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• Media Archive System with branches?

  - by Ian McEwen

  In short, how can I get VCS features (revisioning, branching, and deduplication) for a media collection that's far too large for most/all VCS systems? Background I have a 300GB music folder; unfortunately, I only have the hard drive space for this on my desktop system. However, a good portion of my collection is FLAC; therefore, I could theoretically have a space-optimized version in which I transcode all the FLAC to mp3 or some other lossy format, and use only that version on the laptop. However, a portion of my collection isn't FLAC. And that which isn't FLAC shouldn't be transcoded to an equivalent format; it won't have any space savings, which is the point. Moreover, it shouldn't be duplicated: the mp3/ogg portions of the collection should probably be exactly the same files. Thoughts One solution is to have format-specific organization of my music folders, and use some script to transcode the FLAC directory to mp3 or such into another directory. Another is some sort of hack using entirely separate copies and symbolic links for deduplication, or something similar. But these also have a disadvantage of lacking versioning; I'd like to be able to reorganize my music collection, retag things, etc. and save history. This isn't key, but would be awfully nice. I can't see it as entirely unreasonable to set up VCS hooks or something equivalent to keep directory structure synced between two copies, update tags, and transcode FLAC automatically into the space-optimized copy. Basically, the system I really want is a version control system. Two branches: one archival/desktop branch including the FLAC, one space-optimized/laptop branch without it; most VCSes would deal well with whole chunks being the same files well by compressing in a reasonable way (i.e. don't keep two copies of the same data). I could also do a lot of what I talk about above with hooks. But I don't know of any VCS that would deal with a 300GB repository with almost 20k files. Many of them would just not even initialize the whole affair; others would just do it inexpressibly slowly or otherwise badly. checkpoint looks like it's designed for something close (it's at least for media), but wouldn't do deduplication well (and I'm not convinced I'd be able to script it to do things like automatic transcoding and directory-structure syncing). So. Is there anything out there that can do all this, or should I consider it a programming project?

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• Songbird FLAC library to iTunes MP3 library

  - by vitto

  Hello, I wonder if there's some way to automatically export my Songbird FLAC music library to my iTunes MP3 library. Did someone know something about?

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• flac2mp3 on mac?

  - by Josef

  What tools are there to convert flac to mp3?

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• OpenSL ES decode 24bit FLAC

  - by yano

  I am trying to decode a FLAC file with 24bit sample format using OpenSL ES on Android. Originally, I had my SLDataFormat_PCM for the SLDataSink setup like this. _pcm.formatType = SL_DATAFORMAT_PCM; _pcm.numChannels = 2; _pcm.samplesPerSec = SL_SAMPLINGRATE_44_1; _pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; _pcm.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16; _pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; _pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; This is working well for basically any data format. Luckily the samplesPerSec is not respected (I don't want resampling). Now I want to support the full bit-depth of a FLAC file with 24bit samples. When using this format, it apparently performs a bit-depth conversion, because once I load the file, and then check the ANDROID_KEY_PCMFORMAT_BITSPERSAMPLE info, it is 16. When I put bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_24; or SL_PCMSAMPLEFORMAT_FIXED_32, then OpenSL ES rejects it E/libOpenSLES(22706): pAudioSnk: bitsPerSample=32 W/libOpenSLES(22706): Leaving Engine::CreateAudioPlayer (SL_RESULT_CONTENT_UNSUPPORTED) Any idea how this is meant to work? Is Android currently restricted to 16 bit int only? I would also accept 32bit float, but I don't suppose that will work either.

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• Why doesn't this for loop work?

  - by evilsoup

  This is on Ubuntu 12.04 I'm trying to figure out how to get ffmpeg to do a batch conversion of FLACs to MP3, recursively. If I cd into a directory and use for f in *.flac; do ffmpeg -i "$f" -c:a libmp3lame -q:a 2 "${f/%flac/mp3}"; done that works perfectly fine. However, when I try this, it doesn't work: for f in "$(find . -type f -name *.flac)"; do ffmpeg -i "$f" -c:a libmp3lame -q:a 2 "${f/%flac/mp3}"; done It doesn't even throw up any useful errors (but here is the output anyway, no need to complain): evilsoup@enchantment:~/Music/Jean Sibelius$ for f in "$(find . -type f -name *.flac)"; do ffmpeg -i "$f" -c:a libmp3lame -q:a 2 "${f/%flac/mp3}"; done ffmpeg version git-2012-12-18-b7e085a Copyright (c) 2000-2012 the FFmpeg developers built on Dec 18 2012 19:23:11 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --enable-gpl --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3 libavutil 52. 12.100 / 52. 12.100 libavcodec 54. 80.100 / 54. 80.100 libavformat 54. 49.102 / 54. 49.102 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 28.100 / 3. 28.100 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/02. Symphony No.1.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/03. Symphony No.1.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/stripped2.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/05. Symphony No.1.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/stripped3.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/09. Andante festivo.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/08. Symphony No.3.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/01. Finlandia.flac ./Symphonies 1, 2, 3 & 5 (Oslo Philharmonic Orchestra Conducted by Mariss Jansons) Disc 1/07. Symphony No.3.flac ./Symphonies 1, 2, 3 & 5 I've tested the find command on its own, and it works as expected, so the problem has to be something to do with the interaction between find and for. I'm aware that I could do something with find's -exec option, but I can't find any way to do string substitution as I can with a bash for loop, and I'd rather not have a bunch of file.flac.mp3s to deal with, even if they could be fixed with a simple rename.

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• How can I make Banshee re-encode FLAC to Ogg Vorbis when copying to my player?

  - by Michael E

  I have most of my music in FLAC on my large storage device, and would like to automatically re-encode it in Ogg Vorbis when copying it to my portable audio player (Sansa Fuze v2). I have set my Fuze to MTP mode and told Banshee to encode to Ogg Vorbis with quality 4 in the Device Properties dialog for the Fuze (I would use MSC mode, but don't have an encoding option in the device properties when I do that). However, when I copy music to the device, either by dragging it from the music library or by syncing a playlist, the full FLAC files are copied rather than transcoded and written as Oggs. How can I get my Banshee setup re-encoding the audio? If StackExchange supported bonus points, I'd give bonus points for a solution that only re-encoded music that was already losslessly encoded, but I don't think that's possible.

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• VLC 2.0.3 on Lubuntu 12.04: No audio?

  - by drezabek

  I am on Lubuntu 12.04, and I have installed VLC media player version 2.0.3. When I try and play an audio file, it appears to load fine, and the media position bar displays the progress, and it says it is playing, but I can't here any thing through my speakers. I can hear game audio, web audio, and audio from SMPlayer just fine, but with VLC, I can't here anything. Below is the "Messages" output with the verbosity option set to "2 (debug)" main debug: processing request item: The Bottom, node: Playlist, skip: 0 main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: starting playback of the new playlist item main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: creating new input thread main debug: Creating an input for 'The Bottom' main debug: TIMER input launching for 'Floex - Machinarium Soundtrack - 01 The Bottom.flac' : 23.706 ms - Total 23.706 ms / 1 intvls (Avg 23.706 ms) main debug: using timeshift granularity of 50 MiB, in path '/tmp' main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' gives access `file' demux `' path `/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access_demux module: 3 candidates main debug: no access_demux module matching "file" could be loaded main debug: TIMER module_need() : 2.332 ms - Total 2.332 ms / 1 intvls (Avg 2.332 ms) main debug: creating access 'file' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac', path='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access module: 2 candidates filesystem debug: opening file `/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: using access module "filesystem" main debug: TIMER module_need() : 0.762 ms - Total 0.762 ms / 1 intvls (Avg 0.762 ms) main debug: Using stream method for AStream* main debug: starting pre-buffering main debug: received first data after 0 ms main debug: pre-buffering done 1024 bytes in 0s - 43478 KiB/s main debug: looking for stream_filter module: 7 candidates main debug: no stream_filter module matching "any" could be loaded main debug: TIMER module_need() : 0.236 ms - Total 0.236 ms / 1 intvls (Avg 0.236 ms) main debug: looking for stream_filter module: 1 candidate main debug: using stream_filter module "stream_filter_record" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for demux module: 54 candidates flacsys debug: Picture type=3 mime=image/png description='' file length=679371 qt4 debug: IM: Setting an input main debug: looking for packetizer module: 21 candidates main debug: using packetizer module "packetizer_flac" main debug: TIMER module_need() : 0.211 ms - Total 0.211 ms / 1 intvls (Avg 0.211 ms) main debug: using demux module "flacsys" main debug: TIMER module_need() : 4.023 ms - Total 4.023 ms / 1 intvls (Avg 4.023 ms) main debug: looking for a subtitle file in /home/doug/Music/unsorted/Floex - Machinarium Soundtrack/ main debug: looking for meta reader module: 2 candidates main debug: using meta reader module "taglib" main debug: TIMER module_need() : 5.245 ms - Total 5.245 ms / 1 intvls (Avg 5.245 ms) main debug: removing module "taglib" main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' successfully opened main debug: selecting program id=0 main debug: looking for decoder module: 30 candidates main debug: using decoder module "flac" main debug: TIMER module_need() : 0.442 ms - Total 0.442 ms / 1 intvls (Avg 0.442 ms) main debug: Buffering 0% flac debug: decode STREAMINFO flac debug: channels:2 samplerate:44100 bitspersamples:16 flac debug: STREAMINFO decoded main debug: Buffering 30% main debug: recycling audio output main debug: looking for audio output module: 3 candidates main debug: Buffering 61% pulse debug: using stereo channel map pulse debug: using library version 1.1.0 pulse debug: (compiled with version 1.1.0, protocol 26) main debug: Buffering 92% main debug: Stream buffering done (371 ms in 2 ms) pulse debug: connected locally to unix:/home/doug/.pulse/dce22254e867f905188a2ce200000003-runtime/native as client #14 pulse debug: using protocol 26, server protocol 26 pulse debug: using buffer metrics: maxlength=4194304, tlength=9880, prebuf=0, minreq=3528 pulse debug: connected to sink 0: alsa_output.pci-0000_00_14.2.analog-stereo main debug: using audio output module "pulse" main debug: TIMER module_need() : 4.571 ms - Total 4.571 ms / 1 intvls (Avg 4.571 ms) main debug: output 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: mixer 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes main debug: filter(s) 'f32l'->'s16l' 44100 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: f32l->s16l, bits per sample: 32->16 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.187 ms - Total 0.187 ms / 1 intvls (Avg 0.187 ms) main debug: conversion pipeline completed main debug: looking for audio mixer module: 2 candidates main debug: using audio mixer module "float32_mixer" main debug: TIMER module_need() : 0.125 ms - Total 0.125 ms / 1 intvls (Avg 0.125 ms) main debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: looking for audio filter module: 1 candidate scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32 scaletempo debug: params: 30 stride, 0.200 overlap, 14 search scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode main debug: using audio filter module "scaletempo" main debug: TIMER module_need() : 0.233 ms - Total 0.233 ms / 1 intvls (Avg 0.233 ms) main debug: filter(s) 's16l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo pulse debug: listing sink alsa_output.pci-0000_00_14.2.analog-stereo (0): Built-in Audio Analog Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: s16l->f32l, bits per sample: 16->32 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.147 ms - Total 0.147 ms / 1 intvls (Avg 0.147 ms) main debug: conversion pipeline completed pulse debug: base volume: 65536 main debug: looking for audio filter module: 1 candidate equalizer debug: equalizer loaded for 44100 Hz with 10 bands 2 pass equalizer debug: 60 Hz -> factor:0.000000 alpha:0.003013 beta:0.993973 gamma:1.993901 equalizer debug: 170 Hz -> factor:0.000000 alpha:0.008490 beta:0.983019 gamma:1.982437 equalizer debug: 310 Hz -> factor:0.000000 alpha:0.015374 beta:0.969252 gamma:1.967331 equalizer debug: 600 Hz -> factor:0.000000 alpha:0.029328 beta:0.941343 gamma:1.934254 equalizer debug: 1000 Hz -> factor:0.000000 alpha:0.047918 beta:0.904163 gamma:1.884869 equalizer debug: 3000 Hz -> factor:0.000000 alpha:0.130408 beta:0.739184 gamma:1.582718 equalizer debug: 6000 Hz -> factor:0.000000 alpha:0.226555 beta:0.546889 gamma:1.015267 equalizer debug: 12000 Hz -> factor:0.000000 alpha:0.344937 beta:0.310127 gamma:-0.181410 equalizer debug: 14000 Hz -> factor:0.000000 alpha:0.366438 beta:0.267123 gamma:-0.521151 equalizer debug: 16000 Hz -> factor:0.000000 alpha:0.379009 beta:0.241981 gamma:-0.808451 main debug: using audio filter module "equalizer" main debug: TIMER module_need() : 0.353 ms - Total 0.353 ms / 1 intvls (Avg 0.353 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: looking for visualization2 module: 1 candidate main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 3.278 ms - Total 3.278 ms / 1 intvls (Avg 3.278 ms) main debug: looking for video filter2 module: 18 candidates swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality) main debug: using video filter2 module "swscale" main debug: TIMER module_need() : 1.037 ms - Total 1.037 ms / 1 intvls (Avg 1.037 ms) main debug: looking for video filter2 module: 18 candidates yuvp debug: YUVP to YUVA converter main debug: using video filter2 module "yuvp" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: Deinterlacing available main debug: deinterlace 0, mode blend, is_needed 0 main debug: Opening vout display wrapper main debug: looking for vout display module: 6 candidates main debug: looking for vout window xid module: 4 candidates qt4 debug: requesting video... qt4 debug: Video was requested 0, 0 main debug: using vout window xid module "qt4" main debug: TIMER module_need() : 61.671 ms - Total 61.671 ms / 1 intvls (Avg 61.671 ms) main debug: looking for inhibit module: 2 candidates main debug: using inhibit module "xdg_screensaver" main debug: TIMER module_need() : 0.336 ms - Total 0.336 ms / 1 intvls (Avg 0.336 ms) xdg_screensaver debug: started xdg-screensaver (PID = 6682) xcb_xv debug: connected to X11.0 server xcb_xv debug: vendor : The X.Org Foundation xcb_xv debug: version: 11103000 xcb_xv debug: using screen 0x15a xcb_xv debug: using XVideo extension v2.2 xcb_xv debug: using adaptor NV17 Video Texture xcb_xv debug: using port 310 xcb_xv debug: using image format 0x30323449 xcb_xv debug: using X11 visual ID 0x21 (depth: 24) xcb_xv debug: using X11 window 0x03400000 xcb_xv debug: using X11 graphic context 0x03400002 main debug: VoutDisplayEvent 'fullscreen' 0 main debug: VoutDisplayEvent 'resize' 800x500 window main debug: using vout display module "xcb_xv" main debug: TIMER module_need() : 69.890 ms - Total 69.890 ms / 1 intvls (Avg 69.890 ms) main debug: original format sz 800x500, of (0,0), vsz 800x500, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0 main debug: removing module "freetype" main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 4.552 ms - Total 4.552 ms / 1 intvls (Avg 4.552 ms) main debug: using visualization2 module "visual" main debug: TIMER module_need() : 84.104 ms - Total 84.104 ms / 1 intvls (Avg 84.104 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 48510 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates main debug: using audio filter module "samplerate" main debug: TIMER module_need() : 0.375 ms - Total 0.375 ms / 1 intvls (Avg 0.375 ms) main debug: conversion pipeline completed main debug: End of audio preroll main debug: Decoder buffering done in 91 ms main warning: PTS is out of range (-9269), dropping buffer pulse debug: deferring start (190703 us) main debug: looking for video blending module: 1 candidate main debug: using video blending module "blend" main debug: TIMER module_need() : 0.275 ms - Total 0.275 ms / 1 intvls (Avg 0.275 ms) main debug: Detected interlaced video main debug: deinterlace 0, mode blend, is_needed 1 xcb_xv debug: display is visible pulse debug: starting deferred pulse warning: too late by 93760 us pulse debug: changed sample rate to 44186 Hz pulse debug: started pulse warning: too late by 94474 us pulse debug: changed sample rate to 44229 Hz pulse warning: too late by 93532 us pulse debug: changed sample rate to 44272 Hz pulse warning: too late by 92829 us pulse debug: changed sample rate to 44315 Hz pulse warning: too late by 92132 us pulse debug: changed sample rate to 44358 Hz xcb_xv debug: display is visible pulse warning: too late by 91534 us pulse debug: changed sample rate to 44401 Hz xcb_xv debug: display is visible pulse warning: too late by 89482 us pulse debug: changed sample rate to 44440 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible pulse warning: too late by 87529 us pulse debug: changed sample rate to 44479 Hz pulse warning: too late by 84577 us pulse debug: changed sample rate to 44504 Hz main debug: auto hiding mouse cursor pulse warning: too late by 78562 us pulse debug: changed sample rate to 44492 Hz pulse warning: too late by 68015 us pulse debug: changed sample rate to 44422 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor pulse debug: changed sample rate to 44336 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor I have had issues with VLC in the past- the audio quality was extremely crackly, as if the headphone jack was plugged in only half way, and the sounds were extremely sharp and caused my speakers to make a ringing/vibrating noise... It would eventually start working after I messed around with the audio settings, but it happened every restart. I eventually switched to SMPlayer, but now I need some of the features that VLC offers, but I still can't use VLC. At this point, the audio can not be heard at all, and the method I used before, messing around with the audio settings, isn't getting me anywhere. (note, I reposted this on VideoLan's forums, link is here: http://forum.videolan.org/viewtopic.php?f=13&t=104726) Please let me know if you need more information, or are confused by something I posted! Thanks!

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• Un groupe de développeurs sort Flac.js, un décodeur JavaScript pour la lecture du contenu audio dans le navigateur sans recours aux codecs

  Un groupe de développeurs sort Flac.js un décodeur audio en JavaScript pour la lecture du contenu audio dans le navigateur sans nécessiter de codecs HTML5, le futur standard du Web introduit la balise audio permettant de créer des applications fournissant le traitement et la synthèse audio dans le navigateur. Les navigateurs récents comme Chrome ou Firefox, intègrent déjà des bibliothèques Javascript qui fournissent des méthodes et propriétés permettant de manipuler l'élément audio. Cependant, les applications HTML 5 manipulant du contenu audio qui fonctionnent normalement dans un navigateur sur un système d'exploitation donné pourraient ne pas marcher correctement lors de...

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• How to compare mp3, flac audio data in a file, ignoring header data (ID3 tag) etc.?

  - by Rob

  I've backed up some audio files up in 2 places and added ID3 tags into one backup but not the other, since time has passed my own memory has faded on whether the backups are actually the same, but now one has ID3 data and the other doesn't, basic binary compare will fail and inspection will be cumbersome. Is there a tool to compare just the audio data (not the header, ID3) in mp3s, flac files, and other files using header data such as ID3. started a thread on beyond compare here: http://www.scootersoftware.com/vbulletin/showthread.php?t=7413 would consider other comparison software that does this task

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• Recursive wildcards in GNU make?

  - by Roger Lipscombe

  It's been a while since I've used make, so bear with me... I've got a directory, flac, containing .FLAC files. I've got a corresponding directory, mp3 containing MP3 files. If a FLAC file is newer than the corresponding MP3 file (or the corresponding MP3 file doesn't exist), then I want to run a bunch of commands to convert the FLAC file to an MP3 file, and copy the tags across. The kicker: I need to search the flac directory recursively, and create corresponding subdirectories in the mp3 directory. And I want to use make to drive this.

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• Recursive wildcards in Rake?

  - by Roger Lipscombe

  Follow up to this question about GNU make: I've got a directory, flac, containing .FLAC files. I've got a corresponding directory, mp3 containing MP3 files. If a FLAC file is newer than the corresponding MP3 file (or the corresponding MP3 file doesn't exist), then I want to run a bunch of commands to convert the FLAC file to an MP3 file, and copy the tags across. The kicker: I need to search the flac directory recursively, and create corresponding subdirectories in the mp3 directory. The directories and files can have spaces in the names, and are named in UTF-8. It turns out that this won't work in make, because of the spaces in the directories and filenames, so I'm wondering how to do it in rake instead...

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• Find directories that DON'T contain a file but YES another one

  - by muixca

  I have quite a large music collection and would like to find the directories in which I still have compressed files (*.rar) unprocessed. Hence looking for a command that lists directories in which i do NOT have *.flac or *.mp3 but YES *.rar present. Working off found examples in this post: Find directories that DON'T contain a file I tried: comm -3 \ <(find ~/Music/ -iname "*.rar" -not -iname "*.flac" -not -iname "*.mp3" -printf '%h\n' | sort -u) \ <(find ~/Music/ -maxdepth 5 -mindepth 2 -type d | sort) \ | sed 's/^.*Music\///' but don' work.

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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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• Using bash to copy file to spec folders

  - by Franko

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

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• Delete a directory with pipe (|) in its name?

  - by Dave Jarvis

  Without booting to Linux, how do you delete a directory that was created in Linux on an NTFS partition that contains a pipe in the file name? For example: f:\flac\foreign\Yoshida_Brothers\Best_of_Yoshida_Brothers_|_Tsugaru_Shamisen Tried and failed: Midnight Commander Recursively deleting the parent folder del /f /s /q Yoshida_Brothers del /f /s /q "\\?f:\flac\foreign\Yoshida_Brothers\" rmdir /s Yoshida_Brothers rmdir Best* FileASSASSIN Cannot delete folder Other ideas?

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• Delete a file with pipe (|) in its name?

  - by Dave Jarvis

  Without booting to Linux, how do you delete a directory that was created in Linux on an NTFS partition that contains a pipe in the file name? For example: f:\flac\foreign\Yoshida_Brothers\Best_of_Yoshida_Brothers_|_Tsugaru_Shamisen Tried and failed: Midnight Commander Recursively deleting the parent folder del /f /s /q Yoshida_Brothers del /f /s /q "\\?f:\flac\foreign\Yoshida_Brothers\ rmdir /s Yoshida_Brothers FileASSASSIN Other ideas?

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• How do I know if my system is capable of playing 24bit/96kHz sound?

  - by Igor Zinov'yev

  Let me state for the record that I'm a total noob when it comes to Hi-Fi sound systems, but I am rather picky about the sound quality. Normally I listen to CD recordings ripped to FLAC in 16/44, but I have several albums that are also ripped from vinyls to FLAC in 24/96. But it seems that I can't tell the difference between 16-bit and 24-bit versions (except for some vinyl noises, of course). That can be due to several reasons: my equipment (onboard audio, monitor headphones) isn't good enough to make any difference, my system is not playing audio in 24-bit 96 kHz, I am physically unable to hear the difference. So here is my question, how do I tell if my system can play 24-bit sound with 96 or 192 kHz resolution? And if it can, how do I tell that it plays it instead of downsampling to 16-bit / 44 kHz? Also, what hardware (audio cards, amplifiers, etc.) would you recommend to play such recordings on Ubuntu?

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• Why touching "d_name" makes calls to readdir() fail?

  - by Sarah Mani

  Hi, I'm trying to write a little helper for Windows which eventually will accept a file extension as an argument and return the number of files of that kind in the current directory. To do so, I'm reading the file entries in the directories and after getting the extension I'd like to convert it to lowercase to compare it with the yet-to-add specified argument. When converting the extension to lowercase I found that touching even a duplicate string of the d_name variable will cause a strange behaviour, like no more calls to readdir are called. Here is the code I'm using right now (the commented code is preliminary) and outputs for a given directory: #include <ctype.h> #include <dirent.h> #include <stdio.h> #include <string.h> char * strrch(char *string, size_t elements, char character) { char *reverse = string + elements; while (--reverse != string) if (*reverse == character) return reverse; return NULL; } void test(char *string) { // Even being a duplicate will make it fail: char *str = strdup(string); printf("Strings: %s %s\n", string, str); *str = 'a'; printf("Strings: %s %s\n", string, str); //unsigned short int i = 0; //for (; str[i] != '\0', str++; i++) // str[i] = tolower((unsigned char) str[i]); //puts(str); } int main(int argc, char **argv) { DIR *directory; struct dirent *element; if (directory = opendir(".")) { while (element = readdir(directory)) test(strrch(element->d_name, element->d_namlen, '.')); closedir(directory); puts(NULL); } else puts("Couldn't open the directory.\n"); } Output without modifying the duplicate (modification and the second printf call commented): Strings: (null) (null) Strings: . . Strings: .exe .exe Strings: .pdf .pdf Strings: .c .c Strings: .ini .ini Strings: .pdf .pdf Strings: .pdf .pdf Strings: .pdf .pdf Strings: .flac .flac Strings: .FLAC .FLAC Strings: .lnk .lnk Strings: .URL .URL Output of the same directory (with the code above, with the 2 printfs): Strings: (null) (null) Is there anything wrong? Is it a compiler issue? I'm using GCC 4.4.3 in Windows (MinGW) right now. Thank you very much for your help. By the way, is there any other way to work with files and directories in a Windows environment not using the POSIX functions?

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• How do i pipe stdout/stderr in .NET?

  - by acidzombie24

  I want to do something like this ffmpeg -i audio.mp3 -f flac - | oggenc2.exe - -o audio.ogg i know how to do ffmpeg -i audio.mp3 -f flac using the process class in .NET but how do i pipe it to oggenc2? Any example of how to do this (it doesnt need to be ffmpeg or oggenc2) would be fine.

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• About redirected stdout in System.Diagnostics.Process

  - by sforester

  I've been recently working on a program that convert flac files to mp3 in C# using flac.exe and lame.exe, here are the code that do the job: ProcessStartInfo piFlac = new ProcessStartInfo( "flac.exe" ); piFlac.CreateNoWindow = true; piFlac.UseShellExecute = false; piFlac.RedirectStandardOutput = true; piFlac.Arguments = string.Format( flacParam, SourceFile ); ProcessStartInfo piLame = new ProcessStartInfo( "lame.exe" ); piLame.CreateNoWindow = true; piLame.UseShellExecute = false; piLame.RedirectStandardInput = true; piLame.RedirectStandardOutput = true; piLame.Arguments = string.Format( lameParam, QualitySetting, ExtractTag( SourceFile ) ); Process flacp = null, lamep = null; byte[] buffer = BufferPool.RequestBuffer(); flacp = Process.Start( piFlac ); lamep = new Process(); lamep.StartInfo = piLame; lamep.OutputDataReceived += new DataReceivedEventHandler( this.ReadStdout ); lamep.Start(); lamep.BeginOutputReadLine(); int count = flacp.StandardOutput.BaseStream.Read( buffer, 0, buffer.Length ); while ( count != 0 ) { lamep.StandardInput.BaseStream.Write( buffer, 0, count ); count = flacp.StandardOutput.BaseStream.Read( buffer, 0, buffer.Length ); } Here I set the command line parameters to tell lame.exe to write its output to stdout, and make use of the Process.OutPutDataRecerved event to gather the output data, which is mostly binary data, but the DataReceivedEventArgs.Data is of type "string" and I have to convert it to byte[] before put it to cache, I think this is ugly and I tried this approach but the result is incorrect. Is there any way that I can read the raw redirected stdout stream, either synchronously or asynchronously, bypassing the OutputDataReceived event? PS: the reason why I don't use lame to write to disk directly is that I'm trying to convert several files in parallel, and direct writing to disk will cause severe fragmentation. Thanks a lot!

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• Cannot copy MP3 files from a CD

  - by MountainX

  I purchased a set of spoken word audio CD's that have MP3 and FLAC audio files; I think they also play as regular audio CD's because I see a CDA directory and .cda files. But I'm only interested in playing the MP3 files by copying them to my phone. Dolphin file manager shows all the files on the CD. However, it will not copy any of them to my hard drive, which is what my goal is. Dolphin shows no error, but the copy progress is zero. Amarok will play the files but not easily. I only tried the flac files. To play a file, I click the file in Dolphin, then I have to cancel a job using KDE's notification system, then Amarok proceeds to copy the file to a tmp directory which takes a long time, then it finally plays. kb3 will rip the audio, but I would prefer to copy the files directly from the CD. Since Dolphin would not copy the files, I thought I would try the terminal, but I can't get that to work either. mount -t auto -o ro /dev/sr0 /mnt/temp that gives the error: wrong fs type, bad option, bad superblock, etc. I get the same error using -t iso9660 and -t udf. so I started troubleshooting: ~$ wodim --devices wodim: Overview of accessible drives (1 found) : ------------------------------------------------------------------------- 0 dev='/dev/sg1' rwrw-- : 'MATSHITA' 'DVD-RAM UJ8A0AS' ------------------------------------------------------------------------- /dev/sg1 is not a block device sudo file -s /dev/sr0 ERROR: cannot read /dev/sr0 (input/output error) sudo file -s /dev/sg1 just hangs How can I copy these files to my computer hard disk?

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• Test your internet connection - Emtel Mobile Internet

  After yesterday's report on Emtel Fixed Broadband (I'm still wondering where the 'fixed' part is), I did the same tests on Emtel Mobile Internet. For this I'm using the Huawei E169G HSDPA USB stick, connected to the same machine. Actually, this is my fail-safe internet connection and the system automatically switches between them if a problem, let's say timeout, etc. has been detected on the main line. For better comparison I used exactly the same servers on Speedtest.net. The results Following are the results of Rose Hill (hosted by Emtel) and respectively Frankfurt, Germany (hosted by Vodafone DE): Speedtest.net result of 31.05.2013 between Flic en Flac and Rose Hill, Mauritius (Emtel - Mobile Internet) Speedtest.net result of 31.05.2013 between Flic en Flac and Frankfurt, Germany (Emtel - Mobile Internet) As you might easily see, there is a big difference in speed between national and international connections. More interestingly are the results related to the download and upload ratio. I'm not sure whether connections over Emtel Mobile Internet are asymmetric or symmetric like the Fixed Broadband. Might be interesting to find out. The first test result actually might give us a clue that the connection could be asymmetric with a ratio of 3:1 but again I'm not sure. I'll find out and post an update on this. It depends on network coverage Later today I was on tour with my tablet, a Samsung Galaxy Tab 10.1 (model GT-P7500) running on Android 4.0.4 (Ice Cream Sandwich), and did some more tests using the Speedtest.net app. The results are actually as expected and in areas with better network coverage you will get better results after all. At least, as long as you stay inside the national networks. For anything abroad, it doesn't really matter. But see for yourselves: Speedtest.net result of 31.05.2013 between Cascavelle and servers in Rose Hill, Mauritius (Emtel - Mobile Internet), Port Louis, Mauritius and Kuala Lumpur, Malaysia It's rather shocking and frustrating to see how the speed on international destinations goes down. And the full capability of the tablet's integrated modem (HSDPA: 21 Mbps; HSUPA: 5.76 Mbps) isn't used, too. I guess, this demands more tests in other areas of the island, like Ebene, Pailles or Port Louis. I'll keep you updated... The question remains: Alternatives? After the publication of the test results on Fixed Broadband I had some exchange with others on Facebook. Sadly, it seems that there are really no alternatives to what Emtel is offering at the moment. There are the various internet packages by Mauritius Telecom feat. Orange, like ADSL, MyT and Mobile Internet, and there is Bharat Telecom with their Bees offer which is currently limited to Ebene and parts of Quatre Bornes.

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• Mass audio encoder

  - by bessman

  I have a few thousand FLAC files which I would like to transcode to OGG Vorbis, but I can't find any suitable tools for the job. To name a few I have tried so far and why they are unsuitable: oggenc is single-threaded and would require me to automate it myself, mencoder requires the input to also contain video, and abcde assumes the input is a CD. The ideal tool should be multi-threaded, and support inputing multiple files located in different directories simultaneously. CLI or GUI makes no matter. Does such a tool exist?

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• How do you edit the "Preferred Format" settings in Rhythmbox?

  - by skyblue

  In Rhythmbox's Preferences, you can change the "Preferred Format" for Music to MPEG Layer 3 Audio, Ogg Vorbis, FLAC, or MPEG 4 Audio. However, despite there being a Settings button, it does not become enabled for any of these choices. (I have installed all of the gstreamer plugins, but this has made no difference.) So how can you change the "Preferred Format", for example to change the bit rate or the quality setting?

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• Merging cuesheet chapter halves into single track for an audiobook

  - by TheSavo

  I have an audiobook that I have ripped and I need some help constructing chapters. I have already made some cue sheets TITLE "Bookname" PERFORMER "the Author" FILE "File1.FLAC" wave ; 23971906.667 milliseconds TRACK 01 AUDIO TITLE "_Intro" INDEX 01 00:00:00 TRACK 02 AUDIO TITLE "CH 01" INDEX 01 24:15:50 TRACK 03 AUDIO TITLE "CH 02" INDEX 01 66:21:00 TRACK 04 AUDIO TITLE "CH 03" INDEX 01 87:05:00 The audio book is in two files. The chapter at the end of the first file is continued in the second file. However, the second file restates: The publisher Book Title List item Blah blah blah I would like to merge the two 'halves' of the chapter in one seamless track. The only way I can think to do this would be be: Bulk cut down the tracks. Drop the junk info into junk track Continue the track listings as normal Take the two "halves" of the target chapter and build a separate cue sheet for it. I know there has to be an easier way. I am ok with making the 'junk' info a 'gap' or something. These are are FLAC files that will be converted to MP3 for my phone and other potable devices. I have read the primers on cue sheets, but I am just not getting it.

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