March 2017 - ffmpeg-user - lists.ffmpeg.org

h264_mp4toannexb unable to split h264_qsv-encoded files?
by Hugo Garbez 03 Mar '17

03 Mar '17

Hello, I'm using FFmpeg 3.2.4 on Debian 8.7, and I have some problems with h264_qsv and HLS. I'm splitting my files for HLS with the following command: $ ffmpeg -i example.mp4 -codec copy -map 0 -bsf:v h264_mp4toannexb -hls_time 5 -hls_init_time 1 -hls_list_size 0 example.m3u8 When example.mp4 is created with libx264, there's no problem with splitting files with HLS: $ ffmpeg -i original.mp4 -codec:v libx264 example.mp4 $ [previous HLS command] $ cat example.m3u8 #EXTM3U #EXT-X-VERSION:3 #EXT-X-TARGETDURATION:5 #EXT-X-MEDIA-SEQUENCE:0 #EXTINF:2.200000, example-0.ts #EXTINF:2.480000, example-1.ts #EXTINF:4.800000, example-2.ts #EXTINF:4.440000, example-3.ts With libx264, -hls_time 5 works (EXT-X-TARGETDURATION and EXTINF look good). But that's not the case if example.mp4 was encoded with h264_qsv instead of libx264: $ ffmpeg -i original.mp4 -codec:v h264_qsv example.mp4 $ [previous HLS command] $ cat example.m3u8 #EXTM3U #EXT-X-VERSION:3 #EXT-X-TARGETDURATION:10 #EXT-X-MEDIA-SEQUENCE:0 #EXTINF:10.000000, example-0.ts #EXTINF:10.000000, example-1.ts #EXTINF:10.000000, example-2.ts #EXTINF:10.000000, example-3.ts #EXTINF:9.640000, As you can see, it looks like "-hls_time 5" has no effect when the video is encoded with h264_qsv instead of libx264. Is there anything I can do about this? Thank you very much.

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Re: [FFmpeg-user] Chroma problem converting mp4 to prores
by David Cecco 03 Mar '17

03 Mar '17

Sorry about the slow response, that has fixed the issue, thanks chronek! The file still seems a bit soft in quality though. I have tried turning the -qscale right down to 1 and adjusting the bits_per_mb up to 2400 and the picture improved but it still wasn’t 100%, but the file grew to 2.5x the size. I tried with the prores codec as opposed to the prores_ks codec and it was better, but again not 100% interestingly the file size didn’t grow when using the bits_per_mb with prores. Here is an example output: $ ffmpeg -y -i "5082197_Soccer Enrique Upd.mp4" -acodec pcm_s16le -vcodec prores_ks -profile:v 1 -vf scale=interl=1 -sws_flags full_chroma_inp+full_chroma_int -copytb -1 "5082197_Soccer Enrique Upd.mov" ffmpeg version 3.2.2-tessus Copyright (c) 2000-2016 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-version3 --disable-ffplay --disable-indev=qtkit --disable-indev=x11grab_xcb libavutil 55. 34.100 / 55. 34.100 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.100 / 57. 56.100 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '5082197_Soccer Enrique Upd.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 creation_time : 2017-03-01T23:52:21.000000Z Duration: 00:02:15.83, start: 0.000000, bitrate: 10170 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], 9915 kb/s, 25 fps, 25 tbr, 50k tbn, 50 tbc (default) Metadata: creation_time : 2017-03-01T23:52:22.000000Z handler_name : Alias Data Handler encoder : AVC Coding Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 253 kb/s (default) Metadata: creation_time : 2017-03-01T23:52:22.000000Z handler_name : Alias Data Handler Output #0, mov, to '5082197_Soccer Enrique Upd.mov': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42mp41 encoder : Lavf57.56.100 Stream #0:0(eng): Video: prores (prores_ks) (apcs / 0x73637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 12800 tbn, 25 tbc (default) Metadata: creation_time : 2017-03-01T23:52:22.000000Z handler_name : Alias Data Handler encoder : Lavc57.64.101 prores_ks Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s (default) Metadata: creation_time : 2017-03-01T23:52:22.000000Z handler_name : Alias Data Handler encoder : Lavc57.64.101 pcm_s16le Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> prores (prores_ks)) Stream #0:1 -> #0:1 (aac (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help frame= 3394 fps=7.7 q=-0.0 Lsize= 1522742kB time=00:02:15.80 bitrate=91852.6kbits/s speed=0.31x video:1497227kB audio:25464kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.003344% Thanks! dave > Try -sws_flags full_chroma_inp+full_chroma_int W dniu 2017-02-23 o 23:05, Dave pisze: > Hi guys, > > > > I am having a issue where all the clips that I convert have a chroma > > problem. The red is saturated on all clips and there are red horizontal > > lines in the chroma. > > > > $ ffmpeg -y -i 5079162_NHL\ Marleau.mp4 -acodec pcm_s16le -vcodec prores_ks > > -profile:v 1 -copytb -1 -vf yadif=1 -qscale:v 10 mp4-yadif1-q10.mov > > I thought this might be the chroma bug but I have experimented with a > > couple of different options replacing yadif=1 with the following: > > > > -vf scale=out_color_matrix=bt709 > > -pix_fmt yuv422p10le > > -vf colormatrix=bt601:bt709 > > -vf yadif > > > > The yadif=1 option provides the closest results and specifying multiple > > options seems to make it worse not better. The problem is more obvious on > > a broadcast monitor. > > > > We are also having a problem where the converted clips are 1 frame shorter > > than the original, missing a black frame at the start of the clip. > > > > Here are the details of the clip I am trying to convert: > > > > $ ffprobe -v error -show_format -show_streams 5079162_NHL\ Marleau.mp4 > > [STREAM] > > index=0 > > codec_name=h264 > > codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 > > profile=Main > > codec_type=video > > codec_time_base=1/50 > > codec_tag_string=avc1 > > codec_tag=0x31637661 > > width=1920 > > height=1080 > > coded_width=1920 > > coded_height=1080 > > has_b_frames=1 > > sample_aspect_ratio=1:1 > > display_aspect_ratio=16:9 > > pix_fmt=yuv420p > > level=41 > > color_range=tv > > color_space=unknown > > color_transfer=unknown > > color_primaries=unknown > > chroma_location=left > > field_order=unknown > > timecode=N/A > > refs=1 > > is_avc=true > > nal_length_size=4 > > id=N/A > > r_frame_rate=25/1 > > avg_frame_rate=25/1 > > time_base=1/50000 > > start_pts=0 > > start_time=0.000000 > > duration_ts=2354000 > > duration=47.080000 > > bit_rate=9081205 > > max_bit_rate=N/A > > bits_per_raw_sample=8 > > nb_frames=1177 > > nb_read_frames=N/A > > nb_read_packets=N/A > > DISPOSITION:default=1 > > DISPOSITION:dub=0 > > DISPOSITION:original=0 > > DISPOSITION:comment=0 > > DISPOSITION:lyrics=0 > > DISPOSITION:karaoke=0 > > DISPOSITION:forced=0 > > DISPOSITION:hearing_impaired=0 > > DISPOSITION:visual_impaired=0 > > DISPOSITION:clean_effects=0 > > DISPOSITION:attached_pic=0 > > DISPOSITION:timed_thumbnails=0 > > TAG:creation_time=2017-02-03T23:55:00.000000Z > > TAG:language=eng > > TAG:handler_name=Alias Data Handler > > TAG:encoder=AVC Coding > > [/STREAM] > > [STREAM] > > index=1 > > codec_name=aac > > codec_long_name=AAC (Advanced Audio Coding) > > profile=LC > > codec_type=audio > > codec_time_base=1/48000 > > codec_tag_string=mp4a > > codec_tag=0x6134706d > > sample_fmt=fltp > > sample_rate=48000 > > channels=2 > > channel_layout=stereo > > bits_per_sample=0 > > id=N/A > > r_frame_rate=0/0 > > avg_frame_rate=0/0 > > time_base=1/48000 > > start_pts=0 > > start_time=0.000000 > > duration_ts=2259840 > > duration=47.080000 > > bit_rate=253262 > > max_bit_rate=283875 > > bits_per_raw_sample=N/A > > nb_frames=2210 > > nb_read_frames=N/A > > nb_read_packets=N/A > > DISPOSITION:default=1 > > DISPOSITION:dub=0 > > DISPOSITION:original=0 > > DISPOSITION:comment=0 > > DISPOSITION:lyrics=0 > > DISPOSITION:karaoke=0 > > DISPOSITION:forced=0 > > DISPOSITION:hearing_impaired=0 > > DISPOSITION:visual_impaired=0 > > DISPOSITION:clean_effects=0 > > DISPOSITION:attached_pic=0 > > DISPOSITION:timed_thumbnails=0 > > TAG:creation_time=2017-02-03T23:55:01.000000Z > > TAG:language=eng > > TAG:handler_name=Alias Data Handler > > [/STREAM] > > [FORMAT] > > filename=5079162_NHL Marleau.mp4 > > nb_streams=2 > > nb_programs=0 > > format_name=mov,mp4,m4a,3gp,3g2,mj2 > > format_long_name=QuickTime / MOV > > start_time=0.000000 > > duration=47.146600 > > size=54956371 > > bit_rate=9325189 > > probe_score=100 > > TAG:major_brand=mp42 > > TAG:minor_version=0 > > TAG:compatible_brands=mp42mp41 > > TAG:creation_time=2017-02-03T23:54:57.000000Z > > [/FORMAT] > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > To unsubscribe, visit link above, or email > > ffmpeg-user-request at ffmpeg.org with subject "unsubscribe".

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Converting raw h264 file to MP4
by Varad Gunjal 03 Mar '17

03 Mar '17

Hello. I am posting this question here as this was a suggestion in the output message that I got while using ffmpeg. I have a file that has recorded an RTP stream in raw h264 format. Unfortunately, the data was stored in an unknown format (mjr, designed for the Janus WebRTC gateway). Can someone educate me as to if and how how this raw data can be converted to an mp4 file? Thanks a lot for your time. -- Best Regards, Varad Gunjal

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filter: subtitles and special chars
by Mike F 03 Mar '17

03 Mar '17

Hello, Due to the fact that I cannot get "-i my.srt" and "sub_charenc" to work, even after ( I guess ) satisfying 'iconv' requirements during the compile of N-83701-g70ebc05-1 20170205, I found the "subtitles" filter and 'charenc=CP1250' perfectly doing the job. The only small issues is that the subtitle filename.srt can not use any characters that need to be escaped, such as '[' ']' etc. When issuing this option: -filter:v scale=960:-1,subtitles=myvidfile\[1of3\].srt:charenc=CP1250:force_style='Fontsize=24:original_size=hd480' the "myvidfile\[1of10\].srt" name causes ffmpeg to error out with: [Parsed_subtitles_1 @ 0x5635ef66d280] Unable to open myvidfile [AVFilterGraph @ 0x5635ef667560] Error initializing filter 'subtitles' with args 'myvidfile' wrapping the myvidfile[1of10].srt in double or single quotes does not help... Thank you! Mike

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Re: [FFmpeg-user] filter: subtitles and special chars
by Mike F 02 Mar '17

02 Mar '17

Hello Nicolas, Let me start with this first, to clarify. < Hello. You just hijacked a thread. It is forbidden on this list. I suggest you re-read the posting guidelines carefully in order to not do anything forbidden again. > I am aware of the rules... and none of: "thread hijacking (that is, replying to a thread and changing the subject line to something completely unrelated that was not being discussed within the original thread)" has been done by me - a simple new "compose" and I pasted in the email address that i got in the initial email, after receiving the confirmation of the mailing list subscription. I am not sure how did you, or the system process that message, but "hijacking" was not done on my side... if there is a "real" problem with the msg showing up as "hijacked", i might think about dumping yahoo mail, or there is an issue on the opposite side. maybe I just subscribed to ffmpeg list, but I am a returning "ffmpeg user questions" mailing list subscriber - I was one between 2012-2014 before. going back to what you mentioned about being precise about the cmdline, and errors... I could provide the information in a "proper" form, I will try this now: ===================== $ ffmpeg -y -i myvid\[1of10\].mkv -c:v libx264 -coder 1 -bf 3 -refs 3 -subq 10 -partitions +parti4x4+parti8x8+partp8x8 -g 100 -keyint_min 25 -qmin 1 -qmax 51 -trellis 2 -sc_threshold 40 -threads 48 -filter:v scale=960:-1,subtitles=myvid\[1of10\].srt:charenc=CP1250:force_style='Fontsize=24:original_size=hd480' -c:a aac -ab 96k -ac 2 test.mp4 ffmpeg version N-83701-g70ebc05-1 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-6) 20170205 <obvious, snip> creation_time : 2017-03-02T16:34:40.000000Z handler_name : GPAC MPEG-4 Scene Description Handler [Parsed_subtitles_1 @ 0x564d1346b020] Shaper: FriBidi 0.19.7 (SIMPLE) HarfBuzz-ng 1.4.2 (COMPLEX) [Parsed_subtitles_1 @ 0x564d1346b020] Unable to open myvid [AVFilterGraph @ 0x564d132e5e40] Error initializing filter 'subtitles' with args 'myvid' Error opening filters! $ ====================== I would apologize, if I would hijack the thread... Thank you, Mike

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How to specify Audio bit depth for test sources?
by Basin Ilya 02 Mar '17

02 Mar '17

Hi. Different test sources have different bit depth: ffplay -f lavfi aevalsrc="0" # Audio: pcm_f64le ffplay -f lavfi anullsrc # Audio: pcm_u8 If I want to do some PCM math in the audio filter, I'll need to adapt the filter for each different input bit depth. There's no filter to change the depth. "aresample" only changes the sample rate. -acodec option won't do here, because it takes effect AFTER filtering and some tools don't have it (ffplay). -isf: I was unable to make it work The only workaround is to use /dev/zero as the source. Then you can specify the needed input format.

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ogg encoder always adds 128-255 extra samples
by Basin Ilya 02 Mar '17

02 Mar '17

Hi. I'm trying to create a set of chainable ogg files out of a long wav file. However, the produced ogg files are always 128-255 samples longer than the source PCM data and the extra data looks like anti-click or fade-out. See the attached image. test0 was made directly. test1 was made via an intermediate ogg file. I tried to choose a duration whose sample count is divisible by 64 or 128, but that didn't help. # produce 10 seconds of noise ffmpeg -y -t 10 -f lavfi -i aevalsrc="random(0)*2-1" -c:a pcm_s16le test.wav ( </dev/null ffmpeg -y -ss 0 -t 0.995567 -i test.wav -af "ashowinfo" test0.wav </dev/null ffmpeg -y -ss 0 -t 0.995567 -i test.wav -af "ashowinfo" test1.ogg # pts:43008 nb_samples:897 </dev/null ffmpeg -y -i "test1.ogg" -af "ashowinfo" test1.wav # pts:44032 nb_samples:128 ) _ The only workaround I see is to make a slightly longer ogg, then use "-acodec copy" and "-t" to discard 1 or 2 unneeded ogg frames.

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Video and Audio is off-sync on Decklink SDI output
by Tung Tran 02 Mar '17

02 Mar '17

Dear all I am testing to setup a decoder with ffmpeg, take live stream in (RTMP/HLS,.) and output to Decklink SDI. Its working but I have the issue of audio and video is off-sync on decklink SDI output, I tried to play around with vf setpts and af asetpts but couldnt make video/audio to sync. Any input will be very appreciated. Here is my command so far * ffmpeg -i rtmp://something -vf "setpts=PTS-STARTPTS" -af "asetpts=PTS-STARTPTS" -f decklink -pix_fmt uyvy422 "DeckLink SDI" * Video and audio of input rtmp is synced on VLC or other software. --- Best regards, *Tung Tran*

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libass : ERROR NOT in Path in ffmpeg compile
by derekg 02 Mar '17

02 Mar '17

Dear Help , Hi , I am trying to set up a fully working version of ffmpeg on my Linux Mint Qiana 17 OS , using thecomprehensive guide https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu . Please find attached a config.log complaining about a libass package not in the path . Also please see a second attached file pkg-configversion.txt showing releases data . I am an English old-age-pensioner and using Mint's Synaptic Package Manager I had Completely Removed ffmpeg version 7:2.4.3-1ubuntu1~trusty before following the guide . Any suggestions will be gratefully received and requests for more data will be acted upon . Thank you , Derek Hopley

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minimal nb_sambles in ogg frame
by Basin Ilya 02 Mar '17

02 Mar '17

Hi. I'm splitting a .wav file into a set of chainable .ogg files all having 1 second duration. I'm using input flags: "-ss n -t 1" When I concatenate these ogg files, the resulting clip is slightly longer than the original. Is it because the minimal nb_sambles in ogg frame is 128 and a set of ogg frames cannot contain exactly 44100 samples? </dev/null ffmpeg -y -ss 0 -t 1 -i test.wav -af "ashowinfo" test1.ogg # pts:44032 nb_samples:68 </dev/null ffmpeg -y -ss 1 -t 1 -i test.wav -af "ashowinfo" test2.ogg # pts:44032 nb_samples:68 </dev/null ffmpeg -y -i "concat:test1.ogg|test2.ogg" -af "ashowinfo" test5.wav # pts:88132 nb_samples:128 88132 + 128 != 88200 ffprobe test5.wav # Duration: 00:00:02.01

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