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• About AMR audio file playing issue on different devices

  - by user352537

  I have got a quite strange problem here. I am developing an IM software and need to play audio files recorded by another client on Android. The same audio file I've got can be played with AVAudioPlayer on 3GS(IOS 4.2.1) device and simulator 4.2. But when I tried by play it on iPhone4(iOS 4.3.3), the function "play" always return NO. I also tried with two iPhone devices, the audio files recorded by iPhone client can be played on both 3GS and iPhone4. So I asked the Android developers about the record parameters they've used. They said that the "AudioEncoder" used by them was "DEFAULT". There are also some other parameters as following: **private AudioEncoder() {} public static final int DEFAULT = 0; /** AMR (Narrowband) audio codec */ public static final int AMR_NB = 1; /** @hide AMR (Wideband) audio codec */ public static final int AMR_WB = 2; /** @hide AAC audio codec */ public static final int AAC = 3; /** @hide enhanced AAC audio codec */ public static final int AAC_PLUS = 4; /** @hide enhanced AAC plus audio codec */ public static final int EAAC_PLUS = 5;** Does anybody know what's the matter?

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• FFMPEG compilation errors

  - by Nitin Sagar

  First of all i am a newbie to Ubuntu Linux and have been trying to install and compile FFMPEG on an Ubuntu machine... I am trying to compile FFMPEG on an Ubuntu machine, using the following link reference: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide I have already install git packages from resource centre whatever it results in search... Whatever i am trying to clone to is showing the below error... and please note that the network is wireless and connected with full bandwidth and i am able to browse through website and not sure why its showing an error as unable to connect and connection timed out.... root@ubuntu:~# cd root@ubuntu:~# git clone --depth 1 git://github.com/mstorsjo/fdk-aac.git Cloning into 'fdk-aac'... fatal: unable to connect to github.com: github.com[0: 207.97.227.239]: errno=Connection timed out Tried these commands as well to install x264 lib: cd git clone --depth 1 git://git.videolan.org/x264 cd x264 I am doing all this as a root user. Any help and comments would be appreciated. Thanks Nitin

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• HTML5 Audio: Which formats? Ditch Ogg Vorbis in favor of Ogg Opus? Is MP3 still needed?

  - by phoibos

  I'm currently working on a website which has to stream audio files. Since bandwidth is always an issue, the file size should be as small as possible. I wonder what audio formats I should provide. MP3 - Most common format but low quality, I don't know if it's even required, since AAC is well supported by the browsers incapable of playing free codecs MP4 AAC - Nice quality / small filesize, supported by Safari / Mobile Devices / IE9 / Flash / Chrome A free codec - well, until recently, there only was Ogg Vorbis, but Ogg Opus is standardized now and it's really good! Questions: Is it time yet to use Opus instead if Vorbis? Firefox supports Opus since version 15, and Opera has support on its roadmap - I guess Chrome will follow in the future too. Do I still have to provide an MP3 file?

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• Syncing Music Everywhere with Google Music and iTunes Match - Will This Work?

  - by dragonmantank

  I have the following devices: Personal Laptop running Windows 7/Ubuntu 11.10 (mostly use Ubuntu) Media Server running Windows 7 with PS3 Media Server and iTunes Work Laptop running OSX Snow Leopard iPad iPhone 4S The iPhone just replaced my Droid 2 Global. What I had been doing was using Google Music to watch the folders iTunes was storing music in and moving any new files up to Google Music. The Droid would pull music down from the cloud via streaming or me telling it to make it available offline, I had folders set up with PS3 Media Server to stream them to TV's via DLNA, and used RDP to play music through my speakers in the office. So far it's worked well. Since I've replaced the Droid 2 though with an iPhone, I've lost the syncing ability with Google Music and have to do it via iTunes (I knew this would happen, no big suprise). I got to thinking though - Apple does offer iTunes Match, which allows your devices to stream/download the music from 'the cloud,' much like Google Music. I could then listen to whatever I Wanted (for the most part) on my phone, iPad, and laptops by syncing via iTunes Match. I don't want to loose my MP3s though, and since I've never used iTunes Match, I wonder if the following is a viable solution: Sign up for iTunes Match on my media server Let it scan my library and make available my songs in AAC in the cloud Not delete the media server MP3s Set up other devices to sync to iTunes Match Continue to get MP3s via Amazon or other services and add to iTunes Let the MP3s sync to Google Music, and let the MP3's add to the AAC versions on my devices I think the main kicker is I don't want to lose the MP3 versions of my songs as those will work just fine on all my devices and I generally rip at 320kbps. I don't mind spending $25/year if it means that I can easily shift the music from device to device without much thinking, but I'm not going to pay $25/year to end up converting my library over to AAC just to save myself the hassle of manually syncing my iPad and iPhone.

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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 to convert dvr-ms file in Ubuntu to DVD?

  - by edmicman

  I have a .dvr-ms file of a recorded TV show from my Vista Media Center. I would like to burn this to a DVD that can play on any standalone DVD player. My main PC that I want to use to convert it to a DVD format is running Ubuntu 10.04. I am able to play the file in Ubuntu using VLC (which surprised me) so I'm assuming I have what I need to decode it. I guess my questions are: What format do I need to convert this file to so that I could burn it to a playable DVD? I started to go through VLC's conversion process and chose I think H264 and AAC or something, and it gave a message about not having an AAC encoder. I'll look into that some more tonight, but is that something I could then burn to a DVD? Thanks for any help!

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• How to enjoy DVD on Apple iPad

  - by user44251

  I believe many people spent a sleepless night yesterday waiting for the new Apple Tablet to come, just a few days ago or perhaps longer I noticed fierce debate about it, its name, size, capacity, processor, main features, price etc. And now, they can take a long breath with the new Apple Tablet named iPad officially released on 28, January, 2010 (Beijing Time). But I know a new battle just begins. iPad, sounds somewhat like iPod and it really shares some similarities in terms of shape like smart, light and portable. It has a 9.7-inch, LED-backlit, IPS display with a remarkable precise Multi-Touch screen. And yet, at just 1.5 lbs and 0.5 inches thin, it's easy to carry and use everywhere. It can greatly facilitates your experience with the web, emails, photos and videos. Right now, it can run almost 140.000 of the apps on the Apple store. It can even run the apps you have downloaded for your iPhone or iPod touch. But so far, I haven't seen any possibility that it can work with DVD, probability there is no built-in DVD-ROM or DVD player which can play DVD directly. As Apple iPad states, the video formats supported are MPEG-4 (MP4, M4V), H.264, MOV etc and audio formats accepted are AAC, Proteceted AAC, MP3, AIFF and WAV etc, those are formats that are commonly used with iMac. This could really a hard nut to crack if you want to watch your favourite DVD on this magic Apple iPad. But don't worry, there is still way out, you just need a few steps for ripping and importing DVD movies to Apple iPad with a simple application DVD to iPad converter What's on DVD to iPad Converter for Mac DVD to iPad converter for Mac is a powerful and professional application designed for the newly released Apple iPad which can rip, convert your DVD contents to Apple iPad compatible MPEG-4 (MP4, M4V), H.264, MOV etc, and other popular file formats like AVI, WMV, MPG, MKV, VOB, 3GP, FLV etc can also be converted so that you can put on your portable devices like iPod, iPhone, iRiver, BlackBerry etc. Besides, it can also extract audio from DVD videos and save as MP3, AIFF, AAC, WAV etc. Mac DVD to iPad converter has also been enhanced that can run both on PowerPC and Intel (Snow Leopard included). It can offer versatile editing features which allows you to make your own DVD videos. For example, you can cut your DVD to whatever length you like by Trim, crop off unwanted parts from DVD clips by Crop, add special effect like Gray, Emboss and Old film to make your videos more artistic. Besides, its built-in merging feature and batch mode allows you to join several DVD clips into a single one and do batch conversion. And more features can be expected if you afford a few minutes to try.

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• MPlayer does not work

  - by Soham Pal

  Using the xubuntu desktop, on Ubuntu Raring updated from Quantal. MPlayer never really worked. No video, no audio, nothing. I really can't be any more helpful, so here's the log: petey@home-pc:~$ mplayer "/home/petey/Downloads/Polar Bear Cafe (480p)HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv" MPlayer SVN-r35984-4.7 (C) 2000-2013 MPlayer Team Playing /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv. libavformat version 55.0.100 (internal) libavformat file format detected. [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (aac), -aid 0 [lavf] stream 2: subtitle (ass), -sid 0 VIDEO: [H264] 848x480 0bpp 23.810 fps 0.0 kbps ( 0.0 kbyte/s) Clip info: creation_time: 2012-04-05 21:36:10 Load subtitles in /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/ Can't open /dev/fb0: Permission denied [fbdev2] Can't open /dev/fb0: Permission denied VO: [v4l2] No such file or directory vo_cvidix: No vidix driver name provided, probing available ones (-v option for details)! [cyberblade] Error occurred during pci scan: Operation not permitted [mach64] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [nvidia_vid] Error occurred during pci scan: Operation not permitted [pm3] Error occurred during pci scan: Operation not permitted [radeon] Error occurred during pci scan: Operation not permitted [rage128] Error occurred during pci scan: Operation not permitted [s3_vid] Error occurred during pci scan: Operation not permitted [SiS] Error occurred during pci scan: Operation not permitted [unichrome] Error occurred during pci scan: Operation not permitted [VO_SUB_VIDIX] Couldn't find working VIDIX driver. ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family libavcodec version 55.0.100 (internal) Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, floatle, 0.0 kbit/0.00% (ratio: 0->352800) Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory DVB card number must be between 1 and 4 AO: [null] 44100Hz 2ch floatle (4 bytes per sample) Starting playback... Movie-Aspect is 1.78:1 - prescaling to correct movie aspect. VO: [null] 848x480 = 854x480 Planar YV12 A: 4.7 V: 4.7 A-V: 0.002 ct: 0.083 0/ 0 22% 0% 0.5% 0 0 MPlayer interrupted by signal 2 in module: sleep_timer A: 4.7 V: 4.7 A-V: 0.001 ct: 0.083 0/ 0 21% 0% 0.5% 0 0 Exiting... (Quit)

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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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• python: creating a list inside a dictionary

  - by user1871081

  I just started using python and I'm trying to create a program that will read a file that looks like this: AAA x 111 AAB x 111 AAA x 112 AAC x 123 ... the file is 50 lines long and I'm trying to make the letters into keys in a dictionary and the numbers lists that correspond with the keys. I want the output to look like this: {AAA: ['111', '112'], AAB: ['111'], AAC: [123], ...} This is what I've tried file = open("filename.txt", "r") readline = file.readline().rstrip() while readline!= "": list = [] list = readline.split(" ") j = list.index("x") k = list[0:j] v = list[p + 1:] d = {} if k in d == False d[k] = [] d[k].append(v) else d[k].append(v) readline = file.readline().rstrip() I keep getting syntax errors on my if statement and I can't figure out what I've done wrong.

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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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• Android 2.3 Gingerbread est disponible pour les développeurs, beaucoup d'améliorations dont un nouveau Garbage Collector

  Comme vous pouvez le constater sur le site Developpeurs d'Android, Android 2.3 Gingerbread est sorti. Au menu, un nouveau Garbage Collector, des améliorations pour l'accès aux fonctionnalités comme OpenGL ES ou les senseurs depuis un programme natif (en C), support des formats VP8, WebM, AAC et AMR, des nouveaux effets audio, la gestion d'une caméra pour la vidéo-conférence, support de SIP/VOIP, support de Near Field Communication

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• How can I maximum compress video files?

  - by EmmyS

  I received 4 .mov files from a client that they want on their mobile website via SlideShowPro. Each original file was between 200 and 400 mb. I've gotten each one down to about 30 mb using transmageddon as described here, but that's still really big for a mobile connection. Is there any way to shrink them even further? Maybe it's the settings; I used Output Format = MPEG4, Audio = AAC, Video = H264 (which is what is suggested by SlideShowPro.)

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• Convert mp4 video to a format xbox 360 can play

  - by Björn Lindqvist

  Here is a video file my Xbox 360 refuses to play: $ MP4Box -info video.mp4 * Movie Info * Timescale 90000 - Duration 02:18:33.365 Fragmented File no - 2 track(s) File Brand mp42 - version 0 Created: GMT Sat Jul 21 07:08:55 2012 File has root IOD (9 bytes) Scene PL 0xff - Graphics PL 0xff - OD PL 0xff Visual PL: ISO Reserved Profile (0x7f) Audio PL: High Quality Audio Profile @ Level 2 (0x0f) No streams included in root OD iTunes Info: Encoder Software: HandBrake 0.9.6 2012022800 Track # 1 Info - TrackID 1 - TimeScale 90000 - Duration 02:18:33.235 Media Info: Language "Undetermined" - Type "vide:avc1" - 199318 samples Visual Track layout: x=0 y=0 width=1280 height=688 MPEG-4 Config: Visual Stream - ObjectTypeIndication 0x21 AVC/H264 Video - Visual Size 1280 x 688 AVC Info: 1 SPS - 1 PPS - Profile High @ Level 4.1 NAL Unit length bits: 32 Self-synchronized Track # 2 Info - TrackID 2 - TimeScale 48000 - Duration 02:18:33.365 Media Info: Language "English" - Type "soun:mp4a" - 389689 samples MPEG-4 Config: Audio Stream - ObjectTypeIndication 0x40 MPEG-4 Audio MPEG-4 Audio AAC LC - 6 Channel(s) - SampleRate 48000 Synchronized on stream 1 $ avconv -i video.mp4 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:33 with gcc 4.6.3 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42isomavc1 creation_time : 2012-07-21 07:08:55 encoder : HandBrake 0.9.6 2012022800 Duration: 02:18:33.36, start: 0.000000, bitrate: 2299 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x688, 1973 kb/s, 23.98 fps, 90k tbr, 90k tbn, 180k tbc Metadata: creation_time : 2012-07-21 07:08:55 Stream #0.1(eng): Audio: aac, 48000 Hz, 5.1, s16, 319 kb/s Metadata: creation_time : 2012-07-21 07:08:55 At least one output file must be specified What tool, such as ffmpeg or mencoder, and what magic command line incantation should I use to transcode this file into a format Xbox 360 can play? I want the transcode process to retain as good video quality as possible.

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• Join mp4 files in linux

  - by Jose Armando

  I want to join two mp4 files to create a single one. The video streams are encoded in h264 and the audio in aac. I can not re-encode the videos to another format due to computational reasons. Also, I cannot use any gui programs, all processing must be performed with linux command line utilities. FFmpeg cannot do this for mpeg4 files so instead I used MP4Box e.g. MP4Box -add video1.mp4 -cat video2.mp4 newvideo.mp4 unfortunately the audio gets all mixed up. I thought that the problem was that the audio was in aac so I transcoded it in mp3 and used again MP4Box. In this case the audio is fine for the first half of newvideo.mp4 (corresponding to video1.mp4) but then their is no audio and I cannot navigate in the video also. My next thought was that the audio and video streams had some small discrepancies in their lengths that I should fix. So for each input video I splitted the video and audio streams and then joined them with the -shortest option in ffmpeg. thus for the first video I ran avconv -y -i video1.mp4 -c copy -map 0:0 videostream1.mp4 avconv -y -i video1.mp4 -c copy -map 0:1 audiostream1.m4a avconv -y -i videostream1.mp4 -i audiostream1.m4a -c copy -shortest video1_aligned.mp4 similarly for the second video and then used MP4Box as previously. Unfortunately this didn't work either. The only success I had was when I joined the video streams separetely (i.e. videostream1.mp4 and videostream2.mp4) and the audio streams (i.e. audiostream1.m4a and audiostream2.m4a) and then joined the video and audio in a final file. However, the synchronization is lost for the second half of the video. Concretelly, there is a 1 sec delay of audio and video. Any suggestions are really welcome.

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• newsubtitles line not working in FFmpeg

  - by godMode

  i'm trying to run the following line on FFmpeg that will basically "re-format" an MKV file to MP4 without doing any re-encoding and also embed SRT subtitles onto the MP4 output: ffmpeg -i test.mkv -i test.srt -newsubtitle -acodec copy -vcodec copy test.mp4 Without the "-i test.srt -nwesubtitle" bit, it seems to work just fine; however, with it I get the following output: Seems stream 0 codec frame rate differs from container frame rate: 47.95 (5000000/104271) - 23.98 (24000/1001) Stream #0.0(eng): Video: h264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc Stream #0.1(eng): Subtitle: 0x0000 Metadata: title : English Stream #0.2(jpn): Audio: aac, 48000 Hz, stereo, s16 Metadata: title : Japanese 2.0 Stream #0.3(eng): Audio: aac, 48000 Hz, stereo, s16 Metadata: title : English 2.0 Stream #0.4(eng): Subtitle: 0x0000 Metadata: title : English Songs & Signs Stream #0.5: Attachment: 0x0000 Metadata: filename : MyriadPro-Bold.ttf Stream #0.6: Attachment: 0x0000 Metadata: filename : MyriadPro-RegularHaruhi.ttf Stream #0.7: Attachment: 0x0000 Metadata: filename : ChaparralPro-BoldIt.ttf Stream #0.8: Attachment: 0x0000 Metadata: filename : ChaparralPro-SemiboldIt.ttf Stream #0.9: Attachment: 0x0000 Metadata: filename : epmgobld_ending.ttf Stream #0.10: Attachment: 0x0000 Metadata: filename : epminbld_opening.ttf Stream #0.11: Attachment: 0x0000 Metadata: filename : Folks-Bold.ttf Stream #0.12: Attachment: 0x0000 Metadata: filename : GosmickSansBold.ttf Stream #0.13: Attachment: 0x0000 Metadata: filename : WarnockPro-LightDisp.ttf Stream #0.14: Attachment: 0x0000 Metadata: filename : epmgobld_ending.ttf Stream #0.15: Attachment: 0x0000 Metadata: filename : GosmickSansBold.ttf Stream #0.16: Attachment: 0x0000 Metadata: filename : Marker SD 1.2.ttf Stream #0.17: Attachment: 0x0000 Metadata: filename : MyriadPro-Bold.ttf Stream #0.18: Attachment: 0x0000 Metadata: filename : MyriadPro-RegularHaruhi.ttf Stream #0.19: Attachment: 0x0000 Metadata: filename : MyriadPro-SemiCn.ttf test.srt: Invalid data found when processing input I tried adding "-r pal", "-r ntsc" or "-r 23.98" thinking it was framerate issue with no change.

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• ffmpeg - h264 to xvid creates large file

  - by fatnic

  I'm trying to use ffmpeg to convert a h264/aac video file to an xvid/mp3 file so I can play it in my ultra-cheap media player. At the moment the converted video file is TWICE the size of the original mp4. Is there any way to get a smaller file size without loosing too much quality? Even a drop to -qmin 1 is pretty awful! The command i'm using is ffmpeg -i input.mp4 -vcodec libxvid -sameq -acodec libmp3lame -ab 128k -ac 2 output.avi And the ffmpeg output is Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4' Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 Duration: 01:34:27.69, start: 0.000000, bitrate: 1520 kb/s Stream #0.0(und): Video: h264, yuv420p, 720x304 [PAR 1:1 DAR 45:19], 1387 kb/s, 25 fps, 25 tbr, 25k tbn, 50 tbc Stream #0.1(und): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s Output #0, avi, to 'output.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0(und): Video: mpeg4, yuv420p, 720x304 [PAR 1:1 DAR 45:19], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1(und): Audio: libmp3lame, 48000 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1

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• VLC Crash when playing MKV files in Windows 7

  - by Phelios

  I'm not sure if the mkv file is corrupted, but when it is opened with VLC player, VLC loads up and displays nothing. Can't even close VLC after that. And also, VLC is running with 50% of the CPU. I have to use End Process to kill it. How do I know if the file is corrupted? How do I solve this? info from mediainfo Format : Matroska File size : 69.4 MiB Duration : 21mn 48s Overall bit rate : 445 Kbps Encoded date : UTC 2009-11-20 18:33:49 Writing application : mkvmerge v2.9.7 ('Tenderness') built on Jul 1 2009 18:43:35 Writing library : libebml v0.7.7 + libmatroska v0.8.1 Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : [email protected] Format settings, CABAC : Yes Format settings, ReFrames : 2 frames Format settings, GOP : N=1 Codec ID : V_MPEG4/ISO/AVC Duration : 21mn 48s Width : 640 pixels Height : 352 pixels Display aspect ratio : 16:9 Frame rate : 23.976 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Audio ID : 2 Format : AAC Format/Info : Advanced Audio Codec Format profile : HE-AAC / LC Codec ID : A_AAC Duration : 21mn 48s Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 48.0 KHz / 24.0 KHz Compression mode : Lossy

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• Playing a .TS file on iOS

  - by Jonathan Grynspan

  We're working with some hardware that produces files in the .TS format, and we'd like to play them on an iOS device. (The files are internally consistent with what iOS supports--MPEG-4 video, AAC audio.) We've been investigating three options so far: Roll our own integrated HTTP Live Streaming server and serve up a faux M3U8 playlist from within the app. This... doesn't seem to want to play nice, and we've had mixed luck actually getting the .TS files to play on devices. Unwrap the MPEG-4 and AAC data from the TS file and re-wrap it as MP4. This, I'm told, is exceedingly difficult to do, and I haven't found anything useful online that could shed light on how to do it. We've got code in the pipeline to do it but it won't be ready until long after we need it. If we could do it, I could easily subclass NSURLProtocol and have it working within a matter of hours minutes. Use FFmpeg to implement option #2. FFmpeg seems like a possible solution but it isn't configured to build for iOS and I don't have the background to get it working (whereas the rest of our engineers don't have the Apple background needed.) I think #2 is our best bet, but as I don't know the ins and outs of MPEG-2 TS and MPEG-4, I don't have the ability to put it together myself. Does anybody have any insight into this problem? Perhaps some experience playing local TS files on iOS, or some tips on converting from TS to MP4?

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• Using bash to copy file to spec folders

  - by Franko

  I have a folder with a fair amount of subfolders. In some of the subfolders do I have a folder.jpg picture. What I try to do find that folder(s) and copy it to all other subfolders that got the same artist and album information then continue on to the next album etc. The structure of all the folders are "artist - year - album - [encoding information]". I have made a really simple one liner that find the folders that got the file but there am I stuck. ls -F | grep / | while read folders;do find "$folders" -name folder.jpg; done Anyone have any good tip or ideas how to solve this or pointers how to proceed? Edit: First of all, i´m real new to this (like you cant tell) so please have patience. Ok, let me break it down even more. I have a folder structure that looks like this: artist1 - year - album - [flac] artist1 - year - album - [mp3] artist1 - year - album - [AAC] artist2 - year - album - [flac] etc I like to loop over the set of folders that have the same artist and album information and look for a folder.jpg file. When I find that file do I like to copy it to all of the other folders in the same set. Ex if I find one folder.jpg in artist1 - year - album - [flac] folder do I like to have that folder.jpg copied to artist1 - year - album - [mp3] & artist1 - year - album - [AAC] but not to artist2 - year - album - [flac]. The continue the loop until all the sets been processed. I really hope that makes it a bit more easy to understand what I try to do :)

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• sum of square of each elements in the vector using for_each

  - by pierr

  Hi, As the function accepted by for_each take only one parameter (the element of the vector), I have to define a static int sum = 0 somewhere so that It can be accessed after calling the for_each . I think this is awkward. Any better way to do this (still use for_each) ? #include <algorithm> #include <vector> #include <iostream> using namespace std; static int sum = 0; void add_f(int i ) { sum += i * i; } void test_using_for_each() { int arr[] = {1,2,3,4}; vector<int> a (arr ,arr + sizeof(arr)/sizeof(arr[0])); for_each( a.begin(),a.end(), add_f); cout << "sum of the square of the element is " << sum << endl; } In Ruby, We can do it this way: sum = 0 [1,2,3,4].each { |i| sum += i*i} #local variable can be used in the callback function puts sum #=> 30 Would you please show more examples how for_each is typically used in practical programming (not just print out each element)? Is it possible use for_each simulate 'programming pattern' like map and inject in Ruby (or map /fold in Haskell). #map in ruby >> [1,2,3,4].map {|i| i*i} => [1, 4, 9, 16] #inject in ruby [1, 4, 9, 16].inject(0) {|aac ,i| aac +=i} #=> 30 EDIT: Thank you all. I have learned so much from your replies. We have so many ways to do the same single thing in C++ , which makes it a little bit difficult to learn. But it's interesting :)

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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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• Best video codec for filmed powerpoint presentation

  - by rslite

  I have some presentations that are filmed. The audio is the presenter and the video is all the Powerpoint slides (size 1024x768, video codec H264, audio codec AAC). I would like to reduce their final file size since a 1 hour presentation is about 800 MB. Most of it is the video part which as I said is mostly powerpoint slides that don't change much over a matter of several seconds. Which codec would be better suited to encode this images and reduce the size of the end file?

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• How do I add another audio stream to an MP4 file?

  - by RandomEngy

  I've got an MP4 video file and I want to add another AAC audio track to it. I've tried YAMB and MeGUI (frontends for MP4Box) and it plays correctly in Zoom Player, but it picks the wrong track in WMP and plays both at once in Quicktime. I think this might have to do with designating the default audio track somehow. Does anyone know how to specify the default audio track with YAMB/MeGUI or know of another way of adding a track to an MP4 file?

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• Youtube "unable to convert video file"

  - by Alexandra

  I encoded some videos from a dvd format (mpeg 2 I think) to h246 (using a .mkv container). When I upload them to youtube, most of them work, but there are a few that don't. After I upload it, I get the message "Failed (unable to convert video file)" What could be the problem because all the videos are the same format, and only a few of them fail. When I click upload details, while uploading, the file seems to be recognized by youtube: Format: MATROSKA Dimensions: 704 x 480 px Video codec: H264 Audio codec: AAC

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