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• Converting from mp4 to Xvid avi using avconv?

  - by Ricardo Gladwell

  I normally use avidemux to convert mp4s to Xvid AVI for my Philips Streamium SLM5500. Normally I select MPEG-4 ASP (Xvid) at Two Pass with an average bitrate f 1500kb/s for video and AC3 (lav) audio and it converts correctly. However, I'm trying to using avconv so I can automate the process with a script, but when I do this the video stutters and stops playing part way through. I have a suspicion its something to do with a faulty audio conversion. The commands I'm using are as follows: avconv -y -i video.mp4 -pass 1 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi /dev/null avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi There is a bewildering array of arguments for avconv. Is there something I'm doing wrong? Is there a way I can script avidemux from a headless server? Please see command line output: $ avconv -y -i video.mp4 -pass 1 -vtag xvid -an -b:v 1500k -f avi /dev/null avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x7f4c40] w:720 h:404 pixfmt:yuv420p Output #0, avi, to '/dev/null': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 1, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Press ctrl-c to stop encoding frame=66227 fps=328 q=2.0 Lsize= 0kB time=2649.16 bitrate= 0.0kbits/s video:401602kB audio:0kB global headers:0kB muxing overhead -100.000000% $ avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x12b4f00] w:720 h:404 pixfmt:yuv420p Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' [mpeg4 @ 0x12b3ec0] [lavc rc] Using all of requested bitrate is not necessary for this video with these parameters. Output #0, avi, to 'video.avi': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 2, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, flt, 128 kb/s Metadata: creation_time : 2013-02-04 13:53:42 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (ac3 -> ac3) Press ctrl-c to stop encoding Input stream #0:1 frame changed from rate:44100 fmt:s16 ch:2 to rate:44100 fmt:flt ch:2 frame=66227 fps=284 q=2.2 Lsize= 458486kB time=2649.13 bitrate=1417.8kbits/s video:413716kB audio:41393kB global headers:0kB muxing overhead 0.741969%

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• How to cut audio file with avconv?

  - by x-yuri

  I have a hard time trying to figure out how to cut a file with avconv. Here's the command I use: avconv -ss 52:13:49 -t 01:13:52 -i RR119Accessibility.wav RR119Accessibility-2.wav But it doesn't work. I get the whole file as a result. Well, almost the whole file. Somehow the resulting file has duration 1:16:31 instead of 1:17:23. Also I believe I executed this command in every possible way: with -ss and -t after -i, with -t specifying ending point, with mp3 files, with specifying audio codec, with ffmpeg. Am I doing it wrong?

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• Howto overlay subtitles onto video using avconv?

  - by hatterman

  I am using the following script to overlay a dvdsub stream and a video stream; avconv -i input.mkv -i subtitles.sub -filter_complex "[0:v:0][1:s:0]overlay[vid]" -map "[vid]" -s hd720 -sws_flags lanczos -vcodec libx264 -crf 20 -preset medium ~/Desktop/test.mkv I get the following error ; Stream specifier ':s:0' in filtergraph description [0:v:0][1:s:0]overlay[vid] matches no streams. No matter what I do I just cannot get this to work. Can any one tell me how to 'burn-in' subtitles using avconv (or ffmpeg) ?

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• Grab sound of a SDL game with ffmpeg/avconv

  - by Peregring-lk

  I'm trying to make a screencast of a SDL game which I developed some years ago, with the following command: sleep 5 && avconv -f x11grab -s 1366x768 -r 25 -i :0.0 -same_quant screen_cast.mkv (in this 5 seconds of sleep, I open the game). But the generated video (screen_cast.mkv) doesn't capture audio. I use for my game the SDL_Mixer library, with default configuration (22050 for frequency, AUDIO_S16SYS for format, and 2 channels). What's the problem? (with options -f alsa -i pulse it doesn't work either).

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• Webm videos converted with avconv not working properly

  - by user285943

  I'm converting mp4 videos to webm with the following command: avconv -i INPUT.mp4 -c:v libvpx -qmin 10 -qmax 42 -maxrate 500k -bufsize 1000k -threads 2 -c:a libvorbis OUTPUT.webm Am I doing something wrong? Should I use all of these parameters? I just want to make the videos compatible with all the browsers/devices. The videos are not working on Firefox on most devices and on Chrome in a specific Android device that I have. EDIT: When i play the embeded videos on firefox, on any android device, it goes straight to the end and doesn't play. It only plays if i click in the progress bar. If i open the videos on firefox without embedding them in a html page, they seem to have a bigger width box outside then, while the video is inside it. I uploaded a video that i converted. Try it in firefox: https://mega.co.nz/#!cg9FVZJD!CpC6X3NXH8rAVdBiS9ZqqwyITdyPb_qkKAKsvQXRpzI This is a comparison between the videos, after and before the conversion: ![Image here]: http://i.imgur.com/V2FijCQ.jpg

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• Streaming desktop with avconv - severe sound issues

  - by Tommy Brunn

  I'm trying to do some live streaming in Ubuntu 12.10, but I'm having some problems with audio. More specifically, the quality is complete garbage and it's at least 10 seconds out of sync with the video. I'm using an excellent guide found here to set up my loopback devices so that I can combine the desktop audio with the microphone input. It seems to work, as I'm able to stream both audio and video to Twitch.tv. But, as I said, the audio quality is terrible. The microphone audio is very, very low, but if I increase it, I get a horrible garbled sound that is absolutely unbearable. Nothing like that is present during VoIP calls or when recording sound alone with the sound recorder, so it's not an issue with the microphone itself. The entire audio stream is also delayed about 10-15 seconds compared to the video stream. I put together an imgur album of my settings. Here is some example output from when I'm streaming: avconv version 0.8.4-6:0.8.4-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:11 with gcc 4.7.2 [x11grab @ 0x162fd80] device: :0.0+570,262 -> display: :0.0 x: 570 y: 262 width: 1280 height: 720 [x11grab @ 0x162fd80] shared memory extension found [x11grab @ 0x162fd80] Estimating duration from bitrate, this may be inaccurate Input #0, x11grab, from ':0.0+570,262': Duration: N/A, start: 1353181686.735113, bitrate: 884736 kb/s Stream #0.0: Video: rawvideo, bgra, 1280x720, 884736 kb/s, 30 tbr, 1000k tbn, 30 tbc [alsa @ 0x163fce0] capture with some ALSA plugins, especially dsnoop, may hang. [alsa @ 0x163fce0] Estimating duration from bitrate, this may be inaccurate Input #1, alsa, from 'pulse': Duration: N/A, start: 1353181686.773841, bitrate: N/A Stream #1.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x1641ec0] w:1280 h:720 pixfmt:bgra [scale @ 0x1642480] w:1280 h:720 fmt:bgra -> w:852 h:480 fmt:yuv420p flags:0x4 [libx264 @ 0x165ae80] VBV maxrate unspecified, assuming CBR [libx264 @ 0x165ae80] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x165ae80] profile Main, level 3.1 [libx264 @ 0x165ae80] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=cbr mbtree=1 bitrate=712 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=712 vbv_bufsize=512 nal_hrd=none ip_ratio=1.25 aq=1:1.00 Output #0, flv, to 'rtmp://live.justin.tv/app/live_23011330_Pt1plSRM0z5WVNJ0QmCHvTPmpUnfC4': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: libx264, yuv420p, 852x480, q=-1--1, 712 kb/s, 1k tbn, 30 tbc Stream #0.1: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 712 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #1:0 -> #0:1 (pcm_s16le -> libmp3lame) Press ctrl-c to stop encoding frame= 17 fps= 0 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 32 fps= 31 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 40 fps= 23 q=29.0 size= 44kB time=0.03 bitrate=13786.2kbits/s dup=frame= 47 fps= 21 q=31.0 size= 93kB time=2.73 bitrate= 277.7kbits/s dup=0frame= 62 fps= 23 q=29.0 size= 160kB time=3.23 bitrate= 406.2kbits/s dup=0frame= 77 fps= 24 q=23.0 size= 209kB time=3.71 bitrate= 462.5kbits/s dup=0frame= 92 fps= 25 q=20.0 size= 267kB time=4.91 bitrate= 445.2kbits/s dup=0frame= 107 fps= 25 q=20.0 size= 318kB time=5.41 bitrate= 482.1kbits/s dup=0frame= 123 fps= 26 q=18.0 size= 368kB time=5.96 bitrate= 505.7kbits/s dup=0frame= 139 fps= 26 q=16.0 size= 419kB time=6.48 bitrate= 529.7kbits/s dup=0frame= 155 fps= 27 q=15.0 size= 473kB time=7.00 bitrate= 553.6kbits/s dup=0frame= 170 fps= 27 q=14.0 size= 525kB time=7.52 bitrate= 571.7kbits/s dup=0 frame= 180 fps= 25 q=-1.0 Lsize= 652kB time=7.97 bitrate= 670.0kbits/s dup=0 drop=32 //Here I stop the streaming video:531kB audio:112kB global headers:0kB muxing overhead 1.345945% [libx264 @ 0x165ae80] frame I:1 Avg QP:30.43 size: 39748 [libx264 @ 0x165ae80] frame P:45 Avg QP:11.37 size: 11110 [libx264 @ 0x165ae80] frame B:134 Avg QP:15.93 size: 27 [libx264 @ 0x165ae80] consecutive B-frames: 0.6% 0.0% 1.7% 97.8% [libx264 @ 0x165ae80] mb I I16..4: 7.3% 0.0% 92.7% [libx264 @ 0x165ae80] mb P I16..4: 0.1% 0.0% 0.1% P16..4: 49.1% 1.2% 2.1% 0.0% 0.0% skip:47.4% [libx264 @ 0x165ae80] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.1% 0.0% 0.0% direct: 0.0% skip:99.9% L0:42.5% L1:56.9% BI: 0.6% [libx264 @ 0x165ae80] coded y,uvDC,uvAC intra: 82.3% 87.4% 71.9% inter: 7.1% 8.4% 7.0% [libx264 @ 0x165ae80] i16 v,h,dc,p: 27% 29% 16% 28% [libx264 @ 0x165ae80] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 21% 14% 8% 8% 8% 7% 5% 7% [libx264 @ 0x165ae80] i8c dc,h,v,p: 47% 22% 20% 11% [libx264 @ 0x165ae80] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x165ae80] ref P L0: 96.4% 3.6% [libx264 @ 0x165ae80] kb/s:474.19 Received signal 2: terminating. Any ideas on how I can resolve this? The video delay is perfectly acceptable, so I wouldn't think that it's a network issue that's causing the delay in the audio. Any help would be appreciated.

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• avconv and ffmpeg - drawtext filter text_w evaluates as 0 in Ubuntu Precise

  - by Chris White

  I'm attempting to draw text onto a video using either the avconv or ffmpeg commands. When specifying x= for where on the final video to place the text, the 'text_w' value is evaluating to 0, rather than the width of the rendered text as it should. I'm using Ubuntu 12.04 I've got avconv version 0.8.3-4:0.8.3-0ubuntu0.12.04.1 and ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1 Example command: avconv -i test.mov -vf "drawtext=fontfile='/usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf':text='test text':x=text_w:y=50:fontsize=24:fontcolor=black" texted.mov This command causes the text to be printed as if x were set to 0. What I'd really like to be able to do is center the text horizontally using something like this: avconv -i test.mov -vf "drawtext=fontfile='/usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf':text='test text':x=(main_w-text_w)/2:y=50:fontsize=24:fontcolor=black" texted.mov Using ffmpeg for to attempt the same ends with the same result ffmpeg -i test.mov -vf "drawtext=fontfile='/usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf':text='test text':x=(main_w-text_w)/2:y=50:fontsize=24:fontcolor=black" texted.mov

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• Encoding multiple video streams with a single avconv invocation

  - by automatthias

  I played with avconv on Ubuntu and I'm now able to e.g. record the desktop with sound from a soundcard. One thing I wanted to do was recording two video inputs at the same time, for instance the desktop and from the webcam. I thought about doing something like this: avconv \ -f alsa \ -i default \ -acodec flac \ -f video4linux2 \ -r 6 \ -i /dev/video0 \ -f x11grab \ -i :0.0 \ out.mkv My thinking was that if you define multiple video inputs, and the .mkv format can handle multiple video streams, avconv will encode 2 video streams and 1 audio stream into one file. But this isn't what happens: avconv version 0.8.4-6:0.8.4-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:11 with gcc 4.7.2 [alsa @ 0x1091bc0] capture with some ALSA plugins, especially dsnoop, may hang. [alsa @ 0x1091bc0] Estimating duration from bitrate, this may be inaccurate Input #0, alsa, from 'default': Duration: N/A, start: 1354364317.020350, bitrate: N/A Stream #0.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s [video4linux2 @ 0x10923e0] Estimating duration from bitrate, this may be inaccurate Input #1, video4linux2, from '/dev/video0': Duration: N/A, start: 100607.724745, bitrate: 29491 kb/s Stream #1.0: Video: rawvideo, yuyv422, 640x480, 29491 kb/s, 6 tbr, 1000k tbn, 6 tbc [x11grab @ 0x107b2a0] device: :0.0+83,87 -> display: :0.0 x: 83 y: 87 width: 854 height: 480 [x11grab @ 0x107b2a0] shared memory extension found [x11grab @ 0x107b2a0] Estimating duration from bitrate, this may be inaccurate Input #2, x11grab, from ':0.0+83,87': Duration: N/A, start: 1354364318.488382, bitrate: 196761 kb/s Stream #2.0: Video: rawvideo, bgra, 854x480, 196761 kb/s, 15 tbr, 1000k tbn, 15 tbc Incompatible pixel format 'bgra' for codec 'mpeg4', auto-selecting format 'yuv420p' [buffer @ 0x107fcc0] w:854 h:480 pixfmt:bgra [avsink @ 0x10bdf00] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x10dc680] w:854 h:480 fmt:bgra -> w:854 h:480 fmt:yuv420p flags:0x4 Output #0, matroska, to '.../out.mkv': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: mpeg4, yuv420p, 854x480, q=2-31, 4000 kb/s, 1k tbn, 15 tbc Stream #0.1: Audio: libvorbis, 48000 Hz, 2 channels, s16 Stream mapping: Stream #2:0 -> #0:0 (rawvideo -> mpeg4) Stream #0:0 -> #0:1 (pcm_s16le -> libvorbis) Press ctrl-c to stop encoding [mpeg4 @ 0x10bd800] rc buffer underflow ^Cframe= 160 fps= 15 q=2.0 Lsize= 3414kB time=10.66 bitrate=2623.0kbits/s video:3273kB audio:131kB global headers:4kB muxing overhead 0.165600% Received signal 2: terminating. I'm not sure if it's the question of mapping (some -map options to add?) or that avconv just can't encode more than 1 video stream at one time. So is it an actual avconv limitation, or a limitation of the available containers, or me simply not finding the right combination of command line options?

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• AVConv increases song duration when converting MP3

  - by chauffch

  I am struggling with the following issue. I want to convert an MP3 ADTS into pure a MP3. I am using AVConv on Ubuntu 12.10. The outcome is a file that has the same size, but the duration is now longer. $ ls -l total 6436 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mpga Blindsided_Bon_Iver.mpga: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo $ avconv -i Blindsided_Bon_Iver.mpga -c copy Blindsided_Bon_Iver.mp3 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:50:25 with gcc 4.6.3 [mp3 @ 0x8c6e240] max_analyze_duration reached Input #0, mp3, from 'Blindsided_Bon_Iver.mpga': Duration: 00:05:29.29, start: 0.000000, bitrate: 160 kb/s Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 160 kb/s Output #0, mp3, to 'Blindsided_Bon_Iver.mp3': Metadata: TSSE : Lavf53.21.0 Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, 160 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press ctrl-c to stop encoding size= 6432kB time=329.30 bitrate= 160.0kbits/s video:0kB audio:6432kB global headers:0kB muxing overhead 0.002080% $ ls -l total 12868 -rw-rw-r-- 1 teuf teuf 6586129 nov. 27 22:26 Blindsided_Bon_Iver.mp3 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mp3 Blindsided_Bon_Iver.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 32 kbps, 44.1 kHz, Stereo Amarok shows the new file has a duration of 25:27 and has a lot of silence. Am I using an incorrect option? Is it a bug in AVConv? Any ideas how to fix it?

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• How to replace avconv with (the real) ffmpeg and have it work right?

  - by u2n

  Anyone know how to install the real, current ffmpeg (not the older version by avconv that comes w/ Ubuntu) without having the problems mentioned here? Reasons: Everything works (or so many have written), Use the new syntax, and After some research -- and without getting into the controversy -- I simply believe the packagers made a mistake going with avconv. This is a fix. Apparently this isn't as easy as it should be due to versioning conflicts, and that avconv and ffmpeg can't coexist on the same machine (noted in the above post). Though at least one guide mentions installing ffmpeg locally (~) to achieve coexistence. My release is Precise 12.04.3 LTS, but it appears this would apply to newer ones also.

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• Record and display Kinect audio and video via avconv

  - by ThomasH

  I'm trying to record audio and video from a Kinect using avconv but I seem to have trouble specifying the right options. avconv -f video4linux2 -video_size 640x480 -c:v h264 -c:a ac3 -i /dev/video1 test.mp4 results in: avconv: /build/buildd/libav-extra-0.8.3ubuntu0.12.04.1/libav/libavutil/mathematics.c:79: av_rescale_rnd: Assertion `c > 0' failed. Aborted (core dumped) Cheese is perfectly happy recording the audio and video from the Kinect, so it's not a setup issue. For added fun, I also need to display the video at the same time.

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• Scripting with variables from file

  - by Nooster

  I have several videos on my PC that I would like to shorten. For instance I have a 30 sec video where I want to have the section from sec 15 to 20 (a 5sec video). To cut this, I use avconv. avconv -i input.mp4 -ss 15 -acodec copy -vcodec copy -t 5 output.mp4 This command works pretty well. I have many videos I want to cut the same way. This is why I created a textfile containing the information: input-name, start of cut, length of cut, output-name. Those are written into in.txt that looks like this: input.mp4 15 5 output.mp4 input1.mp4 32 10 output1.mp4 input2.mp4 10 7 output2.mp4 ... My question is: How do I have to modify the avconv-command to cut my videos automatically? What I tried was this, but it didn't work at all: avconv -i $1 -ss $2 -acodec copy -vcodec copy -t $3 $4 < in.txt Any idea?

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• avconv - not working with movflags

  - by MarKa

  With the Parameter "-movflags frag_custom" my avconv says [mp4 muxer @ 0x8d05f60] [Eval @ 0x7fffcc763f00] Undefined constant or missing '(' in 'frag_custom' [mp4 muxer @ 0x8d05f60] Unable to parse option value "frag_custom" [mp4 muxer @ 0x8d05f60] Error setting option movflags to value frag_custom. But with avconv is compiled with a mp4 muxer, so whats the problem ? Version is avconv 0.8.1-4:0.8.1-0ubuntu1

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• Using avconv (ffmpeg) to concatenate a bunch of .bmps into a mkv/avi video

  - by user1509246

  Hoi, Trying to figure out how to get avconv to concatenate a bunch of .bmps together into a video file. Here's what I've got so far: avconv -f image2 -i Capture/%d.bmp -vcodec mpeg4 -r 24 -b:v 20M Capture.mkv While this does work, the quality is terrible - there are tons of artifacts that are visible, the colours are distorted and everything is blurred. I've trawled through the documentation for avconv and ffmpeg, but can't find anything that increases the quality. Any ideas as to how I can get the quality as close to the original bmps as possible? Thanks for lending me your brains, - Alex

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• how to convert avi (xvid) to mkv or mp4 (h264)

  - by bcsteeve

  Very noob when it comes to video. I'm trying to make sense of what I"m finding via Google... but its mostly Greek to me. I have a bunch of Avi files that won't play in my WD TV Play box. Mediainfo tells me they are xvid. Specs for the box show that should be fine... but digging through forums says its hit-and-miss. So I'd like to try converting them to h264 encoded MKV or mp4 files. I gather avconv is the tool, but reading the manual just has me really really confused. I tried the very basic example of: avconv -i file.avi -c copy file.mp4 it took less than 4 seconds. And it worked... sort of. It "played" in that something came up on the screen... but there was horrible artifacting and scenes would just sort of melt into each other. I want to preserve quality if possible. I'm not concerned about file size. I'm not terribly concerned with the time it takes either, provided I can do them in a batch. Can someone familiar with the process please give me a command with the options? Thank you for your help. I'm posting the mediainfo in case it helps: General Complete name : \\SERVER\Video\Public\test.avi Format : AVI Format/Info : Audio Video Interleave File size : 189 MiB Duration : 11mn 18s Overall bit rate : 2 335 Kbps Writing application : Lavf52.32.0 Video ID : 0 Format : MPEG-4 Visual Format profile : Advanced Simple@L5 Format settings, BVOP : 2 Format settings, QPel : No Format settings, GMC : No warppoints Format settings, Matrix : Default (H.263) Muxing mode : Packed bitstream Codec ID : XVID Codec ID/Hint : XviD Duration : 11mn 18s Bit rate : 2 129 Kbps Width : 720 pixels Height : 480 pixels Display aspect ratio : 16:9 Frame rate : 29.970 fps Standard : NTSC Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Compression mode : Lossy Bits/(Pixel*Frame) : 0.206 Stream size : 172 MiB (91%) Writing library : XviD 1.2.1 (UTC 2008-12-04) Audio ID : 1 Format : MPEG Audio Format version : Version 1 Format profile : Layer 3 Mode : Joint stereo Mode extension : MS Stereo Codec ID : 55 Codec ID/Hint : MP3 Duration : 11mn 18s Bit rate mode : Constant Bit rate : 192 Kbps Channel(s) : 2 channels Sampling rate : 48.0 KHz Compression mode : Lossy Stream size : 15.5 MiB (8%) Alignment : Aligned on interleaves Interleave, duration : 24 ms (0.72 video frame)

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• avconv gets killed if mkv has subtitles

  - by Lukas Knuth

  What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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• Incorrect durations mp4 file created by ffmpeg (avconv)

  - by Ruslan Sharipov

  Example usage: avconv -i rtmp://maps.lo.ufanet.ru/live/10e227922b473e91f37474fa084107af -vcodec copy -an -sn -map 0 -f segment -segment_format mp4 -segment_time 60 -y %05d.mp4 avconv version 0.8.3-6:0.8.3-1+b1, Copyright (c) 2000-2012 the Libav developers built on Jun 15 2012 13:54:35 with gcc 4.7.0 HandShake: client signature does not match! Metadata: height 480.00 remote_addr: sdp_session {sdp_session,0, {sdp_o,"-","1289703354974145","1289703354974145",inet4, "10.1.12.99"}, "Media Presentation", {inet4,"0.0.0.0"}, {0,0}, [{"control","*"},{"range","npt=0.0 start 30400239.52 timeshift_duration 319250.58 timeshift_size 120000.00 width 640.00 [flv @ 0x1d36a40] Estimating duration from bitrate, this may be inaccurate Input #0, flv, from 'rtmp://maps.lo.ufanet.ru/live/10e227922b473e91f37474fa084107af': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0.0: Video: h264 (Baseline), yuvj420p, 640x480 [PAR 1:1 DAR 4:3], 1k tbr, 1k tbn, 2k tbc Output #0, segment, to '%05d.mp4': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: libx264, yuvj420p, 640x480 [PAR 1:1 DAR 4:3], q=2-31, 1k tbn, 1k tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press ctrl-c to stop encoding ^Cframe= 9566 fps= 36 q=-1.0 Lsize= -0kB time=318.25 bitrate= -0.0kbits/s video:30348kB audio:0kB global headers:0kB muxing overhead -100.000071% Received signal 2: terminating. Result: serafim@yard:~/video2$ ls 00000.mp4 00001.mp4 00002.mp4 00003.mp4 00004.mp4 00005.mp4 Now try to play the files in the player, such as VLC. And that's what we get: the first fragment (00000.mp4) played well, no problems, but the second (00001.mp4 and beyond) starts the bug manifests itself, namely the file 00001.mp4 first 60 seconds black screen, but since 61 seconds starts playing the video. Attachments: https://dl.dropbox.com/u/760901/rtmp_and_mp4.zip How to get rid of the delay with black screen at the beginning of the segments? Maybe ffmpeg to pass parameters, or third-party software is able to correct the obtained segments mp4?

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• How do I capture my second monitor using avconv?

  - by Codemonkey

  With this command: avconv -f x11grab -s 2560x1440 -i :0.0 I can stream video from my main monitor. I also have a second, 1080p monitor on which I do my gaming. I want to stream from that monitor. This doesn't work: avconv -f x11grab -s 1920x1080 -i :0.1 I assume I have to use -i :0.0 and somehow specify that it should capture 1920x1080 pixels from X position 2560 and Y position 0. My gaming monitor is placed to the right of my main monitor. Unfortunately the man page for avconv is miles long, so I haven't had any luck figuring this out on my own. I have tried Using -vf with crop like this: -vcodec libx264 ... -vf "crop=$IN_WIDTH:$IN_HEIGHT:2560:360" But that only displayed 1080p video from the top left corner of my main display.

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• Tool to bulk speed up/convert an audio file

  - by User1

  I want to listen to certain podcasts on my phone but I have two common problems: The audio is in some weird format (some don't play on my phone). The audio is slow. I want to use something like sox or avconv to bulk convert the files. Since this is just voice and going on a cell phone, small low-quality files would be best for me. I had some good success using avconv: avconv -i weird.wma normal.ogg Unforunately, this command creates an enormous ogg file and I can't get it play faster. Ideally, this particular file would play at 170% of the original speed.

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• Join mp4 files in linux

  - by Jose Armando

  I want to join two mp4 files to create a single one. The video streams are encoded in h264 and the audio in aac. I can not re-encode the videos to another format due to computational reasons. Also, I cannot use any gui programs, all processing must be performed with linux command line utilities. FFmpeg cannot do this for mpeg4 files so instead I used MP4Box e.g. MP4Box -add video1.mp4 -cat video2.mp4 newvideo.mp4 unfortunately the audio gets all mixed up. I thought that the problem was that the audio was in aac so I transcoded it in mp3 and used again MP4Box. In this case the audio is fine for the first half of newvideo.mp4 (corresponding to video1.mp4) but then their is no audio and I cannot navigate in the video also. My next thought was that the audio and video streams had some small discrepancies in their lengths that I should fix. So for each input video I splitted the video and audio streams and then joined them with the -shortest option in ffmpeg. thus for the first video I ran avconv -y -i video1.mp4 -c copy -map 0:0 videostream1.mp4 avconv -y -i video1.mp4 -c copy -map 0:1 audiostream1.m4a avconv -y -i videostream1.mp4 -i audiostream1.m4a -c copy -shortest video1_aligned.mp4 similarly for the second video and then used MP4Box as previously. Unfortunately this didn't work either. The only success I had was when I joined the video streams separetely (i.e. videostream1.mp4 and videostream2.mp4) and the audio streams (i.e. audiostream1.m4a and audiostream2.m4a) and then joined the video and audio in a final file. However, the synchronization is lost for the second half of the video. Concretelly, there is a 1 sec delay of audio and video. Any suggestions are really welcome.

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• Convert mp4 video to a format xbox 360 can play

  - by Björn Lindqvist

  Here is a video file my Xbox 360 refuses to play: $ MP4Box -info video.mp4 * Movie Info * Timescale 90000 - Duration 02:18:33.365 Fragmented File no - 2 track(s) File Brand mp42 - version 0 Created: GMT Sat Jul 21 07:08:55 2012 File has root IOD (9 bytes) Scene PL 0xff - Graphics PL 0xff - OD PL 0xff Visual PL: ISO Reserved Profile (0x7f) Audio PL: High Quality Audio Profile @ Level 2 (0x0f) No streams included in root OD iTunes Info: Encoder Software: HandBrake 0.9.6 2012022800 Track # 1 Info - TrackID 1 - TimeScale 90000 - Duration 02:18:33.235 Media Info: Language "Undetermined" - Type "vide:avc1" - 199318 samples Visual Track layout: x=0 y=0 width=1280 height=688 MPEG-4 Config: Visual Stream - ObjectTypeIndication 0x21 AVC/H264 Video - Visual Size 1280 x 688 AVC Info: 1 SPS - 1 PPS - Profile High @ Level 4.1 NAL Unit length bits: 32 Self-synchronized Track # 2 Info - TrackID 2 - TimeScale 48000 - Duration 02:18:33.365 Media Info: Language "English" - Type "soun:mp4a" - 389689 samples MPEG-4 Config: Audio Stream - ObjectTypeIndication 0x40 MPEG-4 Audio MPEG-4 Audio AAC LC - 6 Channel(s) - SampleRate 48000 Synchronized on stream 1 $ avconv -i video.mp4 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:33 with gcc 4.6.3 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42isomavc1 creation_time : 2012-07-21 07:08:55 encoder : HandBrake 0.9.6 2012022800 Duration: 02:18:33.36, start: 0.000000, bitrate: 2299 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x688, 1973 kb/s, 23.98 fps, 90k tbr, 90k tbn, 180k tbc Metadata: creation_time : 2012-07-21 07:08:55 Stream #0.1(eng): Audio: aac, 48000 Hz, 5.1, s16, 319 kb/s Metadata: creation_time : 2012-07-21 07:08:55 At least one output file must be specified What tool, such as ffmpeg or mencoder, and what magic command line incantation should I use to transcode this file into a format Xbox 360 can play? I want the transcode process to retain as good video quality as possible.

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• How do I add and/or keep subtitles when converting video?

  - by JoeSteiger

  I have a mkv video I want to convert to mp4, but every which way I try and convert it (Handbrake, WinFF, ffmpeg, mencoder,...I lose the video's subtitles. How can I convert the video,keeping the subtitles, or add a subtitles.srt? I also would like 2 pass encoding with a video bitrate of 4054 and audio bitrate of 160. Thanks. I was asked for the ffmpeg -i: joe@joe-Leopard-Extreme:/media/Elements/Home Folder/Videos$ ffmpeg -i iron.mkv ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Jun 12 2012 16:52:09 with gcc 4.6.3 *** THIS PROGRAM IS DEPRECATED *** This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. [matroska,webm @ 0x1a319a0] Estimating duration from bitrate, this may be inaccurate Input #0, matroska,webm, from 'iron.mkv': Metadata: title : Iron Duration: 02:06:01.67, start: 0.000000, bitrate: 1280 kb/s Chapter #0.0: start 0.000000, end 546.170622 Metadata: title : Chapter 00 Chapter #0.1: start 546.170622, end 1080.579489 Metadata: title : Chapter 01 Chapter #0.2: start 1080.579489, end 1609.941667 Metadata: title : Chapter 02 Chapter #0.3: start 1609.941667, end 2101.849733 Metadata: title : Chapter 03 Chapter #0.4: start 2101.849733, end 2595.259333 Metadata: title : Chapter 04 Chapter #0.5: start 2595.259333, end 3158.488667 Metadata: title : Chapter 05 Chapter #0.6: start 3158.488667, end 3564.644400 Metadata: title : Chapter 06 Chapter #0.7: start 3564.644400, end 4052.423356 Metadata: title : Chapter 07 Chapter #0.8: start 4052.423356, end 4304.300000 Metadata: title : Chapter 08 Chapter #0.9: start 4304.300000, end 4711.206489 Metadata: title : Chapter 09 Chapter #0.10: start 4711.206489, end 5080.575489 Metadata: title : Chapter 10 Chapter #0.11: start 5080.575489, end 5700.111067 Metadata: title : Chapter 11 Chapter #0.12: start 5700.111067, end 6269.346400 Metadata: title : Chapter 12 Chapter #0.13: start 6269.346400, end 6811.471333 Metadata: title : Chapter 13 Chapter #0.14: start 6811.471333, end 7561.679000 Metadata: title : Chapter 14 Stream #0.0(eng): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 640 kb/s (default) Metadata: title : 3/2+1 Stream #0.2(ita): Audio: ac3, 48000 Hz, 5.1, s16, 640 kb/s Metadata: title : 3/2+1 Stream #0.3(eng): Subtitle: pgssub (default) Stream #0.4(eng): Subtitle: pgssub Stream #0.5(eng): Subtitle: pgssub Stream #0.6(eng): Subtitle: pgssub At least one output file must be specified joe@joe-Leopard-Extreme:/media/Elements/Home Folder/Videos

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• Removing specific part of filename (what's after the second dash) for all files in a folder

  - by Bodo

  I use the command line utility youtube-dl to download videos from YouTube and make mp3s from them with avconv. I'm doing this under Ubuntu 14.04 and very happy with it. The utility downloads the files and saves them with the following name scheme: TITLE(artist-track)-ID.mp3 So an actual filename looks like: EPIC RAP BATTLE of MANLINESS-_EzDRpkfaO4.mp3 Some other file names in the folder look like: EPIC RAP BATTLE of MANLINESS-_EzDRpkfaO4.mp3 Martin Garrix - Animals (Official Video)-gCYcHz2k5x0.mp3 Stromae - Papaoutai-oiKj0Z_Xnjc.mp3 At first, this was no problem. It didn't bother me while listening to my music in Rhytmbox. But when moving to phone or other devices it is pretty confusing to see a so long name, and some players, like the Samsung ones, treat that last part (id after second dash) of the name as Album or something. I'd like to create a bash script that removes what's after the second dash in the name for all files, so it'll make them like this: From: Martin Garrix - Animals (Official Video)-gCYcHz2k5x0.mp3 To: Martin Garrix - Animals (Official Video).mp3 Is it also possible to instruct youtube-dl to exclude the ID from now on? I am currently downloading with the command: youtube-dl --extract-audio --audio-quality 0 --audio-format mp3 URL

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• How do I convert a video to GIF using ffmpeg, with reasonable quality?

  - by Kamil Hismatullin

  I'm converting .flv movie to .gif file with ffmpeg. ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif It works great, but output gif file has a very law quality. Any ideas how can I improve quality of converted gif? Output of command: $ ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif ffmpeg version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:52:53 with gcc 4.7.2 *** THIS PROGRAM IS DEPRECATED *** This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.flv': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 Duration: 00:00:18.85, start: 0.000000, bitrate: 3098 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x720, 2905 kb/s, 25 fps, 25 tbr, 50 tbn, 50 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 192 kb/s Metadata: creation_time : 2013-02-14 04:00:07 [buffer @ 0x92a8ea0] w:1280 h:720 pixfmt:yuv420p [scale @ 0x9215100] w:1280 h:720 fmt:yuv420p -> w:320 h:240 fmt:rgb24 flags:0x4 Output #0, gif, to 'output.gif': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 encoder : Lavf53.21.1 Stream #0.0(und): Video: rawvideo, rgb24, 320x240, q=2-31, 200 kb/s, 90k tbn, 10 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream mapping: Stream #0.0 -> #0.0 Press ctrl-c to stop encoding frame= 101 fps= 32 q=0.0 Lsize= 8686kB time=10.10 bitrate=7045.0kbits/s dup=0 drop=149 video:22725kB audio:0kB global headers:0kB muxing overhead -61.778676% Thanks.

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• ffmpeg cutting video duration

  - by Steve Spence

  When using ffmpeg on linux, my 4.3GB 2.21 second video is being chopped down to 1.56 duration. I'm trying to reduce file size, but not lose frames. steve@steve-OptiPlex-170L:~/Desktop$ ffmpeg -i microbe.avi microbe.mp4 ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Jun 12 2012 16:37:58 with gcc 4.6.3 * THIS PROGRAM IS DEPRECATED * This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. Input #0, avi, from 'microbe.avi': Duration: 00:02:21.80, start: 0.000000, bitrate: 242311 kb/s Stream #0.0: Video: rawvideo, bgr24, 1280x960, 10 tbr, 10 tbn, 10 tbc Incompatible pixel format 'bgr24' for codec 'mpeg4', auto-selecting format 'yuv420p' [buffer @ 0x9f861e0] w:1280 h:960 pixfmt:bgr24 [avsink @ 0x9f86440] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x9f7d800] w:1280 h:960 fmt:bgr24 - w:1280 h:960 fmt:yuv420p flags:0x4 Output #0, mp4, to 'microbe.mp4': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: mpeg4, yuv420p, 1280x960, q=2-31, 200 kb/s, 10 tbn, 10 tbc Stream mapping: Stream #0.0 - #0.0 Press ctrl-c to stop encoding frame= 1164 fps= 6 q=31.0 Lsize= 3775kB time=116.40 bitrate= 265.7kbits/s video:3765kB audio:0kB global headers:0kB muxing overhead 0.272870% steve@steve-OptiPlex-170L:~/Desktop$

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