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  • Why shouldnt i use flash again?

    - by acidzombie24
    I heard many times i should avoid flash for my website. Yet no one has told me a good reason. I searched for reasons and i see many that are not true (such as text in flash are not indexable by search engines) or may not necessarily be true or significant enough (eating more bandwidth. Would a JS equivalent be bigger or smaller?). My site uses flash to playback sound (m4a). I dont have to worry about indexing, the back button not working, etc. But i have feeling there may be other reasons. What are reasons i shouldnt use flash on my website. I'll note one, the fact iphone/itouch and mobile devices does not support it. Not a big deal for most sites and is obvious. What are reason to avoid flash on my site?

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  • Using system Sound to play sounds

    - by Shoaibi
    Here is the code: -(void)stop { NSLog(@"Disposing Sounds"); AudioServicesDisposeSystemSoundID (soundID); //AudioServicesRemoveSystemSoundCompletion (soundID); } static void completionCallback (SystemSoundID mySSID, void* myself) { NSLog(@"completion Callback"); } - (void) playall: (id) sender { [self stop]; AudioServicesAddSystemSoundCompletion (soundID,NULL,NULL, completionCallback, (void*) self); OSStatus err = kAudioServicesNoError; NSString *aiffPath = [[NSBundle mainBundle] pathForResource:@"slide1" ofType:@"m4a"]; NSURL *aiffURL = [NSURL fileURLWithPath:aiffPath]; err = AudioServicesCreateSystemSoundID((CFURLRef) aiffURL, &soundID); AudioServicesPlaySystemSound (soundID); NSLog(@"Done Playing"); } Output: Disposing Sounds Done Playing In actual no sound gets play at all and completion call back isn't called as well. Any idea what could be wrong here? I want to stop any previous sound before playing current.

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  • Why shouldn't I use Flash?

    - by acidzombie24
    I heard many times i should avoid flash for my website. Yet no one has told me a good reason. I searched for reasons and i see many that are not true (such as text in flash are not indexable by search engines) or may not necessarily be true or significant enough (eating more bandwidth. Would a JS equivalent be bigger or smaller?). My site uses flash to playback sound (m4a). I dont have to worry about indexing, the back button not working, etc. But i have feeling there may be other reasons. What are reasons i shouldnt use flash on my website. I'll note one, the fact iphone/itouch and mobile devices does not support it. Not a big deal for most sites and is obvious. What are reason to avoid flash on my site?

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  • ipod touch crashing after uploading app to device

    - by MaKo
    hi, I installed a new device (the second out of the 100), on xcode, an iPod touch but when I upload the app, the iPod crashes, apple logo shows, and gets frozen for a while, and then resusitates, in the xcode, I get the message on console: The Debugger has exited due to signal 15 (SIGTERM). I tried a simple app I made, and it loaded it, (some bouncing ball) after starting again, but tried the same with another app that plays some sounds and it shows normally, but doesnt play the sounds, questions: how to fix this issue? (in MyApp-info.plist, in bundle identifier, I have: com.yourcompany.${PRODUCT_NAME:rfc1034identifier} havent changed this, is this a problem?? 1.b. I used that conf to upload to an iPad with no problem?? Do the apps play normally sounds *.m4a, in the simulator it works!, not in the iPod, is this due to the crash or not? Thank you,, edit Im using AudioToolbox framework, the question after 1.b is 2 in my editor, but appears as 1 in the post??

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  • Splitting an MP4 file

    - by Asaf Chertkoff
    what is the fastest and less resource consuming method for splitting an MP4 file? @Alex: it didn't work, i don't know why. see the out put here: asafche@asafche-laptop:~$ ffmpeg -vcodec copy -ss 0 -t 00:10:00 -i /home/asafche/Videos/myVideos/MAH00124.MP4 /home/asafche/Videos/myVideos/eh.mp4 FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.1-1ubuntu1.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Mar 31 2011 18:53:20, gcc: 4.4.3 Seems stream 0 codec frame rate differs from container frame rate: 119.88 (120000/1001) -> 59.94 (60000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/asafche/Videos/myVideos/MAH00124.MP4': Duration: 00:15:35.96, start: 0.000000, bitrate: 5664 kb/s Stream #0.0(und): Video: h264, yuv420p, 1280x720, 59.94 tbr, 59.94 tbn, 119.88 tbc Stream #0.1(und): Audio: aac, 48000 Hz, stereo, s16 Output #0, mp4, to '/home/asafche/Videos/myVideos/eh.mp4': Stream #0.0(und): Video: libx264, yuv420p, 1280x720, q=2-31, 90k tbn, 59.94 tbc Stream #0.1(und): Audio: 0x0000, 48000 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Unsupported codec for output stream #0.1 it says something about different frame rate...

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  • How to write files in specific order?

    - by Bernie
    Okay, here's a weird problem -- My wife just bought a 2014 Nissan Altima. So, I took her iTunes library and converted the .m4a files to .mp3, since the car audio system only supports .mp3 and .wma. So far so good. Then I copied the files to a DOS FAT-32 formatted USB thumb drive, and connected the drive to the car's USB port, only to find all of the tracks were out of sequence. All tracks begin with a two digit numeric prefix, i.e., 01, 02, 03, etc. So you would think they would be in order. So I called Nissan Connect support and the rep told me that there is a known problem with reading files in the correct order. He said the files are read in the same order they are written. So, I manually copied a few albums with the tracks in a predetermined order, and sure enough he was correct. So I copied about 6 albums for testing, then changed to the top level directory and did a "find . music.txt". Then I passed this file to rsync like this: rsync -av --files-from=music.txt . ../Marys\ Music\ Sequenced/ The files looked like they were copied in order, but when I listed the files in order of modified time, they were in the same sequence as the original files: ../Marys Music Sequenced/Air Supply/Air Supply Greatest Hits ls -1rt 01 Lost In Love.mp3 04 Every Woman In The World.mp3 03 Chances.mp3 02 All Out Of Love.mp3 06 Here I Am (Just When I Thought I Was Over You).mp3 05 The One That You Love.mp3 08 I Want To Give It All.mp3 07 Sweet Dreams.mp3 11 Young Love.mp3 So the question is, how can I copy files listed in a file named music.txt, and copy them to a destination, and ensure the modification times are in the same sequence as the files are listed?

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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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  • Why do most songs in my media collection play twice? - Corrupt media?

    - by Dean
    Problem: Whether I'm playing the media with Rhythmbox on Ubuntu, Winamp on Windows, or my Nokia N95's media player, most of my audio files (OK, maybe only 40%) play twice. Info: I have a 500GB external 2.5" WD HDD, with a 150GB primary FAT32 partition labeled MUSIC. Inside this, I have about 500 folders containing about 10,000 MP3/WMA/M4A/WAV files. I manage the drive using Ubuntu 9.10, and frequently copy data to/from it using RSYNC, or on windows, TotalCopy. The visual output is different in each media player, but it behaves as if the 1 MP3 has the same song on it twice, and as soon as it ends it begins again. Winamp shows that the song goes for 2x as long as it should, The N95's media player shows the progress bar off the right-hand-side of the screen when it begins playing (then jumps back to the left, then continues along...). Rhythmbox doesn't show me how long the song is, nor does the progress bar move along the screen. Plea: It seams to me somewhere along the lines my collection has become corrupt... but where? And how? and please someone tell me I can fix it!! TIA, Dean.

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  • FFMPEG: how to add watermark to video?

    - by DocWiki
    My Platform: Ubuntu 10.10 + FFMPEG 0.5.3(I installed ffmpeg from source) I try to add Watermark to a .MOV video with FFMPEG 0.5.3 imlib2.so (Please note FFMPEG 0.6+ dont support imlib2.so, so I use ffmpeg 0.5.3) Here is my code: ffmpeg -sameq -i example.mov -vhook '/usr/local/lib/vhook/imlib2.so -x 0 -y 0 -i /var/www/files/watermark.png' newexample.mov Here is the output: FFmpeg version 0.5.3, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-avfilter --enable-filter=movie --enable-avfilter-lavf libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 built on Jul 3 2011 12:05:08, gcc: 4.4.5 Seems stream 1 codec frame rate differs from container frame rate: 59.94 (5994/100) - 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'example.mov': Duration: 00:03:14.06, start: 0.000000, bitrate: 3350 kb/s Stream #0.0(eng): Audio: aac, 48000 Hz, stereo, s16 Stream #0.1(eng): Video: h264, yuv420p, 1150x647, 29.97 tbr, 29.97 tbn, 59.94 tbc Output #0, mov, to 'newexample.mov': Stream #0.0(eng): Video: mpeg4, yuv420p, 1150x647, q=2-31, 200 kb/s, 90k tbn, 29.97 tbc Stream #0.1(eng): Audio: 0x0000, 48000 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.1 - #0.0 Stream #0.0 - #0.1 Unsupported codec for output stream #0.1 What could be the possible problem? Is that AAC or H264 that is not supported? I installed libavcodec-extra-52, linfaac, libfaad and etc. but the error is the same. Do I have to install following this instruction? HOWTO: Install and use the latest FFmpeg and x264 or there is a simpler solution?

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  • How do I convert a video to GIF using ffmpeg, with reasonable quality?

    - by Kamil Hismatullin
    I'm converting .flv movie to .gif file with ffmpeg. ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif It works great, but output gif file has a very law quality. Any ideas how can I improve quality of converted gif? Output of command: $ ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif ffmpeg version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:52:53 with gcc 4.7.2 *** THIS PROGRAM IS DEPRECATED *** This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.flv': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 Duration: 00:00:18.85, start: 0.000000, bitrate: 3098 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x720, 2905 kb/s, 25 fps, 25 tbr, 50 tbn, 50 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 192 kb/s Metadata: creation_time : 2013-02-14 04:00:07 [buffer @ 0x92a8ea0] w:1280 h:720 pixfmt:yuv420p [scale @ 0x9215100] w:1280 h:720 fmt:yuv420p -> w:320 h:240 fmt:rgb24 flags:0x4 Output #0, gif, to 'output.gif': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 encoder : Lavf53.21.1 Stream #0.0(und): Video: rawvideo, rgb24, 320x240, q=2-31, 200 kb/s, 90k tbn, 10 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream mapping: Stream #0.0 -> #0.0 Press ctrl-c to stop encoding frame= 101 fps= 32 q=0.0 Lsize= 8686kB time=10.10 bitrate=7045.0kbits/s dup=0 drop=149 video:22725kB audio:0kB global headers:0kB muxing overhead -61.778676% Thanks.

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  • ffmpeg conversion problem

    - by user33126
    installed ffmpeg and it shows version and all correctly. but even info ffmpeg command itself shows ffmpeg -i Alice_In_Wonderland.mp4 gives messgae like FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --extra-cflags=-fPIC --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:11:04, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 1 codec frame rate differs from container frame rate: 49.93 (9986/200) - 49.92 (599/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Alice_In_Wonderland.mp4': Duration: 00:01:39.65, start: 0.000000, bitrate: 542 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Stream #0.1(und): Video: h264, yuv420p, 480x270, 49.92 tbr, 24.96 tbn, 49.93 tbc At least one output file must be specified Please tell me whats the problem

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  • ffmpeg conversion problem

    - by Elamurugan
    installed ffmpeg and it shows version and all correctly. but even info ffmpeg command itself shows ffmpeg -i Alice_In_Wonderland.mp4 gives messgae like FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --extra-cflags=-fPIC --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:11:04, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 1 codec frame rate differs from container frame rate: 49.93 (9986/200) - 49.92 (599/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Alice_In_Wonderland.mp4': Duration: 00:01:39.65, start: 0.000000, bitrate: 542 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Stream #0.1(und): Video: h264, yuv420p, 480x270, 49.92 tbr, 24.96 tbn, 49.93 tbc At least one output file must be specified Please tell me whats the problem

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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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  • Would like to change audio codec, but keep video settings with ffmpeg

    - by Craig Tataryn
    I have a video for which I'd like to convert the audio codec to AAC 320 kbps / 44.100 kHz. What would I use for ffmpeg switches such that all the video settings and codec remain the same, but only the audio codec and settings change? Here's my video: $ ffmpeg -i Winnipeg.rb\ Scala-Talk.mov FFmpeg version SVN-r25375, Copyright (c) 2000-2010 the FFmpeg developers built on Oct 6 2010 13:02:41 with gcc 4.2.1 (Apple Inc. build 5664) configuration: --enable-libmp3lame --enable-shared --disable-mmx --arch=x86_64 libavutil 50.32. 2 / 50.32. 2 libavcore 0. 9. 1 / 0. 9. 1 libavcodec 52.92. 0 / 52.92. 0 libavformat 52.80. 0 / 52.80. 0 libavdevice 52. 2. 2 / 52. 2. 2 libavfilter 1.48. 0 / 1.48. 0 libswscale 0.12. 0 / 0.12. 0 Seems stream 0 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 10.00 (10/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Winnipeg.rb Scala-Talk.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt Duration: 01:10:53.00, start: 0.000000, bitrate: 283 kb/s Stream #0.0(eng): Video: h264, yuv420p, 800x598, 94 kb/s, 10 fps, 10 tbr, 1k tbn, 2k tbc Stream #0.1(eng): Audio: adpcm_ima_qt, 22050 Hz, 1 channels, s16 Stream #0.2(eng): Audio: adpcm_ima_qt, 22050 Hz, 1 channels, s16 At least one output file must be specified Many thanks in advance! One with with ffmpeg I've never been able to grok is how to just "tweak" files without having to regurgitate every little setting for things you don't want changes.

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  • WordPress not resizing images with Nginx + php-fpm and other issues

    - by Julian Fernandes
    Recently i setup a Ubuntu 12.04 VPS with 512mb/1ghz CPU, Nginx + php-fpm + Varnish + APC + Percona's MySQL server + CloudFlare Pro for our Ubuntu LoCo Team's WordPress blog. The blog get about 3~4k daily hits, use about 180MB and 8~20% CPU. Everything seems to be working insanely fast... page load is really good and is about 16x faster than any of our competitors... but there is one problem. When we upload a image, WordPress don't resize it, so all we can do it insert the full image in the post. If the imagem have, let's say, 30kb, it resize fine... but if the image have 100kb+, it won't... In nginx error logs i see this: upstream timed out (110: Connection timed out) while reading response header from upstream, client: 150.162.216.64, server: www.ubuntubrsc.com, request: "POST /wp-admin/async-upload.php HTTP/1.1", upstream: "fastcgi://unix:/var/run/php5-fpm.sock:", host: "www.ubuntubrsc.com", referrer: "http://www.ubuntubrsc.com/wp-admin/media-upload.php?post_id=2668&" It seems to be related with the issue, but i dunno. When that timeout happens, i started to get it when i'm trying to view a post too: upstream timed out (110: Connection timed out) while reading response header from upstream, client: 150.162.216.64, server: www.ubuntubrsc.com, request: "GET /tutoriais-gimp-6-adicionando-aplicando-novos-pinceis.html HTTP/1.1", upstream: "fastcgi://unix:/var/run/php5-fpm.sock:", host: "www.ubuntubrsc.com", referrer: "http://www.ubuntubrsc.com/" And only a restart of php5-fpm fix it. I tryed increasing some timeouts and stuffs but it did not worked, so i guess it's some kind of limitation i did not figured yet. Could someone help me with it, please? /etc/nginx/nginx.conf: user www-data; worker_processes 1; pid /var/run/nginx.pid; events { worker_connections 1024; use epoll; multi_accept on; } http { ## # Basic Settings ## sendfile on; tcp_nopush on; tcp_nodelay off; keepalive_timeout 15; keepalive_requests 2000; types_hash_max_size 2048; server_tokens off; server_name_in_redirect off; open_file_cache max=1000 inactive=300s; open_file_cache_valid 360s; open_file_cache_min_uses 2; open_file_cache_errors off; server_names_hash_bucket_size 64; # server_name_in_redirect off; client_body_buffer_size 128K; client_header_buffer_size 1k; client_max_body_size 2m; large_client_header_buffers 4 8k; client_body_timeout 10m; client_header_timeout 10m; send_timeout 10m; include /etc/nginx/mime.types; default_type application/octet-stream; ## # Logging Settings ## error_log /var/log/nginx/error.log; access_log off; ## # CloudFlare's IPs (uncomment when site goes live) ## set_real_ip_from 204.93.240.0/24; set_real_ip_from 204.93.177.0/24; set_real_ip_from 199.27.128.0/21; set_real_ip_from 173.245.48.0/20; set_real_ip_from 103.22.200.0/22; set_real_ip_from 141.101.64.0/18; set_real_ip_from 108.162.192.0/18; set_real_ip_from 190.93.240.0/20; real_ip_header CF-Connecting-IP; set_real_ip_from 127.0.0.1/32; ## # Gzip Settings ## gzip on; gzip_disable "msie6"; gzip_vary on; gzip_proxied any; gzip_comp_level 9; gzip_min_length 1000; gzip_proxied expired no-cache no-store private auth; gzip_buffers 32 8k; # gzip_http_version 1.1; gzip_types text/plain text/css application/json application/x-javascript text/xml application/xml application/xml+rss text/javascript; ## # nginx-naxsi config ## # Uncomment it if you installed nginx-naxsi ## #include /etc/nginx/naxsi_core.rules; ## # nginx-passenger config ## # Uncomment it if you installed nginx-passenger ## #passenger_root /usr; #passenger_ruby /usr/bin/ruby; ## # Virtual Host Configs ## include /etc/nginx/conf.d/*.conf; include /etc/nginx/sites-enabled/*; } /etc/nginx/fastcgi_params: fastcgi_param QUERY_STRING $query_string; fastcgi_param REQUEST_METHOD $request_method; fastcgi_param CONTENT_TYPE $content_type; fastcgi_param CONTENT_LENGTH $content_length; fastcgi_param SCRIPT_FILENAME $request_filename; fastcgi_param SCRIPT_NAME $fastcgi_script_name; fastcgi_param REQUEST_URI $request_uri; fastcgi_param DOCUMENT_URI $document_uri; fastcgi_param DOCUMENT_ROOT $document_root; fastcgi_param SERVER_PROTOCOL $server_protocol; fastcgi_param GATEWAY_INTERFACE CGI/1.1; fastcgi_param SERVER_SOFTWARE nginx/$nginx_version; fastcgi_param REMOTE_ADDR $remote_addr; fastcgi_param REMOTE_PORT $remote_port; fastcgi_param SERVER_ADDR $server_addr; fastcgi_param SERVER_PORT $server_port; fastcgi_param SERVER_NAME $server_name; fastcgi_param HTTPS $https; fastcgi_send_timeout 180; fastcgi_read_timeout 180; fastcgi_buffer_size 128k; fastcgi_buffers 256 4k; # PHP only, required if PHP was built with --enable-force-cgi-redirect fastcgi_param REDIRECT_STATUS 200; /etc/nginx/sites-avaiable/default: ## # DEFAULT HANDLER # ubuntubrsc.com ## server { listen 8080; # Make site available from main domain server_name www.ubuntubrsc.com; # Root directory root /var/www; index index.php index.html index.htm; include /var/www/nginx.conf; access_log off; location / { try_files $uri $uri/ /index.php?q=$uri&$args; } location = /favicon.ico { log_not_found off; access_log off; } location = /robots.txt { allow all; log_not_found off; access_log off; } location ~ /\. { deny all; access_log off; log_not_found off; } location ~* ^/wp-content/uploads/.*.php$ { deny all; access_log off; log_not_found off; } rewrite /wp-admin$ $scheme://$host$uri/ permanent; error_page 404 = @wordpress; log_not_found off; location @wordpress { include /etc/nginx/fastcgi_params; fastcgi_pass unix:/var/run/php5-fpm.sock; fastcgi_param SCRIPT_NAME /index.php; fastcgi_param SCRIPT_FILENAME $document_root/index.php; } location ~ \.php$ { try_files $uri =404; include /etc/nginx/fastcgi_params; fastcgi_index index.php; fastcgi_param SCRIPT_FILENAME $document_root$fastcgi_script_name; if (-f $request_filename) { fastcgi_pass unix:/var/run/php5-fpm.sock; } } } server { listen 8080; server_name ubuntubrsc.* www.ubuntubrsc.net www.ubuntubrsc.org www.ubuntubrsc.com.br www.ubuntubrsc.info www.ubuntubrsc.in; return 301 $scheme://www.ubuntubrsc.com$request_uri; } /var/www/nginx.conf: # BEGIN W3TC Minify cache location ~ /wp-content/w3tc/min.*\.js$ { types {} default_type application/x-javascript; expires modified 31536000s; add_header X-Powered-By "W3 Total Cache/0.9.2.5b"; add_header Vary "Accept-Encoding"; add_header Pragma "public"; add_header Cache-Control "max-age=31536000, public, must-revalidate, proxy-revalidate"; } location ~ /wp-content/w3tc/min.*\.css$ { types {} default_type text/css; expires modified 31536000s; add_header X-Powered-By "W3 Total Cache/0.9.2.5b"; add_header Vary "Accept-Encoding"; add_header Pragma "public"; add_header Cache-Control "max-age=31536000, public, must-revalidate, proxy-revalidate"; } location ~ /wp-content/w3tc/min.*js\.gzip$ { gzip off; types {} default_type application/x-javascript; expires modified 31536000s; add_header X-Powered-By "W3 Total Cache/0.9.2.5b"; add_header Vary "Accept-Encoding"; add_header Pragma "public"; add_header Cache-Control "max-age=31536000, public, must-revalidate, proxy-revalidate"; add_header Content-Encoding gzip; } location ~ /wp-content/w3tc/min.*css\.gzip$ { gzip off; types {} default_type text/css; expires modified 31536000s; add_header X-Powered-By "W3 Total Cache/0.9.2.5b"; add_header Vary "Accept-Encoding"; add_header Pragma "public"; add_header Cache-Control "max-age=31536000, public, must-revalidate, proxy-revalidate"; add_header Content-Encoding gzip; } # END W3TC Minify cache # BEGIN W3TC Browser Cache gzip on; gzip_types text/css application/x-javascript text/x-component text/richtext image/svg+xml text/plain text/xsd text/xsl text/xml image/x-icon; location ~ \.(css|js|htc)$ { expires 31536000s; add_header Pragma "public"; add_header Cache-Control "max-age=31536000, public, must-revalidate, proxy-revalidate"; add_header X-Powered-By "W3 Total Cache/0.9.2.5b"; } location ~ \.(html|htm|rtf|rtx|svg|svgz|txt|xsd|xsl|xml)$ { expires 3600s; add_header Pragma "public"; add_header Cache-Control "max-age=3600, public, must-revalidate, proxy-revalidate"; add_header X-Powered-By "W3 Total Cache/0.9.2.5b"; try_files $uri $uri/ $uri.html /index.php?$args; } location ~ \.(asf|asx|wax|wmv|wmx|avi|bmp|class|divx|doc|docx|eot|exe|gif|gz|gzip|ico|jpg|jpeg|jpe|mdb|mid|midi|mov|qt|mp3|m4a|mp4|m4v|mpeg|mpg|mpe|mpp|otf|odb|odc|odf|odg|odp|ods|odt|ogg|pdf|png|pot|pps|ppt|pptx|ra|ram|svg|svgz|swf|tar|tif|tiff|ttf|ttc|wav|wma|wri|xla|xls|xlsx|xlt|xlw|zip)$ { expires 31536000s; add_header Pragma "public"; add_header Cache-Control "max-age=31536000, public, must-revalidate, proxy-revalidate"; add_header X-Powered-By "W3 Total Cache/0.9.2.5b"; } # END W3TC Browser Cache # BEGIN W3TC Minify core rewrite ^/wp-content/w3tc/min/w3tc_rewrite_test$ /wp-content/w3tc/min/index.php?w3tc_rewrite_test=1 last; set $w3tc_enc ""; if ($http_accept_encoding ~ gzip) { set $w3tc_enc .gzip; } if (-f $request_filename$w3tc_enc) { rewrite (.*) $1$w3tc_enc break; } rewrite ^/wp-content/w3tc/min/(.+\.(css|js))$ /wp-content/w3tc/min/index.php?file=$1 last; # END W3TC Minify core # BEGIN W3TC Skip 404 error handling by WordPress for static files if (-f $request_filename) { break; } if (-d $request_filename) { break; } if ($request_uri ~ "(robots\.txt|sitemap(_index)?\.xml(\.gz)?|[a-z0-9_\-]+-sitemap([0-9]+)?\.xml(\.gz)?)") { break; } if ($request_uri ~* \.(css|js|htc|htm|rtf|rtx|svg|svgz|txt|xsd|xsl|xml|asf|asx|wax|wmv|wmx|avi|bmp|class|divx|doc|docx|eot|exe|gif|gz|gzip|ico|jpg|jpeg|jpe|mdb|mid|midi|mov|qt|mp3|m4a|mp4|m4v|mpeg|mpg|mpe|mpp|otf|odb|odc|odf|odg|odp|ods|odt|ogg|pdf|png|pot|pps|ppt|pptx|ra|ram|svg|svgz|swf|tar|tif|tiff|ttf|ttc|wav|wma|wri|xla|xls|xlsx|xlt|xlw|zip)$) { return 404; } # END W3TC Skip 404 error handling by WordPress for static files # BEGIN Better WP Security location ~ /\.ht { deny all; } location ~ wp-config.php { deny all; } location ~ readme.html { deny all; } location ~ readme.txt { deny all; } location ~ /install.php { deny all; } set $susquery 0; set $rule_2 0; set $rule_3 0; rewrite ^wp-includes/(.*).php /not_found last; rewrite ^/wp-admin/includes(.*)$ /not_found last; if ($request_method ~* "^(TRACE|DELETE|TRACK)"){ return 403; } set $rule_0 0; if ($request_method ~ "POST"){ set $rule_0 1; } if ($uri ~ "^(.*)wp-comments-post.php*"){ set $rule_0 2$rule_0; } if ($http_user_agent ~ "^$"){ set $rule_0 4$rule_0; } if ($rule_0 = "421"){ return 403; } if ($args ~* "\.\./") { set $susquery 1; } if ($args ~* "boot.ini") { set $susquery 1; } if ($args ~* "tag=") { set $susquery 1; } if ($args ~* "ftp:") { set $susquery 1; } if ($args ~* "http:") { set $susquery 1; } if ($args ~* "https:") { set $susquery 1; } if ($args ~* "(<|%3C).*script.*(>|%3E)") { set $susquery 1; } if ($args ~* "mosConfig_[a-zA-Z_]{1,21}(=|%3D)") { set $susquery 1; } if ($args ~* "base64_encode") { set $susquery 1; } if ($args ~* "(%24&x)") { set $susquery 1; } if ($args ~* "(\[|\]|\(|\)|<|>|ê|\"|;|\?|\*|=$)"){ set $susquery 1; } if ($args ~* "(&#x22;|&#x27;|&#x3C;|&#x3E;|&#x5C;|&#x7B;|&#x7C;|%24&x)"){ set $susquery 1; } if ($args ~* "(%0|%A|%B|%C|%D|%E|%F|127.0)") { set $susquery 1; } if ($args ~* "(globals|encode|localhost|loopback)") { set $susquery 1; } if ($args ~* "(request|select|insert|concat|union|declare)") { set $susquery 1; } if ($http_cookie !~* "wordpress_logged_in_" ) { set $susquery "${susquery}2"; set $rule_2 1; set $rule_3 1; } if ($susquery = 12) { return 403; } # END Better WP Security /etc/php5/fpm/php-fpm.conf: pid = /var/run/php5-fpm.pid error_log = /var/log/php5-fpm.log emergency_restart_threshold = 3 emergency_restart_interval = 1m process_control_timeout = 10s events.mechanism = epoll /etc/php5/fpm/php.ini (only options i changed): open_basedir ="/var/www/" disable_functions = pcntl_alarm,pcntl_fork,pcntl_waitpid,pcntl_wait,pcntl_wifexited,pcntl_wifstopped,pcntl_wifsignaled,pcntl_wexitstatus,pcntl_wtermsig,pcntl_wstopsig,pcntl_signal,pcntl_signal_dispatch,pcntl_get_last_error,pcntl_strerror,pcntl_sigprocmask,pcntl_sigwaitinfo,pcntl_sigtimedwait,pcntl_exec,pcntl_getpriority,pcntl_setpriority,dl,system,shell_exec,fsockopen,parse_ini_file,passthru,popen,proc_open,proc_close,shell_exec,show_source,symlink,proc_close,proc_get_status,proc_nice,proc_open,proc_terminate,shell_exec ,highlight_file,escapeshellcmd,define_syslog_variables,posix_uname,posix_getpwuid,apache_child_terminate,posix_kill,posix_mkfifo,posix_setpgid,posix_setsid,posix_setuid,escapeshellarg,posix_uname,ftp_exec,ftp_connect,ftp_login,ftp_get,ftp_put,ftp_nb_fput,ftp_raw,ftp_rawlist,ini_alter,ini_restore,inject_code,syslog,openlog,define_syslog_variables,apache_setenv,mysql_pconnect,eval,phpAds_XmlRpc,phpA ds_remoteInfo,phpAds_xmlrpcEncode,phpAds_xmlrpcDecode,xmlrpc_entity_decode,fp,fput,virtual,show_source,pclose,readfile,wget expose_php = off max_execution_time = 30 max_input_time = 60 memory_limit = 128M display_errors = Off post_max_size = 2M allow_url_fopen = off default_socket_timeout = 60 APC settings: [APC] apc.enabled = 1 apc.shm_segments = 1 apc.shm_size = 64M apc.optimization = 0 apc.num_files_hint = 4096 apc.ttl = 60 apc.user_ttl = 7200 apc.gc_ttl = 0 apc.cache_by_default = 1 apc.filters = "" apc.mmap_file_mask = "/tmp/apc.XXXXXX" apc.slam_defense = 0 apc.file_update_protection = 2 apc.enable_cli = 0 apc.max_file_size = 10M apc.stat = 1 apc.write_lock = 1 apc.report_autofilter = 0 apc.include_once_override = 0 apc.localcache = 0 apc.localcache.size = 512 apc.coredump_unmap = 0 apc.stat_ctime = 0 /etc/php5/fpm/pool.d/www.conf user = www-data group = www-data listen = /var/run/php5-fpm.sock listen.owner = www-data listen.group = www-data listen.mode = 0666 pm = ondemand pm.max_children = 5 pm.process_idle_timeout = 3s; pm.max_requests = 50 I also started to get 404 errors in front page if i use W3 Total Cache's Page Cache (Disk Enhanced). It worked fine untill somedays ago, and then, out of nowhere, it started to happen. Tonight i will disable my mobile plugin and activate only W3 Total Cache to see if it's a conflict with them... And to finish all this, i have been getting this error: PHP Warning: apc_store(): Unable to allocate memory for pool. in /var/www/wp-content/plugins/w3-total-cache/lib/W3/Cache/Apc.php on line 41 I already modifed my APC settings, but no sucess. So... could anyone help me with those issuees, please? Ooohh... if it helps, i instaled PHP like this: sudo apt-get install php5-fpm php5-suhosin php-apc php5-gd php5-imagick php5-curl And Nginx from the official PPA. Sorry for my bad english and thanks for your time people! (:

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  • Bash scripting problem

    - by komidore64
    I'm writing a bash script to sync my iTunes music directory to a directory on a removable hard drive. The script works fine when there is absolutely nothing in the folder on the external hard drive. Once all files have been copied to the external drive, then the script begins to act strange. Even though i just sync'd everything over, it proceeds to recopy certain files again. After the initial sync, it chooses the same files to resync each consecutive time the script is executed without any changes being made to the source directory. #!/bin/bash # shell script to sync music with gigabeat and/or firewire drive musicdir="/Users/komidore64/Music/iTunes/iTunes Media/Music" gigadir="/Volumes/GIGABEAT/music" # fwdir="/Volumes/" remove() { find "$1" \ ! \( -name "*.wav" \ -o -name "*.ogg" \ -o -name "*.flac" \ -o -name "*.aac" \ -o -name "*.mp3" \ -o -name "*.m4a" \ -o -name "*.wma" \ -o -name "*.m4p" \ -o -name "*.ape" \ -o -type d \) \ -exec rm -i {} \; } if [ $# == 0 ]; then echo "no device argument present" echo "specify '-g' for gigabeat" echo "or '-f' for firewire drive" else remove "$musicdir" while [ $1 ]; do case $1 in -g | --gigabeat ) rsync --archive --verbose --delete "$musicdir/" "$gigadir" ;; -f | --firewire ) rsync --archive --verbose --delete "$musicdir/" "$fwdir" esac shift done echo "music synced" fi

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  • How to Watch Youtube Videos on PSP with iMoviesoft FLV Converter

    - by user312417
    Do you have worried about it? You can not watch Youtube videos anytime, anywhere.It is so boring on the way to work and home.How you want to be able to enjoy the wonderful Youtube Video on PSP that you can watch them on the way to home, home on bus. This artice will tell you about how to convert Youtube VIdeos to PSP Player, take "Alice.in.Wonderland" as an example, We can use iMoviesoft FLV Converter to convert it to PSP video file. iMoviesoft FLV Converter is a powerful FLV Converter which can convert FLV and YouTube Videos to almost any video formats, with excellent conversion speed and quality, such as converting FLV to MP4, FLV to AVI, FLV to WMV, FLV to MPEG etc. Furthermore, it can also easily convert video files to some popular audio formats, such as WMA, MP3, M4A, AAC, etc. You can convert FLV and YouTube videos to PSP, iPod, iPhone, Zune video player and other portable video players. After easy and wonderful conversion, you can fully enjoy videos on your PSP, iPod, iPhone and some other portable video players. Besides, you can also use it to join videos. Merge several videos into one output PSP video and enjoy them conveniently. You can also trim your favarite clips or remove the video black edges by [iMoviesoft FLV Converter. Hope to help every Video Enthusiasts.

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  • AVAudioPlayer only initializes with some files

    - by Brendan
    Hi everyone, I'm having trouble playing some files with AVAudioPlayer. When I try to play a certain m4a, it works fine. It also works with an mp3 that I try. However it fails on one particular mp3 every time (15 Step, by Radiohead), regardless of the order in which I try to play them. The audio just does not play, though the view loading and everything that happens concurrently happens correctly. The code is below. I get the "Player loaded." log output on the other two songs, but not on 15 Step. I know the file path is correct (I have it log outputted earlier in the app, and it is correct). Any ideas? NSData *musicData = [NSData dataWithContentsOfURL:[[NSURL alloc] initFileURLWithPath:[[NSBundle mainBundle] pathForResource:[song filename] ofType:nil]]]; NSLog([[NSBundle mainBundle] pathForResource:[song filename] ofType:nil]); if(musicData) { NSLog(@"File found."); } self.songView.player = [[AVAudioPlayer alloc] initWithData:musicData error:nil]; if(self.songView.player) { NSLog(@"Player loaded."); } [self.songView.player play]; NSLog(@"You should be hearing something now."); Thanks, Brendan

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  • Play multiple audio files using AVAudioPlayer

    - by inScript09
    Hi all, I am planning on releasing 10 of my song recordings for free but bundled in an iphone app. They are not available on web or itunes or anywhere as of now. I am new to iphone sdk (latest) as you can imagine, so I have been going through the developer documentation, various forums and stackoverflow to learn. Apple's avTouch sample application was a great start. But I want my app to play all the 10 tracks one by one. All the songs are added to resources folder and are named as track1, track2...track10. In the avTouch app code I can see the following 2 parts which is where I think I need to make changes to achieve what I am looking for. But I am lost. // Load the array with the sample file NSURL *fileURL = [[NSURL alloc] initFileURLWithPath: [[NSBundle mainBundle] pathForResource:@"sample" ofType:@"m4a"]]; - (void)audioPlayerDidFinishPlaying:(AVAudioPlayer *)player successfully:(BOOL)flag { if (flag == NO) NSLog(@"Playback finished unsuccessfully"); [player setCurrentTime:0.]; [self updateViewForPlayerState]; } can anyone please help me on 1. how to load the array with all the 10 tracks which are added to resources folder 2. and when I hit play, player should start the first track. when the 1st track ends 2nd track should start and so on for the remaining tracks. Thank You

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  • c# regex split and extract multiple parts from a string

    - by nLL
    Hi, I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) - 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to diffrent string objects instead of single

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  • Applying for .net jobs as a "self learner"

    - by DeanMc
    Hi All, I have recently started applying for .Net jobs. I currently work in a sales role with a large telco. I found out quite late that I like programming and as such bought my house and made commitments that mean college is not an option. What I would like to know is, is it harder to get a junior job as a self learner? I have gotten a few enquiries regarding my C.V but nothing concrete yet. I try to be involved in projects as I get the chance and tend to put up any worthwhile projects as I develop them. Some examples of my work are: A Xaml lexer and parser: http://www.xlight.mendhak.com A font obfuscation tool: http://www.silverlightforums.com/showthread.php?1516-Font-Obsfucation-Tool-ALPHA A tagger for m4a: http://projectaudiophile.codeplex.com/SourceControl/list/changesets I, of course think that these are great examples of my work but that is my opinion based on self learning. The other query is how much should I actually know? I've never used linked lists but I know that strings are immutable and I understand what that means. I am only touching on T-SQL but I understand things like how properties function in IL (as two standard methods :) ). I suppose I understand a lot of concepts but specific features need some looking up to implement as I may not know the syntax off the top of my head.

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  • AVAudioPlayer not unloading cached memory after each new allocation

    - by Rob
    I am seeing in Instruments that when I play a sound via the standard "AddMusic" example method that Apple provides, it allocates 32kb of memory via the prepareToPlay call (which references the AudioToolBox framework's Cache_DataSource::ReadBytes function) each time a new player is allocated (i.e. each time a different sound is played). However, that cached data never gets released. This obviously poses a huge problem if it doesn't get released and you have a lot of sound files to play, since it tends to keep allocating memory and eventually crashes if you have enough unique sound files (which I unfortunately do). Have any of you run across this or what am I doing wrong in my code? I've had this issue for a while now and it's really bugging me since my code is verbatim of what Apple's is (I think). How I call the function: - (void)playOnce:(NSString *)aSound { // Gets the file system path to the sound to play. NSString *soundFilePath = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; // Converts the sound's file path to an NSURL object NSURL *soundURL = [[NSURL alloc] initFileURLWithPath: soundFilePath]; self.soundFileURL = soundURL; [soundURL release]; AVAudioPlayer * newAudio=[[AVAudioPlayer alloc] initWithContentsOfURL: soundFileURL error:nil]; self.theAudio = newAudio; // automatically retain audio and dealloc old file if new m4a file is loaded [newAudio release]; // release the audio safely // this is where the prior cached data never gets released [theAudio prepareToPlay]; // set it up and play [theAudio setNumberOfLoops:0]; [theAudio setVolume: volumeLevel]; [theAudio setDelegate: self]; [theAudio play]; } and then theAudio gets released in the dealloc method of course.

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  • regex split and extract multiple parts from a string

    - by nLL
    I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) -> 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to different string objects instead of single

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  • ffmpeg hangs when creating a video

    - by FearUs
    I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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  • FFmpeg extract clip - stream frame rate differs from container frame rate (x264, aac)

    - by fideli
    Summary H.264 video seems to have a really high frame rate that requires a scaling factor to the applied to the duration of video that I'm trying to extract (900x lower). Body I'm trying to extract a clip from a movie that I have in MP4 format (created using Handbrake). After trying mencoder and VLC, I decided to give FFmpeg a shot since it was the least troublesome when it came to copying the codecs. That is, compared to mencoder and VLC, the resulting file was still playable in QuickTime (I know about Perian, etc, I'm just trying to learn how all this works). Anyway, my command was as follows: ffmpeg -ss 01:15:51 -t 00:05:59 -i outofsight.mp4 \ -acodec copy -vcodec copy clip.mp4 During the copy, The following comes up: Seems stream 0 codec frame rate differs from container frame rate: 45000.00 (45000/1) -> 25.00 (25/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from outofsight.mp4': Duration: 01:57:42.10, start: 0.000000, bitrate: 830 kb/s Stream #0.0(und): Video: h264, yuv420p, 720x384, 25 tbr, 22500 tbn, 45k tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16 Output #0, mp4, to 'out.mp4': Stream #0.0(und): Video: libx264, yuv420p, 720x384, q=2-31, 90k tbn, 22500 tbc Stream #0.1(eng): Audio: libfaac, 48000 Hz, stereo, s16 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 2591 fps=2349 q=-1.0 size= 8144kB time=101.60 bitrate= 656.7kbits/s … Instead of a 5:59 duration clip, I get the entire rest of the movie. So, to test this, I ran the ffmpeg command with -t 00:00:01. What I got was exactly a 15:00 minute clip. So I did some black box engineering and decided to scale my -t option by calculating what value to enter given that 1 second was interpreted as 900 s. For my desired 359 s clip, I calculated 0.399 s and so my ffmpeg command became: ffmpeg -ss 01:15.51 -t 00:00:00.399 -i outofsight.mp4 \ -acodec copy -vcodec copy clip.mp4 This works, but I have no idea why the duration is scaled by 900. Investigating further, each ffmpeg run has the line: Seems stream 0 codec frame rate differs from container frame rate: 45000.00 (45000/1) -> 25.00 (25/1) 45000/25 = 1800. Must be a relation somewhere. Somehow, the obscenely high frame rate is causing issues with the timing. How is that frame rate so high? The best part about this is that the resulting clip.mp4 has the exact same feature (due to the copied video codec), and taking further clips from this needs the same scaling for the -t duration option. Therefore, I've made it available for anyone willing to check this out. Appendix The preamble for ffmpeg on my system (built using MacPorts ffmpeg port): FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/opt/local --disable-vhook --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libfaac --enable-libfaad --enable-libxvid --enable-libx264 --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/gcc-4.2 --arch=x86_64 libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 1. 4. 0 / 1. 4. 0 libswscale 1. 7. 1 / 1. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jan 4 2010 21:51:51, gcc: 4.2.1 (Apple Inc. build 5646) (dot 1)

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