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  • Quality gets worse using ffmpeg and Flash

    - by HOpety
    I have bunch of flash videos and am adding my brand to all of them. The problem is quality gets worse. I am doing with this command: ffmpeg -i /input.flv -vhook "/usr/loca/vhook/drawtext.so -f /usr/share/fonts/somefont.ttf -x 5 -y 5 t MyBrand" -f flv -s 320x240 - | flvtools2 -U stdin /output.flv Please tell me what I am doing wrong. I need the same quality.

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  • ffmpeg logo blur

    - by ime
    i add logo on video with ffmpeg and in some videos it blurs. how to make logo independent from videos? i mean so that it no matter what quality of video logo will be good.

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  • ffmpeg help converting

    - by ellman121
    so I've been trying to reencode a .mp4 video I have into a .avi so my cousin can use it on his Windows machine. He's not very tech-savvy, and doesn't want to deal with downloading any new programs to open .mp4 videos, but thats beside the point. The current string I'm using is ffmpeg -i Courage.Under.Fire.1996.BRRip.H264.AAC.5.1ch.Gopo.mp4 -sameq -acodec copy -vcodec copy CourageUnderFire.avi It produces the video, however doesn't give me any audio. Any assistance?

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  • X Error of failed request: BadMatch (invalid parameter attributes) ffmpeg error

    - by Evan Carroll
    I'm getting the following error message in ffmpeg: X Error of failed request: BadMatch (invalid parameter attributes) Major opcode of failed request: 140 (MIT-SHM) Minor opcode of failed request: 4 (X_ShmGetImage) Serial number of failed request: 11 Current serial number in output stream: 11 I turns up when I run the bash function mentioned in a forum post about streaming in Linux. What does it mean and how can I fix it?

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  • Video for HTML5 -- ffmpeg commands

    - by StackOverflowNewbie
    I'd like to allow my users to upload their videos (no clue what format they'll be) and convert them into the following formats: MP4 Ogg Webm I think it's those 3 formats I need in order to support HTML5, at least per http://mediaelementjs.com/. I've tried various commands that I found on the web. Some of them worked, some of them used old syntax, some of them worked only on my computer and not others, some of them gave conflicting information, etc. Are there any FFMPEG experts here that can provide the "proper" commands? I'm not particularly trying to achieve anything special. I just want to be able to convert the video into something playable on the web. Highest quality, smallest filesize, etc. are the basic goals. Something that works without a lot of special configurations would be ideal, too. I ran into a lot of "missing presets" problems.

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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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  • have ffmpeg scan and report correct time

    - by acidzombie24
    I am encoding a section of a song. I used -ss offset -t 30 (duration). When i use -i file.acc i see it says the audio is 31, 32 and once 36 seconds long. Opening it in vlc showed it as 30sec after a few seconds of playback. My code needs to filter sounds more then 30 seconds. I can fudge it and allow 30.99 (maybe 20.48 is better) however 2 seconds too long is not good and i would need to filter this out even though playback is 30seconds long. How do i get ffmpeg to scan the file and report an accurate time?

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  • Best Configuration for Performance - ffmpeg streaming / mp4 /flv

    - by Sam Alex
    I have some mpeg video files and a web page. A visitor comes to that web page and according to the selected options, a php script calls ffmpeg and combines the different mpeg files and then converts it to a mp4 file. That mp4 file is then shown to the visitor using flowplayer. The MP4 creation takes some time and flowplayer takes some time to load the file. What do you think is the best way to accomplish this task ? Should i go for streaming server ? I need to reduce the time taken for conversion. Will converting to FLV be faster ?

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  • ffmpeg volume parameter format

    - by tanon
    ffmpeg's -vol parameter is confusing me. 256 => normal (i guess meaning same as input volume, no change) 512 => (double the volume - read this somewhere). So what to do for 3 times the volume? 1.5 times the volume? Basically, lets say I have the max sound amplitudes (audacity levels) in 3 files as: 0.8 0.6 0.9 I want to amplify in the first two files, so that max=0.9 in all files. What parameters of -vol I would use?

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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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  • Using avconv (ffmpeg) to concatenate a bunch of .bmps into a mkv/avi video

    - by user1509246
    Hoi, Trying to figure out how to get avconv to concatenate a bunch of .bmps together into a video file. Here's what I've got so far: avconv -f image2 -i Capture/%d.bmp -vcodec mpeg4 -r 24 -b:v 20M Capture.mkv While this does work, the quality is terrible - there are tons of artifacts that are visible, the colours are distorted and everything is blurred. I've trawled through the documentation for avconv and ffmpeg, but can't find anything that increases the quality. Any ideas as to how I can get the quality as close to the original bmps as possible? Thanks for lending me your brains, - Alex

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  • FFMPEG Splitting MP4 with Same Quality

    - by Pragmatic
    I have one large MP4 file. I am attempting to split it into smaller files. ffmpeg -i largefile.mp4 -sameq -ss 00:00:00 -t 00:50:00 smallfile.mp4 I thought using -sameq would keep the same quality settings. However, I must not understand what that does. I'm looking to keep the same quality (audio/video) and compression with the split files. However, this setting makes the split files much larger. What flag(s) do I need to set to keep the same quality and attributes in the split files while maintaining the same quality to size ratio? For instance if my original file is about 12 GB and is 1920x1080 with a bitrate of 10617kbps and a framerate of 23 frames/sec and 6 channel audio with 317kbps, I would like the split files to be the same only a third of this size (if i split it into three pieces).

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  • FFmpeg add multiple audio files to video at specific points

    - by Arran
    I have two audio files, each about 3 minutes long. I want to take the first 10 seconds of each file and add them each to a video file at specific points - 0 seconds and 10 seconds. So the resulting video should be 20 seconds long. I've got this far: ffmpeg -i video.mov -ss 0 -t 20 -itsoffset 0 -i audio1.mp3 -itsoffset 10 -i audio2.mp3 -acodec copy -vcodec copy out.mov ...but the resulting video has 20 seconds of the first audio file only, the second audio file doesn't start at 10 seconds like it should. Any help would be appreciated, thanks!

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  • FFMPEG settings for Youtube and facebook video uploads

    - by eco_bach
    Can any FFMPEG experts share their preferred settings for video conversion to both Youtube and Facebook? For youtube I am following these guidelines and my video size is 480P @ 24 fps Audio Codec: AAC-LC Channels: Stereo or Stereo + 5.1<br> Sample rate 96khz or 48 khz<br> Video Codec: H.264 Progressive scan (no interlacing)<br> High Profile<br> 2 consecutive B frames<br> Closed GOP. GOP of half the frame rate.<br> CABAC<br> Variable bitrate. No bitrate limit required Color Space: 4.2.0 http://support.google.com/youtube/bin/static.py?hl=en&topic=1728573&guide=1728585&page=guide.cs

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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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  • Using FFMPEG to reliably convert videos to mp4 for iphone/ipod and flash players

    - by Jake Stevenson
    I need to convert videos for use in both a flash player and the iphone/ipod touch. I'm using the following batch script with ffmpeg: @echo off ffmpeg.exe -i %1 -s qvga -acodec libfaac -ar 22050 -ab 128k -vcodec libx264 -threads 0 -f ipod %2 This always outputs an mp4 file, and I can always play it on my PC. The videos also seem to play fine on my iphone 3GS. But with some input files it won't work for older iphone versions (3G and iPod touch). Here's the ffmpeg output from one such file: D:\ffmpeg>encode.bat d:\temp\recording.flv d:\temp\out.m4v FFmpeg version SVN-r18709, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-memalign-hack --prefix=/mingw --cross-prefix=i686-ming w32- --cc=ccache-i686-mingw32-gcc --target-os=mingw32 --arch=i686 --cpu=i686 --e nable-avisynth --enable-gpl --enable-zlib --enable-bzlib --enable-libgsm --enabl e-libfaac --enable-libfaad --enable-pthreads --enable-libvorbis --enable-libtheo ra --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libxvid - -enable-libschroedinger --enable-libx264 libavutil 50. 3. 0 / 50. 3. 0 libavcodec 52.27. 0 / 52.27. 0 libavformat 52.32. 0 / 52.32. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0. 7. 1 / 0. 7. 1 built on Apr 28 2009 04:04:42, gcc: 4.2.4 [flv @ 0x187d650]skipping flv packet: type 18, size 164, flags 0 Input #0, flv, from 'd:\temp\recording.flv': Duration: 00:00:07.17, start: 0.001000, bitrate: N/A Stream #0.0: Video: flv, yuv420p, 320x240, 1k tbr, 1k tbn, 1k tbc Stream #0.1: Audio: nellymoser, 44100 Hz, mono, s16 [libx264 @ 0x13518b0]using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE 4.2 [libx264 @ 0x13518b0]profile Baseline, level 4.2 Output #0, ipod, to 'd:\temp\out.m4v': Stream #0.0: Video: libx264, yuv420p, 320x240, q=2-31, 200 kb/s, 1k tbn, 1k tbc Stream #0.1: Audio: libfaac, 22050 Hz, mono, s16, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 90 fps= 0 q=-1.0 Lsize= 128kB time=6.87 bitrate= 152.4kbits/s video:92kB audio:32kB global headers:1kB muxing overhead 2.620892% [libx264 @ 0x13518b0]slice I:8 Avg QP:29.62 size: 7047 [libx264 @ 0x13518b0]slice P:82 Avg QP:30.83 size: 467 [libx264 @ 0x13518b0]mb I I16..4: 17.9% 0.0% 82.1% [libx264 @ 0x13518b0]mb P I16..4: 0.6% 0.0% 0.0% P16..4: 23.1% 0.0% 0.0% 0.0% 0.0% skip:76.3% [libx264 @ 0x13518b0]final ratefactor: 57.50 [libx264 @ 0x13518b0]SSIM Mean Y:0.9544735 [libx264 @ 0x13518b0]kb/s:8412.6 My suspicion is that it has something to do with the audio encoding. If so, does anyone know how to force it to reencode the audio to the proper format? Any other ideas?

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  • How to install FFMpeg in WampServer 2.0 (Windows XP)

    - by Richard Knop
    I need to install the ffmpeg PHP extension on my localhost so I can test few of my scripts but I am having troubles figuring out how to do that. I have WampServer 2.0 with PHP 5.2.9-2, my OS is Windows XP. Please somebody give me step by step instructions. I have found some Windows builds here: http://sourceforge.net/projects/ffmpeg-php/files/ But I don't know which one to download and what to do with files. EDITED: What I have done so far: Download ffmpeg_new Copy php_ffmpeg.dll from the php5 folder to the C:\wamp\bin\php\php5.2.9-2\ext Copy files from common to the windows/system32 folder Add extension=php_ffmpeg.dll to php.ini file Restarted all services (Apache, PHP...) I am gettings an error after using this code: $extension = 'ffmpeg'; $extension_soname = 'php_ffmpeg.dll'; $extension_fullname = PHP_EXTENSION_DIR . "/" . $extension_soname; // load extension if(false === extension_loaded($extension)) { if (false === dl($extension_soname)) throw new Exception("Can't load extension $extension_fullname\n"); } The error: Warning: dl() [function.dl]: Not supported in multithreaded Web servers - use extension=ffmpeg.dll in your php.ini in C:\wamp\www\hunnyhive\application\modules\default\controllers\MyAccountController.php on line 314 Plus I also get the exception from above.

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  • How to convert .flv file to .3gp using ffmpeg?

    - by Chetana
    I have converted any video format to 3gp file format using ffmpeg on one server. But on another server it not works. Following is my script: exec("ffmpeg -i test.flv -sameq -acodec libmp3lame -ar 22050 -ab 96000 -deinterlace -nr 500 -s 320x240 -aspect 4:3 -r 20 -g 500 -me_range 20 -b 270k -deinterlace -f flv -y test.3gp "); Can anyone tell me what is wrong in script? Following is my ffmpeg setting: root@ninja [~]# ffmpeg -formats ffmpeg version CVS, build 3277056, Copyright (c) 2000-2004 Fabrice Bellard configuration: --enable-mp3lame --enable-libogg --enable-gpl --disable-mmx --enable-shared built on Jun 17 2009 10:51:43, gcc: 4.1.2 20080704 (Red Hat 4.1.2-44)

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  • Adding audio channel using ffmpeg

    - by Raj
    Hi all, I am working on ffmpeg and trying to add a audio stream on the fly. I am using AudioQueues and I get raw audio buffer. I am encoding audio with linear PCM and hence the audio I get will be of raw format, which I know ffmpeg does accept it. But I cannot figure out how. I have looked into AVStream, where in we have to create a new stream for this audio channel but how do I encode it to a video which is already initialized in another AVStream structure. Overall, I would like to have an idea of the architecture of ffmpeg. I found it difficult to work since it is least documented. Any pointers or details are appreciated. Thanks and Regards, Raj Pawan G

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  • Problem with recording audio in Flash (Red5, ffmpeg)

    - by AT
    I'm trying to implement a small program with Flash and php that records audio and converts it to mp3. Currently I have Red5 server up and running, I can connect to it with no problems and I can publish flv recordings to the server. When I listen to the flv with Wimpy FLV player it seems to be fine. The problem comes when I'm trying to convert it with ffmpeg on the command line. I'm simply using a command ffmpeg -i but the output wav is about 50% slower than the input. When I record 10sec, the output is 15sec and pitched down. I've also tried all kinds of bitrate settings, -nv option, etc. but nothing seems to work. I have a recent version of ffmpeg that supports nellymoser format.. Don't know what to do. Anyone have any ideas?

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  • FFmpeg and qt, Unable to find a suitable output format for '>'

    - by Spredzy
    Hi all, I'm trying to execute a ffmpeg operation through Qt I would like to execute this line : ./ffmpeg -t 10 -i temp1 -f mpeg - > temp2 When I execute through the terminal, it works perfectly fine. How ever when I launch it through Qt like this : QProcess *process = new QProcess(); QString parameters("./ffmpeg -t 10 -i temp1 -f mpeg - > temp2"); std::cout << process->execute(parameters) << std::endl; I get an Unable to find a suitable output format for '>' any body has the idea of why ?

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  • Pipe ffmpeg to oggenc(2) with .NET

    - by acidzombie24
    I am trying to encode ogg files at -q 6/192k. ffmpeg doesnt seem to be listenting to my command ffmpeg -i blah -acodec vorbis -ab 192k -y out.ogg So i would like to use the recommended ogg encoder. It says the input must be wav or similar. I would like to pipe a wav from ffmpeg to ogg. However not only am i unsure if oggenc2 will accept input from stdin, i have no idea how to pipe one process to another inside of .net using the Process class.

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  • Solid FFmpeg wrapper for C#/.NET

    - by Lillemanden
    I have been searching the web for some time for a solid FFmpeg wrapper for C#/.NET. But I have yet to come up with something useful. I have found the following three projects, but all of them apears to be dead in early alpha stage. FFmpeg.NET ffmpeg-sharp FFLIB.NET So my question is if anyone knows of a wrapper project that is more mature? I am not looking for a full transcoding engine with job queues and more. Just a simple wrapper so I do not have to make a command line call and then parse the console output, but can make method calls and use eventlisteners for progress. And please feel free to mention any active projects, even if they are stil in the early stages.

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