[FFmpeg-trac] #6658(avformat:new): Dash demuxer segfault

[FFmpeg]

#6658: Dash demuxer segfault
+--
 Reporter:  Detergent   | Type:  defect
   Status:  new | Priority:  normal
Component:  avformat|  Version:  git-master
 Keywords:  dash demux  |   Blocked By:
 Blocking:  |  Reproduced by developer:  0
Analyzed by developer:  0   |
+--
 Summary of the bug:
 How to reproduce:
 {{{
 % ffmpeg -i http://download.tsi.telecom-paristech.fr/gpac/dataset/dash/uhd
 /dashevc-main-6s/dashevc-main-6s.mpd
 ffmpeg version N-87286-g6ce4a635ed Copyright (c) 2000-2017 the FFmpeg
 developers
   built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
   configuration: --enable-gpl --enable-nonfree --enable-libx264 --enable-
 libopus --enable-shared --enable-libcaca --enable-openssl --enable-libxml2
   libavutil  55. 74.100 / 55. 74.100
   libavcodec 57.105.100 / 57.105.100
   libavformat57. 82.100 / 57. 82.100
   libavdevice57.  8.100 / 57.  8.100
   libavfilter 6.105.100 /  6.105.100
   libswscale  4.  7.103 /  4.  7.103
   libswresample   2.  8.100 /  2.  8.100
   libpostproc54.  6.100 / 54.  6.100
 [2]26794 segmentation fault (core dumped)  ffmpeg -i
 }}}

 Backtrace:

 {{{
 Program received signal SIGSEGV, Segmentation fault.
 0x7fbf08d83ce3 in ff_dash_fill_tmpl_params
 (dst=dst@entry=0x7fffb7b0 " ",
 buffer_size=buffer_size@entry=4096, template=0x0,
 rep_id=rep_id@entry=0, number=0,
 bit_rate=bit_rate@entry=0, time=0) at libavformat/dash.c:102
 102 while (dst_pos < buffer_size - 1 && *t_cur) {
 (gdb) bt
 #0  0x7fbf08d83ce3 in ff_dash_fill_tmpl_params
 (dst=dst@entry=0x7fffb7b0 " ",
 buffer_size=buffer_size@entry=4096, template=0x0,
 rep_id=rep_id@entry=0, number=0,
 bit_rate=bit_rate@entry=0, time=0) at libavformat/dash.c:102
 #1  0x7fbf08d87c5d in get_current_fragment (pls=pls@entry=0x66e6e0)
 at libavformat/dashdec.c:1253
 #2  0x7fbf08d87f24 in read_data (opaque=0x66e6e0, buf=0x67e060
 "Ȓ\030\006\277\177",
 buf_size=32768) at libavformat/dashdec.c:1406
 #3  0x7fbf08d78157 in fill_buffer (s=0x66e6e8) at
 libavformat/aviobuf.c:573
 #4  avio_read (s=s@entry=0x66e6e8, buf=0x66ef00
 "\270\222\030\006\277\177",
 size=size@entry=2048) at libavformat/aviobuf.c:667
 #5  0x7fbf08da1ed3 in av_probe_input_buffer2 (pb=pb@entry=0x66e6e8,
 fmt=fmt@entry=0x7fffc968, filename=filename@entry=0x7fbf08ee021c
 "",
 logctx=logctx@entry=0x0, offset=offset@entry=0,
 max_probe_size=1048576,
 max_probe_size@entry=0) at libavformat/format.c:314
 #6  0x7fbf08da20e9 in av_probe_input_buffer (pb=pb@entry=0x66e6e8,
 fmt=fmt@entry=0x7fffc968, filename=filename@entry=0x7fbf08ee021c
 "",
 logctx=logctx@entry=0x0, offset=offset@entry=0,
 max_probe_size=max_probe_size@entry=0)
 at libavformat/format.c:368
 #7  0x7fbf08d84536 in reopen_demux_for_component (s=s@entry=0x669e20,
 pls=pls@entry=0x66e6e0) at libavformat/dashdec.c:1530
 #8  0x7fbf08d84dd5 in open_demux_for_component (s=s@entry=0x669e20,
 pls=0x66e6e0)
 at libavformat/dashdec.c:1565
 #9  0x7fbf08d8841d in dash_read_header (s=0x669e20) at
 libavformat/dashdec.c:1615
 #10 0x7fbf08eaf0ee in avformat_open_input (ps=ps@entry=0x7fffcb88,
 filename=filename@entry=0x7fffde40 "http://download.tsi.telecom-
 paristech.fr/gpac/dataset/dash/uhd/dashevc-main-6s/dashevc-main-6s.mpd",
 fmt=fmt@entry=0x0, options=0x669c28)
 at libavformat/utils.c:595
 #11 0x00417b55 in open_input_file (o=o@entry=0x7fffcd40,
 filename=) at ffmpeg_opt.c:1060
 #12 0x00419a23 in open_files (l=0x669bd8, l=0x669bd8,
 open_file=0x4160d0 , inout=0x43299f "input") at
 ffmpeg_opt.c:3278
 #13 ffmpeg_parse_options (argc=argc@entry=3,
 argv=argv@entry=0x7fffda98)
 at ffmpeg_opt.c:3318
 #14 0x00408542 in main (argc=3, argv=0x7fffda98) at
 ffmpeg.c:4786
 }}}

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Re: [FFmpeg-trac] #6657(avformat:new): DVB-T2 audio support

[FFmpeg]

#6657: DVB-T2 audio support
-+-
 Reporter:   |Owner:
  Igor_Selivanov |   Status:  new
 Type:  defect   |Component:  avformat
 Priority:  normal   |   Resolution:
  Version:  git-master   |   Blocked By:
 Keywords:  mpegts latm  |  Reproduced by developer:  0
 Blocking:   |
Analyzed by developer:  0|
-+-
Changes (by cehoyos):

 * keywords:   => mpegts latm
 * version:  unspecified => git-master
 * component:  undetermined => avformat


Comment:

 For future tickets:
 Please always test current FFmpeg git head (the only version supported
 here) and provide the command line you tested together with the complete,
 uncut console output, do not provide excerpts that you consider important.

 Do you know for sure that this is a problem with ARD and not a problem
 with your LOEWE tv?

 Patch sent.

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Re: [FFmpeg-trac] #6656(avdevice:new): jack output support

[FFmpeg]

#6656: jack output support
-+
 Reporter:  dimbulb  |Owner:
 Type:  enhancement  |   Status:  new
 Priority:  wish |Component:  avdevice
  Version:  unspecified  |   Resolution:
 Keywords:  jack |   Blocked By:
 Blocking:   |  Reproduced by developer:  0
Analyzed by developer:  0|
-+
Changes (by llogan):

 * keywords:   => jack
 * priority:  normal => wish
 * component:  undetermined => avdevice


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[FFmpeg-trac] #6659(avformat:new): Feature Request - Range header in RTSP protocol implementation

[FFmpeg]

#6659: Feature Request - Range header in RTSP protocol implementation
-+-
 Reporter:   | Type:
  walterzielony  |  enhancement
   Status:  new  | Priority:  important
Component:  avformat |  Version:
 Keywords:  rtsp |  unspecified
 Blocking:   |   Blocked By:
Analyzed by developer:  0|  Reproduced by developer:  0
-+-
 The ability to specify "Range" header in time format for RTSP protocols is
 desired, in order to allow syncronized starts for multiple RTSP stream
 connection.

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[FFmpeg-trac] #6660(ffmpeg:new): RTSP timeout for GET_PARAMETER request is hardcoded

[FFmpeg]

#6660: RTSP timeout for GET_PARAMETER request is hardcoded
-+-
 Reporter:   | Type:
  walterzielony  |  enhancement
   Status:  new  | Priority:  normal
Component:  ffmpeg   |  Version:
 Keywords:   |  unspecified
 Blocking:   |   Blocked By:
Analyzed by developer:  0|  Reproduced by developer:  0
-+-
 The timeout for keep-alive requests made to an RTSP stream is not
 customizable. Instead, it is hardcoded in rtspc.c file to 60 seconds.

 The current parameter -timeout only stands for incomming connections
 (implies the ‘rtsp_flags’ set to ‘listen’).

 Also the value seems to be overwriten later with the response from the
 SETUP request made at the RTSP stream start. This should be avoided also.

 This is needed to allow for larger timeout values preventing ffmpeg to
 drop out.

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Re: [FFmpeg-trac] #6660(ffmpeg:new): RTSP timeout for GET_PARAMETER request is hardcoded

[FFmpeg]

#6660: RTSP timeout for GET_PARAMETER request is hardcoded
---+--
 Reporter:  walterzielony  |Owner:
 Type:  enhancement|   Status:  new
 Priority:  normal |Component:  ffmpeg
  Version:  unspecified|   Resolution:
 Keywords: |   Blocked By:
 Blocking: |  Reproduced by developer:  1
Analyzed by developer:  1  |
---+--
Changes (by walterzielony):

 * analyzed:  0 => 1
 * reproduced:  0 => 1


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Re: [FFmpeg-trac] #6651(ffmpeg:new): cut file without specifying codecs

[FFmpeg]

#6651: cut file without specifying codecs
---+--
 Reporter:  Xoristzatziki  |Owner:
 Type:  enhancement|   Status:  new
 Priority:  wish   |Component:  ffmpeg
  Version:  git-master |   Resolution:
 Keywords: |   Blocked By:
 Blocking: |  Reproduced by developer:  0
Analyzed by developer:  0  |
---+--

Comment (by Xoristzatziki):

 That's my main request. Thanks

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Re: [FFmpeg-trac] #6659(avformat:new): Feature Request - Range header in RTSP protocol implementation

[FFmpeg]

#6659: Feature Request - Range header in RTSP protocol implementation
-+-
 Reporter:   |Owner:
  walterzielony  |   Status:  new
 Type:  enhancement  |Component:  avformat
 Priority:  wish |   Resolution:
  Version:  unspecified  |   Blocked By:
 Keywords:  rtsp |  Reproduced by developer:  0
 Blocking:   |
Analyzed by developer:  0|
-+-
Changes (by cehoyos):

 * priority:  important => wish


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Re: [FFmpeg-trac] #6660(avformat:new): RTSP timeout for GET_PARAMETER request is hardcoded

[FFmpeg]

#6660: RTSP timeout for GET_PARAMETER request is hardcoded
-+-
 Reporter:   |Owner:
  walterzielony  |   Status:  new
 Type:  enhancement  |Component:  avformat
 Priority:  wish |   Resolution:
  Version:  unspecified  |   Blocked By:
 Keywords:  rtsp |  Reproduced by developer:  0
 Blocking:   |
Analyzed by developer:  0|
-+-
Changes (by cehoyos):

 * analyzed:  1 => 0
 * keywords:   => rtsp
 * component:  ffmpeg => avformat
 * reproduced:  1 => 0
 * priority:  normal => wish


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[FFmpeg-trac] #6661(undetermined:new): Missing streams when cutting M2TS file

[FFmpeg]

#6661: Missing streams when cutting M2TS file
-+-
 Reporter:  xamarin  | Type:  defect
   Status:  new  | Priority:  normal
Component:   |  Version:
  undetermined   |  unspecified
 Keywords:   |   Blocked By:
 Blocking:   |  Reproduced by developer:  0
Analyzed by developer:  0|
-+-
 Summary of the bug:
 When cutting a M2TS file to get a portion of it, the portion is missing
 all subtitle tracks and all audio except one.
 The original file has 6 audio and 5 subtitles.

 How to reproduce:
 {{{
 % ffmpeg.exe -i 1.m2ts -vcodec copy -acodec copy -ss 00:09:23 -to
 00:10:00 outputFile.m2ts
 ffmpeg-20170910-8e17cd2-win64-static
 }}}


 {{{
 ffmpeg version N-87257-g8e17cd2 Copyright (c) 2000-2017 the FFmpeg
 developers
   built with gcc 7.1.0 (GCC)
   configuration: --enable-gpl --enable-version3 --enable-cuda --enable-
 cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc
 --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
 --enable-gnutls --enable-iconv --enable-libass --enable-libbluray
 --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme
 --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264
 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy
 --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame
 --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-
 libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-
 libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma
 --enable-zlib
   libavutil  55. 74.100 / 55. 74.100
   libavcodec 57.105.100 / 57.105.100
   libavformat57. 82.100 / 57. 82.100
   libavdevice57.  8.100 / 57.  8.100
   libavfilter 6.105.100 /  6.105.100
   libswscale  4.  7.103 /  4.  7.103
   libswresample   2.  8.100 /  2.  8.100
   libpostproc54.  6.100 / 54.  6.100
 [mpegts @ 025d24a0] start time for stream 7 is not set in
 estimate_timings_from_pts
 [mpegts @ 025d24a0] start time for stream 8 is not set in
 estimate_timings_from_pts
 [mpegts @ 025d24a0] start time for stream 9 is not set in
 estimate_timings_from_pts
 [mpegts @ 025d24a0] start time for stream 10 is not set in
 estimate_timings_from_pts
 [mpegts @ 025d24a0] start time for stream 11 is not set in
 estimate_timings_from_pts
 [mpegts @ 025d24a0] Could not find codec parameters for stream 7
 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size
 Consider increasing the value for the 'analyzeduration' and 'probesize'
 options
 [mpegts @ 025d24a0] Could not find codec parameters for stream 8
 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size
 Consider increasing the value for the 'analyzeduration' and 'probesize'
 options
 [mpegts @ 025d24a0] Could not find codec parameters for stream 9
 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size
 Consider increasing the value for the 'analyzeduration' and 'probesize'
 options
 [mpegts @ 025d24a0] Could not find codec parameters for stream 10
 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size
 Consider increasing the value for the 'analyzeduration' and 'probesize'
 options
 [mpegts @ 025d24a0] Could not find codec parameters for stream 11
 (Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090)): unspecified size
 Consider increasing the value for the 'analyzeduration' and 'probesize'
 options
 Input #0, mpegts, from '.\1.m2ts':
   Duration: 01:44:54.31, start: 11.642667, bitrate: 72514 kb/s
   Program 1
 Stream #0:0[0x1011]: Video: hevc (Main 10) (HDMV / 0x564D4448),
 yuv420p10le(tv, bt2020nc/bt2020/smpte2084), 3840x2160 [SAR 1:1 DAR 16:9],
 23.98 fps, 23.98 tbr, 90k tbn, 23.98 tbc
 Stream #0:1[0x1100]: Audio: truehd (AC-3 / 0x332D4341), 48000 Hz, 7.1,
 s32 (24 bit)
 Stream #0:2[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz,
 5.1(side), fltp, 640 kb/s
 Stream #0:3[0x1101]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz,
 5.1(side), fltp, 640 kb/s
 Stream #0:4[0x1102]: Audio: dts (DTS-HD MA) ([134][0][0][0] / 0x0086),
 48000 Hz, 5.1(side), s16p
 Stream #0:5[0x1103]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo,
 fltp, 192 kb/s
 Stream #0:6[0x1104]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz,
 5.1(side), fltp, 640 kb/s
 Stream #0:7[0x12a0]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] /
 0x0090)
 Stream #0:8[0x12a1]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] /
 0x0090)
 

Re: [FFmpeg-trac] #6661(undetermined:closed): Missing streams when cutting M2TS file

[FFmpeg]

#6661: Missing streams when cutting M2TS file
-+-
 Reporter:  xamarin  |Owner:
 Type:  defect   |   Status:  closed
 Priority:  normal   |Component:
  Version:  unspecified  |  undetermined
 Keywords:   |   Resolution:  invalid
 Blocking:   |   Blocked By:
Analyzed by developer:  0|  Reproduced by developer:  0
-+-
Changes (by llogan):

 * status:  new => closed
 * resolution:   => invalid


Comment:

 This is not a bug. The default stream selection behavior is to select one
 stream per stream type. See:
 https://ffmpeg.org/ffmpeg.html#Stream-selection

 If you want to override this behavior use the `-map` option, such as `-map
 0` to map all streams from input 0 (the first and only input in your
 case).

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[FFmpeg-trac] #6657(undetermined:new): DVB-T2 audio support

[FFmpeg]

#6657: DVB-T2 audio support
-+-
 Reporter:   | Type:  defect
  Igor_Selivanov | Priority:  normal
   Status:  new  |  Version:
Component:   |  unspecified
  undetermined   |   Blocked By:
 Keywords:   |  Reproduced by developer:  0
 Blocking:   |
Analyzed by developer:  0|
-+-
 Summary of the bug:
 Audio is not supported. The video samples dvbt2_small.ts is recording of
 free German DVB-T2 TV channel "Das Erste HD". Unfortunately sound track
 AAC format is not supported/implemented with FFMPEG version 3.3.2. Since
 this year (March 2017) new DVB-T2 standard is broadcasting in Germany and
 France. Unfortunately FFMPEG doesn't implement support of some audio
 formats of this standard.
 How to reproduce:
 {{{
 ffmpeg -i dvbt2_small.ts
 }}}
 or
 {{{
 ffplay dvbt2_small.ts
 }}}
 The problem related to audio track:
 Stream #0:1[0x552]: Audio: aac (LTP) ([15][0][0][0] / 0x000F), 22050 Hz,
 5.1, fltp, 1077 kb/s
 Errors are:
 - More than one AAC RDB per ADTS frame is not implemented. Update your
 FFmpeg version to the newest one from Git. If the problem still occurs, it
 means that your file has a feature which has not been implemented.
 - Too large remapped id is not implemented. Update your FFmpeg version to
 the newest one from Git. If the problem still occurs, it means that your
 file has a feature which has not been implemented

 FFmpeg version:
 OS: Arch Linux, ffmpeg version is taken from Arch Linux repository.
 ffmpeg version 3.3.2 Copyright (c) 2000-2017 the FFmpeg developers
   built with gcc 7.1.1 (GCC) 20170516
   configuration: --prefix=/usr --disable-debug --disable-static --disable-
 stripping --enable-avisynth --enable-avresample --enable-fontconfig
 --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libass
 --enable-libbluray --enable-libfreetype --enable-libfribidi --enable-
 libgsm --enable-libiec61883 --enable-libmodplug --enable-libmp3lame
 --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg
 --enable-libopus --enable-libpulse --enable-libschroedinger --enable-
 libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-
 libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-
 libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxvid
 --enable-shared --enable-version3
   libavutil  55. 58.100 / 55. 58.100
   libavcodec 57. 89.100 / 57. 89.100
   libavformat57. 71.100 / 57. 71.100
   libavdevice57.  6.100 / 57.  6.100
   libavfilter 6. 82.100 /  6. 82.100
   libavresample   3.  5.  0 /  3.  5.  0
   libswscale  4.  6.100 /  4.  6.100
   libswresample   2.  7.100 /  2.  7.100
   libpostproc54.  5.100 / 54.  5.100

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Re: [FFmpeg-trac] #6627(avformat:closed): av_read_frame got incorrect packet after av_seek_frame

[FFmpeg]

#6627: av_read_frame got incorrect packet after av_seek_frame
-+-
 Reporter:  sorayuki |Owner:
 Type:  defect   |   Status:  closed
 Priority:  normal   |Component:  avformat
  Version:  git-master   |   Resolution:
 Keywords:  mov  |  needs_more_info
 Blocking:   |   Blocked By:
Analyzed by developer:  0|  Reproduced by developer:  0
-+-
Changes (by sorayuki):

 * status:  new => closed
 * resolution:   => needs_more_info


Comment:

 In second turn, that the audio packet with the same pts is larger
 is caused by the merging of side data.

 Decoding error may be caused by other problem.

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Re: [FFmpeg-trac] #6654(undetermined:new): AAC-file stay on pause through LAVSplitterSource right after start.

[FFmpeg]

#6654: AAC-file stay on pause through LAVSplitterSource right after start.
-+-
 Reporter:  max79|Owner:
 Type:  defect   |   Status:  new
 Priority:  important|Component:
  Version:  git-master   |  undetermined
 Keywords:  aac  |   Resolution:
  regression |   Blocked By:
 Blocking:   |  Reproduced by developer:  0
Analyzed by developer:  0|
-+-

Comment (by max79):

 Replying to [comment:1 cehoyos]:
 > Possibly a duplicate of ticket #6634
 yeah seems as duplicate, please sorry for my inattention. But issue still
 not fixed. thank you for pointing!

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