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  • How to keep source frame rate with mencoder/ffmpeg?

    - by Sandra
    I would like to crop and rotate a video, and then encode it to mp4 or mkv. mencoder video.mp4 -vf rotate=1,crop=720:1280:0:0 -oac pcm -ovc x264 -x264encopts preset=veryslow:tune=film:crf=15:frameref=15:fast_pskip=0:threads=auto -lavfopts format=matroska -o test.mkv But when I do the above encoding, the frame rate is way too fast. The encoding options were something I found, so I don't know if that is the problem. Question All I want is to crop and rotate the video, and keep the audio/video quality as good as possible. Have anyone tried this?

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  • FFMEPG permission problem, is there any alternative solution instead of FFMPEG?

    - by caglaror
    We have some kind of permenant permission problem on FFMPEG.exe to write JPEG file on to HDD. We are using IIS and try all permission methods to ffmepg.exe, its including folder,folders, cmd.exe and any executable file, related folder, file. Also we tried many many command examples. But never went beyond the "permission denied" error messages. We give up. Do you know another alternative solution to pick images from flv, f4v movie files? Or %100 quaranteed method to achive this permission control on IIS? Thank you. ---last code we try variables etc. aren't shown here. jpegYapKomutu = videoEditorKlasoru &"\ffmpeg.exe -i " & videoEditorKlasoru & "\deneme.flv" &" -s 480×360 -ss 00:00:"&saniyesi&" -vframes 1 -f mjpeg "& "C:\Webhome\normworks\caglarorhan\deneme.jpg" WScript.Run "%COMSPEC% /C dir" & jpegYapKomutu

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  • Recording slow web stream

    - by Budric
    I'm trying to record an mpeg2 video stream from a website that doesn't have the greatest bandwidth. The video often buffers. I want to download the stream and watch it offline. The extract stream format received is: Stream #0.0[0x44]: Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s Stream #0.1[0x45]: Video: mpeg2video (Main), yuv420p, 704x576 [PAR 16:11 DAR 16:9], 15000 kb/s, 27.19 fps, 25 tbr, 90k tbn, 50 tbc I use the following tool to transocde the stream: ffmpeg -i "http://url" -y -vcodec libx264 -b 3000k -acodec copy /tmp/stream.mp4 Unfortunately after a few seconds ffmpeg stops recording with an error [mpegts @ 0x1f0b9c0] PES packet size mismatch [mp2 @ 0x1f14640] incomplete frame Error while decoding stream #0.0 [mpeg2video @ 0x1f16860] ac-tex damaged at 0 26 [mpeg2video @ 0x1f16860] Warning MVs not available I've tried encoding with vlc as well with similar issues. Although vlc doesn't stop encoding, the output video has regions where it hangs. vlc -I dummy "http://url" --network-caching="1000" --sout="#transcode{vcodec=h264,vb=3000,acodec=mp3,ab=192}:std{access=file,mux=mp4,dst=/tmp/stream.mp4}" [mpeg2video @ 0x7f2d4c001e20] ac-tex damaged at 9 33 [mpeg2video @ 0x7f2d4c001e20] Warning MVs not available [mpeg2video @ 0x7f2d4c001e20] concealing 132 DC, 132 AC, 132 MV errors [mpeg2video @ 0x7f2d4c001e20] ac-tex damaged at 16 17 [mpeg2video @ 0x7f2d4c001e20] Warning MVs not available [mpeg2video @ 0x7f2d4c001e20] concealing 836 DC, 836 AC, 836 MV errors libdvbpsi error (PSI decoder): TS discontinuity (received 4, expected 3) for PID 0 I also tried flv transcoding and it shows up with its own set of issues, like output flv file hangs in certain parts. Anyone know what's wrong or how to fix this?

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  • Convert mkv/h264 video so it can be played on a "mid-range" Sony Ericsson phone. (using Ubuntu).

    - by Johan
    Hi As a little experiment I thinking of converting some video/movies/tv-series into a format that could be playable on my K850, but to be a little bit more generic in this question let's say "mid range Sony Ericsson" phone since they all more or less behave the same and has the same screen resolution (240 x 320). I am looking for command line based tools (for Ubuntu), since I am thinking about writing a "convert and move" script later if it is successful. A lot of the video I have is encoded in mkv/h264, but since that is not supported by the phone I guess that I need to convert it into some mp4/mpeg4 low quality video. After some googling it seems like a good candidate for the job is ffmpeg, but that seems to be a very versatile tool with a lot of magic tricks. Am I on the right track? And if so how do I use ffmpeg to do this? Thanks Johan Update: After plating a little bit with ffmeg I noticed that it only uses 1 of my 4 cores, so the transcoding takes forever. I found a arg called -threads but that did not change much, maybe I got it wrong. I also found that something like this plays in the phone. ffmpeg -i Mythbusters\ S1D1_1.mkv -threads 4 -t 180 -vcodec mpeg4 -r 15 -s 320x240 Mythbusters\ S1D1_1_mini.mp4 It was possible to use 3gp/h263, but the quality was really useless. ffmpeg -i Mythbusters\ S1D1_1.mkv -t 180 -vcodec h263 -acodec libfaac -s cif Mythbusters\ S1D1_1_cif.3gp And it seems like mp4/h264 is also possible and the result is ok, thanks to this question, this one seem to use more than one core as well so it was a little bit faster for me. ffmpeg -i Mythbusters_S1D1_1.mkv -t 180 -acodec libfaac -ab 60k -s 320x240 -vcodec libx264 -b 500k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -flags2 +mixed_refs -me_method umh -subq 6 -trellis 1 -refs 5 -coder 0 -me_range 16 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 500k -maxrate 768k -bufsize 2M -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 13 -threads 0 -f mp4 Mythbusters_S1D1_1_qvga.mp4 Update: I have tried to use HandBrakeCLI and it is no problem creating a new file that seem to be the same as the one created with ffmpeg with something like this. HandBrakeCLI -i Mythbusters_S1D1_1.mkv --size 100 -E faac -B 60 --maxHeight 240 -r 15 -e x264 -o Mythbusters_S1D1_1_hand.mp4 But that one did not play in the phone... I found this in the official manual: If you transfer video clips using another program than Media Go™, we recommend that you select H.264 Baseline profile video, up to QVGA at 30 fps, VBR 384 kbps (max 768 kps) with AAC+ audio at 128 kbps (max 255 kbps), 48 kHz and stereo audio in mp4 file format. So the idea to use H264 seems to be correct.

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  • Make exact mp4 (H264) format for uploading to youtube

    - by WHITECOLOR
    With ffmpeg I'm converting video from mp3 and picture to upload it to youtube. After upload, conversion fails. Reasons are unknown. I believe the problem is in format. By the way If I'm uploading file 5 minutes length, it fails if I upload 30 seconds of this file it succeeds. I have donwload mp4 file from youtube. Then I uploaded it, it is done very fast. So a nice solution would be to convert videos to the same format that is done by google. I got the following output by mpeg: ffmpeg version N-44264-g070b0e1 Copyright (c) 2000-2012 the FFmpeg developers built on Sep 7 2012 17:38:57 with gcc 4.7.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runt ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass - -enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-l ibfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenj peg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheo ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena ble-zlib libavutil 51. 72.100 / 51. 72.100 libavcodec 54. 55.100 / 54. 55.100 libavformat 54. 25.105 / 54. 25.105 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 16.100 / 3. 16.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'youtubetrack0.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2012-10-02 22:58:57 Duration: 00:06:46.66, start: 0.000000, bitrate: 176 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yu v420p, 450x360, 78 kb/s, 6 fps, 6 tbr, 12 tbn, 12 tbc Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 95 kb/s Metadata: creation_time : 2012-10-02 22:58:57 handler_name : IsoMedia File Produced by Google, 5-11-2011 Is it possible to construct ffmpeg parameters so that that would give the same format that google internally does? Is the information above sufficient? I couldn't construct needed params. For example I don't understand how to set tbn and what 95 kb/s mean in "Stream #0:1(und): Audio:". Now I just do: ffmpeg -i videoimage.jpg -i audio.mp3 video.mp4 Info I've got: ffmpeg version N-44998-gdf82454 Copyright (c) 2000-2012 the FFmpeg developers built on Oct 2 2012 23:03:12 with gcc 4.7.1 (GCC) configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libcelt --enable-libopencore-amrnb --en able-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-librtmp --enable-libschroedinger - -enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc -- enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enab le-libxavs --enable-libxvid --enable-zlib libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 63.100 / 54. 63.100 libavformat 54. 29.105 / 54. 29.105 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 19.102 / 3. 19.102 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.25.105 Duration: 00:06:46.81, start: 0.000000, bitrate: 129 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p, 450x360, 3392 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 127 kb/s Metadata: handler_name : SoundHandler This video fails the conversion on youtube. I also tried to use other vcode parmam and extensions of output file (mp4, wmv, avi) but failed too. Would be greatful for help.

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  • How do I reduce RAM usage on my server?

    - by Abs
    I have recently launched a site that is very popular but I am having trouble with scalability. My site makes heavy use of FFmpeg and at peak times RAM usage hits the 2 GB point quickly and the swap file starts getting used. CPU usage starts rising too. Users complain that the site is slow. They say this because all FFmpeg instances run very slow because of the number running at the same time. Users make use of FFmpeg on my server in real time. Is there anything I can consider or do to ease down the usage of the server and RAM just shooting up? Maybe there is something better than FFmpeg (!). Is the only solution "throwing some cash" at a more powerful server? I have given little information, please ask for more, so this problem can be solved.

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  • aspect ratio is changed using ffmpeg sameq and codec copy

    - by Vishal Parekh
    i am using ffmpeg to extract clip from mp4 video, i tried with "-acodec copy -vcodec copy" and "-sameq" in both, aspect ration of generated file is changed. ( ffmpeg -sameq -i "input file" "output file" ffmpeg -i "input file" -acodec copy -vcodec copy "outputfile" ) source file is of aspect ratio sar=4:3 dar=4:3 new file is has aspect ratio sar=4:3 dar=1:1 please help me to solve this problem, one weird thing is when i see details in another video tool, it shows me sar=4:3 dar=4:3 of source video but when i use command ffmpeg -i sourcefile, it shows me sar=300:400 dar=1:1 thanks vishal parekh

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  • Iphone sdk code to capture images at 15 FPS on 3g mobile and sending to ffmpeg server .

    - by user286517
    I'm writing an app code to do video recording on iPhone much like all the available apps :) ... All im trying to do is capture screen from iPhone camera on 3g mobile and sending them to server .. but want some time efficient approach for sending to server and capturing image sequences ... its like i want to send 15 images / second to server in one single go :P i've set up server with FFMPEG and other codecs so their is no issue in generation of live video / stream .... Help me ...special reward for best helping answer

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  • FFmpeg & Installation on phpmyadmin

    - by Vivek
    I am attempting to have an interface in which people can upload music files and listen to them through the site. The biggest problem obviously is that someone who uploads an audio track in mp3 format into Mozilla wouldn't be able to play it back (since MF doesn't support mp3 playback since I'm using jPlayer). I did some research and found out that I could use command line php using FFmpeg to convert the mp3 to ogg or some other supportable format. I believe I understand (a little bit) how command line php works but I was wondering how I could install it onto phpmyadmin on my hosting service? Could anyone link me to a tutorial or care to explain? I tried googling it but I just couldn't find it.

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  • Is FFmpeg missing from the official repositories in 14.04?

    - by user254877
    I tried to install ffmpeg in trusty/Ubuntu 14.04 and got the following message: $sudo apt-get install ffmpeg Reading package lists... Done Building dependency tree Reading state information... Done Package ffmpeg is not available, but is referred to by another package. This may mean that the package is missing, has been obsoleted, or is only available from another source E: Package 'ffmpeg' has no installation candidate Why isn't the package available?

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  • Why doesn't this for loop work?

    - by evilsoup
    This is on Ubuntu 12.04 I'm trying to figure out how to get ffmpeg to do a batch conversion of FLACs to MP3, recursively. If I cd into a directory and use for f in *.flac; do ffmpeg -i "$f" -c:a libmp3lame -q:a 2 "${f/%flac/mp3}"; done that works perfectly fine. However, when I try this, it doesn't work: for f in "$(find . -type f -name *.flac)"; do ffmpeg -i "$f" -c:a libmp3lame -q:a 2 "${f/%flac/mp3}"; done It doesn't even throw up any useful errors (but here is the output anyway, no need to complain): evilsoup@enchantment:~/Music/Jean Sibelius$ for f in "$(find . -type f -name *.flac)"; do ffmpeg -i "$f" -c:a libmp3lame -q:a 2 "${f/%flac/mp3}"; done ffmpeg version git-2012-12-18-b7e085a Copyright (c) 2000-2012 the FFmpeg developers built on Dec 18 2012 19:23:11 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --enable-gpl --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3 libavutil 52. 12.100 / 52. 12.100 libavcodec 54. 80.100 / 54. 80.100 libavformat 54. 49.102 / 54. 49.102 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 28.100 / 3. 28.100 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/02. Symphony No.1.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/03. Symphony No.1.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/stripped2.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/05. Symphony No.1.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/stripped3.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/09. Andante festivo.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/08. Symphony No.3.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/01. Finlandia.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/07. Symphony No.3.flac ./Symphonies 1, 2, 3 & 5 I've tested the find command on its own, and it works as expected, so the problem has to be something to do with the interaction between find and for. I'm aware that I could do something with find's -exec option, but I can't find any way to do string substitution as I can with a bash for loop, and I'd rather not have a bunch of file.flac.mp3s to deal with, even if they could be fixed with a simple rename.

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  • How to escape this in the bash script?

    - by allenskd
    I'm trying to complete a batch of 3 videos to leave it there till morning processing but it seems there are special characters in it... I try it "raw" in the terminal and it works but in bash script it stops working Example: args1="-r 29.97 -t 00:13:30 -vsync 0 -vpre libx264-medium -i" args12="-r 29.97 -ss 00:40:30 -vsync 0 -vpre libx264-medium -i" args2="[in] scale=580:380 [T1],[T1] pad=720:530:0:50 (other arguments with lots of [ and ]" In the output it says Unable to find a suitable output format for 'scale=580:380' not sure why... like I said, the command runs fine in the command-line, just not in the script /usr/local/bin/ffmpeg "$args1" "${file}" -vf "$args2" "$args3" "${args[0]}_${startingfrom}_0001_02.mp4"

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  • Does FFMpeg support gpu acceleration of media encoding/decoding?

    - by Jason123
    I was wondering if ffmpeg supported gpu acceleration. I was reading on their websites and came across contradicting information. http://www.ffmpeg.org/general.html#Video-Codecs -H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration) http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide -Will a graphics card make x264 encode faster? No. libx264 doesn't use them (at least not yet). There are some proprietary encoders that utilize the GPU, but that does not mean they are well optimized, though encoding time may be faster; and they might be worse than x264 anyway, and possibly slower. Regardless, FFmpeg today doesn't support any means of gpu encoding, outside of libx264. If not, is there any way to add gpu acceleration to h.264 encoding/decoding?

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  • Blu-ray player?

    - by Dox
    I'd like to play bluray discs in my laptop. I found the official documentation, and there it's explained that one should use mplayer and ffmpeg. Looking at the repositories, there exist two different mplayer packages (in conflict with each other) mplayer mplayer2 Any ideas with of them should I install? On the other hand the official documentation seems to be out of date since no mention to Ubuntu after 9.04 is done. Does the DumpHD package from the repositories work? Finally, Where could the keydb.cfg keys be found? I'm open to suggestions, specially of people who had done the job of making it work. Cheers

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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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  • How to change the volume of left channel using pulseaudio

    - by user2622247
    I am recording the video using logitech camera and bluetooth microphone. Logitech can used for recording both audio and video. When I turn on bluetooth microphone in the middle of recording, it is replacing the logitech audio channel due to this we are getting the bluetooth audio from the left channel. But when I turn off the bluetooth then I get the logitech audio on left channel but the volume is very low and also getting the some noise. I am using PulseAudio and ffmpeg for recording purpose. So how can I increase / change the volume of left channel during runtime?

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  • Concatenation of a 2 second silence audio with a normal audio not working

    - by user1665130
    I have a code for concatenation of files using ffmpeg.Here silence.wav is a mute audio file with 2 seconds length. I need to prepend this mut audio file to REC00096_Jun-06-2014 16.47.28.wav. I tried the folowing code. ffmpeg -i D:\vishnu\silence.wav -i D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav \-filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' \-map '[out]' output.wav Following is the error i am getting. D:\vishnu>ffmpeg -i silence.wav -i "D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav" -filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' -map '[out]' outp ut.wav ffmpeg version N-59036-g5d8e4f6 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 12 2013 22:01:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 58.100 / 52. 58.100 libavcodec 55. 45.101 / 55. 45.101 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, wav, from 'silence.wav': Metadata: encoder : Lavf55.22.100 Duration: 00:00:02.02, bitrate: 4234 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 5.1, s16, 4 233 kb/s Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav': Duration: 00:00:08.04, bitrate: 384 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, mono, s16, 384 kb/s [wav @ 036f5e40] Invalid stream specifier: '[out]'. Last message repeated 1 times Stream map ''[out]'' matches no streams. D:\vishnu>

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  • How to extract a Vorbis stream from a WAVE file?

    - by H.B.
    I would like to move the Vorbis stream into an ogg container but ffmpeg does not seem to recognize the stream. Even though MPlayer gives this output upon playback: Opening audio decoder: [acm] Win32/ACM decoders Loading codec DLL: 'vorbis.acm' Loaded DLL driver vorbis.acm at 10000000 Warning! ACM codec reports srcsize=0 AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000-176400) Selected audio codec: [vorbisacm] afm: acm (OggVorbis ACM) ffmpeg: ffmpeg -i Source.wav -acodec copy Target.ogg Input #0, wav, from 'Source.wav': Duration: 00:02:15.17, bitrate: 128 kb/s Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s [ogg @ 00000000003096C0] Unsupported codec id in stream 0 Output #0, ogg, to 'Target.ogg': Metadata: encoder : Lavf53.6.0 Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Could not write header for output file #0 (incorrect codec parameters ?) Of course this does not necessarily need to be done via ffmpeg, any method that is workable would be fine... I have cut down one of the files to 512KB: sample.wav (Changed two chunk size fields in the wave header to account for this, the embedded stream is cut "without notice")

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  • How can I avoid a few seconds of blank video when using -vcodec copy?

    - by arlomedia
    I'm processing user-uploaded videos on a CentOS web server with ffmpeg. I need to convert each video to a standard size and format, then extract a 30-second sample clip from each video. I want to use the "-vcodec copy" flag in the extraction command to avoid encoding a second time. This command works for my initial conversion: ffmpeg -i uploaded.mov -f mp4 -vcodec libx264 -vpre medium -acodec libfaac -r 15 -b 360k -ab 48k -ar 22050 -s 480x320 formatted.mp4 And this sometimes works for the extraction: ffmpeg -i formatted.mp4 -vcodec copy -acodec copy -ss 0 -t 30 formatted_sample.mp4 However, when I run the extraction command on some videos, the extracted sample clip starts with several seconds of blank video. The audio starts right away but the video doesn't start for 3-6 seconds. To demonstrate the problem, I've uploaded two video clips and run the above commands on them. I created the first clip in Final Cut Express and encoded it with Handbrake before uploading to the web server: 1a) uploaded clip 1b) converted with first command 1c) extracted with second command, missing first six seconds By comparison, this second clip comes from Apple's website and does not show the problem: 2a) uploaded clip 2b) converted with first command 2c) extracted with second command, no problem Can anyone see what's different about the two source clips? And if so, is there anything I can do in my conversion command so that when the extraction command runs, the clip is set up to avoid the missing video? By the way, I initially had the problem with ffmpeg 0.6.1 installed from yum, but I upgraded to the latest git version and the problem remains.

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  • DTS to AC3 conversion for LG TV using mediatomb DLNA server

    - by prion crawler
    I want to convert a MKV video file containing DTS audio to a stream with AC3 audio. I want to pass this resulting stream to mediatomb's transcoding feature. Mediatomb will transfer the stream via DLNA to a LG TV, which does not support DTS audio. I have tried the VLC command below but the TV does not recognize the stream, and playing the destination stream on PC does not produce sound. vlc -vvv -I dummy INPUT.file --sout \ '#transcode{acodec=ac3,ab=256k,channels=2,threads=4} \ :std{mux=ts,access=file,dst=DEST.file}' The following ffmpeg command give a stream that plays on the TV with sound, but the ffmpeg process gets killed (with signal 15) within 10-15 seconds, and then the TV restarts the playback from the beginning. This goes on in loops. ffmpeg -i INPUT.file -acodec ac3 -ab 384k -vcodec copy \ -vbsf h264_mp4toannexb -f mpegts -y DEST.file I want to have a working DLNA server which transcodes DTS to AC3, any help is appreciated.

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  • How to install two packages that write the same file

    - by Joel E Salas
    Sirs, a conundrum. I have two packages that each create /usr/bin/ffprobe. One of them is ffmpeg from the Deb Multimedia repository, while the other is ffmbc 0.7-rc5 built from source. The hand-rolled one is business-critical, and we used to just install it from source wherever it was necessary. I can only assume it would clobber the ffmpeg file, and there were never any ill effects. In theory, it should be acceptable for our ffmbc package to overwrite the file from the ffmpeg package. Is there any easy way to reconcile this?

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  • Out of sync audio video using mencoder

    - by 1ch1g0
    hi i converted a mkv (matroska) file to avi using ffmpeg: ffmpeg -i input.mkv -f mp4 -vcodec mpeg4 -sameq -r 29.97 -b 512kb -acodec ac3 -ab 128kb -vol 512 output.avi the output file plays fine using mplayer. after that, i using mencoder to insert subtitles: mencoder output.avi -o new.avi -oac pcm -ovc lavc -subfont-text-scale 3 -sub subtitle.srt however, after i play back the video "new.avi", the video and audio is out of sync. What options can i put into mencoder to sync the A/V. ? I have also tried ffmpeg -newsubtitle option but can't get it work. Any examples of usage of -newsubtitle would be greatly appreciated. thanks

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