May 2011 - ffmpeg-user - lists.ffmpeg.org

How do I encode in H264 Baseline but change the GOP?
by Carl Partridge 28 May '11

28 May '11

Hi everyone, I've just joined the mailing list, so thank you in advance for having me. I am trying to encode to H264/MP3, mux'd into Mpeg1-TS. It's for HTTP Live Streaming, but the decoder only supports baseline profile. After some time spent looking at the options, I realised that I could just do: ffmpeg -profile baseline -i input.avi -y -async 1 -f mpegts -vcodec libx264 -b 1200k -bt 100k -r 25 -acodec libmp3lame -ab 64k -ar 48000 -ac 2 output.ts ....to generate my stream. (then segment with an appropriate app) Unfortunately, this is no good for segmenting, since I have no control over the GOP length. Ideally I would like an IDR (index) frame at the beginning of each 10-second segment. I tried: -g 10 (before AND after the -profile parameter) ...unfortunately it seems to be ignored, presumably because the default settings that "-profile baseline" applies wipe out everything else. Is there any way to specify a GOP interval while forcing everything else to constrained baseline, or will I have to manually specify all the various X264 flags. I tried doing this, but wasted hours, since I couldn't find out the exact ones that corresponded to constrained baseline profile, and something kept bumping the decoder. Thanks in advance, Carl PS: This is all for 'SilverLive', an HTTP Live Streaming client I have developed for MS Silverlight. -- ________________________________ *Carl Partridge* FatAttitude Ltd www.fatattitude.com | follow on twitter <http://www.twitter.com/fatattitude>

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Converted file size is bigger than source
by Vishal Nikam 28 May '11

28 May '11

Hello ffmpeg team, I am transcoding one .wmv file to .mp4 file, it gets successfully transcode into h264/aac encoder. But, my problem is that Converted file size is twice than Source file size. Here is my command. ffmpeg -i CCC2009Sampler.wmv -acodec aac -vcodec libx264 -fpre "libx264-lossless_ultrafast.ffpreset" -crf 18 -strict experimental -y CCC2009Sampler.mp4 Details of source file: FileFormat :: asf AvgBitRate :: 2289 kb/s Duration :: 3020(Sec) AudioCodec :: wmav2 AudioBitRate:: 128 kb/s Channel :: Stereo SamplingRate :: 44100(Hz) VideoCodec :: wmv3 (Main) VideoBitRate :: 2200 kb/s FrameRate :: -1.0(fps) VideoSize :: 640*480 FileSize :: 864231560(bytes) Details of target file :: FileFormat :: mov AvgBitRate :: 3420 kb/s Duration :: 3020 AudioDetails :: aac VideoCodec :: h264 VideoBitRate :: 3368 kb/s FrameRate :: 29.97(fps) VideoSize ::640*480 FileSize :: 1291278946(bytes) Please consider my issue. I am waiting for your reply. Thanks, Vishal T. Nikam

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Re: [FFmpeg-user] to 64 or 32
by Andrey m0shbear Vul 28 May '11

28 May '11

-original message- Subject: [FFmpeg-user] to 64 or 32 From: Jim Shupert <jshupert(a)pps-inc.com> Date: 2011-05-27 16:50 friends, I am about to put together a system and was wondering what might be suggested 32 bit or 64 I am thinking of a ubuntu 10.4 machine hp wx8400 with a black magic decklink card if anyone has an opinion... ____ x86_64 is faster because there are 8 more general-purpose registers. Seeing as ffmpeg x86 asm is mostly sse, this may be somewhat less relevant.

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Non monotonic timestamp issues
by Jon Drukman 28 May '11

28 May '11

I've got a .MOV file that I'm having a real bear of a time converting. Here is the file: http://tm-ami.s3.amazonaws.com/clip.mov % ffmpeg -y -i clip.mov -s 480x360 -vcodec libx264 -acodec libfaac clip.mp4 ffmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 25 2011 11:47:40 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --enable-pthreads --enable-libfaac --enable-libmp3lame --enable-libx264 --enable-gpl --enable-nonfree --disable-network --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 1. 0 / 53. 1. 0 libavformat 53. 0. 3 / 53. 0. 3 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'clip.mov': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2011-05-22 00:47:28 Duration: 00:01:32.94, start: 0.073946, bitrate: 5878 kb/s Stream #0.0(und): Audio: pcm_s24le, 44100 Hz, 2 channels, s32, 2116 kb/s Metadata: creation_time : 2011-05-22 00:47:28 Stream #0.1(und): Video: mjpeg, yuvj420p, 640x480 [PAR 72:72 DAR 4:3], 3684 kb/s, 14.51 fps, 30k tbr, 30k tbn, 30k tbc Metadata: creation_time : 2011-05-22 00:47:28 Incompatible sample format 's32' for codec 'libfaac', auto-selecting format 's16' [buffer @ 0x1010037e0] w:640 h:480 pixfmt:yuvj420p [scale @ 0x101003b20] w:640 h:480 fmt:yuvj420p -> w:480 h:360 fmt:yuvj420p flags:0xa0000004 [libx264 @ 0x101829a00] Default settings detected, using medium profile [libx264 @ 0x101829a00] using SAR=1/1 [libx264 @ 0x101829a00] MB rate (20700000) > level limit (983040) [libx264 @ 0x101829a00] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x101829a00] profile High, level 5.1 [libx264 @ 0x101829a00] 264 - core 104 - H.264/MPEG-4 AVC codec - Copyleft 2003-2010 - 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=1 sliced_threads=0 nr=0 decimate=1 interlaced=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=abr mbtree=1 bitrate=200 ratetol=1.0 qcomp=0.60 qpmin=10 qpmax=51 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'clip.mp4': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2011-05-22 00:47:28 encoder : Lavf53.0.3 Stream #0.0(und): Video: libx264, yuvj420p, 480x360 [PAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 30k tbn, 30k tbc Metadata: creation_time : 2011-05-22 00:47:28 Stream #0.1(und): Audio: libfaac, 44100 Hz, 2 channels, s16, 64 kb/s Metadata: creation_time : 2011-05-22 00:47:28 Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 4327 fps=222 q=51.0 size= 118kB time=0.14 bitrate=6780.1kbits/s dup=4325 drop=0 With these settings, it takes forever to encode. It's only a 90 second clip but I gave up waiting for the encode after a few hours. Also notice the really high number of 'dups'. I tried setting vsync to 0 but ffmpeg aborts almost immediately with: [mp4 @ 0x101829400] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 2218 >= 2218 av_interleaved_write_frame(): Invalid argument I tried converting the same clip with Handbrake and it took a few seconds on my laptop (MacBook Pro, 4 cores). Anybody got any ideas how to make ffmpeg play happily with this file?

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trying to work out ffmpeg and avi conversion
by mroper 28 May '11

28 May '11

hi, hoping someone can help, i'm fairly new to ffmpeg, trying to work out how to convert an HD 1280x720 AVI file to mpeg2 so my tv via linux media tomb can play it. When i try doing.... ffmpeg -i "$filename" -target pal-dvd -b 25000k /destpath/$filename i get .....Error while opening encoder for output stream #0.0- maybe incorrect parameters such as bit_rate, rate, width or height i've tried searching on the error and drawn a blank, lots of information but nothing I can relate to it. likewise the man pages don't give me a clue. reason why I have the -b 25000k is I'm trying to maintain the quality of the orginal avi, if I don't set this it works fine. likewise if I set a lower rate it also works fine (but at lower quality). The orginal avi when doing a ffmpeg -i filename shows up Stream #0.0: Video: mjpeg, yuvj422p, 1280x720, 30 tbr, 30 tbn, 30 tbc Metadata: strn : FUJIFILM AVI STREAM 0100 Stream #0.1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s Where as the best I can get is via ffmpeg -i "$filename" -target pal-dvd -s hd720 /destpath/$filename and produces Duration: 00:00:02.00, start: 0.500000, bitrate: 7176 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 448 kb/s Which has a much lower quality/bitrate (you can see the difference on screen). Any ideas? Thanks Miles

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to 64 or 32
by Jim Shupert 27 May '11

27 May '11

friends, I am about to put together a system and was wondering what might be suggested 32 bit or 64 I am thinking of a ubuntu 10.4 machine hp wx8400 with a black magic decklink card if anyone has an opinion... thanks

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Overlay video sync
by Riley Schutt 27 May '11

27 May '11

Hello, I have two videos that I would like to combine side-by-side using the overlay filter. The two videos are both h264 @60fps and capture the same event from different angles. I would like to get frame-exact syncing of the videos but because the I-frames do not line up, it seems the videos cannot be clipped or started with frame-exact precision. I have tried the following without success: ffmpeg -i clip_01_Bow.mp4 \ -acodec copy \ -vcodec libx264 -preset slow -crf 20 -threads 0 \ -vf "[in] setpts=PTS-STARTPTS,transpose=1,scale=404:720,pad=1280:720:0:0 [b1] ; movie=clip_01_Mast.mp4:seek_point=0.66 [m1]; [m1] setpts=PTS-STARTPTS,crop=876:720:150:0 [m2];[b1][m2] overlay=404:0 [out]" \ test.mp4 How can I go about shifting the overlayed (or input) video in time by 2 or 3 frames? Thanks, Riley

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clock gap in ffmpeg
by jose＠netlab.tkk.fi 27 May '11

27 May '11

Hello, We are using ffmpeg from VLC for doing transcoding and it seems that ffmpeg gets stuck. Snippet from the log ====================== 2011-05-18 00:24:53 [0x1021970] main input warning: clock gap, unexpected stream discontinuity 0x1021970] main input warning: feeding synchro with a new reference point trying to recover from clock gap 2011-05-18 13:16:20 mb incr damaged 2011-05-18 13:16:20 [0x7fab300343e0] avcodec encoder warning: almost fed libavcodec with a frame in the past (current: 1695556067861, last: 1699689600301) ==================== It seems that this 'almost fed libavcodec..' error at VLC comes from modules/codec/avcodec/encoder.c:907 check, iirc in libav/ffmpeg didn't like if pts-values weren't monotonic. We could comment out that if clause in code and recompile that module but it might not be good idea as it is doing this timing check. Could you indicate what might be the reason for this problem and what could be the solution? Thanks, Rgds Jose

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multiple drawtext parameters
by mromey 27 May '11

27 May '11

Want to use ffmpeg to publish quickTimes and thumbnails for our asset management system. Evaluating how viable ffmpeg would be as a replacement to our current system. In the past I have done it lots of different ways. I have used After Effects, Nuke, RVIO, python and pil. Thus far I am impressed with ffmpeg but having some difficulty with some parameters. Exploring the drawtext filter. I have the following example. ffmpeg.exe -i C:/temp/input.mov -vf drawtext="arial.ttf=':text='TEST'" C:/temp/output.mov Have a few questions: 1. Is it possible to process multiple drawtext in the same IO stream. I would rather load the movie or image sequence once into memory once then apply each drawtext overlay rather than process the addition of each drawtext and load the file each time. If this is possible can somebody help by showing me the syntax. 2. One of the drawtext burn ins needs to be a frame counter. Is it possible to send some type of internal variable/flag to the value of text that would use the frame number in lieu of static text? 3. Was hoping to get some direction as to how I would take a single frame slate file and append it to a quicktime via ffmpeg. Thanks for the help in advanced. Romey -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/multiple-drawtext-parameters-tp355… Sent from the FFmpeg-users mailing list archive at Nabble.com.

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output lagging at startup
by Karoly Horvath 27 May '11

27 May '11

hi, I'm running ffmpeg, it is reading from stdin and writes to stdout. When I start it for a couple of seconds it doesn't output anything. If I use an output codec with small bitrate this can take minutes which is really bad, because I want to segment the stream and stream it with a web server. I tried several command line parameteres related to various buffers but none of them seems to solve the problem. Is there a parameter to fix this? -- Karoly Horvath software engineer

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