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  • How to find the leaky faucet that loads into Malloc 32kb

    - by Rob
    I have been messing around with Leaks trying to find which function is not being deallocated (I am still new to this) and could really use some experienced insight. I have this bit of code that seems to be the culprit. Every time I press the button that calls this code, 32kb of memory is additionally allocated to memory and when the button is released that memory does not get deallocated. What I found was that everytime that AVAudioPlayer is called to play an m4a file, the final function to parse the m4a file is MP4BoxParser::Initialize() and this in turn allocates 32kb of memory through Cached_DataSource::ReadBytes My question is, how do I go about deallocating that after it is finished so that it doesn't keep allocating 32kb every time the button is pressed? Any help you could provide is greatly appreciated! - (void)touchesBegan:(NSSet *)touches withEvent:(UIEvent *)event { //stop playing theAudio.stop; // cancel any pending handleSingleTap messages [NSObject cancelPreviousPerformRequestsWithTarget:self selector:@selector(handleSingleTap) object:nil]; UITouch* touch = [[event allTouches] anyObject]; NSString* filename = [g_AppsList objectAtIndex: [touch view].tag]; NSString *path = [[NSBundle mainBundle] pathForResource: filename ofType:@"m4a"]; theAudio=[[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:path] error:NULL]; theAudio.delegate = self; [theAudio prepareToPlay]; [theAudio setNumberOfLoops:-1]; [theAudio setVolume: g_Volume]; [theAudio play]; }

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  • Encoding with FFmpeg using a FIFO

    - by Ashot Martirosyan
    Hello everyone. I'm trying to convert Flac audio file to AAC file using command line. So I wrote this ffmpeg -i input.flac temp.wav faac -q 120 -o output.m4a temp.wav It's working fine. Now I want to do the same using fifo, so I'm writing this mkfifo temp.wav ffmpeg -i input.flac temp.wav & faac -q 120 -o output.m4a temp.wav And it's freezing. So could you tall me what I'm doing wrong. Thanks a lot, and sorry for my English.

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  • How do I get media players to play ripped dvd files using Handbrake?

    - by LibraryGeekAdam
    I want to rip my dvd collection to put on a server for my tv. I started this with trying 4 dvds using Handbrake. The first 3 ripped perfectly fine according to the directions I used from LifeHacker (http://lifehacker.com/5773000/how-to-rip-dvds-with-handbrake). The 4th dvd took about 2 secs to rip and then stopped saying it was complete. It was not. Now with the 3 files I have I tried to open and play with VLC and Miro. Neither one would play the file. It says its open but there is no sound or video and it doesn't actually show the movie playing in the progress bar. I ran it from the terminal and this is the error message I get. [0xb04c2f38] mp4 demux error: MP4 plugin discarded (no moov,foov,moof box) [mov,mp4,m4a,3gp,3g2,mj2 @ 0xa032980] moov atom not found [mov,mp4,m4a,3gp,3g2,mj2 @ 0xb0efd500] moov atom not found [0xb04c2f38] avformat demux error: Could not open ????e/boris/Videos/Movies/Taras Bulba 1962.m4v: Unknown error 1094995529 I have also changed a file name to end .mp4 instead of .m4v which is how I originally ripped them.

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  • Reproduce PIPE functionality in IronPython

    - by Muppet Geoff
    Hi, I am hoping some genious out there can help me out with this... I am using sox to merge and resample a group of WAV files, and pipe the output directly to the input of NeroAACEnc for encoding to AAC format. I originally ran the process in a script, which included: sox.exe d:\audio\1.wav d:\audio\2.wav d:\audio\3.wav -c 1 -r 22050 -t wav - | neroAacEnc.exe -q 0.5 -if - -of test.m4a This worked as expected. The '-' in the comand line translates as 'Pipe/redirect input/output (stdin/stdout)' - So Sox pipes to stdout, and NeroAACEnc reads from stdin, the | joins them together. I then migrated the whole solution to Python, and the equivalent command became: from subprocess import call, Popen, PIPE runwav = Popen(['sox.exe', 'd:\audio\1.wav', 'd:\audio\2.wav', 'd:\audio\3.wav', '-c', '1', '-r', '22050', '-t', 'wav', '-'], shell=False, stdout=PIPE) runm4b = call(['neroAacEnc.exe', '-q', '0.5', '-if', '-', '-of', 'test.m4a'], shell=False, stdin=runwav.stdout) This also worked like a charm, exactly as expected. Slightly more convoluted, but hey :) Well now I have to move it to IronPython, and the Subprocess module isn't available (the partial implementation that is, doesn't have Popen/PIPE support - plus it seems silly to add a custom library when there is probably a native alternative). I should mention here, that I opted for IronPython over C#, because I am comfortable with Python now - however, there is a chance of moving it again later to C# native, and I am using IronPython to ease myself into it :) I have no C# or .net experience. So far I have the following equivalent, that sets up the 2 processes: from System.Diagnostics import Process wav = Process() wav.StartInfo.UseShellExecute = False wav.StartInfo.RedirectStandardOutput = True wav.StartInfo.FileName = 'sox.exe' wav.StartInfo.Arguments = 'd:\audio\1.wav d:\audio\2.wav d:\audio\3.wav -c 1 -r 22050 -t wav -' wav.Start() m4b = Process() m4b.StartInfo.UseShellExecute = False m4b.StartInfo.RedirectStandardInput = True m4b.StartInfo.FileName = 'neroAacEnc.exe' m4b.StartInfo.Arguments = '-q 0.5 -if - -of test.m4a' m4b.Start() I know that these 2 processes start (I can see Nero and Sox in the task manager) but what I can't figure out (for the life of me) is how to string the two output/input streams together, as with the previous two solutions. I have searched and searched, so I thought I'd ask! If anyone knows either: How to join the two streams with the same net result as the Python and Commandline versions; or A better way to acheive what I am trying to do. I would be extremely grateful! Many thanks in advance, Geoff P.S. A code sample based off the above would be awesome :) or a specific code example of a similar process that I can easily translate... this has broked my brayne.

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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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  • 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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  • unzip and maintain directory structure of archives

    - by Ramy
    On fedora-13, I tried using: unzip -j [nameof.zip] but this doesn't seem to maintain the folder structure of the original archive. I REALLY need to maintain this structure because the archive is a backup of all my m4a's which are being converted to mp3. If I just convert it as is, then i'll just have a single massive directory full of mp3's, but they won't be in their respective "artist" folder.

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  • unzip and maintain directory structure or archives

    - by Ramy
    On fedora-13, I tried using: unzip -j [nameof.zip] but this doesn't seem to maintain the folder structure of the original archive. I REALLY need to maintain this structure because the archive is a backup of all my m4a's which are being converted to mp3. If I just convert it as is, then i'll just have a single massive directory full of mp3's, but they won't be in their respective "artist" folder.

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  • How to Play FLAC Files in Windows 7 Media Center & Player

    - by Mysticgeek
    An annoyance for music lovers who enjoy FLAC format, is there’s no native support for WMP or WMC. If you’re a music enthusiast who prefers FLAC format, we’ll look at adding support to Windows 7 Media Center and Player. For the following article we are using Windows 7 Home Premium 32-bit edition. Download and Install madFLAC v1.8 The first thing we need to do is download and install the madFLAC v1.8 decoder (link below). Just unzip the file and run install.bat… You’ll get a message that it has been successfully registered, click Ok. To verify everything is working, open up one of your FLAC files with WMP, and you’ll get the following message. Check the box Don’t ask me again for this extension and click Yes. Now Media Player should play the track you’ve chosen.   Delete Current Music Library But what if you want to add your entire collection of FLAC files to the Library? If you already have it set up as your default music player, unfortunately we need to remove the current library and delete the database. The best way to manage the music library in Windows 7 is via WMP 12. Since we don’t want to delete songs from the computer we need to Open WMP, press “Alt+T” and navigate to Tools \ Options \ Library.   Now uncheck the box Delete files from computer when deleted from library and click Ok. Now in your Library click “Ctrl + A” to highlight all of the songs in the Library, then hit the “Delete” key. If you have a lot of songs in your library (like on our system) you’ll see the following dialog box while it collects all of the information.   After all of the data is collected, make sure the radio button next to Delete from library only is marked and click Ok. Again you’ll see the Working progress window while the songs are deleted. Deleting Current Database Now we need to make sure we’re starting out fresh. Close out of Media Player, then we’ll basically follow the same directions The Geek pointed out for fixing the WMP Library. Click on Start and type in services.msc into the search box and hit Enter. Now scroll down and stop the service named Windows Media Player Network Sharing Service. Now, navigate to the following directory and the main file to delete CurrentDatabase_372.wmdb %USERPROFILE%\Local Settings\Application Data\Microsoft\Media Player\ Again, the main file to delete is CurrentDatabase_372.wmdb, though if you want, you can delete them all. If you’re uneasy about deleting these files, make sure to back them up first. Now after you restart WMP you can begin adding your FLAC files. For those of us with large collections, it’s extremely annoying to see WMP try to pick up all of your media by default. To delete the other directories go to Organize \ Manage Libraries then open the directories you want to remove. For example here we’re removing the default libraries it tries to check for music. Remove the directories you don’t want it to gather contents from in each of the categories. We removed all of the other collections and only added the FLAC music directory from our home server. SoftPointer Tag Support Plugin Even though we were able to get FLAC files to play in WMP and WMC at this point, there’s another utility from SoftPointer to add. It enables FLAC (and other file formats) to be picked up in the library much easier. It has a long name but is effective –M4a/FLAC/Ogg/Ape/Mpc Tag Support Plugin for Media Player and Media Center (link below). Just install it by accepting the defaults, and you’ll be glad you did. After installing it, and re-launching Media Player, give it some time to collect all of the data from your FLAC directory…it can take a while. In fact, if your collection is huge, just walk away and let it do its thing. If you try to use it right away, WMP slows down considerably while updating the library.   Once the library is setup you’ll be able to play your FLAC tunes in Windows 7 Media Center as well and Windows Media Player 12.   Album Art One caveat is that some of our albums didn’t show any cover art. But we were usually able to get it by right-clicking the album and selecting Find album info.   Then confirming the album information is correct…   Conclusion Although this seems like several steps to go through to play FLAC files in Windows 7 Media Center and Player, it seems to work really well after it’s set up. We haven’t tried this with a 64-bit machine, but the process should be similar, but you might want to make sure the codecs you use are 64-bit. We’re sure there are other methods out there that some of you use, and if so leave us a comment and tell us about it. Download madFlac V1.8  M4a/FLAC/Ogg/Ape/Mpc Tag Support Plugin for Media Player and Media Center from SoftPointer Similar Articles Productive Geek Tips How to Play .OGM Video Files in Windows VistaFixing When Windows Media Player Library Won’t Let You Add FilesUsing Netflix Watchnow in Windows Vista Media Center (Gmedia)Kantaris is a Unique Media Player Based on VLCEasily Change Audio File Formats with XRECODE TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 PCmover Professional OutSync will Sync Photos of your Friends on Facebook and Outlook Windows 7 Easter Theme YoWindoW, a real time weather screensaver Optimize your computer the Microsoft way Stormpulse provides slick, real time weather data Geek Parents – Did you try Parental Controls in Windows 7?

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  • Another ubuntu one music problem with ID3-Tags ("unknown artist" problem)

    - by Andi
    I started using ubuntu one a few days ago. I put some MP3s into the cloud music folder and I can play them just fine in the music web and andriod applications. The problem is that all files are sorted under "unknown artist" and "unknown album" and the title is either the file name or a part of it (which is from the service "guessing the title" I guess). It seems the problem happened before. I looked in the FAQ, which said this happens with m4a files, but I use mp3 files. The ID3 tags are correct and are tagged with ID3v1 and ID3v2. I read to wait, until the service can catch up with the tagging, so I waited 24 hours, still nothing. Every single file is still listed under unknown artist/unknown album. I'm running out of options here :/

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  • Ubuntu One Music - 'Unable to Parse '2006-12-12T08:00:00Z' as integer (iTunes Non-DRM AAC)

    - by Scott
    Good Morning; Title says most of it, uploaded a new song to my U1 Music (via my Android using the Files app). Which is an recently purchased iTunes .m4a song, so is non-DRM AAC. Uploaded fine, and browsing in U1 Music I see artist "Spray" fine, and then the Album, but attempting to open the Album to the song, returns: "'Unable to Parse '2006-12-12T08:00:00Z' as integer" Not sure if it's a problem with the file itself, or just how Android uploaded the file, as that is clearly a weird date code. All my other music is fine, no errors, and the service works awesome. My U1 account is under the e-mail address used for this question. Thanks!

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  • DropVox Records Voice Memos Right to Your Dropbox Account

    - by Jason Fitzpatrick
    DropVox is a clever and highly specialized application that, quite effectively, turns your iOS device into a voice recorder with Dropbox-based storage. Install the app, launch it, hit the record button, and your recording is uploaded to your Dropbox account in .m4a format as soon as you’re finished creating it. You can also configure DropVox to start recording immediately after launch and to continue recording if the device is locked or other applications are in use. Hit up the link to grab a copy. DropVox is currently $0.99 (50% off for a limited time) and works on the iPhone, iPad, and iPod Touch with microphone attached. DropVox [via Download Squad] HTG Explains: What’s the Difference Between the Windows 7 HomeGroups and XP-style Networking?Internet Explorer 9 Released: Here’s What You Need To KnowHTG Explains: How Does Email Work?

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  • trouble with boost::filesystem::wrecursive_directory_iterator

    - by Dogmatixed
    I'm trying to write a program to help me manage my iTunes library, including removing duplicates and cataloging certain things. At this point I'm still just trying to get it to walk through all the folders, and have run into a problem: I have a small amount of Japanese music, where the artist and/or album is written in Japanese characters. Because of how iTunes arranges things in its library the directories contain these characters. "shouldn't be a problem, though." I thought, because the boost::filesystem library has a wide character version of its recursive iterator. but when I actually try to use it, it seems to completely stop when it hits the first Japanese char. complete stop as in it doesn't finish printing the line, no carriage return or anything. now, I'm still pretty new to programming, so I'm assuming it's my mistake, anyone know why this is happening? here's what I think is the relevant code: fs::wrecursive_directory_iterator end_it; int i; try { for(fs::wrecursive_directory_iterator rec_it(full_path); rec_it != end_it; ++rec_it) { for(i = 0; i < rec_it.level(); i++) { out << "\t"; } out << rec_it->string() << std::endl; } } catch(std::exception e) { out << "something went wrong: " << e.what(); } and from my output file, minus some of the path: /Test Libs/Combine /Test Libs/Lib1 /Test Libs/Lib1/02 Too Long.m4a /Test Libs/Lib1/03 Like a Hitman, Like a Dancer.mp3 /Test Libs/Lib1/A Certain Ratio /Test Libs/Lib1/A Certain Ratio/Beyond Punk! /Test Libs/Lib1/A Certain Ratio/Unknown Album /Test Libs/Lib1/A Certain Ratio/Unknown Album/Do The Du.mp3 /Test Libs/Lib1/A Certain Ratio/Unknown Album/Shack Up.mp3 /Test Libs/Lib1/ finally, what I expect: /Test Libs/Combine /Test Libs/Lib1 /Test Libs/Lib1/02 Too Long.m4a /Test Libs/Lib1/03 Like a Hitman, Like a Dancer.mp3 /Test Libs/Lib1/A Certain Ratio /Test Libs/Lib1/A Certain Ratio/Beyond Punk! /Test Libs/Lib1/A Certain Ratio/Unknown Album /Test Libs/Lib1/A Certain Ratio/Unknown Album/Do The Du.mp3 /Test Libs/Lib1/A Certain Ratio/Unknown Album/Shack Up.mp3 /Test Libs/Lib1/??? /Test Libs/Lib1/Bring it on /Test Libs/Lib1/04 Bring it on.mp3 any thoughts? Thanks.

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  • Make exact mp4 (H264) format for uploading to youtube

    - by WHITECOLOR
    With ffmpeg I'm converting video from mp3 and picture to upload it to youtube. After upload, conversion fails. Reasons are unknown. I believe the problem is in format. By the way If I'm uploading file 5 minutes length, it fails if I upload 30 seconds of this file it succeeds. I have donwload mp4 file from youtube. Then I uploaded it, it is done very fast. So a nice solution would be to convert videos to the same format that is done by google. I got the following output by mpeg: ffmpeg version N-44264-g070b0e1 Copyright (c) 2000-2012 the FFmpeg developers built on Sep 7 2012 17:38:57 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. 72.100 / 51. 72.100 libavcodec 54. 55.100 / 54. 55.100 libavformat 54. 25.105 / 54. 25.105 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 16.100 / 3. 16.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'youtubetrack0.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2012-10-02 22:58:57 Duration: 00:06:46.66, start: 0.000000, bitrate: 176 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yu v420p, 450x360, 78 kb/s, 6 fps, 6 tbr, 12 tbn, 12 tbc Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 95 kb/s Metadata: creation_time : 2012-10-02 22:58:57 handler_name : IsoMedia File Produced by Google, 5-11-2011 Is it possible to construct ffmpeg parameters so that that would give the same format that google internally does? Is the information above sufficient? I couldn't construct needed params. For example I don't understand how to set tbn and what 95 kb/s mean in "Stream #0:1(und): Audio:". Now I just do: ffmpeg -i videoimage.jpg -i audio.mp3 video.mp4 Info I've got: ffmpeg version N-44998-gdf82454 Copyright (c) 2000-2012 the FFmpeg developers built on Oct 2 2012 23:03:12 with gcc 4.7.1 (GCC) configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libcelt --enable-libopencore-amrnb --en able-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-librtmp --enable-libschroedinger - -enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc -- enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enab le-libxavs --enable-libxvid --enable-zlib libavutil 51. 73.101 / 51. 73.101 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, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.25.105 Duration: 00:06:46.81, start: 0.000000, bitrate: 129 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p, 450x360, 3392 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 127 kb/s Metadata: handler_name : SoundHandler This video fails the conversion on youtube. I also tried to use other vcode parmam and extensions of output file (mp4, wmv, avi) but failed too. Would be greatful for help.

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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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  • Beeping Hard Disk - Seagate 250GB Momentus 5400.6

    - by Pez Cuckow
    I have been trying to repair a laptop that simply beeps instead of booting. After taking it apart I have now realised that it is the hard disk beeping. I know that sound strange but I guarantee that is what it is! (Currently powered on it's own with a Sata Mains lead). The beeping is slightly faster than one per second there is a link to a recording below: http://www.pezcuckow.com/files/BeepingHardDisk.m4a This recording was made resting the mic on the hard disk while it was sat on a table on it's own, there are no speakers anywhere near, the sound is coming from the hard disk. Does anyone know what this beep means? Is the hard drive just dead, or is it fixable and the data recoverable? Many thanks,

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  • What is the @ sign on the end of file permission on terminal?

    - by shannoga
    I have a sound file in my app that the iPhone does not play. After checking other problems I checked the file permission in terminal. What I can see is that the file permission of this file has a- @ at the end of it. I don't know if that is the problem but this is the only difference from the other sound files that plays fine. What is this sign ? Could it cause a problem ? EDIT Thanks this is what I get: com.apple.FinderInfo: 00000000 4D 34 41 20 68 6F 6F 6B 00 00 00 00 00 00 00 00 |M4A hook........| 00000010 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 |................| 00000020 Thanks Shani

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  • Game Center Leaderboard not dismissing

    - by FireStorm
    I was implementing Game Center into my app and all was going well except for the leaderboard done button not dismissing the leaderboard even with gameCenterControllerDidFinish added in. I call up the leaderboard with the touch of a button in the .m file as so: - (void)touchesBegan:(NSSet *)touches withEvent:(UIEvent *)event { UITouch *touch = [touches anyObject]; CGPoint location = [touch locationInNode:self]; SKNode *node = [self nodeAtPoint:location]; if ([node.name isEqualToString:@"rankButton"]) { [self runAction:[SKAction playSoundFileNamed:@"fishtran.m4a" waitForCompletion: NO]]; GKGameCenterViewController *gameCenterController = [[GKGameCenterViewController alloc] init]; if (gameCenterController != nil) { gameCenterController.viewState = GKGameCenterViewControllerStateAchievements; UIViewController *vc = self.view.window.rootViewController; [vc presentViewController: gameCenterController animated: YES completion:nil]; } } else if ([node.name isEqualToString:@"Leaderboard"]) { GKGameCenterViewController *gameCenterController = [[GKGameCenterViewController alloc] init]; if (gameCenterController != nil) { gameCenterController.viewState = GKGameCenterViewControllerStateLeaderboards; UIViewController *vc = self.view.window.rootViewController; [vc presentViewController: gameCenterController animated: YES completion:nil]; } } ... and then I added thegameCenterControllerDidFinish immediately after as so: - (void)gameCenterControllerDidFinish:(GKGameCenterViewController*)gameCenterController { UIViewController *vc = self.view.window.rootViewController; [vc dismissViewControllerAnimated:YES completion:nil]; } and the done button still doesn't work and i haven't been able to find any solutions. And yes, I do have GKGameCenterControllerDelegate in my .h file. Any help would be greatly appreciated, Thanks!

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  • Flash/Flex: play embedded AAC audio?

    - by aaaidan
    I'm pretty sure of the answer, but just wanted to check with you all. Is it possible to play an embedded AAC file in Flash/Flex somehow? I know you can playback embedded MP3 files, but I hear that you can't do that with AAC. Anyone know any sneaky ways to get around this? By way of illustration, here's come code. [Embed(source='../../audio/music02.m4a', mimeType="audio/aac")] private static const __ExampleMp4File:Class; public var myMp4Sound:Sound = new __ExampleMp4File(); public function EmbeddedAudioTest() { myMp4Sound.play(); }

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  • Microphone audio streaming from Cocoa mac app to iPhone

    - by Benzamin
    Hi devs, I'm trying to build a microphone audio streamer to iPhone. The server software will be a mac desktop app and the client will be iPhone, and they are connected via tcp port. I've successfully connected the mac app and iPhone, and tried to send a fixed test.m4a audio file first. But at the iPhone i grabbed the data well, when tried to play it i used AVAudioPlayer and its returning OSStatus error. I played around with the audio queue service but its very tricky and i only got some example for fixed length audio playing like http://cocoawithlove.com/2009/06/revisiting-old-post-streaming-and.html Now i need help on two things, how can i continuously grab audio data from Mac desktop microphone? And then after grabbing the data how i can play this unfixed length audio data in the iPhone. What exactly i need to do? Please please help me on this......

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  • Why does ffmpeg stop randomly in the middle of a process?

    - by acidzombie24
    ffmpeg feels like its taking a long time. I then look at my output file and i see it stops between 6 and 8mbs. A fully encoded file is about 14mb. Why does ffmpeg stop? My code locks up on StandardOutput.ReadToEnd();. I had to kill the process (after seeing it not move for more then 10 seconds when i see it update every second previously) then i get the results of stdout and err. stdout is "" stderr is below. The output msg shows the filesize ended. I also see a drop in my CPU usage when it stops. I copyed the argument from visual studios. CD to the same working directory and ran the cmd (bin/ffmpeg) and pasted the argument. It was able to complete. int soundProcess(string infn, string outfn) { string aa, aa2; aa = aa2 = "DEAD"; var app = new Process(); app.StartInfo.UseShellExecute = false; app.StartInfo.RedirectStandardOutput = true; app.StartInfo.RedirectStandardError = true; //*/ app.StartInfo.FileName = @"bin\ffmpeg.exe"; app.StartInfo.Arguments = string.Format(@"-i ""{0}"" -ab 192k -y {2} ""{1}""", infn, outfn, param); app.Start(); try { app.PriorityClass = ProcessPriorityClass.BelowNormal; } catch (Exception ex) { if (!Regex.IsMatch(ex.Message, @"Cannot process request because the process .*has exited")) throw ex; } aa = app.StandardOutput.ReadToEnd(); aa2 = app.StandardError.ReadToEnd(); app.WaitForExit(); if (aa2.IndexOf("could not find codec parameters") != -1) return 1; else if (aa == "DEAD" || aa2 == "DEAD") return -1; else if (aa2.Length != 0) return -2; else return 0; } The output of stderr. stdout is empty. FFmpeg version SVN-r15815, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-memalign-hack --enable-postproc --enable-swscale --enable-gpl --enable-libfaac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libx264 --enable-libxvid --disable-ffserver --disable-vhook --enable-avisynth --enable-pthreads libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 6. 1 / 0. 6. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 13 2008 10:28:29, gcc: 4.2.4 (TDM-1 for MinGW) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\dev\src\trunk\prjname\prjname\App_Data/temp/m/o/6304266424778814852': Duration: 00:12:53.36, start: 0.000000, bitrate: 154 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Output #0, ipod, to 'C:\dev\src\trunk\prjname\prjname\App_Data\temp\m\o\2.m4a': Stream #0.0(und): Audio: libfaac, 44100 Hz, stereo, s16, 192 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size= 87kB time=4.74 bitrate= 150.7kbits/s size= 168kB time=9.06 bitrate= 151.9kbits/s size= 265kB time=14.28 bitrate= 151.8kbits/s size= 377kB time=20.29 bitrate= 152.1kbits/s size= 487kB time=26.22 bitrate= 152.1kbits/s size= 594kB time=32.02 bitrate= 152.1kbits/s size= 699kB time=37.64 bitrate= 152.1kbits/s size= 808kB time=43.54 bitrate= 152.0kbits/s size= 930kB time=50.09 bitrate= 152.2kbits/s size= 1058kB time=57.05 bitrate= 152.0kbits/s size= 1193kB time=64.23 bitrate= 152.1kbits/s size= 1329kB time=71.63 bitrate= 152.0kbits/s size= 1450kB time=78.16 bitrate= 152.0kbits/s size= 1578kB time=85.05 bitrate= 152.0kbits/s size= 1706kB time=92.00 bitrate= 152.0kbits/s size= 1836kB time=98.94 bitrate= 152.0kbits/s size= 1971kB time=106.25 bitrate= 151.9kbits/s size= 2107kB time=113.57 bitrate= 152.0kbits/s size= 2214kB time=119.33 bitrate= 152.0kbits/s size= 2345kB time=126.39 bitrate= 152.0kbits/s size= 2479kB time=133.56 bitrate= 152.0kbits/s size= 2611kB time=140.76 bitrate= 152.0kbits/s size= 2745kB time=147.91 bitrate= 152.1kbits/s size= 2880kB time=155.20 bitrate= 152.0kbits/s size= 3013kB time=162.40 bitrate= 152.0kbits/s size= 3146kB time=169.58 bitrate= 152.0kbits/s size= 3277kB time=176.61 bitrate= 152.0kbits/s size= 3412kB time=183.90 bitrate= 152.0kbits/s size= 3540kB time=190.80 bitrate= 152.0kbits/s size= 3670kB time=197.81 bitrate= 152.0kbits/s size= 3805kB time=205.08 bitrate= 152.0kbits/s size= 3932kB time=211.93 bitrate= 152.0kbits/s size= 4052kB time=218.38 bitrate= 152.0kbits/s size= 4171kB time=224.82 bitrate= 152.0kbits/s size= 4277kB time=230.55 bitrate= 152.0kbits/s size= 4378kB time=235.96 bitrate= 152.0kbits/s size= 4486kB time=241.79 bitrate= 152.0kbits/s size= 4592kB time=247.50 bitrate= 152.0kbits/s size= 4698kB time=253.21 bitrate= 152.0kbits/s size= 4804kB time=258.95 bitrate= 152.0kbits/s size= 4906kB time=264.41 bitrate= 152.0kbits/s size= 5012kB time=270.09 bitrate= 152.0kbits/s size= 5118kB time=275.85 bitrate= 152.0kbits/s size= 5234kB time=282.10 bitrate= 152.0kbits/s size= 5331kB time=287.39 bitrate= 151.9kbits/s size= 5445kB time=293.55 bitrate= 152.0kbits/s size= 5555kB time=299.40 bitrate= 152.0kbits/s size= 5665kB time=305.37 bitrate= 152.0kbits/s size= 5766kB time=310.80 bitrate= 152.0kbits/s size= 5876kB time=316.70 bitrate= 152.0kbits/s size= 5984kB time=322.50 bitrate= 152.0kbits/s size= 6094kB time=328.49 bitrate= 152.0kbits/s size= 6212kB time=334.76 bitrate= 152.0kbits/s size= 6327kB time=340.99 bitrate= 152.0kbits/s

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  • Python File Meta Tag reading

    - by Jeff
    Anyone know of a Python module that can pull Tag data from multiple media formats? Trying to build an app that allows for manipulation of ASF (Windows Media Player files, ie WMA, WMV, etc), ID3, including both ID3v1 and ID3v2 (MPEG files, ie MP3), MPEG Audio Bit Stream (ie ABS, MP1, MP2, MP3), MPEG Program Stream (MPEG movies, and DVD and HD DVD video discs, ie MPG, MPEG, VOB, EVO), and ISO Base Media File Format (eg QuickTime, MPEG-4 and iTunes AAC files, ie QT, MOV, MP4, M4A, M4B, M4P, M4V, etc). Don't need ALL of that but just most standard consumer formats like mov and mpeg. I can't seem to find a good module to support that or a library. Any recommendations?

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