GStreamer issueshttps://gitlab.freedesktop.org/groups/gstreamer/-/issues2021-09-24T13:30:10Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/74avidemux: zeta_Giga_1.avi doesn't preroll, demuxer should send GAP events2021-09-24T13:30:10ZBugzilla Migration Useravidemux: zeta_Giga_1.avi doesn't preroll, demuxer should send GAP events## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#685708)](https://bugzilla.gnome.org/show_bug.cgi?id=685708)**
## Description
$ gst-launch-1.0 playbin uri=file:///home/data/test-files/movies/zeta_Giga_1.avi
Sett...## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#685708)](https://bugzilla.gnome.org/show_bug.cgi?id=685708)**
## Description
$ gst-launch-1.0 playbin uri=file:///home/data/test-files/movies/zeta_Giga_1.avi
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Redistribute latency...
[and wait]https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/72pulsesrc: add support for renegotiation2021-09-24T13:30:10ZBugzilla Migration Userpulsesrc: add support for renegotiation## Submitted by Sjoerd Simons `@sjoerd`
**[Link to original bug (#683902)](https://bugzilla.gnome.org/show_bug.cgi?id=683902)**
## Description
Just after renegotiation started a lot of warnings are logged (which are probably harmles...## Submitted by Sjoerd Simons `@sjoerd`
**[Link to original bug (#683902)](https://bugzilla.gnome.org/show_bug.cgi?id=683902)**
## Description
Just after renegotiation started a lot of warnings are logged (which are probably harmless), unfortunately afterwards pa_stream_connect_record apparently fails.
0:00:06.580594086 1317 0x2ab7770 WARN audiosrc gstaudiosrc.c:245:audioringbuffer_thread_func:`<pulsesrc0>` error reading data -1 (reason: Success), skipping segment
0:00:06.580608406 1317 0x2ab7770 WARN audiosrc gstaudiosrc.c:245:audioringbuffer_thread_func:`<pulsesrc0>` error reading data -1 (reason: Success), skipping segment
0:00:06.580622683 1317 0x2ab7770 WARN audiosrc gstaudiosrc.c:245:audioringbuffer_thread_func:`<pulsesrc0>` error reading data -1 (reason: Success), skipping segment
0:00:06.580664593 1317 0x2ab7770 WARN audiosrc gstaudiosrc.c:245:audioringbuffer_thread_func:`<pulsesrc0>` error reading data -1 (reason: Success), skipping segment
0:00:06.580678688 1317 0x2ab7770 ERROR ringbuffer gstaudioringbuffer.c:1970:gst_audio_ring_buffer_set_timestamp:`<audiosrcringbuffer0>` Could not store timestamp, no timestamps buffer
0:00:06.580798741 1317 0x2ac91e0 INFO pulse pulsesrc.c:1407:gst_pulsesrc_prepare:`<pulsesrc0>` maxlength: -1
0:00:06.580818444 1317 0x2ac91e0 INFO pulse pulsesrc.c:1408:gst_pulsesrc_prepare:`<pulsesrc0>` tlength: -1
0:00:06.580832238 1317 0x2ac91e0 INFO pulse pulsesrc.c:1409:gst_pulsesrc_prepare:`<pulsesrc0>` prebuf: 0
0:00:06.580845753 1317 0x2ac91e0 INFO pulse pulsesrc.c:1410:gst_pulsesrc_prepare:`<pulsesrc0>` minreq: -1
0:00:06.580858797 1317 0x2ac91e0 INFO pulse pulsesrc.c:1411:gst_pulsesrc_prepare:`<pulsesrc0>` fragsize: 320
0:00:06.585881211 1317 0x2ac91e0 WARN pulse pulsesrc.c:1488:gst_pulsesrc_prepare:`<pulsesrc0>` error: Failed to connect stream: Invalid argument
empathy-Message: Element error: Failed to connect stream: Invalid argument -- pulsesrc.c(1488): gst_pulsesrc_prepare (): /GstPipeline:pipeline0/EmpathyGstAudioSrc:empathygstaudiosrc0/GstPulseSrc:pulsesrc0https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/71qtdemux: avoid reading parts of file not used (skip reading video data if onl...2021-09-24T13:30:09ZBugzilla Migration Userqtdemux: avoid reading parts of file not used (skip reading video data if only audio is active)## Submitted by David Jordan
**[Link to original bug (#682172)](https://bugzilla.gnome.org/show_bug.cgi?id=682172)**
## Description
Currently Gstreamer will read all of a file from disk even if certain streams are not needed and cou...## Submitted by David Jordan
**[Link to original bug (#682172)](https://bugzilla.gnome.org/show_bug.cgi?id=682172)**
## Description
Currently Gstreamer will read all of a file from disk even if certain streams are not needed and could be skipped. This can be an issue when, for example, one wants to process only the audio stream from a file containing both audio and high-bitrate video, since performance will be bottlenecked by hard disk throughput.
If elements like qtdemux could determine which streams aren't being used, avoid reading data for unused streams, and instead only read the index and the streams which are being used; this would hugely increase performance in those use cases.
A good test uses footage from a Canon DSLR, which is commonly used for professional video. One 12 minute file is 4GB in size, of which less than 150MB is audio.
The following pipeline takes approximately as long as simply reading the entire file from disk. (about 40 seconds *uncached* on a Sandybridge laptop using the file below)
time gst-launch-1.0 filesrc location="$INPUT_FILE" ! qtdemux ! audioconvert ! filesink location="OUTPUT_FILE"
When I profiled it, most time was spent in functions for reading from disk (kernel -> system_call_fastpath -> sys_read), with CPU usage remaining low.
After extracting the audio only and placing it in another qt container, the time required was dramatically reduced.
Example DLSR file can be found here (about 4GB).
http://uds-o.novacut.com/GWQ5M6GPPNVEY3SAGUORKMLUAMQGD6O4.mov
We should figure out a way to signal this behavior properly for elements like qtdemux.https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/70wavparse: add support for reverse playback2021-09-24T13:30:09ZBugzilla Migration Userwavparse: add support for reverse playback## Submitted by Tim Müller `@tpm`
**[Link to original bug (#681612)](https://bugzilla.gnome.org/show_bug.cgi?id=681612)**
## Description
Don't know if it's much different than in 0.10..## Submitted by Tim Müller `@tpm`
**[Link to original bug (#681612)](https://bugzilla.gnome.org/show_bug.cgi?id=681612)**
## Description
Don't know if it's much different than in 0.10..https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/69matroskademux negative rate in gap2021-09-24T13:30:08ZBugzilla Migration Usermatroskademux negative rate in gap## Submitted by Farkas Levente `@lfarkas`
**[Link to original bug (#680444)](https://bugzilla.gnome.org/show_bug.cgi?id=680444)**
## Description
in the latest update matroskademux is extended to be able to use new-gap-time is bigger...## Submitted by Farkas Levente `@lfarkas`
**[Link to original bug (#680444)](https://bugzilla.gnome.org/show_bug.cgi?id=680444)**
## Description
in the latest update matroskademux is extended to be able to use new-gap-time is bigger then max-gap-time but why it's not working for negative rate? in the code:
----------------------------------
if (demux->max_gap_time &&
GST_CLOCK_TIME_IS_VALID (demux->last_stop_end) &&
demux->common.segment.rate > 0.0) {
----------------------------------
why is the last condition?
Version: 0.10.xhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/68matroskademux: Support reverse playback in files without index2021-09-24T13:30:08ZBugzilla Migration Usermatroskademux: Support reverse playback in files without index## Submitted by Farkas Levente `@lfarkas`
**[Link to original bug (#680443)](https://bugzilla.gnome.org/show_bug.cgi?id=680443)**
## Description
in the latest update matroskademux is extended to be able to seek in a not yet closed (...## Submitted by Farkas Levente `@lfarkas`
**[Link to original bug (#680443)](https://bugzilla.gnome.org/show_bug.cgi?id=680443)**
## Description
in the latest update matroskademux is extended to be able to seek in a not yet closed (ie. indexed) matroska file. but it's still not possible to seek after a negative rate (ie backward) although it'd be very useful...https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/67flvdemux: seek in push mode makes it start from beginning2021-09-24T13:30:07ZBugzilla Migration Userflvdemux: seek in push mode makes it start from beginning## Submitted by Kyrylo V. Polezhaiev
**[Link to original bug (#680087)](https://bugzilla.gnome.org/show_bug.cgi?id=680087)**
## Description
I am using Fedora 17, GStreamer 0.10.36 and Good plugins 0.10.31-6.
When trying to seek FL...## Submitted by Kyrylo V. Polezhaiev
**[Link to original bug (#680087)](https://bugzilla.gnome.org/show_bug.cgi?id=680087)**
## Description
I am using Fedora 17, GStreamer 0.10.36 and Good plugins 0.10.31-6.
When trying to seek FLV file via HTTP, get following error from flvdemux:
"failed to find an index, seeking back to beginning".
To reproduce, please open any GStreamer player with GST_DEBUG=flvdemux:3 and open FLV file located in HTTP server.
Version: 1.xhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/66isomp4: Export transformation matrix2021-09-24T13:30:06ZBugzilla Migration Userisomp4: Export transformation matrix## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#679522)](https://bugzilla.gnome.org/show_bug.cgi?id=679522)**
## Description
http://people.gnome.org/~hadess/IMG_0205.MOV
QuickTime can export a tranformation m...## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#679522)](https://bugzilla.gnome.org/show_bug.cgi?id=679522)**
## Description
http://people.gnome.org/~hadess/IMG_0205.MOV
QuickTime can export a tranformation matrix in the mvhd "Movie Header" atom.
See:
http://developer.apple.com/library/mac/#documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP40000939-CH204-BBCBEAIF
This could be exported, and naively interpreted by movie players to rotate the video.
### Depends on
* [Bug 738914](https://bugzilla.gnome.org/show_bug.cgi?id=738914)https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/65matroska: Reverse playback not working with webm file2021-09-24T13:30:05ZBugzilla Migration Usermatroska: Reverse playback not working with webm file## Submitted by Xavi Artigas `@xartigas`
**[Link to original bug (#679250)](https://bugzilla.gnome.org/show_bug.cgi?id=679250)**
## Description
Created attachment 217815
Basic tutorial 13 from the GStreamer SDK tutorials
The ...## Submitted by Xavi Artigas `@xartigas`
**[Link to original bug (#679250)](https://bugzilla.gnome.org/show_bug.cgi?id=679250)**
## Description
Created attachment 217815
Basic tutorial 13 from the GStreamer SDK tutorials
The attached code exercises multiple trick modes.
With a webm file, frame stepping and fast forward seem to work correctly, but, going backwards (press 'D') is completely ignored for HTTP files and hangs with this message for local files:
CRITICAL **: gst_matroska_demux_seek_to_previous_keyframe: assertion `demux->seek_index' failed
gst-playback-test is reported as not working nicely either.
**Attachment 217815**, "Basic tutorial 13 from the GStreamer SDK tutorials":
[basic-tutorial-13.c](/uploads/635483f4013f212c7ba9d2c89b86d6eb/basic-tutorial-13.c)https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/64apedemux: APE tags should be preffered over ID3v12021-09-24T13:30:05ZBugzilla Migration Userapedemux: APE tags should be preffered over ID3v1## Submitted by k.p..@..il.com
**[Link to original bug (#677919)](https://bugzilla.gnome.org/show_bug.cgi?id=677919)**
## Description
If for example an MP3 file includes both APE tags and ID3v1 tags at the end of the file, typefind ...## Submitted by k.p..@..il.com
**[Link to original bug (#677919)](https://bugzilla.gnome.org/show_bug.cgi?id=677919)**
## Description
If for example an MP3 file includes both APE tags and ID3v1 tags at the end of the file, typefind first finds the ID3v1 tag at the end and after id3demux having parsed that the APE tag that comes before the ID3v1 tag is found and read.
Now with ID3v1 being a very limited format and not having a clear definition of the character set used, reading non-ASCII titles (especially Shift-JIS or GB18030) from ID3v1 mostly results in garbled characters. As it is parsed before the APE tag, media players etc. will usually display the not very useful ID3v1 title. I would therefore propose that in case of ID3v1 and APE tag both being present, the APE tag should be preffered.https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/63[2.0] qtdemux/qtmux: merge image/x-j2c and image/x-jpc and drop non-standard ...2021-09-24T13:30:04ZBugzilla Migration User[2.0] qtdemux/qtmux: merge image/x-j2c and image/x-jpc and drop non-standard boxing in the jpeg2000 elements## Submitted by Tim Müller `@tpm`
**[Link to original bug (#677754)](https://bugzilla.gnome.org/show_bug.cgi?id=677754)**
## Description
From commit message:
From 426b2db2cba849ab4f64a6ba91047380ff338837 Mon Sep 17 00:00:00 200...## Submitted by Tim Müller `@tpm`
**[Link to original bug (#677754)](https://bugzilla.gnome.org/show_bug.cgi?id=677754)**
## Description
From commit message:
From 426b2db2cba849ab4f64a6ba91047380ff338837 Mon Sep 17 00:00:00 2001
From: Sebastian Dröge <slomo@circular-chaos.org>
Date: Mon, 01 Dec 2008 15:48:13 +0000
Subject: ext/jp2k/: Add image/x-jpc caps name for real, raw JPEG2000 codestream data.
Original commit message from CVS:
* ext/jp2k/gstjasperdec.c: (gst_jasper_dec_sink_setcaps):
* ext/jp2k/gstjasperenc.c: (gst_jasper_enc_reset),
(gst_jasper_enc_set_src_caps), (gst_jasper_enc_init_encoder),
(gst_jasper_enc_sink_setcaps), (gst_jasper_enc_get_data):
* ext/jp2k/gstjasperenc.h:
Add image/x-jpc caps name for real, raw JPEG2000 codestream data.
In 0.11 we should merge image/x-j2c and image/x-jpc and simply drop
the non-standard boxing in the jasper elements and handle it in
qtmux/qtdemux.
image/x-jpc will be used by mxfdemux later.
Also add support for JP2 output in jp2kenc.2.xhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/62matroskademux, avidemux: implement accurate keyframe snapping seeks2021-09-24T13:30:04ZBugzilla Migration Usermatroskademux, avidemux: implement accurate keyframe snapping seeks## Submitted by Vincent Penquerc'h `@vincent`
**[Link to original bug (#676630)](https://bugzilla.gnome.org/show_bug.cgi?id=676630)**
## Description
When the ACURATE flag is set for snapping keyframe seeks, we seek to the
previous...## Submitted by Vincent Penquerc'h `@vincent`
**[Link to original bug (#676630)](https://bugzilla.gnome.org/show_bug.cgi?id=676630)**
## Description
When the ACURATE flag is set for snapping keyframe seeks, we seek to the
previous known keyframe and scan forward till the required keyframe is
found. For backward snapping, this will require an extra seek.
Note that this will be unneeded if all keyframes are indexed. I have no
idea whether Matroska requires this or not.https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/61rgvolume does'nt apply gain using fallback-gain (without Track and Album tags)2021-09-24T13:30:03ZBugzilla Migration Userrgvolume does'nt apply gain using fallback-gain (without Track and Album tags)## Submitted by anthony
**[Link to original bug (#673970)](https://bugzilla.gnome.org/show_bug.cgi?id=673970)**
## Description
Created attachment 211909
rgvolume does'nt apply gain using fallback-gain (without Track and Album tags...## Submitted by anthony
**[Link to original bug (#673970)](https://bugzilla.gnome.org/show_bug.cgi?id=673970)**
## Description
Created attachment 211909
rgvolume does'nt apply gain using fallback-gain (without Track and Album tags)
When trying to apply the advised gain (that is the result of rganalysis) by using the fallback-gain property (no ReplayGain tags), the result gain is always near 0 and no normalization is being applied.
The reason is that in this case, the plugin assumes a peak value of 1.0, which makes the clipping prevention mecanism prevent any normalization (because gain + linear_to_db(peak) is always > headroom, in fact in our case here, linear_to_db(1.0) is 0 every time), so as long as the gain is positive, the result gain will always be reduced.
To sum up, the anti-clipping mecanism is counter-productive when the media has no ReplayGain tags: assuming a peak of 1.0 means that the track has full volume, therefore normalization cannot be safely applied (according to the anti-clipping feature).
I propose a patch to change peak value from 1.0 to 0.000001 (linear_to_db(0.000001) = -120, which equals to the maximum appliable gain, i.e. pre-amp + fallback-gain), so that in the case of using fallback-gain, the gain will always be applied.
**Patch 211909**, "rgvolume does'nt apply gain using fallback-gain (without Track and Album tags)":
[0001-BugFix-Change-peak-value-to-normalize-audio-file-wit.patch](/uploads/27c633c4fe487b12470bf35d237ec3ad/0001-BugFix-Change-peak-value-to-normalize-audio-file-wit.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/60udpsink: implement timestamp smoothing / sender throttling (for rtprawpay and...2021-09-24T13:30:03ZBugzilla Migration Userudpsink: implement timestamp smoothing / sender throttling (for rtprawpay and gigabit ethernet)## Submitted by kwisp
**[Link to original bug (#673794)](https://bugzilla.gnome.org/show_bug.cgi?id=673794)**
## Description
Created attachment 211668
good
It is some problem with rtpvrawdepay and gigabit ethernet.
All righ...## Submitted by kwisp
**[Link to original bug (#673794)](https://bugzilla.gnome.org/show_bug.cgi?id=673794)**
## Description
Created attachment 211668
good
It is some problem with rtpvrawdepay and gigabit ethernet.
All right, if we use Fast Ethernet mode:
sender:
kwisp@klochkov ~ $ LANG=en.en GST_PLUGIN_PATH=/usr/local/lib/gstreamer-0.10/ GST_PLUGIN_SYSTEM_PATH=/usr/lib/gstreamer-0.10/ gst-launch-0.10 -v videotestsrc ! video/x-raw-yuv,format=\(fourcc\)I420,width=320,height=240 ! rtpvrawpay ! udpsink host="192.168.136.130" port=5000
Setting pipeline to PAUSED ...
/GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0.GstPad:src: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)320, height=(int)240, color-matrix=(string)sdtv, chroma-site=(string)mpeg2, framerate=(fraction)30/1
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)320, height=(int)240, color-matrix=(string)sdtv, chroma-site=(string)mpeg2, framerate=(fraction)30/1
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)320, height=(int)240, color-matrix=(string)sdtv, chroma-site=(string)mpeg2, framerate=(fraction)30/1
/GstPipeline:pipeline0/GstRtpVRawPay:rtpvrawpay0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)BT601-5, payload=(int)96, ssrc=(uint)2837818414, clock-base=(uint)2969841640, seqnum-base=(uint)10940
/GstPipeline:pipeline0/GstRtpVRawPay:rtpvrawpay0.GstPad:sink: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)320, height=(int)240, color-matrix=(string)sdtv, chroma-site=(string)mpeg2, framerate=(fraction)30/1
/GstPipeline:pipeline0/GstRtpVRawPay:rtpvrawpay0: timestamp = 2969841640
/GstPipeline:pipeline0/GstRtpVRawPay:rtpvrawpay0: seqnum = 10940
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)BT601-5, payload=(int)96, ssrc=(uint)2837818414, clock-base=(uint)2969841640, seqnum-base=(uint)10940
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
receiver:
unit29@calligraphy2:~$ LANG=en.en gst-launch -v udpsrc uri="udp://192.168.136.130:5000" caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)BT601-5, payload=(int)96, ssrc=(uint)3179834474, clock-base=(uint)2843576415, seqnum-base=(uint)64658" ! rtpvrawdepay ! xvimagesink
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstRtpVRawDepay:rtpvrawdepay0.GstPad:src: caps = video/x-raw-yuv, width=(int)320, height=(int)240, format=(fourcc)I420, framerate=(fraction)0/1
/GstPipeline:pipeline0/GstRtpVRawDepay:rtpvrawdepay0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)BT601-5, payload=(int)96, ssrc=(uint)3179834474, clock-base=(uint)2843576415, seqnum-base=(uint)64658
/GstPipeline:pipeline0/GstXvImageSink:xvimagesink0.GstPad:sink: caps = video/x-raw-yuv, width=(int)320, height=(int)240, format=(fourcc)I420, framerate=(fraction)0/1
`<good>`
If we use Gigabit ethernet mode:
sender:
LANG=en.us gst-launch -v videotestsrc ! video/x-raw-yuv,format=\(fourcc\)I420,width=320,height=240 ! rtpvrawpay ! udpsink host="192.168.192.2" port=5000
Setting pipeline to PAUSED ...
/GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0.GstPad:src: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)320, height=(int)240, color-matrix=(string)sdtv, chroma-site=(string)mpeg2, framerate=(fraction)30/1
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)320, height=(int)240, color-matrix=(string)sdtv, chroma-site=(string)mpeg2, framerate=(fraction)30/1
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)320, height=(int)240, color-matrix=(string)sdtv, chroma-site=(string)mpeg2, framerate=(fraction)30/1
/GstPipeline:pipeline0/GstRtpVRawPay:rtpvrawpay0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)BT601-5, payload=(int)96, ssrc=(uint)2151192507, clock-base=(uint)1835557868, seqnum-base=(uint)14436
/GstPipeline:pipeline0/GstRtpVRawPay:rtpvrawpay0.GstPad:sink: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)320, height=(int)240, color-matrix=(string)sdtv, chroma-site=(string)mpeg2, framerate=(fraction)30/1
/GstPipeline:pipeline0/GstRtpVRawPay:rtpvrawpay0: timestamp = 1835557868
/GstPipeline:pipeline0/GstRtpVRawPay:rtpvrawpay0: seqnum = 14436
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)BT601-5, payload=(int)96, ssrc=(uint)2151192507, clock-base=(uint)1835557868, seqnum-base=(uint)14436
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
receiver2:
unit29@calligraphy2:~$ LANG=en.en gst-launch -v udpsrc uri="udp://192.168.192.2:5000" caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)BT601-5, payload=(int)96, ssrc=(uint)2151192507, clock-base=(uint)1835557868, seqnum-base=(uint)14436" ! rtpvrawdepay ! ffmpegcolorspace ! ximagesink
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstRtpVRawDepay:rtpvrawdepay0.GstPad:src: caps = video/x-raw-yuv, width=(int)320, height=(int)240, format=(fourcc)I420, framerate=(fraction)0/1
/GstPipeline:pipeline0/GstRtpVRawDepay:rtpvrawdepay0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)RAW, sampling=(string)YCbCr-4:2:0, depth=(string)8, width=(string)320, height=(string)240, colorimetry=(string)BT601-5, payload=(int)96, ssrc=(uint)2151192507, clock-base=(uint)1835557868, seqnum-base=(uint)14436
/GstPipeline:pipeline0/GstFFMpegCsp:ffmpegcsp0.GstPad:src: caps = video/x-raw-rgb, bpp=(int)16, depth=(int)16, endianness=(int)1234, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, width=(int)320, height=(int)240, framerate=(fraction)0/1, pixel-aspect-ratio=(fraction)1/1
/GstPipeline:pipeline0/GstFFMpegCsp:ffmpegcsp0.GstPad:sink: caps = video/x-raw-yuv, width=(int)320, height=(int)240, format=(fourcc)I420, framerate=(fraction)0/1
/GstPipeline:pipeline0/GstXImageSink:ximagesink0.GstPad:sink: caps = video/x-raw-rgb, bpp=(int)16, depth=(int)16, endianness=(int)1234, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, width=(int)320, height=(int)240, framerate=(fraction)0/1, pixel-aspect-ratio=(fraction)1/1
`<bad>`
We need show video more then 320x240 resolution
1024x768
`<very bad>`
It is Intel Atom 1.6GHz on the receiver side.
Gigabit ethernet maximum load is 6Mb/sec
Maximum CPU load is 30%.
gstrtpjitterbuffer dont save us too.
mailto: kwispost@gmail.com
**Attachment 211668**, "good":
![good](/uploads/edbb70980ebf02d7694c583073365ac4/good.png)https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/59qtdemux: better handling of unknown QuickTime tag warnings2021-09-24T13:30:03ZBugzilla Migration Userqtdemux: better handling of unknown QuickTime tag warnings## Submitted by Matej `@Knopp`
**[Link to original bug (#670214)](https://bugzilla.gnome.org/show_bug.cgi?id=670214)**
## Description
I'm getting lot of when opening a quicktime file, perhaps these should be added?
WARN gst.qt...## Submitted by Matej `@Knopp`
**[Link to original bug (#670214)](https://bugzilla.gnome.org/show_bug.cgi?id=670214)**
## Description
I'm getting lot of when opening a quicktime file, perhaps these should be added?
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type elng
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type pinf
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type UUID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type elng
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type avc1
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type avcC
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type pasp
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type pinf
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type UUID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type fall
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type elng
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type ac-3
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type elng
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type c608
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type apID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type cnID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type atID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type plID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type geID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type sfID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type akID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type hdvd
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type tvnn
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type stik
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type purd
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type xid
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type flvr
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type mean
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type name
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type ldes
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type sdes
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type mean
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type name
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type elng
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type pinf
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type UUID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type elng
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type avc1
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type avcC
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type pasp
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type pinf
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type UUID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type fall
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type elng
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type ac-3
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type elng
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type c608
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type apID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type cnID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type atID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type plID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type geID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type sfID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type akID
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type hdvd
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type tvnn
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type stik
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type purd
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type xid
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type flvr
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type mean
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type name
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type ldes
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type sdes
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type mean
WARN gst.qtdemux qtdemux_types.c:191 unknown QuickTime node type name
WARN gst.qtdemux qtdemux.c:7053 - - - unknown version 00000000
WARN gst.qtdemux qtdemux.c:7053 - - - unknown version 00000000
WARN gst.qtdemux qtdemux.c:8381 - - - This tag com.apple.iTunes:iTunEXTC type:1 is not mapped, file a bug at bugzilla.gnome.org
WARN gst.qtdemux qtdemux.c:8381 - - - This tag com.apple.iTunes:iTunMOVI type:1 is not mapped, file a bug at bugzilla.gnome.orghttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/57rtspsrc enhancement with events2021-09-24T13:30:02ZBugzilla Migration Userrtspsrc enhancement with events## Submitted by Farkas Levente `@lfarkas`
Assigned to **Farkas Levente `@lfarkas`**
**[Link to original bug (#669746)](https://bugzilla.gnome.org/show_bug.cgi?id=669746)**
## Description
hi,
currently the rtspsrc in gstreamer ar...## Submitted by Farkas Levente `@lfarkas`
Assigned to **Farkas Levente `@lfarkas`**
**[Link to original bug (#669746)](https://bugzilla.gnome.org/show_bug.cgi?id=669746)**
## Description
hi,
currently the rtspsrc in gstreamer are not able to handle any kind of play related events except seek. even seek was implemented at the rtsp level (send a pause then seek and play). all other events are dropped in
http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtsp/gstrtspsrc.c
in gst_rtspsrc_handle_src_event then in gst_rtspsrc_handle_internal_src_event function which simple drop these events.
we'd like to implement at least SEEK, RATE, STEP and NAVIGATION event in rtspsrc to be able to send these events through rtspsrc and the server get it. so we'd like to play with different rate, backward and frame by frame through rtsp. and if the server implements it than it can handle it. (of course we'd like to implement the server side too).
of course we'd like to add the patches to gstreamer, but before we start to implement it we'd like to ask core gstreamer developers which would be the preferred way you ie. how to implement it to be easily accepted by you to inclusion. we thought the best would be to send these events through the rtcp protocol which is something for it.
we'd like to do it first in the 0.10 branch.
what do you think about it?
thanks in advance.https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/56jack: Mono streams are not played as stereo - just in left channel2021-09-24T13:30:02ZBugzilla Migration Userjack: Mono streams are not played as stereo - just in left channel## Submitted by Jiri Prochazka
**[Link to original bug (#668467)](https://bugzilla.gnome.org/show_bug.cgi?id=668467)**
## Description
http://code.google.com/p/clementine-player/issues/detail?id=2646
Mono streams are not played ...## Submitted by Jiri Prochazka
**[Link to original bug (#668467)](https://bugzilla.gnome.org/show_bug.cgi?id=668467)**
## Description
http://code.google.com/p/clementine-player/issues/detail?id=2646
Mono streams are not played as stereo - just in left channel.
For other audio sink/outputs it works, just for JACK it doesn't.https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/54rtpsession: Add thread safe method to retrieve sdes and stats from source.2021-09-24T13:30:01ZBugzilla Migration Userrtpsession: Add thread safe method to retrieve sdes and stats from source.## Submitted by Pascal Buhler
**[Link to original bug (#667848)](https://bugzilla.gnome.org/show_bug.cgi?id=667848)**
## Description
Created attachment 205169
patch
The previous way of retrieving stats/sdes was by first getti...## Submitted by Pascal Buhler
**[Link to original bug (#667848)](https://bugzilla.gnome.org/show_bug.cgi?id=667848)**
## Description
Created attachment 205169
patch
The previous way of retrieving stats/sdes was by first getting a ref to the rtpsource and then querying the property on the source. That is unsafe as there are no locks in rtpsource, the rtpsession needs to be locked. So add accessible function to access the properties of the rtpsource through the rtpsession.
**Patch 205169**, "patch":
[0033-rtpsession-Add-thread-safe-method-to-retrieve-sdes-a.patch](/uploads/be6333285f59d7b329e0e7b5f47c236c/0033-rtpsession-Add-thread-safe-method-to-retrieve-sdes-a.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/53rtpmp4vpay: VOP only stream is not passing through the payloader2021-09-24T13:30:01ZBugzilla Migration Userrtpmp4vpay: VOP only stream is not passing through the payloader## Submitted by Marc Leeman `@den_erpel`
**[Link to original bug (#667565)](https://bugzilla.gnome.org/show_bug.cgi?id=667565)**
## Description
Data is decoded correctly is the mpeg4 video parser is bypassed
clip of 30":
http...## Submitted by Marc Leeman `@den_erpel`
**[Link to original bug (#667565)](https://bugzilla.gnome.org/show_bug.cgi?id=667565)**
## Description
Data is decoded correctly is the mpeg4 video parser is bypassed
clip of 30":
http://crichton.homelinux.org/~marc/downloads/mango-dsp-ivs-raven-m.gdp.gz
This bug also affects mpeg4videoparse.
see https://bugzilla.gnome.org/show_bug.cgi?id=667564
### See also
* [Bug 667564](https://bugzilla.gnome.org/show_bug.cgi?id=667564)https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/51mp4mux: should add support for PSP ftyp2021-09-24T13:30:00ZBugzilla Migration Usermp4mux: should add support for PSP ftyp## Submitted by an unknown user
**[Link to original bug (#660783)](https://bugzilla.gnome.org/show_bug.cgi?id=660783)**
## Description
the Playstation portable uses its own ftyp for what is basically a mp42 file with some size/fps a...## Submitted by an unknown user
**[Link to original bug (#660783)](https://bugzilla.gnome.org/show_bug.cgi?id=660783)**
## Description
the Playstation portable uses its own ftyp for what is basically a mp42 file with some size/fps and codec limitations. List below of file information as provided by the mediainfo tool.
General
Complete name : PSPGO_FINAL.MP4
Format : MPEG-4
Format profile : Sony PSP
Codec ID : MSNV
File size : 32.6 MiB
Duration : 1mn 44s
Overall bit rate : 2 610 Kbps
Encoded date : UTC 2009-05-26 23:23:08
Tagged date : UTC 2009-05-26 23:23:08
Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Main@L2.1
Format settings, CABAC : Yes
Format settings, ReFrames : 2 frames
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 1mn 44s
Bit rate mode : Variable
Bit rate : 1 568 Kbps
Width : 480 pixels
Height : 270 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 29.970 fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.404
Stream size : 19.6 MiB (60%)
Language : English
Encoded date : UTC 2009-05-26 23:23:08
Tagged date : UTC 2009-05-26 23:23:08
Color primaries : BT.601-6 525, BT.1358 525, BT.1700 NTSC, SMPTE 170M
Transfer characteristics : BT.601-6 525, BT.601-6 625, BT.1358 525, BT.1358 625, BT.1700 NTSC, SMPTE 170M
Matrix coefficients : BT.601-6 525, BT.1358 525, BT.1700 NTSC, SMPTE 170M
Audio` #1`
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 1mn 44s
Bit rate mode : Variable
Bit rate : 128 Kbps
Maximum bit rate : 384 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 1.60 MiB (5%)
Language : English
Encoded date : UTC 2009-05-26 23:23:08
Tagged date : UTC 2009-05-26 23:23:08
Correct example file:
http://gstreamer.freedesktop.org/media/large/PSPGO_FINAL.MP4