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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
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Sur d’autres sites (9173)
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FFMpeg generating screenshots drop frames
6 octobre 2011, par MorganI'm using FFMpeg to extract frames from a video file, and for most videos it works flawlessly. But certain videos it seems to have to seek to the position that I want to take the screenshot from, and drops all the frames on the way there taking quite a while if the position i'm generating a screenshot from is near the end of the video. Here is the command I'm using to extract my screenshot.
ffmpeg -ss 60 -i input.mkv -vframes 1 -f image2 -an -r 1/5 output.jpg
and the output I receive
ffmpeg version 0.8.3, Copyright (c) 2000-2011 the FFmpeg developers
built on Sep 20 2011 00:46:47 with clang 2.1 (tags/Apple/clang-163.7.1)
configuration: --prefix=/Users/Morgan/Downloads/installed/ffmpeg --extra-libs=/Users/Morgan/Downloads/installed/lib/libfaac.a --enable-gpl --extra-cflags=-I/Users/Morgan/Downloads/installed/include --disable-ffplay --disable-ffserver --disable-shared --disable-debug --extra-ldflags=-L/Users/Morgan/Downloads/installed/lib --enable-libmp3lame --enable-libfaac --enable-libx264 --enable-nonfree --cc=clang
libavutil 51. 9. 1 / 51. 9. 1
libavcodec 53. 7. 0 / 53. 7. 0
libavformat 53. 4. 0 / 53. 4. 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
[matroska,webm @ 0x7f9dd3007c00] Estimating duration from bitrate, this may be inaccurate
Seems stream 0 codec frame rate differs from container frame rate: 47.95 (500000/10427) -> 23.98 (24000/1001)
Input #0, matroska,webm, from 'input.mkv':
Duration: 01:01:36.16, start: 0.000000, bitrate: 576 kb/s
Stream #0.0(eng): Video: h264 (High), yuv420p, 1280x720, PAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
Metadata:
title : Input Video
Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s (default)
Metadata:
title : English Dolby Digital 5.1 AC3
Stream #0.2(eng): Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s
Metadata:
title : Commentary
Stream #0.3(eng): Subtitle: [0][0][0][0] / 0x0000 (default)
Metadata:
title : English
Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p'
[buffer @ 0x106c10c00] w:1280 h:720 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param:
[buffersink @ 0x106c10e40] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out'
[scale @ 0x106c11140] w:1280 h:720 fmt:yuv420p -> w:1280 h:720 fmt:yuvj420p flags:0x4
Output #0, image2, to 'output.jpg':
Metadata:
encoder : Lavf53.4.0
Stream #0.0(eng): Video: mjpeg, yuvj420p, 1280x720 [PAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 0.20 tbc (default)
Metadata:
title : Input Video
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop, [?] for help
frame= 0 fps= 0 q=0.0 Lsize= -0kB time=00:00:00.00 bitrate= 0.0kbits/s dup=0 drop=275
video:0kB audio:0kB global headers:0kB muxing overhead -inf% -
Chrome times out on streaming FFMPEG output from ASP.NET Web Api
3 août 2014, par Hayden McAfeeI’ve got a unique problem here !
UPDATE 2 So it turns out the development below is FALSE, the inconsistency of the bug made it seem like not closing the stream made it work... but in fact the same issue persists !
UPDATE Interesting development ; if I comment outffmpegBufferedIn.Close();below, the entire stream always goes through fine... the request just never ends. What could be going on here ?I’m writing a web service that stores audio files in Azure Blob Storage, and converts them to MP3 live when requested through my ASP.NET Web API endpoint. I accomplish this by using ’DownloadToStream’ via the Azure Storage API, feeding that stream through the STDIN of an FFMPEG process, and sending the STDOUT stream as the request response.
The block of code that does this looks like this :
public HttpResponseMessage Get(Guid songid)
{
// This could take awhile.
HttpContext.Current.Server.ScriptTimeout = 600;
Process ffmpeg = new Process();
ProcessStartInfo startinfo = new ProcessStartInfo(HostingEnvironment.MapPath("~/App_Data/executables/ffmpeg.exe"), "-i - -vn -ar 44100 -ac 2 -ab 192k -f mp3 - ");
startinfo.RedirectStandardError = true;
startinfo.RedirectStandardOutput = true;
startinfo.RedirectStandardInput = true;
startinfo.UseShellExecute = false;
startinfo.CreateNoWindow = true;
ffmpeg.StartInfo = startinfo;
ffmpeg.ErrorDataReceived += ffmpeg_ErrorDataReceived;
// Our response is a stream
var response = Request.CreateResponse();
response.StatusCode = HttpStatusCode.OK;
// Retrieve storage account from connection string.
CloudStorageAccount storageAccount = CloudStorageAccount.Parse(
CloudConfigurationManager.GetSetting("StorageConnectionString"));
// Create the blob client.
CloudBlobClient blobClient = storageAccount.CreateCloudBlobClient();
// Retrieve reference to a previously created container.
CloudBlobContainer container = blobClient.GetContainerReference("songs");
// Retrieve reference to a blob
CloudBlockBlob blockBlob = container.GetBlockBlobReference(songid.ToString());
ffmpeg.Start();
ffmpeg.BeginErrorReadLine();
// Buffer the streams
var ffmpegBufferedIn = new BufferedStream(ffmpeg.StandardInput.BaseStream);
var ffmpegBufferedOut = new BufferedStream(ffmpeg.StandardOutput.BaseStream);
blockBlob.DownloadToStreamAsync(ffmpegBufferedIn).ContinueWith((t) => {
ffmpegBufferedIn.Flush();
ffmpegBufferedIn.Close();
});
response.Content = new StreamContent(ffmpegBufferedOut);
response.Content.Headers.ContentType = new MediaTypeHeaderValue("audio/mpeg");
System.Diagnostics.Debug.WriteLine("Returned response.");
return response;
}This works quite well in all browsers - all except for Chrome, which has an interesting way of buffering audio streams. Chrome will buffer the first 2 megabytes of a stream, then keep the connection open and wait until the user gets closer to playing the next segment of a file before consuming the rest of the stream. This should be fine - and for some songs it is. For others, I get this :
At first I thought this was due to some kind of timeout - But it happens at a different time and size for each file. It is consistent within about 15 seconds on the same songs, however. The output on the server side is normal - no exceptions thrown, and FFMpeg finishes encoding the song successfully.
Here’s the server-side output of the above request :
ffmpeg version N-64919-ga613257 Copyright (c) 2000-2014 the FFmpeg developers
built on Jul 23 2014 00:27:32 with gcc 4.8.3 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
libavutil 52. 92.101 / 52. 92.101
libavcodec 55. 69.100 / 55. 69.100
libavformat 55. 48.101 / 55. 48.101
libavdevice 55. 13.102 / 55. 13.102
libavfilter 4. 11.102 / 4. 11.102
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100
libpostproc 52. 3.100 / 52. 3.100
Input #0, mp3, from 'pipe:':
Metadata:
TSRC : AUUM71001516
title : Sunlight
track : 2
artist : Bag Raiders
copyright : 2010 Modular Recordings
genre : Electronic
album : Bag Raiders
album_artist : Bag Raiders
disc : 1/1
publisher : Modular Recordings
composer : Chris Stracey/Jack Glass/Dan Black
date : 2010
Duration: N/A, start: 0.000000, bitrate: 320 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg), 600x600 [SAR 300:300 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
Metadata:
title :
comment : Other
Output #0, mp3, to 'pipe:':
Metadata:
TSRC : AUUM71001516
TIT2 : Sunlight
TRCK : 2
TPE1 : Bag Raiders
TCOP : 2010 Modular Recordings
TCON : Electronic
TALB : Bag Raiders
TPE2 : Bag Raiders
TPOS : 1/1
TPUB : Modular Recordings
TCOM : Chris Stracey/Jack Glass/Dan Black
TDRL : 2010
TSSE : Lavf55.48.101
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 192 kb/s
Metadata:
encoder : Lavc55.69.100 libmp3lame
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (native) -> mp3 (libmp3lame))
size= 6kB time=00:00:00.21 bitrate= 227.6kbits/s
size= 102kB time=00:00:04.31 bitrate= 193.7kbits/s
size= 202kB time=00:00:08.56 bitrate= 192.9kbits/s
size= 341kB time=00:00:14.49 bitrate= 192.5kbits/s
size= 489kB time=00:00:20.82 bitrate= 192.4kbits/s
size= 642kB time=00:00:27.35 bitrate= 192.3kbits/s
size= 792kB time=00:00:33.75 bitrate= 192.2kbits/s
size= 950kB time=00:00:40.49 bitrate= 192.2kbits/s
size= 1106kB time=00:00:47.15 bitrate= 192.2kbits/s
size= 1258kB time=00:00:53.63 bitrate= 192.1kbits/s
size= 1415kB time=00:01:00.31 bitrate= 192.1kbits/s
size= 1563kB time=00:01:06.66 bitrate= 192.1kbits/s
size= 1710kB time=00:01:12.90 bitrate= 192.1kbits/s
size= 1857kB time=00:01:19.17 bitrate= 192.1kbits/s
size= 2008kB time=00:01:25.63 bitrate= 192.1kbits/s
size= 2162kB time=00:01:32.21 bitrate= 192.1kbits/s
size= 2299kB time=00:01:38.03 bitrate= 192.1kbits/s
size= 2457kB time=00:01:44.80 bitrate= 192.1kbits/s
size= 2600kB time=00:01:50.89 bitrate= 192.1kbits/s
size= 2755kB time=00:01:57.52 bitrate= 192.1kbits/s
size= 2864kB time=00:02:02.17 bitrate= 192.1kbits/s
size= 3022kB time=00:02:08.88 bitrate= 192.1kbits/s
size= 3172kB time=00:02:15.31 bitrate= 192.1kbits/s
size= 3284kB time=00:02:20.06 bitrate= 192.1kbits/s
size= 3385kB time=00:02:24.40 bitrate= 192.1kbits/s
size= 3529kB time=00:02:30.51 bitrate= 192.0kbits/s
size= 3687kB time=00:02:37.25 bitrate= 192.0kbits/s
size= 3838kB time=00:02:43.71 bitrate= 192.0kbits/s
size= 3988kB time=00:02:50.11 bitrate= 192.0kbits/s
size= 4138kB time=00:02:56.53 bitrate= 192.0kbits/s
size= 4279kB time=00:03:02.54 bitrate= 192.0kbits/s
size= 4408kB time=00:03:08.03 bitrate= 192.0kbits/s
size= 4544kB time=00:03:13.85 bitrate= 192.0kbits/s
size= 4683kB time=00:03:19.78 bitrate= 192.0kbits/s
size= 4805kB time=00:03:24.95 bitrate= 192.0kbits/s
size= 4939kB time=00:03:30.67 bitrate= 192.0kbits/s
size= 5049kB time=00:03:35.38 bitrate= 192.0kbits/s
size= 5141kB time=00:03:39.32 bitrate= 192.0kbits/s
size= 5263kB time=00:03:44.49 bitrate= 192.0kbits/s
size= 5372kB time=00:03:49.17 bitrate= 192.0kbits/s
The thread 0xb24 has exited with code 259 (0x103).
size= 5436kB time=00:03:51.91 bitrate= 192.0kbits/s
size= 5509kB time=00:03:55.02 bitrate= 192.0kbits/s
size= 5657kB time=00:04:01.32 bitrate= 192.0kbits/s
size= 5702kB time=00:04:03.22 bitrate= 192.0kbits/s
video:0kB audio:5701kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.005738%Any ideas ? I’m grateful for suggestions - I’ve been chasing this for a week now !
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ffmpeg reencoding of flv files from MAC
22 septembre 2011, par Nick MitinI've recorded a video from a webcam on mac and now i'm trying to reencode it, but ffmpeg does not recognize audio stream :
FFmpeg version git-120610e, Copyright (c) 2000-2010 the FFmpeg developers built on Sep 21 2010 15:56:57 with gcc 4.4.1 configuration : —enable-gpl —enable-version3 —enable-nonfree —enable-postproc —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-libtheora —enable-libvorbis —enable-libmp3lame —enable-libx264 libavutil 50.27. 0 / 50.27. 0 libavcore 0. 9. 0 / 0. 9. 0 libavcodec 52.89. 0 / 52.89. 0 libavformat 52.78. 5 / 52.78. 5 libavdevice 52. 2. 2 / 52. 2. 2 libavfilter 1.39. 0 / 1.39. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [flv @ 0x1e79470] Estimating duration from bitrate, this may be inaccurate Input #0, flv, from '10125174c09241f6536ccbe503ebbc00.flv' : Duration : 00:00:22.68, start : 0.000000, bitrate : N/A Stream #0.0 : Video : flv, yuv420p, 640x480, 1k tbr, 1k tbn, 1k tbc Stream #0.1 : Audio : [0][0][0][0] / 0x0000, 0 channelsIs it possible to make ffmpeg to support audio stream form flv recorded on MAC ?
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