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• How to convert .avi video to .mp4(for Motorola Milestone, Android 2.3.4) with Avidemux

  - by kv1dr

  I open .avi video with Avidemux and I set Video format to MPEG-4 AVC(under Configure, Bitrate tab I choose "Single Pass - Bitrate (Average)" and Target bitrate to 256 kb/s, under Filters I choose MPlayer resize to 480x360 and I also add a subtitles) audio format to AAC (Faac)(Under Configure, I choose Bitrate 96) and format to MP4(like a image below). When Avidemux convert video to .mp4 format I can play the file on my copmuter, but on my phone I can't. When I want to play it on my phone with native video player, it just show the error something like "Can't play this video". So the question is how to convert .avi video to .mp4 with Avidemux(because I want to have subtitles inside movie) to be playable with android phone(Android version 2.3.4) with native player. Any help will be highly appreciated. :)

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• Is there now any way to convert mp3 files to m4a or aac 192kbit?

  - by piedro

  Since about two years now I am trying to find a way to convert high quality mp3 files to m4a or aac files with a fixed bitrate 192k. Please don't suggest using another format - i thought this through as far as it goes. The problem here is: ffmpeg obvioulsy can't convert to a higher bitrate than 152k. Even when it says it does so the resulting files still have 152k instead of 192k. ffmpeg also has/had a bug not writing the bitrate into the audio file tags which means when testing you have to calculate the bitrate manually by dividing the filesize by the length of the audio in seconds (resulting in 152k - see above) choosing faac as converter gets me the same results other programs don't work reliably (see this thread Howto convert audio files to *.m4a? I know that this is not an original new problem but I am wondering if there is still no way to convert with ubuntu/kubuntu 12.04 after a lot time passed and I can't find some of the bug issues mentioned in the other thread anymore. So: Is there a solution after all?

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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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• 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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• 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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• 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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• Sound Juicer doesn't respect Lame's codec settings when ripping CDs

  - by Takkat

  Using Sound Juicer I am able to rip CDs very conveniently. I would like to rip them in about 256 kbit/s variable bitrate. To accomplish this I have defined the settings for mp3 in gnome-audio-profiles-properties as follows: audio/x-raw-int,rate=44100,channels=2 ! lame name=enc mode=0 vbr-quality=0 ! id3v2mux where vbr-quality=0 should give me a variable bitrate averaging 245 kbit/s. The resulting files however always say they are in 128 kbit/s. Is this only a tagging bug or is indeed the bitrate that low? How could I find out?

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• Why have mp3 files ripped with Lame always have 128 kbit/s irrespect of settings?

  - by Takkat

  Using Sound Juicer I am able to rip Cds very conveniently. I would like to rip them in about 256 kbit/s variable bitrate. To accomplish this I have defined the settings for mp3 in gnome-audio-profiles-properties as follows: audio/x-raw-int,rate=44100,channels=2 ! lame name=enc mode=0 vbr-quality=0 ! id3v2mux where vbr-quality=0 should give me a variable bitrate averaging 245 kbit/s. The resulting files however always say they are in 128 kbit/s. Is this only a tagging bug or is indeed the bitrate that low? How could I find out?

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• What is some good lossless video codec for recording gameplay?

  - by Don Salva

  I'm an avid gamer and I like to record my gameplay. Usually I've been using Fraps to do it, however I'm thinking of switching to Dxtory as it allows to write on multiple HDDs at once. Say I have 3 HDDs with the following write speeds: HDD1 with 50 mb/s, HDD2 with 22 mb/s and HDD3 with 45 mb/s. Combined write speed would be: 117 mb/s. Dxtory allows you to utilize all 3 HDD's at once while recording your gameplay. Using this formula: RGB24 YUV24: Width x Height x 3 x fps = bitrate (byte/sec) YUV420: Width x Height x 3 / 2 x fps = bitrate (byte/sec) YUV410: Width x Height x 9 / 8 x fps = bitrate (byte/sec) And recording in YUV420 colorspace at 1920x1080 with 30 fps I'd need about 95 mb/s write speed. Dxtory is good because it allows me to play with constant 60 fps while recording in 30 fps. Fraps does not (even though they say it does), once you start recording with Fraps, the game's fps drops. So I'm looking for a codec that doesn't need a very high write speed (bitrate) yet records in good (lossless) quality. Dxtory comes with its own codec, the Dxtory codec. Which allows me some experimentation. Fraps has it's own codec which I can use in Dxtory to expirement around. I also came across http://lags.leetcode.net/codec.html . Are there more lossless codecs out there (besides Fraps' and Dxtory's) which are good for what I want to do? Edit: To clarify, yes, I'm aware a lossless codec always has "good" quality. But that's not what I'm looking for. Let me take the Fraps codec and Dxtory codec to clarify what I'm looking for. When I record with the Dxtory codec in RGB colorspace at 1920x1080 with targeted 30 fps, I can play the game at 60 fps, BUT I'm recording with 10-15 fps, that's because RGB with Dxtory needs much, much more write speed than my hdd can handle. When recording with Dxtory codec in YUV410 colorspace at 1920x1080 with targeted 30 fps, I can play at 60 fps and record at 30 fps, again, that's because YUV410 in Dxtory's codec takes much, much less write speed than RGB When recording with Fraps codec in ??? (I dunno the color space Fraps records in, I guess YUV420), I can play with 60 fps and record with 30 fps. What I'm looking for is a lossless codec that can record in YUV420 (or even RGB??) which does not exceed a write speed (or bitrate if you will) of 100 mb/s in 1920x1080 or in other words, which will allow me to record in constant 30fps. Obviously the best solution would be to buy an SDD, but that's not what I'm after.

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• ffmpeg options: -acodec libfaac -ab 192k produce 150kbit files?

  - by piedro

  Hello! When I use ffmpeg to convert an audiofile with the option -acodec libfaac -ab 192k and use ffmpeg -i on that file afterwards to get the audio file information, it tells me bitrate: 152 kb/s Why ist this? Do I miss something here? If I want to convert a file with a bitrate of 192kb it should give me 192 kbit after the conversion, shouldn't it? Or: How do I get the 192 kbit rate then?

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• How to embed/hardcode SRT subtitles into mp4 videos with VLC?

  - by Jens Bannmann

  I'm looking for a way to "burn in" or render/rembed/hardcode subtitles (from an SRT file) into an MP4 video with VLC. But no matter what options I use, it never works properly. I get a file that plays video way too fast (audio is normal), or one that plays normally, but actually does not have embedded subtitles. Also, with some options (like the one below) it does not play in QuickTime, only in VLC. So the main question is: how can I make this work in VLC? Secondary questions are: How do I decide which options I should set? Which settings are best if I want to leave the file bitrate etc. the same as much as possible, only embed subtitles? It seems I cannot leave the field empty or Video/Audio unchecked, so I guess I would first need to figure out the original audio and video bitrate. What do the "Scale" and "Channels" options mean? ... none of which are answered within the VLC documentation. For example, this is one set of options I used in the "Advanced Open File…" dialog: Advanced Open File… myFileName.mp4 [ ] Treat as a pipe rather than as a file [x] Load subtitles file: mySubtitleFileName.srt [ ] Play another media synchronously [x] Streaming/Saving Streaming and Transcoding Options [ ] Display the stream locally (o) File [outputFileName.mp4 ] [ ] Dump raw input Encapsulation Method: (MPEG 4 ) Transcoding options [x] Video (mp4v ) Bitrate (kb/s) [256 ] Scale [1 ] [x] Audio (mp3 ) Bitrate (kb/s) [128 ] Channels [1 ]

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• ffmpeg add two audio streams to video

  - by Tossin Hausen

  I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

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• mix audio with h264 mp4 video with ffmpeg

  - by user2362912

  I have 2 files : Input #0, wav, from '105426_1.wav': Duration: 00:00:09.98, bitrate: 1312 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 41000 Hz, stereo, s16, 1312 kb/s and: Duration: 00:00:41.29, start: 0.000000, bitrate: 1313 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 1211 kb/s, 24.42 fps, 25 tbr, 90k tbn, 48 tbc Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 99 kb/s Metadata: handler_name : SoundHandler I want to insert first audio file into video in special place (for example in 10 secunde of video) and mix it with audio stream of video file. I try to /usr/local/bin/ffmpeg -i 105426_1.wav -i 105426.mp4 -map 0:0 -map 1:1 -map 1:0 video_finale.mp4 but result is : Duration: 00:00:41.31, start: 0.046440, bitrate: 755 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s Metadata: handler_name : SoundHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s Metadata: handler_name : SoundHandler Stream #0:2(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 588 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc Metadata: handler_name : VideoHandler I need only one audio stream and first stream play not from beginig but from 10 sec

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• FFSERVER - streaming an ASF video as Webm output

  - by Emmanuel Brunet

  I'm trying to stream an IP webcam ASF live stream to a ffserver to output a webm video format. The server starts successfully but the ffserver commands used to feed the ffserver fails and generates a core dump. Environment Debian 7.5 ffmpeg 2.2 Input stream $ ffprobe http://account:password@webcam/videostream.asf Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, 1 channels, s16p, 32 kb/s ffserver configuration my ffserver configuration is : Port 8091 RTSPPort 554 BindAddress 192.168.1.62 MaxHTTPConnections 1000 MaxClients 100 MaxBandwidth 1000 CustomLog - <Feed webcam.ffm> File /tmp/webcam.ffm FileMaxSize 500M ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Feed> <Stream webcam.webm> # Output stream URL definition Feed webcam.ffm # Feed from which to receive video Format webm # Audio settings AudioCodec vorbis AudioBitRate 64 # Audio bitrate # Video settings VideoCodec libvpx VideoSize 640x480 # Video resolution VideoFrameRate 25 # Video FPS AVOptionVideo flags +global_header # Parameters passed to encoder # (same as ffmpeg command-line parameters) AVOptionVideo cpu-used 0 AVOptionVideo qmin 10 AVOptionVideo qmax 42 AVOptionVideo quality good AVOptionAudio flags +global_header PreRoll 15 StartSendOnKey # VideoBitRate 32 # Video bitrate </Stream> <Stream status.html> Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> ffmpeg feed I run the following command that fails $ ffmpeg -i http://account:password@webcam/videostream.asf http://192.168.1.62:8091/webcam.ffm http://192.168.1.62:8091/webcam.ffm Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x36a80c0] deprecated pixel format used, make sure you did set range correctly Segmentation fault I tryed $ ffmpeg -i http://account:password@webcam/videostream.asf -pix_fmt yuv420p http://192.168.1.62:8091/webcam.ffm But it raises the same error. Thanks for your help Edit For an easy testing (I thought), I tried to publish the whole ASF stream as is, meaning connecting the ASF webcam output stream to the ffserver that outputs ASF format too. And thus with mirrored encoding so I changed the ffserver configuration to ... <Stream webcam.asf> Feed webcam.ffm Format asf VideoFrameRate 25 VideoSize 640X480 VideoBitRate 256 VideoBufferSize 1000 VideoGopSize 30 AudioBitRate 32 StartSendOnKey </Stream> ... And the output is now : Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 1k tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x3d620c0] deprecated pixel format used, make sure you did set range correctly Output #0, ffm, to 'http://192.168.1.62:8091/webcam.ffm': Metadata: creation_time : now encoder : Lavf55.40.100 Stream #0:0: Audio: wmav2, 22050 Hz, mono, fltp, 32 kb/s Metadata: encoder : Lavc55.64.100 wmav2 Stream #0:1: Video: msmpeg4v3 (msmpeg4), yuv420p, 640x480, q=2-31, 256 kb/s, 1k fps, 1000k tbn, 1k tbc Metadata: Stream mapping: Stream #0:1 -> #0:0 (adpcm_ima_wav -> wmav2) Stream #0:0 -> #0:1 (mjpeg -> msmpeg4) Press [q] to stop, [?] for help Segmentation fault I can't even forward the stream.

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• ffserver-2.2 - streaming an ASF video as Webm output with ffserver on Debian 7.5

  - by Emmanuel Brunet

  I'm trying to stream an IP webcam ASF live stream to a ffserver to output a webm video format. The server starts successfully but the ffserver commands used to feed the ffserver fails and generates a core dump. Input stream $ ffprobe http://account:password@webcam/videostream.asf Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, 1 channels, s16p, 32 kb/s ffserver configuration my ffserver configuration is : Port 8091 RTSPPort 554 BindAddress 192.168.1.62 MaxHTTPConnections 1000 MaxClients 100 MaxBandwidth 1000 CustomLog - <Feed webcam.ffm> File /tmp/webcam.ffm FileMaxSize 500M ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Feed> <Stream webcam.webm> # Output stream URL definition Feed webcam.ffm # Feed from which to receive video Format webm # Audio settings AudioCodec vorbis AudioBitRate 64 # Audio bitrate # Video settings VideoCodec libvpx VideoSize 640x480 # Video resolution VideoFrameRate 25 # Video FPS AVOptionVideo flags +global_header # Parameters passed to encoder # (same as ffmpeg command-line parameters) AVOptionVideo cpu-used 0 AVOptionVideo qmin 10 AVOptionVideo qmax 42 AVOptionVideo quality good AVOptionAudio flags +global_header PreRoll 15 StartSendOnKey # VideoBitRate 32 # Video bitrate </Stream> <Stream status.html> Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> ffmpeg feed I run the following command that fails $ ffmpeg -i http://account:password@webcam/videostream.asf http://ffserver_ip:port/webcam.ffm http://192.168.1.62:8091/webcam.ffm Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x36a80c0] deprecated pixel format used, make sure you did set range correctly Segmentation fault I tryed $ ffmpeg -i http://account:password@webcam/videostream.asf -pix_fmt yuv420p http://ffserver_ip:port/webcam.ffm But it raises the same error. Thanks for your help Edit For an easy testing (I thought), I tried to publish the whole ASF stream as is, meaning connecting the ASF webcam output stream to the ffserver that outputs ASF format too. And thus with mirrored encoding so I changed the ffserver configuration to ... <Stream webcam.asf> Feed webcam.ffm Format asf VideoFrameRate 25 VideoSize 640X480 VideoBitRate 256 VideoBufferSize 1000 VideoGopSize 30 AudioBitRate 32 StartSendOnKey </Stream> ... And the output is now : Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 1k tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x3d620c0] deprecated pixel format used, make sure you did set range correctly Output #0, ffm, to 'http://192.168.1.62:8091/webcam.ffm': Metadata: creation_time : now encoder : Lavf55.40.100 Stream #0:0: Audio: wmav2, 22050 Hz, mono, fltp, 32 kb/s Metadata: encoder : Lavc55.64.100 wmav2 Stream #0:1: Video: msmpeg4v3 (msmpeg4), yuv420p, 640x480, q=2-31, 256 kb/s, 1k fps, 1000k tbn, 1k tbc Metadata: Stream mapping: Stream #0:1 -> #0:0 (adpcm_ima_wav -> wmav2) Stream #0:0 -> #0:1 (mjpeg -> msmpeg4) Press [q] to stop, [?] for help Segmentation fault I can't even forward the stream. Thanks for your help again.

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• Pymedia video encoding failed

  - by user1474837

  I am using Python 2.5 with Windows XP. I am trying to make a list of pygame images into a video file using this function. I found the function on the internet and edited it. It worked at first, than it stopped working. This is what it printed out: Making video... Formating 114 Frames... starting loop making encoder Frame 1 process 1 Frame 1 process 2 Frame 1 process 2.5 This is the error: Traceback (most recent call last): File "ScreenCapture.py", line 202, in <module> makeVideoUpdated(record_files, video_file) File "ScreenCapture.py", line 151, in makeVideoUpdated d = enc.encode(da) pymedia.video.vcodec.VCodecError: Failed to encode frame( error code is 0 ) This is my code: def makeVideoUpdated(files, outFile, outCodec='mpeg1video', info1=0.1): fw = open(outFile, 'wb') if (fw == None) : print "Cannot open file " + outFile return if outCodec == 'mpeg1video' : bitrate= 2700000 else: bitrate= 9800000 start = time.time() enc = None frame = 1 print "Formating "+str(len(files))+" Frames..." print "starting loop" for img in files: if enc == None: print "making encoder" params= {'type': 0, 'gop_size': 12, 'frame_rate_base': 125, 'max_b_frames': 90, 'height': img.get_height(), 'width': img.get_width(), 'frame_rate': 90, 'deinterlace': 0, 'bitrate': bitrate, 'id': vcodec.getCodecID(outCodec) } enc = vcodec.Encoder(params) # Create VFrame print "Frame "+str(frame)+" process 1" bmpFrame= vcodec.VFrame(vcodec.formats.PIX_FMT_RGB24, img.get_size(), # Covert image to 24bit RGB (pygame.image.tostring(img, "RGB"), None, None) ) print "Frame "+str(frame)+" process 2" # Convert to YUV, then codec da = bmpFrame.convert(vcodec.formats.PIX_FMT_YUV420P) print "Frame "+str(frame)+" process 2.5" d = enc.encode(da) #THIS IS WHERE IT STOPS print "Frame "+str(frame)+" process 3" fw.write(d.data) print "Frame "+str(frame)+" process 4" frame += 1 print "savng file" fw.close() Could somebody tell me why I have this error and possibly how to fix it? The files argument is a list of pygame images, outFile is a path, outCodec is default, and info1 is not used anymore. UPDATE 1 This is the code I used to make that list of pygame images. from PIL import ImageGrab import time, pygame pygame.init() f = [] #This is the list that contains the images fps = 1 for n in range(1, 100): info = ImageGrab.grab() size = info.size mode = info.mode data = info.tostring() info = pygame.image.fromstring(data, size, mode) f.append(info) time.sleep(fps)

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• AAC Sample Rate and Bit Rate for High Quality Audio?

  - by marco.ragogna

  What are the AAC Sample Rate and Bit Rate settings to set in order to encode an audio track with a quality comparable to MP3 320kbps? I need to backup a DVD movie, the default settings for AAC are Bitrate (KB/s) 128 Sample Rate (HZ) 44100 should I set Bitrate (KB/s) 320 Sample Rate (HZ) 48000 or the default are already good?

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• good, simple music streaming for windows?

  - by The Journeyman geek

  I've been using itunes to stream music over lan, and quite frankly the quality is terrible. On the other hand, its ease of use - having a simple client on either end, and it just 'working' with full control on my end is nice- the quality however is terrible - I'm wondering if there's any controllable streaming remote music serve for windows that will let me pick the bitrate it sends at, or uses whatever bitrate the original file is in, and just sends it, and has itself controllable at the client end, and will work acceptably over a wlan connection

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• Matching ID3 Tags to Existing Files

  - by SLaks

  I have a collection of ripped CDs that were transcoded to a very low bitrate. (and I lost the original MP3s) I'd like to re-rip the CDs to a higher bitrate, and apply the ID3 tags from the existing rips to the new files. (These are custom tags; they will not be found in online databases) Is there any way to automatically copy ID3 tags by length and track number, or do I need to write one myself?

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• Need to convert a video file from mp4 to xvid

  - by Shawn

  I checked out the questions with similar titles and didn't find anything that I thought would help. I am attempting to convert a video into an avi, preferably xvid. The video file's Video and Audio Properties are as follows: Video Dimensions: 1280x544 Codec H.264/AVC Framerate: 24 frames per second Bitrate: 774 kpbs Audio Codec: MPEG-4 AAC audio Channels: Stereo Sample Rate: 48000 Hz Bitrate: 32 kpbs I have tried numerous times to convert this into an Xvid codec AVI but I have had no luck successfully getting the audio to sync properly. I am using Openshot to attempt conversion, using the libxvid codec and AVI format, but I am unsure of the proper audio settings I should use. What settings should I use to convert this video with Openshot? If it is not possible with Openshot, or if there is a better application to use, I would be grateful to know that as well.

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• ffmpeg - creating DNxHD MFX files with alphas

  - by Hugh

  Hi all, I'm struggling with something in FFMpeg at the moment... I'm trying to make DNxHD 1080p/24, 36Mb/s MXF files from a sequence of PNG files. My current command-line is: ffmpeg -y -f image2 -i /tmp/temp.%04d.png -s 1920x1080 -r 24 -vcodec dnxhd -f mxf -pix_fmt rgb32 -b 36Mb /tmp/temp.mxf To which ffmpeg gives me the output: Input #0, image2, from '/tmp/temp.%04d.png': Duration: 00:00:01.60, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, rgb32, 1920x1080, 25 tbr, 25 tbn, 25 tbc Output #0, mxf, to '/tmp/temp.mxf': Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 90k tbn, 24 tbc Stream mapping: Stream #0.0 -> #0.0 [mxf @ 0x1005800]unsupported video frame rate Could not write header for output file #0 (incorrect codec parameters ?) There are a few things in here that concern me: The output stream is insisting on being yuv422p, which doesn't support alpha. 24fps is an unsupported video frame rate? I've tried 23.976 too, and get the same thing. I then tried the same thing, but writing to a quicktime (still DNxHD, though) with: ffmpeg -y -f image2 -i /tmp/temp.%04d.png -s 1920x1080 -r 24 -vcodec dnxhd -f mov -pix_fmt rgb32 -b 36Mb /tmp/temp.mov This gives me the output: Input #0, image2, from '/tmp/1274263259.28098.%04d.png': Duration: 00:00:01.60, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, rgb32, 1920x1080, 25 tbr, 25 tbn, 25 tbc Output #0, mov, to '/tmp/1274263259.28098.mov': Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 90k tbn, 24 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 39 fps= 9 q=1.0 Lsize= 7177kB time=1.62 bitrate=36180.8kbits/s video:7176kB audio:0kB global headers:0kB muxing overhead 0.013636% Which obviously works, to a certain extent, but still has the issue of being yuv422p, and therefore losing the alpha. If I'm going to QuickTime, then I can get what I need using Shake, but my main aim here is to be able to generate .mxf files. Any thoughts? Thanks

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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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• ffmpeg - creating DNxHD MFX files with alphas

  - by Hugh

  I'm struggling with something in FFMpeg at the moment... I'm trying to make DNxHD 1080p/24, 36Mb/s MXF files from a sequence of PNG files. My current command-line is: ffmpeg -y -f image2 -i /tmp/temp.%04d.png -s 1920x1080 -r 24 -vcodec dnxhd -f mxf -pix_fmt rgb32 -b 36Mb /tmp/temp.mxf To which ffmpeg gives me the output: Input #0, image2, from '/tmp/temp.%04d.png': Duration: 00:00:01.60, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, rgb32, 1920x1080, 25 tbr, 25 tbn, 25 tbc Output #0, mxf, to '/tmp/temp.mxf': Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 90k tbn, 24 tbc Stream mapping: Stream #0.0 -> #0.0 [mxf @ 0x1005800]unsupported video frame rate Could not write header for output file #0 (incorrect codec parameters ?) There are a few things in here that concern me: The output stream is insisting on being yuv422p, which doesn't support alpha. 24fps is an unsupported video frame rate? I've tried 23.976 too, and get the same thing. I then tried the same thing, but writing to a quicktime (still DNxHD, though) with: ffmpeg -y -f image2 -i /tmp/temp.%04d.png -s 1920x1080 -r 24 -vcodec dnxhd -f mov -pix_fmt rgb32 -b 36Mb /tmp/temp.mov This gives me the output: Input #0, image2, from '/tmp/1274263259.28098.%04d.png': Duration: 00:00:01.60, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, rgb32, 1920x1080, 25 tbr, 25 tbn, 25 tbc Output #0, mov, to '/tmp/1274263259.28098.mov': Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 90k tbn, 24 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 39 fps= 9 q=1.0 Lsize= 7177kB time=1.62 bitrate=36180.8kbits/s video:7176kB audio:0kB global headers:0kB muxing overhead 0.013636% Which obviously works, to a certain extent, but still has the issue of being yuv422p, and therefore losing the alpha. If I'm going to QuickTime, then I can get what I need using Shake, but my main aim here is to be able to generate .mxf files. Any thoughts? Thanks

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• Reduce size of MP4

  - by testing

  I have a MP4 file with a lengh of 22:44. Here are the details: Video: width: 720 px height: 404 px data bitrate: 1022 kBit/s overall bitrate: 1182 kBit/s fps: 24 codec: H264 - MPEG4 AVC (part 10) (avc1) Audio: bitrate: 159 kBit/s stereo sample rate: 48 kHz codec: MPEG AAC Audio (mp4a) I thought I can reduce the current filesize (about 200 MB) by reducing the width and the height (420 x 236). I tried different programs: Handbrake, Format Factory, Next Video Converter and Super. The first three didn't worked as expected: Handbrake has a bug by setting the width and the height, the next two doesn't allow the fine setting of the videosize (only presets of width and height). Super seems to be the best, but I didn't found a setting which reduces the file size. I reduced the width and the height but only got 20 MB less. Now I tried the xth setting and I still get a too high file size. I want to reduce the filesize to 100 MB or less. The ouput format should be FLV or MP4, because I need this for flowplayer. Which settings of SUPER or which program should I use to reduce the file size? Of course the video should still be viewable.

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• Media Information Extractor for Java

  - by eyazici

  I need a media information extraction library (pure Java or JNI wrapper) that can handle common media formats. I primarily use it for video files and I need at least these information: Video length (Runtime) Video bitrate Video framerate Video format and codec Video size (width X height) Audio channels Audio format Audio bitrate and sampling rate There are several libraries and tools around but I couldn't find for Java.

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