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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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• Is ffmpeg incorrectly interpreting .aif files?

  - by marue

  Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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• getAudioInputStream can not convert [stereo, 4 bytes/frame] stream to [mono, 2 bytes/frame]

  - by brian_d

  Hello. I am using javasound and have an AudioInputStream of format PCM_SIGNED 8000.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian Using AudioSystem.getAudioInputStream(target_format, original_stream) produces an 'IllegalArgumentException: Unsupported Conversion' when the target_format is PCM_SIGNED 8000.0 Hz, 16 bit, mono, 2 bytes/frame, little-endian Is it possible to convert this stream manually after every read() call? And if yes, how? In general, how can you compare two formats and tell if a conversion is possible?

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• Simple wave generator with SDL in c++

  - by Vlad Popescu

  i am having problems understanding how the audio part of the sdl library works now, i know that when you initialize it, you have to specify the frequency and a callback<< function, which i think is then called automatically at the given frequency. can anyone who worked with the sdl library write a simple example that would use sdl_audio to generate a 440 hz square wave (since it is the simplest waveform) at a sampling frequency of 44000 hz? thanks in advance

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• Django Getting RequestContext in custom tag

  - by greggory.hz

  I'm trying to create a custom tag. Inside this custom tag, I want to be able to have some logic that checks if the user is logged in, and then have the tag rendered accordingly. This is what I have: class UserActionNode(template.Node): def __init__(self): pass def render(self, context): if context.user.is_authenticated(): return render_to_string('layout_elements/sign_in_register.html'); else: return render_to_string('layout_elements/logout_settings.html'); def user_actions(parser, test): return UserActionNode() register.tag('user_actions', user_actions) When I run this, I get this error: Caught AttributeError while rendering: 'Context' object has no attribute 'user' The view that renders this looks like this: return render_to_response('start/home.html', {}, context_instance=RequestContext(request)) Why doesn't the tag get a RequestContext object instead of the Context object? How can I get the tag to receive the RequestContext instead of the Context? EDIT: Whether or not it's possible to get a RequestContext inside a custom tag, I'd still be interested to know the "correct" or best way to determine a user's authentication state from within the custom tag. If that's not possible, then perhaps that kind of logic belongs elsewhere? Where?

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• Django Find Out if User is Authenticated in Custom Tag

  - by greggory.hz

  I'm trying to create a custom tag. Inside this custom tag, I want to be able to have some logic that checks if the user is logged in, and then have the tag rendered accordingly. This is what I have: def user_actions(context): request = template.Variable('request').resolve(context) return { 'auth': request['user'].is_athenticated() } register.inclusion_tag('layout_elements/user_actions.html', takes_context=True)(user_actions) When I run this, I get this error: Caught VariableDoesNotExist while rendering: Failed lookup for key [request] in u'[{}]' The view that renders this ends like this: return render_to_response('start/home.html', {}, context_instance=RequestContext(request)) Why doesn't the tag get a RequestContext object instead of the Context object? How can I get the tag to receive the RequestContext instead of the Context? EDIT: Whether or not it's possible to get a RequestContext inside a custom tag, I'd still be interested to know the "correct" or best way to determine a user's authentication state from within the custom tag. If that's not possible, then perhaps that kind of logic belongs elsewhere? Where?

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• Can you write files in Chrome 8?

  - by greggory.hz

  I'm wondering if, with the new File API exposed in Chrome (I'm not concerned with cross-browser support at this time), it would be possible to write back to files opened via a file input. You can see an example of what I'm trying to accomplish here: http://www.grehz.com/ide. I know I can use server side scripts to dynamically create the files and allow the user to download them normally. I'm hoping that there's a way to accomplish this purely client side. I had read somewhere that you can write to files opened via a file input. I haven't been able to find any examples of this, though I have seen passing references to a FileWriter class. I would be completely not surprised if this wasn't possible though (it seems likely that there are security issues with this). Just looking for some guidance or resources. UPDATE: I was reading here: http://dev.w3.org/2009/dap/file-system/file-writer.html As I was playing around in Chrome, it looks like FileSaver and FileWriter are not implemented, but BlobBuilder is. I can call getBlob() on the BB object, is there any way I can then save that without FileSave or FileWriter?

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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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• 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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• Segmentation fault while feeding in an mpeg file through ffmpeg

  - by angel6

  Hi, I've set up FFserver as the streaming server. I'm trying to feed in an mpeg file. But it comes up with a segmentation fault. Does anyone know how to fix this? The following is the command-line output I get $ ./ffmpeg -i test1.mpg http://localhost:8090/feed1.ffm FFmpeg version SVN-r22945, Copyright (c) 2000-2010 the FFmpeg developers built on Apr 22 2010 19:18:45 with gcc 4.4.1 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libx264 --enable-libxvid --enable-x11grab libavutil 50.14. 0 / 50.14. 0 libavcodec 52.66. 0 / 52.66. 0 libavformat 52.61. 0 / 52.61. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0xab0c420]max_analyze_duration reached Input #0, mpeg, from 'test1.mpg': Duration: 00:00:20.96, start: 0.768300, bitrate: 269 kb/s Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 160x120 [PAR 1:1 DAR 4:3], 104857 kb/s, 30 fps, 30 tbr, 90k tbn, 30 tbc Stream #0.1[0x1c0]: Audio: mp2, 32000 Hz, 2 channels, s16, 64 kb/s Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Metadata: encoder : Lavf52.61.0 Stream #0.0: Audio: mp2, 22050 Hz, 1 channels, s16, 48 kb/s Stream #0.1: Video: mpeg1video, yuv420p, 160x128, q=2-31, 40 kb/s, 1000k tbn, 50 tbc Stream #0.2: Audio: libmp3lame, 22050 Hz, 1 channels, s16, 64 kb/s Stream #0.3: Video: msmpeg4, yuv420p, 352x240, q=2-31, 256 kb/s, 1000k tbn, 15 tbc Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Stream #0.1 -> #0.2 Stream #0.0 -> #0.3 Press [q] to stop encoding Segmentation fault

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• Convert swf file to mp4 file using FFMPEG

  - by user1624004

  I now want to show an html5 video on a html page. Now I have an sample.swf file, I want to convert it to .mp4 or .ogg or .webm file. I have tried: ffmpeg -i sample.swf sample.mp4 But I got this error: [swf @ 0000000001feef40] Could not find codec parameters for stream 0 (Audio: pcm_s16le, 5512 Hz, 1 channels, 88 kb/s): unspecified sample format Consider increasing the value for the 'analyzeduration' and 'probesize' options [swf @ 0000000001feef40] Estimating duration from bitrate, this may be inaccurate Guessed Channel Layout for Input Stream #0.0 : mono Input #0, swf, from 'sample.swf': Duration: N/A, bitrate: N/A Stream #0:0: Audio: pcm_s16le, 5512 Hz, mono, 88 kb/s Stream #0:1: Video: mjpeg, yuvj444p, 1024x768 [SAR 100:100 DAR 4:3], 16 fps, 16 tbr, 16 tbn File 'sample.mp4' already exists. Overwrite ? [y/N] y Invalid sample format '(null)' Error opening filters!

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• map based on specific type of stream

  - by Steven Penny

  Consider the following file Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'infile.mp4': Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1038 Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 386 kb/s Stream #0:2(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 124 kb/s I would like to run a command that uses the video and only the stereo audio. With this particular file I could just run ffmpeg -i infile.mp4 -c copy -map v -map :2 outfile.mp4 However this will not work if the audio streams are switched. How can I tell FFmpeg to use the 2 channel audio only?

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• How can I convert audio files to this format?

  - by jeffamaphone

  I have a bunch of audio files that are named .wav but it seems not all .wavs are created equal. For example: $ file * file1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz file2.wav: Audio file with ID3 version 2.2.0, contains: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo file3.wav: Claris clip art? file4.wav: Audio file with ID3 version 2.2.0, contains: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo And for good measure, a non-wav: file5.m4a: ISO Media, MPEG v4 system, iTunes AAC-LC I would like to convert all of these files to the format that file1.wav is: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz What is the proper set of arguments to pass to afconvert to make that happen?

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• Set Display Refresh Rate in OSX w/o External Utilities

  - by codedonut

  I have an iMac and an LG Flatron connected as a secondary monitor. The recommended resolution for the flatron is 1680x1050 @ 65.290 Hz (horiz), 59.954 Hz (vert). For some reason, OSX is choosing a slightly different set of scan rates and this is currently my best guess of why the monitor goes into power saving mode when connected to the iMac (but works fine on a PC). Now, I resolved this by installing switchResX and fudging the scan rates according to the specs in the manual. But how does one change these rates w/o 3rd party tools? Which config files need editing? Thanks

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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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• Meaning of tcp_delack_min

  - by Phi

  Hi, the current Linux Kernel (e.g. 2.6.36) uses Delayed Acknowledgments (delack). In /include/net/tcp.h it says: define TCP_DELACK_MIN ((unsigned)(HZ/25)) So, for a Kernel using a HZ value of 1000, an ACK should be delayed by a minimum of 40 ms. However, RFC 2581 says a TCP implementation should acknowledge every second full sized segment without further delay. Does anybody know whether the Linux Kernel follows that 'should' or whether the TCP_DELACK_MIN value means that even after a full sized segment was received, the ACK continues to be delayed until 40 ms have passed?

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• Monitor resolution messed up somehow

  - by Kelp

  I purchased the Westinghouse 22" LCD LCM-22w3 a few years ago, and now it's been acting up on me. I just booted into Windows 7(without changing any settings), and the default resolution is 1600x1024, and it allows me to select a refresh rate of up to 85 Hz(it didn't let me do that). I usually have my resolution set to 1680x1050 with a refresh rate of 60 Hz. Now, that resolution does not even appear in the list. Does anyone have any idea of what could be the problem and how to fix it? Edit: I am not sure if this will help, but when I go to change the screen resolution, the monitor is known as "Generic Non-PnP Monitor". It used to be referred to as "Generic PnP Monitor). I tried to disable Generic Non-PnP Monitor, but when I restart, it uses that monitor again. Edit 2: I created a custom .inf file using Powerstrip, but that does not work either. The monitor settings are being stubborn.

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• How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab [closed]

  - by martin

  This is all done in MatLab 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the various sampling at different frequencies?

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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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• When spliting MP4s with ffmpeg how do I include metadata?

  - by Josh

  I have a few MP4s that i want to upload to my flickr account but they have a maximum size of 500mb as mine is only about 550 i was planing to simply split them in half then upload them, but i want to make sure all the meta data is included but it does not seem to be. I have tried each of the following with no luck, (at the end of this post i have the original and the new ffprobe outputs): ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_metadata 0:0 SANY0069A.MP4 ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_meta_data SANY0069.MP4:SANY0069A.MP4 SANY0069A.MP4 with the this one I manually produced the individual meta tags that i took from this command ffmpeg -i SANY0069A.MP4 -f ffmetadata meta.txt ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -metadata major_brand="mp42" -metadata minor_version="1" -metadata compatible_brands="mp42avc1" -metadata creation_time="2012-09-29 09:05:50" -metadata comment="SANYO DIGITAL CAMERA CA9" -metadata comment-eng="SANYO DIGITAL CAMERA CA9" SANY0069A.MP4 using the output of the former command i also tried this: ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -f ffmetadata -i meta.txt SANY0069A.MP4 Output: sample output from my first command: ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_metadata 0:0 SANY0069A.MP4 ffmpeg version 0.8.12, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 Duration: 00:08:38.71, start: 0.000000, bitrate: 9142 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9007 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 File 'SANY0069A.MP4' already exists. Overwrite ? [y/N] y Output #0, mp4, to 'SANY0069A.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 encoder : Lavf53.5.0 Stream #0.0(eng): Video: libx264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], q=2-31, 9007 kb/s, 30k tbn, 29.97 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help frame= 7773 fps=4644 q=-1.0 Lsize= 289607kB time=00:04:19.35 bitrate=9147.4kbits/s video:285416kB audio:4033kB global headers:0kB muxing overhead 0.054571% and finaly, when i compare the ffprobe of the original and the first split part i get the 2 following outputs: original ffprobe version 0.8.12, Copyright (c) 2007-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 Duration: 00:08:38.71, start: 0.000000, bitrate: 9142 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9007 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 Split ffprobe version 0.8.12, Copyright (c) 2007-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069A.MP4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf53.5.0 comment : SANYO DIGITAL CAMERA CA9 Duration: 00:04:19.37, start: 0.000000, bitrate: 9146 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9015 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 1970-01-01 00:00:00 I know this is incredibly long but its actually a quite simple question. I thought it would be best to provide as much detail as possible. any advice here would be great, Thanks

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• ffmpeg 0.5 flv to wav conversion creates wav files that other programs won't open.

  - by superrebel

  Hi, I am using the following command to convert FLV files to audio files to feed into julian, a speech to text program. cat ./jon2.flv | ffmpeg -i - -vn -acodec pcm_s16le -ar 16000 -ac 1 -f wav - | cat - > jon2.wav The cat's are there for debugging purposes as the final use will be a running program that will pipe FLV into ffmpeg's stdin and the stdout going to julian. The resulting wave files are identified by "file" as: jon3.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz VLC (based on ffmpeg) plays the file, but no other tools will open/see the data. They show empty wav files or won't open/play. For example Sound Booth from CS4. Has anyone else had similar problems? Julian requires wav files 16bit mono at 16000 Hz. Julian does seem to read the file, but doesn't seem to go through the entire file (may be unrelated). Thanks, -rr

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• Microsecond (or one ms) time resolution on an embedded device (Linux Kernel)

  - by ChrisDiRulli

  Hey guys, I have a kernel module I've built that requires at least 1 ms time resolution. I currently use do_gettimeofday() but I'm concerned that this won't work once I move my module to an embedded device. The device has a 180 Mz processor (MIPS) and the default HZ value in the kernel is 100. Thus using jiffies will only give me at best 10 ms resolution. That won't cut it. What I'd like to know is if do_gettimeofday() is based on the timer interrupt (HZ). Can it be guaranteed to provide at least 1 ms of resolution? Thanks!

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• [C++] Wrong EOF when unzipping binary file

  - by djzmo

  Hello there, I tried to unzip a binary file to a membuf from a zip archive using Lucian Wischik's Zip Utils: http://www.wischik.com/lu/programmer/zip_utils.html http://www.codeproject.com/KB/files/zip_utils.aspx FindZipItem(hz, filename.c_str(), true, &j, &ze); char *content = new char[ze.unc_size]; UnzipItem(hz, j, content, ze.unc_size); delete[] content; But it didn't unzip the file correctly. It stopped at the first 0x00 of the file. For example when I unzip an MP3 file, it will only unzip the first 4 bytes: 0x49443303 (ID3\0) because the 5th to 8th byte is 0x00000000. I also tried to capture the ZR_RESULT, and it always return ZR_OK (which means completed without errors). I think this guy also had the same problem, but no one replied to his question: http://www.codeproject.com/KB/files/zip_utils.aspx?msg=2876222#xx2876222xx Any kind of help would be appreciated :)

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• Why can't I convert FLV to MP4 format using FFmpeg when MP3 works?

  - by hugemeow

  In fact I have succeeded to convert FLV to MP3: D:\tmp\ffmpeg-20121005-git-d9dfe9a-win64-static\ffmpeg-20121005-git-d9dfe9a-win 4-static\bin>ffmpeg.exe -i a.flv -acodec mp3 a.mp3 ffmpeg version N-45080-gd9dfe9a Copyright (c) 2000-2012 the FFmpeg developers built on Oct 5 2012 16:49:01 with gcc 4.7.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-run ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass -enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable- ibfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopen peg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libthe ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-l bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --en ble-zlib libavutil 51. 73.102 / 51. 73.102 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, flv, from 'a.flv': Metadata: metadatacreator : iku hasKeyframes : true hasVideo : true hasAudio : true hasMetadata : true canSeekToEnd : false datasize : 16906383 videosize : 14558526 audiosize : 2270465 lasttimestamp : 530 lastkeyframetimestamp: 529 lastkeyframelocation: 16893721 Duration: 00:08:49.73, start: 0.000000, bitrate: 255 kb/s Stream #0:0: Video: h264 (Main), yuv420p, 448x336 [SAR 1:1 DAR 4:3], 218 kb s, 15 tbr, 1k tbn, 30 tbc Stream #0:1: Audio: aac, 44100 Hz, stereo, s16, 32 kb/s File 'a.mp3' already exists. Overwrite ? [y/N] y Output #0, mp3, to 'a.mp3': Metadata: metadatacreator : iku hasKeyframes : true hasVideo : true hasAudio : true hasMetadata : true canSeekToEnd : false datasize : 16906383 videosize : 14558526 audiosize : 2270465 lasttimestamp : 530 lastkeyframetimestamp: 529 lastkeyframelocation: 16893721 TSSE : Lavf54.29.105 Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16 Stream mapping: Stream #0:1 -> #0:0 (aac -> libmp3lame) Press [q] to stop, [?] for help size= 8279kB time=00:08:49.78 bitrate= 128.0kbits/s video:0kB audio:8278kB subtitle:0 global headers:0kB muxing overhead 0.006842% But I failed to convert FLV to MP4. Why is the encoder 'mp4' unknown? What's more, how can I find the codecs which are already supported by my FFmpeg? D:\tmp\ffmpeg-20121005-git-d9dfe9a-win64-static\ffmpeg-20121005-git-d9dfe9a-win6 4-static\bin>ffmpeg.exe -i a.flv -acodec mp4 aa.mp4 ffmpeg version N-45080-gd9dfe9a Copyright (c) 2000-2012 the FFmpeg developers built on Oct 5 2012 16:49:01 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. 73.102 / 51. 73.102 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, flv, from 'a.flv': Metadata: metadatacreator : iku hasKeyframes : true hasVideo : true hasAudio : true hasMetadata : true canSeekToEnd : false datasize : 16906383 videosize : 14558526 audiosize : 2270465 lasttimestamp : 530 lastkeyframetimestamp: 529 lastkeyframelocation: 16893721 Duration: 00:08:49.73, start: 0.000000, bitrate: 255 kb/s Stream #0:0: Video: h264 (Main), yuv420p, 448x336 [SAR 1:1 DAR 4:3], 218 kb/ s, 15 tbr, 1k tbn, 30 tbc Stream #0:1: Audio: aac, 44100 Hz, stereo, s16, 32 kb/s Unknown encoder 'mp4'

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