January 2018 - ffmpeg-user - lists.ffmpeg.org

Strange results with -preset
by Cecil Westerhof 30 Jan '18

30 Jan '18

I need to upload a set of videos, but before I do that I need to shrink them. But I get a very strange result. Of-course the preset veryfast is faster as faster, fast, medium, slow, slower and veryslow. But the strange thing is that it also create a smaller version of the file. My original file is 733.8 MB. It is a mts file I also convert to mp4. The results: |-----------+-------+------| | preset | size | time | |-----------+-------+------| | veryslow | 83.8 | 52 | | slower | 94.6 | 27 | | slow | 100.2 | 15 | | medium | 109.0 | 13 | | fast | 117.7 | 9 | | faster | 106.2 | 7 | | veryfast | 83.2 | 5 | | superfast | 177.4 | 4 | | ultrafast | 328.5 | 3 | |-----------+-------+------| So it looks like I should always use superfast. (But probably a good idea to test it with a few other videos.) Anyone an idea what could be happening here? The conversion is done with: nice -n 20 ionice -t -c3 \ ffmpeg -y \ -i 000006.mts \ -vcodec libx264 \ -crf 26 \ -acodec copy \ -preset ultrafast \ -vf drawtext="box=1:\ boxborderw=6:\ boxcolor=black@0.2:\ fontcolor=white@0.4:\ fontfile=FreeSerifBold.ttf:\ fontsize=32: text='${_copyright}':\ x=((main_w- text_w) / 2):\ y=main_h - (text_h * 2)" \ 000006Shrinked.mp4 The log starts with: ffmpeg version 3.2.10-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-18) 20170516 configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 Input #0, mpegts, from '00006.MTS': Duration: 00:07:58.40, start: 1.040000, bitrate: 12271 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p(top first), 1440x1080 [SAR 4:3 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 256 kb/s Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090), 1920x1080 [Parsed_drawtext_0 @ 0x557648e381e0] Using "/usr/share/fonts/truetype/dejavu/DejaVuSans.ttf" [libx264 @ 0x557648e2bf80] using SAR=4/3 [libx264 @ 0x557648e2bf80] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX XOP FMA4 FMA3 LZCNT BMI1 [libx264 @ 0x557648e2bf80] profile Constrained Baseline, level 4.0 [libx264 @ 0x557648e2bf80] 264 - core 148 r2748 97eaef2 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=26.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0 [mp4 @ 0x557648de4080] track 1: codec frame size is not set Output #0, mp4, to '00006Shrinked.mp4': Metadata: encoder : Lavf57.56.101 Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], q=-1--1, 25 fps, 12800 tbn, 25 tbc Metadata: encoder : Lavc57.64.101 libx264 Side data: cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 Stream #0:1: Audio: ac3 ([165][0][0][0] / 0x00A5), 48000 Hz, stereo, 256 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof

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Error converting .m4a to .mp3
by Bernhard Döbler 30 Jan '18

30 Jan '18

Hi, I want to convert an mp4 file from iTunes to MP3 using ffmpeg. For on certain file I get error: ~~~ Program: c:\Program Files\ffmpeg-20180121-78e884f-win64-shared\bin\ffmpeg.exe File: ../../src/lame-3.100/libmp3lame/vbrquantize.c, Line 783 Expression: vbrsf[sfb] >= vbrsfmin[sfb] This application has requested the Runtime to terminate it in an unusual way. Please contact the application's support team for more information. ~~~ I understand the exceptioon is thrown my lame; not ffmpeg. I asked this question on the lame page over at sourceforge. They told me to convert the .m4a file to .wav and then use LAME.exe to generate the .mp3 file and see if this errors comes up. I tried that and it did not come up. Can you give me a hint what might be the cause of the error? Best, Bernhard Döbler

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Need help Axis encoder
by Cushman, Aaron 29 Jan '18

29 Jan '18

I have Axis q7406 axis video encoders that send h.264. I am trying to get them to work with Blonder Tongue and they will not as axis does not send H.264 over RTP (RFC 3984). I am trying to see if there is a way to use FFMPEG to take the stream from the axis and resend UDP/RTP do you know if that would be possible.

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Avid opAtom trouble
by Bouke (VideoToolShed) 29 Jan '18

29 Jan '18

Hi guys, Toying around with Avid opAtom mxf, generating test clips. When I look at the clip in Avid, the bin info shows a raster dimension of 1920x1088, pixel aspect ratio of 1.007 (expected 1920x1080 / 1.000) And of course, when editing the clip into a seqence, frameflex kicks in. same line but output to Mov and AMA the clip in does give the expected result. Am I missing something or is this a bug? Next, what kind of metadata besides the clip name / timecode is supported? thx, Bouke ffmpeg -f lavfi -i "testsrc=size=1920x1080:rate=25" -frames:v 10 -f mxf_opatom -vcodec dnxhd -pix_fmt yuv422p -vb 120M -timecode 01:02:03:04 -metadata material_package_name="rastertest" -flags +ildct -top 1 -y "M:\Avid Mediafiles\MXF\2\test.mxf" ffmpeg version N-85653-gb4330a0 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 6.3.0 (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. 61.100 / 55. 61.100 libavcodec 57. 93.100 / 57. 93.100 libavformat 57. 72.101 / 57. 72.101 libavdevice 57. 7.100 / 57. 7.100 libavfilter 6. 87.100 / 6. 87.100 libswscale 4. 7.101 / 4. 7.101 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 Input #0, lavfi, from 'testsrc=size=1920x1080:rate=25': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 1920x1080 [SAR 1:1 DAR 16:9], 25 tbr, 25 tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo (native) -> dnxhd (native)) Press [q] to stop, [?] for help Output #0, mxf_opatom, to 'M:\Avid MediaFiles\MXF\2\tst1.mxf': Metadata: timecode : 01:02:03:04 material_package_name: raster test encoder : Lavf57.72.101 Stream #0:0: Video: dnxhd (DNXHD), yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-1024, 120000 kb/s, 25 fps, 25 tbn, 25 tbc Metadata: encoder : Lavc57.93.100 dnxhd frame= 10 fps=0.0 q=1.0 Lsize= 5927kB time=00:00:00.40 bitrate=121375.9kbits/s speed=2.38x video:5920kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.110738% --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

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HLS: multiple retries on HTTP 404 errors
by den.bkh＠yandex.ru 29 Jan '18

29 Jan '18

Hello I copy live HLS stream with the command like this: ffmpeg -probesize 10000000 -y -i "https://rfe-lh.akamaihd.net/i/rfe_tvmc5@383630/index_0720_av-p.m3u8?sd=10&r…" \ -t 7200 -map 0:0 -map 0:1 -c copy o.mkv It works most of the time, but sometimes some requests for new HLS segments result in "HTTP error 404 Not Found" errors: ... [hls,applehttp @ 0x55e993a7ef80] Opening 'https://rfe-lh.akamaihd.net/i/rfe_tvmc5@383630/segment151719160_0540_av-p.t…' for reading [hls,applehttp @ 0x55e993a7ef80] Opening 'https://rfe-lh.akamaihd.net/i/rfe_tvmc5@383630/segment151719161_0540_av-p.t…' for reading [https @ 0x55e9942e2420] HTTP error 404 Not Found [hls,applehttp @ 0x55e993a7ef80] Failed to open segment of playlist 0 [hls,applehttp @ 0x55e993a7ef80] Opening 'https://rfe-lh.akamaihd.net/i/rfe_tvmc5@383630/segment151719162_0540_av-p.t…' for reading frame=101178 fps= 25 q=-1.0 size= 1048779kB time=01:07:39.98 bitrate=2116.2kbits/s speed=1.01x ... >From this log we can see that the request for the segment with name "segment151719161_0540_av-p.ts" results in HTTP 404 error. It looks like a bug of the streaming service, but I also think that this segment could be available if we make another request(maybe after some timeout). Is it possible to configure ffmpeg to do multiple retries on HTTP errors for buggy/missed/not-yet-created HLS segments?

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FFmpeg dumps audio raw data of more size than expected
by m.kamalasubha m.kamalasubha 29 Jan '18

29 Jan '18

Hello all, I need to generate audio raw buffer for some 30 seconds from a clip whose sample rate is 44100 Hz. I used the following FFmpeg command to dump the raw data. ffmpeg -i <input> -ss 00:00:00 -t 00:00:30 -f s16le -acodec pcm_s16le -ac 1 -ar 44100 audi.raw Expected audio raw file size : 44100 * 2 * 30 = 2646000 Actual audio raw file size : 2646176 I am wondering from where that extra 176 bytes arrive for 30 secs samples with sample rate 44100 Hz and sample size 2 bytes respectively. Kindly help on those lines. Thanks, Shalini

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FFmpeg doing slow encoding
by Sai Krishna Kothapalli 29 Jan '18

29 Jan '18

Hello all, My set up includes a input live RTMP stream, and apply a custom video filter and output RTMP stream which is converted to HLS and served to clients. Now my issue is that FFmpeg is taking a lot of time to do this process (like it is encoding at 0.3 - 0.4X speed). If I give a mp4 file as input and output a mp4 file for testing purposes it is taking nearly 1:30 sec for a 25 sec 4K video clip. The problem is caused when by the time the client finished watching that 25 sec .ts file and by that time the next .ts file is produced the video player just buffers. The video player buffers for 1:05 sec to play the next chunk and this continues. How should I overcome this problem? i am using 8 core aws server for now. Should I use a GPU instance and compile FFmpeg with GPU compatibility or can I solve this issue by using more number of CPUs. I want it to encode at 1x times so that it doesn't buffer on the client video player. Thank you. -- K. Mohan Sai Krishna Roll No: 130101039 Computer Science and Engineering IIT Guwahati India

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How do I get the presets I can use
by Cecil Westerhof 28 Jan '18

28 Jan '18

I want to play a bit with -preset, but how do I get the correct list with parameters? At one place I found: - veryslow - slow - normal - fast - faster - veryfast - ultrafast But at another I also found: - superfast - medium - slower - placebo but was missing normal. I tried ffmpeg -h, but this only gives: -pre preset preset name I tried: ffmpeg -h -pre and: ffmpeg -h pre but that just gives the same as: ffmpeg -h Version information: ffmpeg version 3.2.10-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-18) 20170516 configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof

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error malloc of size ... failed
by JONI JONI 28 Jan '18

28 Jan '18

when I save rtsp strem to 264 file i get error malloc of size ... failed, why? My ffmpeg commnd is : ffmpeg -i rtsp://<rtsp stream> test.264 this error accur too when I want to save mjpeg stream to 264 file My ffmpeg commnd is : ffmpeg -i http://<mjpeg stream> test.264

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Sample types 's32 (24 bit)' and 's16 (24 bit)' are confusing to me
by Zak 28 Jan '18

28 Jan '18

Dear FFmpeg Community, I have seen two sample types (similar to the output of -sample_fmts but maybe different) that confuse me because they are unclear about the bit depth of the samples involved. The two strange sample types are: s32 (24 bit) s16 (24 bit) Those are verbatim strings. I have seen both of them, I am copy-pasting them from my log files. I can see these either on calls to ffprobe or on calls to ffmpeg. Here is an ffprobe call that always reproduces the behavior: bash> ffprobe "01 - Let It Go.flac" ffprobe version 3.0.2 Copyright (c) 2007-2016 the FFmpeg developers built with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --enable-libvidstab --enable-libx265 --disable-doc --arch=x86_64 --enable-runtime-cpudetect libavutil 55. 17.103 / 55. 17.103 libavcodec 57. 24.102 / 57. 24.102 libavformat 57. 25.100 / 57. 25.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 31.100 / 6. 31.100 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100 Input #0, flac, from '01 - Let It Go.flac': Metadata: ALBUM : Megahits 2013 ARTIST : Idina Menzel COMMENT : Vinyl rip by Radioactive Squirrel DATE : 2013 TITLE : Let It Go TRACKTOTAL : 18 track : 1 Duration: 00:03:48.29, start: 0.000000, bitrate: 2701 kb/s Stream #0:0: Audio: flac, 96000 Hz, stereo, s32 (24 bit) There it is, at the bottom of the ffprobe output, "s32 (24 bit)". Here is how you reproduce it from ffmpeg, but this is not actually copy-pasted: bash> ffmpeg -i "01 - Let It Go.flac" output.wav # In the first chunk of printed output, about the input FLAC: Stream #0:0: Audio: flac, 96000 Hz, stereo, s32 (24 bit) # In the second chunk of printed output, about the WAV it is generating: Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, stereo, s16 (24 bit), 3072 kb/s I am fairly certain the input FLAC is 24-bit (i.e. 24 bits per sample), and I am 99.9% certain the output WAV is 16-bit. Why does the FLAC say "s32", which refers to a signed 32-bit integer? Why does the WAV say "(24 bit)", which is simply not true? Notice that I didn't actually supply a -sample_fmt argument to the ffmpeg call above. By default it chooses pcm_s16le and "s16 (24 bit)". I am not sure what happens if I call ffmpeg -i input.flac -sample_fmt s16 output.wav but I could test that. Thank you for any words of wisdom, Zak

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