001 (4 bytes) instead of 0x000001 (3 bytes). + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1073> + +2021-08-27 14:32:45 +0200 Edward Hervey <edward@centricular.com> + + * gst/isomp4/qtdemux.c: + qtdemux: Force stream-start push when re-using EOS'd streams + When re-using streams, we *do* need to push a `stream-start` event downstream if + we previously were EOS'd. Failure to do that would never remove the EOS status + on all downstream elements and cause weird issues. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1068> + +2021-08-24 13:28:22 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/isomp4/qtdemux.c: + qtdemux: add depth for ProRes 4:4:4:4 variants if available + Might be 24bpp in case an alpha channel is coded but + the image is always opaque. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1062> + +2021-08-22 23:16:26 +0000 Ruslan Khamidullin <ruslank@borisfx.com> + + * gst/isomp4/gstqtmux.c: + qtmux: for Apple ProRes, allow overriding pixel bit depth for 4:4:4:4 variants + e.g. when exporting an opaque image, yet with alpha channel. + Apple ProRes certification requires that, when a ProRes-writing + application *knows* that the entire frame is opaque, the application + writes only RGB without alpha even when the clip is RGBA. For that, + this tiny change allows the app to override pixel depth when writing ProRes. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1062> + +2021-06-16 16:30:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: update codec_data in caps regardless of format + Updating of codec_data in the caps is important to propagate changes + in sps/pps/vps via NALs. Without this, downstream does not renegotiate + when upstream changes resolution. + The comment referring to rtph264pay is from 2015 and is out of date. + rtph264pay stopped doing that in 2017 with commit + dabeed52a995d27e16eba9e4617e61eb0bcd44c4 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1025> + +2021-08-15 12:26:38 +0300 Sebastian Dröge <sebastian@centricular.com> + + * ext/soup/gstsouphttpsrc.c: + souphttpsrc: Always use the content decoder but set `Accept-Encoding: identity` if no compression should be used + Some servers respond with gzip-encoded responses regardless of whether + the request allowed it to be used in the response. By always having the + content decoder enabled, these invalid responses can be decoded + correctly while for well-behaving servers the `compress` property + selects between allowing compressed responses or not. + Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/833 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1054> + +2021-07-01 13:18:45 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpssrcdemux.c: + rtpssrcdemux: Remove pads and reset the element also in READY->NULL + Mostly for completeness. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1027> + +2021-07-01 13:18:09 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpptdemux.c: + rtpptdemux: Remove pads also in PAUSED->READY + They're based on per-stream information and that should be reset + whenever going to READY state. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1027> + +2021-06-26 20:00:03 +0900 Seungha Yang <seungha@centricular.com> + + * gst/udp/gstmultiudpsink.c: + multiudpsink: Fix broken SO_SNDBUF get/set on Windows + SO_SNDBUF has been undefined on Windows because of missing WinSock2.h + include. And don't use native socket functions (e.g., setsockopt()) + if code is expected to be built on Windows. We don't link ws2_32.lib + for this plugin. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1026> + +2020-11-03 15:58:30 +0200 Sebastian Dröge <sebastian@centricular.com> + + * ext/qt/gstqsgtexture.cc: + * ext/qt/gstqsgtexture.h: + * ext/qt/qtitem.cc: + qmlglsink: Keep old buffers around a bit longer if they were bound by QML + We don't know exactly when QML will stop using them but it should be + safe to unref them after at least 2 more buffers were bound. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1009> + +2021-06-03 20:33:45 +1000 Matthew Waters <matthew@centricular.com> + + * ext/qt/qtitem.cc: + qtitem: don't potentially leak a large number of buffers + The only other place where these queued buffers are removed, is in + setCaps() but that is not called at all on shutdown so this list of + buffers could not be removed. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1005> + +2021-05-29 12:54:22 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpjpegpay.c: + rtpjpegpay: fix image corruption when compiled with MSVC on Windows + On Windows with MSVC, jpeg_header_size would end up 2 bytes larger + than it should be. This then leads to the first 2 bytes of the + actual jpeg image data to be dropped, because we think those + belong to the header, which results in an undecodable image when + reconstructed in the depayloader. + What happens is that when the compiler evaluates + jpeg_header_size = mem.offset + read_u16_and_inc_offset_by_2(&mem); + it actually uses the mem.offset value after it has been increased + by the function call on the right hand size of the equation. + From section 6.5 of the C99 spec: + 3. The grouping of operators and operands is indicated by the syntax [74]. + Except as specified later (for the function-call (), &&, ||, ?:, and + comma operators), the order of evaluation of subexpressions and the + order in which side effects take place are both unspecified. + Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/889 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1003> + +2021-05-26 00:23:56 +0900 Seungha Yang <seungha@centricular.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Drop "field-order" field while transforming caps + Like other basetransform subclasses are doing, drop field + which can be converted by deinterlace. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1001> + +2021-05-25 20:10:34 +0900 Seungha Yang <seungha@centricular.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Drop field-order field if outputting progressive + Progressive with field-order doesn't make sense + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1001> + +2021-05-23 15:14:11 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/wavparse/gstwavparse.c: + wavparse: use g_strndup() for copying text data + So we don't rely on NUL terminators inside the data. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1000> + +2021-05-23 13:29:07 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/wavparse/gstwavparse.c: + wavparse: clean up adtl/note/labl chunk parsing + We were passing the size of the adtl chunk to the note/labl + sub-chunk parsing function, which means we may memdup lots of + data after the chunk string's NUL terminator that doesn't + really belong to it. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1000> + +2021-05-23 13:24:21 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/wavparse/gstwavparse.c: + wavparse: guard against overflow when comparing chunk sizes + Could be rewritten as lsize > (size - 8) a well, but the + extra check seems clearer. Doesn't look like it was problematic, + lsize wasn't actually used when parsing the sub-chunks. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1000> + +2021-05-07 11:16:47 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com> + + * gst/udp/gstudpsrc.c: + udpsrc: Plug leaks of saddr in error cases + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/986> + +2021-05-07 11:16:21 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com> + + * gst/udp/gstudpsrc.c: + udpsrc: Whitespace + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/986> + +2021-05-07 00:43:44 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Plug a method subobject leak + Changing the method would leak the previous method. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/985> + +2021-04-30 08:12:47 +1000 Jan Schmidt <jan@centricular.com> + + * gst/isomp4/atoms.c: + * gst/isomp4/atoms.h: + qtmux: Make sure to write 64-bit STCO table when needed. + qtmux attempts to choose between writing a 32-bit stco chunk offset table + when it can, but switch to a 64-bit co64 table when file offsets go over + 4GB. + This patch fixes a problem where the atom handling code was checking + mdat-relative offsets instead of the final file offset (computed by + adding the mdat position plus the mdat-relative offset) - leading to + problems where files with a size between 4GB and 4GB+offset-of-the-mdat + would write incorrect STCO tables with some samples having truncated + 32-bit offsets. + Smaller files write STCO correctly, larger files would switch to + co64 and also output correctly. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/972> + +2021-04-22 15:01:32 +0800 Hou Qi <qi.hou@nxp.com> + + * sys/v4l2/gstv4l2object.c: + v4l2object: Add interlace-mode back to caps for camera + skip_try_fmt_probes is set to TRUE for v4l2src to skip interlace-mode and + colorimetry when probe caps. gst_v4l2_object_set_format_full() will add + colorimetry back to caps when iterating over the negotiated caps. There is + one case that v4l2src is first in preview state then starts recording. + v4l2src caps will change with an additional interlace-mode structure after + renegotiation, then v4l2src needs to reset. But this camera driver can't + orphan buffer pool, it causes require buffer failed as streaming is still + in active state. + To fix this, also need to add interlace-mode back to caps for camera to + avoid reset. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/971> + +2021-04-22 08:57:23 +0200 Edward Hervey <edward@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Avoid generation of invalid timestamps + When updating timestamps and timer timeouts with a new offset, make sure that + the resulting value is valid (and not a negative (signed) value which ends up in + a massive (unsigned) value). + Fixes #571 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/963> + +2021-04-19 01:29:33 -0400 Doug Nazar <nazard@nazar.ca> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Fix race saving seek event seqnum. + We need to save the seek seqnum before the flush stop event + since that will start the basesrc task which may send the segment + event before we're ready. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/957> + +2021-04-13 11:30:51 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Check srcresult before waiting on the condition variable too + It might've been set to FLUSHING between the last check and the waiting, + and in that case we'd be waiting here forever now. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/948> + +2021-04-10 01:55:28 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Using multicast UDP has no relation to seekability + The transport has no relation to whether a media can be seeked. The + range response having a duration is the correct thing to check for. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/947> + +2021-04-10 01:54:48 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Add more logging for range parsing and seekable + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/947> + +2021-04-06 17:02:34 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtpmanager/rtpjitterbuffer.c: + rtpjitterbuffer: More logging when calculating rfc7273 timestamps + This code can be fragile, since it is very exacting in the timestamps + that it will accept. Add more logging so it's easier to debug issues + and figure out whether it's a bug in the calculation or something + wrong in the incoming buffers. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/946> + +2021-03-18 19:52:53 +1100 Matthew Waters <matthew@centricular.com> + + * ext/jack/gstjack.c: + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + * ext/pulse/pulsesink.h: + * ext/qt/gstqsgtexture.cc: + * ext/qt/gstqtglutility.cc: + * ext/qt/qtglrenderer.cc: + * ext/qt/qtitem.cc: + * ext/qt/qtwindow.cc: + * ext/vpx/gstvpxdec.c: + * ext/vpx/gstvpxenc.c: + * gst/audioparsers/gstac3parse.h: + * sys/rpicamsrc/gstrpicamsrc.c: + * sys/ximage/ximageutil.c: + gst: don't use volatile to mean atomic + volatile is not sufficient to provide atomic guarantees and real atomics + should be used instead. GCC 11 has started warning about using volatile + with atomic operations. + https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719 + Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/941> + +2021-03-26 16:26:22 +0800 Hou Qi <qi.hou@nxp.com> + + * sys/v4l2/gstv4l2object.c: + v4l2object: Use default colorimetry if that in caps is unknown + Some streams have unknown colorimetry in caps, but v4l2object sets + default values for each primaries. It will cause check colorimetry + fail when do gst_v4l2_video_colorimetry_matches(). + To fix this, need to keep the unknown colorimetry in caps same as + the default value set by v4l2object. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/930> + +2021-03-23 16:59:28 +0800 Hou Qi <qi.hou@nxp.com> + + * sys/v4l2/gstv4l2object.c: + v4l2object: Avoid colorimetry mismatch for streams with invalid colorimetry + video-info sets gst colorimetry to default value when colorimetry in caps + is unparsable or invalid. Then v4l2object uses this gst colorimetry to do + mapping with v4l2 colorimetry. This may cause colorimetry mismatch when + check mapped gst colorimetry with that read from caps directly. + To fix this, need to correct gst colorimetry as that parsed from video-info + when check gst_v4l2_video_colorimetry_matches(). + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/922> + +2021-03-19 10:52:26 +0800 Hou Qi <qi.hou@nxp.com> + + * sys/v4l2/gstv4l2object.c: + v4l2object: Add support for hdr10 stream playback + Colorimetry of hdr10 video is bt2100-pq with transfer as + GST_VIDEO_TRANSFER_SMPTE2084. So map GST_VIDEO_TRANSFER_SMPTE2084 + to V4L2_XFER_FUNC_SMPTE2084 to support hdr10 stream playback. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/921> + +2021-03-18 17:42:02 +0000 Alba Mendez <me@alba.sh> + + * docs/gst_plugins_cache.json: + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Fix more signals + Behaviour change in GLib causes select-stream signal to discard + the value returned by handlers. See !909 for more info. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/920> + +2021-03-17 15:54:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * docs/gst_plugins_cache.json: + * gst/rtsp/gstrtspsrc.c: + Update docs cache and fix before-send signal doc syntax + The docs for before-send were missing because of this + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/918> + +2021-03-17 13:18:34 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: Fix accumulation of before-send signal return values + Since glib 2.62, the accumulated return values in RUN_CLEANUP override the + accumulated return values in RUN_FIRST. Since: + 1. We have a default handler that always returns TRUE, and + 2. User handlers are only run in RUN_FIRST, and + 3. Our accumulator just takes the latest return value + We were discarding the return value from the user handler and always + sending messages even if the user handler said not to. See + https://gitlab.gnome.org/GNOME/glib/-/issues/2352 for more details. + This signal does not need RUN_CLEANUP or RUN_FIRST, so just change it + to RUN_LAST so that it's emitted exactly once and accumulated once. + With this fix, this signal can now be used to intercept PAUSE when + going to GST_STATE_NULL so that the server does a TEARDOWN (if + necessary) and not a PAUSE, which will confuse other RTSP clients when + playing shared media. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/918> + +2021-03-16 19:25:36 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + rtpjitterbuffer: Fix parsing of the mediaclk:direct= field + Due to an off-by-one when parsing the string, the most significant digit + or the clock offset was skipped when parsing the offset. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/910> + +2021-03-15 19:49:05 +0000 Tim-Philipp Müller <tim@centricular.com> + + * docs/gst_plugins_cache.json: + * meson.build: + Back to development + === release 1.18.4 === 2021-03-15 17:48:28 +0000 Tim-Philipp Müller <tim@centricular.com> @@ -5,6 +441,7 @@ * ChangeLog: * NEWS: * RELEASE: + * docs/gst_plugins_cache.json: * gst-plugins-good.doap: * meson.build: Release 1.18.4 |