May 2017 - ffmpeg-user - lists.ffmpeg.org

could not find codec parameters for stream 0 unspecified size
by Ashish Lal 09 Aug '17

09 Aug '17

I am trying to convert a live rtsp stream (H264 + MPEG2 TS ) into MP4 using ffmpeg. I am getting an error "could not find codec parameters for stream 0 unspecified size". Any help in fixing this is appreciated. ffmpeg -rtsp_transport tcp -i rtsp://AA.BB.CC.DD:5555/mpeg2TransportStreamFr omUDPSourceTest -analyzeduration 999999999 -probesize 99999999 -vcodec libx264 -s 176x144 -b:v 256k -f mp4 -movflags frag_keyframe+empty_moov op.mp4 ffmpeg version 2.2.2 Copyright (c) 2000-2014 the FFmpeg developers built on May 27 2014 09:12:51 with gcc 4.7 (Debian 4.7.2-5) configuration: --enable-libx264 --enable-encoder=libx264 --enable-libvpx --pre fix=/home/ashish/ffmpeg-2.2.2/mybuild --enable-gpl --enable-debug --disable-stri pping --disable-optimizations --enable-memalign-hack libavutil 52. 66.100 / 52. 66.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 2.100 / 4. 2.100 libswscale 2. 5.102 / 2. 5.102 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 [NULL @ 0x235cd60] non-existing PPS 0 referenced [h264 @ 0x235cd60] non-existing PPS 0 referenced [h264 @ 0x235cd60] decode_slice_header error [h264 @ 0x235cd60] no frame! [h264 @ 0x235cd60] non-existing PPS 0 referenced ......... ......... Last message repeated 1 times [h264 @ 0x235cd60] decode_slice_header error [h264 @ 0x235cd60] no frame! [h264 @ 0x235cd60] non-existing PPS 0 referenced [rtsp @ 0x232fc40] decoding for stream 0 failed [rtsp @ 0x232fc40] Could not find codec parameters for stream 0 (Video: h264 ([2 7][0][0][0] / 0x001B)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options rtsp://AA.BB.CC.DD:5555/mpeg2TransportStreamFromUDPSourceTest: could not find codec parameters Conversion failed! Regards Ashish

3 3

Create an AAC stream matching the Core Media Audio packet format / priming etc?
by Mark Burton 18 Jul '17

18 Jul '17

I appreciate this is a tricky area and there appear to be different ways that some encoders create AAC streams with regards to the padding and remaining samples etc. I won’t pretend to fully understand all the factors, but I would like to ask a genuine question which purely comes from the point of view of wanting to create a file for my working environment - an environment dominated by Quicktime 7 and X playback / decoding tools. I’m no great fan of Quicktime and appreciate its not well loved here also. In my industry it is still very much the defecto playback engine though, so if I’m able to tailor a file for this decoder, it would be an enormous help. Let me also say, I am not accusing ffmpeg of having an issue, I have read a number of ‘bug’ reports surrounding ffmpeg and AAC priming and there being a sync discrepancy in the resultant encode when played back in certain decoders. I happen to see the exact same issue, but from some of the developers replies, I accept their position is that they feel ffmpeg is doing it the right way and its the decoders that are at fault. With that said, I’d like to approach this question purely from the point of view of finding out whether there is a way to tweak a command in order to change this way the aac stream is created to produce an mp4 or mov file using the native aac encoder which decodes in Quicktime 7 or X, in sync. Currently an encoded file plays back 1 frame out of sync (audio is early by approx. 1 frame). In VLC its about 1/2 a frame out of sync. The source file is a .mov, DNx115 24p (true 24p, not 23.976), PCM 24bit 48khz audio, which is in sync. This is film material where sync is crucial and always expected. Here is the basic command to reproduce. I have attached the uncut loglevel 99 console output for this command: ffmpeg -i SyncTest24p.mov -c:v libx264 -pix_fmt yuv420p -movflags faststart -c:a aac -b:a 128k ffmpeg.mp4 ffmpeg -i SyncTest24p.mov -c:v libx264 -pix_fmt yuv420p -movflags faststart -c:a aac -b:a 128k ffmpeg.mp4 ffmpeg version N-85343-gd0a3143-tessus Copyright (c) 2000-2017 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzmq --enable-libzvbi --enable-version3 --disable-ffplay --disable-indev=qtkit libavutil 55. 60.100 / 55. 60.100 libavcodec 57. 92.100 / 57. 92.100 libavformat 57. 72.100 / 57. 72.100 libavdevice 57. 7.100 / 57. 7.100 libavfilter 6. 84.100 / 6. 84.100 libswscale 4. 7.100 / 4. 7.100 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SyncTest24p.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2017-04-14T13:37:46.000000Z timecode : 01:00:00:00 Duration: 00:00:02.00, start: 0.000000, bitrate: 118705 kb/s Stream #0:0(eng): Video: dnxhd (DNXHD) (AVdn / 0x6E645641), yuv422p(tv, bt709/unknown/unknown), 1920x1080, 116391 kb/s, SAR 1:1 DAR 16:9, 24 fps, 24 tbr, 24k tbn, 24k tbc (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler encoder : Avid DNxHD Codec Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, stereo, s32 (24 bit), 2304 kb/s (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler timecode : 01:00:00:00 Stream mapping: Stream #0:0 -> #0:0 (dnxhd (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (pcm_s24le (native) -> aac (native)) Press [q] to stop, [?] for help [libx264 @ 0x7ff1ae809200] using SAR=1/1 [libx264 @ 0x7ff1ae809200] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 0x7ff1ae809200] profile High, level 4.0 [libx264 @ 0x7ff1ae809200] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=24 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'ffmpeg.mp4': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt timecode : 01:00:00:00 encoder : Lavf57.72.100 Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 24 fps, 12288 tbn, 24 tbc (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler encoder : Lavc57.92.100 libx264 Side data: cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 Stream #0:1(eng): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp (24 bit), 128 kb/s (default) Metadata: creation_time : 2017-04-14T13:37:46.000000Z handler_name : Apple Alias Data Handler encoder : Lavc57.92.100 aac [mp4 @ 0x7ff1ae81ec00] Starting second pass: moving the moov atom to the beginning of the file frame= 48 fps=0.0 q=-1.0 Lsize= 24kB time=00:00:02.00 bitrate= 97.2kbits/s speed= 3.6x video:17kB audio:3kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 15.142209% [libx264 @ 0x7ff1ae809200] frame I:1 Avg QP:18.80 size: 4448 [libx264 @ 0x7ff1ae809200] frame P:14 Avg QP:16.87 size: 374 [libx264 @ 0x7ff1ae809200] frame B:33 Avg QP:23.31 size: 227 [libx264 @ 0x7ff1ae809200] consecutive B-frames: 2.1% 16.7% 6.2% 75.0% [libx264 @ 0x7ff1ae809200] mb I I16..4: 52.6% 46.2% 1.2% [libx264 @ 0x7ff1ae809200] mb P I16..4: 0.0% 0.0% 0.1% P16..4: 0.2% 0.1% 0.0% 0.0% 0.0% skip:99.6% [libx264 @ 0x7ff1ae809200] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.2% 0.1% 0.0% direct: 0.0% skip:99.7% L0:60.9% L1:38.1% BI: 1.0% [libx264 @ 0x7ff1ae809200] 8x8 transform intra:45.8% inter:14.5% [libx264 @ 0x7ff1ae809200] coded y,uvDC,uvAC intra: 1.4% 0.0% 0.0% inter: 0.0% 0.0% 0.0% [libx264 @ 0x7ff1ae809200] i16 v,h,dc,p: 97% 0% 3% 0% [libx264 @ 0x7ff1ae809200] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 62% 2% 36% 0% 0% 0% 0% 0% 0% [libx264 @ 0x7ff1ae809200] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 30% 16% 33% 5% 2% 4% 2% 6% 2% [libx264 @ 0x7ff1ae809200] i8c dc,h,v,p: 100% 0% 0% 0% [libx264 @ 0x7ff1ae809200] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x7ff1ae809200] ref P L0: 70.5% 1.8% 16.3% 11.3% [libx264 @ 0x7ff1ae809200] ref B L0: 58.5% 32.1% 9.4% [libx264 @ 0x7ff1ae809200] ref B L1: 90.6% 9.4% [libx264 @ 0x7ff1ae809200] kb/s:68.72 [aac @ 0x7ff1ae80aa00] Qavg: 61514.836 Attached is an ffprobe log file showing the audio packets for this ffmpeg encode. Also, I ran the same source file through Apple Compressor v4.3.1 to the same output file spec and through YouTube and have attached ffprobe logs of their audio packets. Apologies if I’ve not gone about this the right way, I’m not trying to demonstrate a bug, simply want to highlight the differences between the initial packets of these files and see if there is a way to produce a file via ffmpeg which uses a packet ‘setup’ similar to the Compressor or even YouTube way as this plays very nice with Quicktime decoders. Perhaps the key here is that ffmpeg has a negative pts for the first packet and this is what Quicktime struggles with I would think. It also has SIDE DATA which Quicktime may also be struggling with, but I’ve not found much info on this and from my understanding Quicktime always assumes 2112 samples of delay at the start. https://developer.apple.com/library/content/technotes/tn2258/_index.html Start of ffmpeg file: [PACKET] codec_type=audio stream_index=1 pts=-1024 pts_time=-0.021333 dts=-1024 dts_time=-0.021333 duration=1024 duration_time=0.021333 convergence_duration=N/A convergence_duration_time=N/A size=300 pos=8722 flags=KD [SIDE_DATA] side_data_type=Skip Samples skip_samples=1024 discard_padding=0 skip_reason=0 discard_reason=0 [/SIDE_DATA] [/PACKET] [PACKET] codec_type=audio stream_index=1 pts=0 pts_time=0.000000 dts=0 dts_time=0.000000 duration=1024 duration_time=0.021333 convergence_duration=N/A convergence_duration_time=N/A size=410 pos=9351 flags=K_ [/PACKET] ... Start of Compressor file: [PACKET] codec_type=audio stream_index=0 pts=0 pts_time=0.000000 dts=0 dts_time=0.000000 duration=1024 duration_time=0.021333 convergence_duration=N/A convergence_duration_time=N/A size=6 pos=2320 flags=K_ [/PACKET] ... Start of YouTube file: [PACKET] codec_type=audio stream_index=1 pts=0 pts_time=0.000000 dts=0 dts_time=0.000000 duration=1024 duration_time=0.023220 convergence_duration=N/A convergence_duration_time=N/A size=557 pos=9043 flags=K_ [/PACKET] … Thank you if you have read this far, I really appreciate it. I’m slightly worried this is going to open a can of worms which will be above my pay grade! Anyway, assuming I have provided enough info (and I can send anyone the source file if they want it), are there any options / flags etc I could add to the command to bring the encode in line with the Compressor or YouTube outputs in terms of the aac audio stream? Many many thanks for any help, Regards Mark

11 50

Converting a cassette audio tape (Lecture) to mp3
by jd1008 14 Jul '17

14 Jul '17

Hi all, I am running fedora 20 with ffmpeg version N-51556-ge278500 Copyright (c) 2000-2013 the FFmpeg developers I also installed in /opt ffmpeg version N-72985-g7c3f7e6 Copyright (c) 2000-2015 the FFmpeg developers I need a set of steps/commands to convert this tape. I have plenty of audio tools installed - but not certain what else I might need. If anyone has done this before, please share your script. Kind regards, JD

6 15

need help in reducing the stream start latency in a network enviroment
by Ratheendran R 14 Jun '17

14 Jun '17

Hi All, I am wrote an app which reads elementary streams of encoded audio/video and mux them into mpegts stream subsequently write to unix domain socket as client, an Unix socket server app using libwebsocket takes this streams and send it to websocket server. Now this works as expected, currently mpegts stream start latency is ~ 10 sec. Also this is an embedded system use case and encoding happens through hardware accelarators on soc fixed properties.I would like to know if we can further reduce the size of library and bypass redundant task like probe and parsing metadata of mpegts streams. options used for configuring ffmpeg --enable-decoder=h264 --enable-decoder=aac --enable-bsf=h264_mp4toannexb ---enable-muxer=mpegts --enable-protocol=unix can any one in the forum share their experience on fixing the high latency on mpegts streaming. Thanks, Ratheendran

2 5

Compiling FFMPEG with ffplay support
by Stas Oskin 13 Jun '17

13 Jun '17

Hi. Did the compilation parameters changed in recent FFMPEG version? Configuring the FFMPEG like this: ./configure --prefix=/usr --enable-pthreads Does not produce ffplay anymore. I tried to add --enable-ffplay, without any result. Any idea what I'm missing? Thanks.

6 9

Create thumbnails from RTMP stream
by Grant Curtis 09 Jun '17

09 Jun '17

If I use Flash Media Live Encoder to send a stream to an RTMP server, and then, on that server, run the below command, I can create thumbnails from the stream. But when I try to do the same when I stream the file to the server using ffmpeg, I get errors, and no thumbnails are created Server command and output: $ /usr/bin/ffmpeg -report -i "rtmp://localhost:1935/osc-live-packager/cumin?offsetFromLive=10,userID=testUser1/mp4:rebroadcast_testUser1" -y -r 29.97 -threads 4 -c:v copy -c:a copy /mnt_ephemeral1/media/cumin/cumin_1000.mp4 -filter:v "scale=w=178:h=100" -f image2 -vf "fps=fps=1/5" "/mnt_ephemeral1/media/cumin/cumin/thumbs/thumb_%04d.png" ffmpeg started on 2016-04-25 at 18:28:39 Report written to "/var/log/ffmpeg/ffmpeg-20160425-182839.log" ffmpeg version 2.7.2-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.3 (Debian 4.9.3-1) configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg --enable-libopus --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --cc=gcc-4.9 libavutil 54. 27.100 / 54. 27.100 libavcodec 56. 41.100 / 56. 41.100 libavformat 56. 36.100 / 56. 36.100 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 16.101 / 5. 16.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.100 / 1. 2.100 libpostproc 53. 3.100 / 53. 3.100 [NULL @ 0x31b65e0] missing picture in access unit with size 110664 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2338 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2645 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2600 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 8421 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2895 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 1759 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 776 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 13391 [h264 @ 0x31b65e0] no frame! -truncated a bunch more similar messages- [flv @ 0x31b3c00] decoding for stream 0 failed [flv @ 0x31b3c00] Could not find codec parameters for stream 0 (Video: h264, none): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, flv, from 'rtmp://localhost:1935/osc-live-packager/cumin?offsetFromLive=10,userID=testUser1/mp4:rebroadcast_testUser1': Metadata: moovPosition : 36 avcprofile : 100 avclevel : 31 videoframerate : 25 audiochannels : 2 timescale : 1000 length : 839 sampletype : .mp3 Duration: 00:00:00.84, start: 0.040000, bitrate: N/A Stream #0:0: Video: h264, none, 25 tbr, 1k tbn, 2k tbc Stream #0:1: Subtitle: text Stream #0:2: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s Output #1, image2, to '/mnt_ephemeral1/media/cumin/cumin/thumbs/thumb_%04d.png': Metadata: moovPosition : 36 avcprofile : 100 avclevel : 31 videoframerate : 25 audiochannels : 2 timescale : 1000 length : 839 sampletype : .mp3 Output file #1 does not contain any stream Received signal 2: terminating. --- Here's how I start the stream from my local machine: --- ffmpeg -re -i testVideo.mp4 -ar 44100 -c:v libx264 -f flv rtmp://xxxxxxxx:1935/xxxxx/xxxxx/xxxxx_1000.mp4 ffmpeg version 2.8.4 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 5.2.0 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --enable-decklink --enable-zlib libavutil 54. 31.100 / 54. 31.100 libavcodec 56. 60.100 / 56. 60.100 libavformat 56. 40.101 / 56. 40.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 40.101 / 5. 40.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc 53. 3.100 / 53. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testVideo.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf53.24.2 Duration: 00:01:02.32, start: 0.000000, bitrate: 1347 kb/s Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 959 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 383 kb/s (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : SoundHandler [libx264 @ 03052340] using SAR=1/1 [libx264 @ 03052340] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 03052340] profile High, level 3.1 [libx264 @ 03052340] 264 - core 148 r2638 7599210 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, flv, to 'rtmp://xxxxxx:1935/xxxx/xxxxx/xxxxx_1000.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.40.101 Stream #0:0(und): Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 25 fps, 1k tbn, 25 tbc (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler encoder : Lavc56.60.100 libx264 Stream #0:1(und): Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : SoundHandler encoder : Lavc56.60.100 libmp3lame Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (aac (native) -> mp3 (libmp3lame)) Press [q] to stop, [?] for help frame= 1336 fps= 25 q=28.0 size= 7310kB time=00:00:53.63 bitrate=1116.6kbits/s

2 2

Re: [FFmpeg-user] FFMPEG & Panasonic OP-1B MXF decode issue
by Jeff Smith 06 Jun '17

06 Jun '17

> 2017-03-10 9:20 GMT+01:00 <jeff at jksmith.com <http://ffmpeg.org/mailman/listinfo/ffmpeg-user>>: > >/Happy to provide a sample. Is there an FTP site or something I /> >/can upload it to? /> > You can usehttp://streams.videolan.org/upload/ but you absolutely > have to provide the file name, the service is extremely impractical. > > Carl Eugen Hi Carl, I'm assuming from radio silence that you couldn't get the sample I uploaded. I've posted it on an anonymous FTP server here: ftp://5.56.49.144/085FV4BR.MXF . Grateful if you could acknowledge if this works for you. Thanks, Jeff --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

2 1

Problems with mpegts and long running stream
by Mathias Schneider 01 Jun '17

01 Jun '17

Hello, I have the following setup: a mpegts stream is built by a gstreamer pipeline and re-muxed by ffmpeg. I want to use the rich feature set of the ffmpeg mpegts-muxer (i.e. fill null-packets into my mpegts stream), therefore I’ve chosen this setup. ffmpeg version: 3.3 (latest static build for Linux from ffmpeg.org) gstreamer version: 1.10 OS: Ubuntu 16.04 Example: Start ffmpeg - it will wait for incoming stream from gstreamer: ffmpeg -re -buffer_size 2560000 -i rtp://127.0.0.1:2335?listen -c copy -muxrate 6000k -f mpegts -mpegts_flags system_b udp://239.200.0.82:1234?pkt_size=1316 Start gstreamer: gst-launch-1.0 -e -m -v videotestsrc is-live=true name=vsource pattern=18 ! videoscale ! videoconvert ! videorate ! video/x-raw,width=1280,height=720,framerate=25/1 ! queue ! videoconvert ! videoscale ! videorate ! x264enc bitrate=5000 threads=0 byte-stream=true key-int-max=50 ! video/x-h264,profile=high ! queue ! h264parse ! mpegtsmux alignment=7 name=strmux ! rtpmp2tpay ! udpsink async=false sync=false host=127.0.0.1 port=2335 Everything works fine for around 1 day - 26,5 hours to be exact. Then, ffmpeg fires a lot of warnings, saying "Non-monotonous DTS in output stream“. The first of these messages gives a hint to the problem: [mpegts @ 0x4f21ba0] Non-monotonous DTS in output stream 0:0; previous: 8584531106, current: -5399886; changing to 8584531107. This may result in incorrect timestamps in the output file. There is a overflow in the mpegts PCR-field (33 bits), I guess. Is there a way to make ffmpeg dealing correctly with this overflow? I’ve already tried the options "-copyts" and "-copytb 1“ - but without success. During my search for a solution, I found this patch here: https://github.com/arut/ffmpeg-patches/blob/master/mpegts-33bit I’ve patched my ffmpeg with it and it seems to solve my issue! Question: Did I something wrong or is there really a problem inside ffmpeg mpegts for long duration streams? And is this patch a correct solution? Thanks, Mathias

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Output file size
by Rodolfo Medina 31 May '17

31 May '17

Hi all. I'm adding id3 tags to a 208M mp3 file, with: $ ffmpeg -i input.mp3 -metadata artist="Caterina Pontrandolfo Quartett" -metadata album="concerto di Fonte Avellana" -metadata genre="popular" -metadata composer="tradizione lucana" output.mp3 . Now, the output.mp3 is only 83M. Why this? Does it mean a quality loss? How can I avoid it? How can I have output.mp3 the same size as input.mp3? Adding `-qscale 0' does not help. Thanks for any help, Rodolfo

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interlaced video
by Joshi, Ashwini M 30 May '17

30 May '17

Hi, Could you please let me know the command to be used to get the picture_structure as field using the ffmpeg? I saw many commands & usage but all are createing interlaced video, but the encoding is not happening with picture_structure as field. Please help me on this. Thanks. ________________________________ Please be advised that this email may contain confidential information. If you are not the intended recipient, please notify us by email by replying to the sender and delete this message. The sender disclaims that the content of this email constitutes an offer to enter into, or the acceptance of, any agreement; provided that the foregoing does not invalidate the binding effect of any digital or other electronic reproduction of a manual signature that is included in any attachment.

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