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  • Convert MP3 to AAC,FLAC to AAC (.NET/C#) FREE :)

    - by PearlFactory
    So I was tasked with looking at converting 10 million tracks from mp3 320k to AAC and also Converting from mp3 320k to mp3 128k After a bit of hunting around the tool you need to use is FFMPEG Download x64 WindowsAlso for the best results get the Nero AACEncoder Download Now the command line STEP 1(From Flac)ffmpeg -i input.flac -f wav - | neroAacEnc -ignorelength -q 0.5 -if - -of output.m4aor (From mp3)ffmpeg -i input.mp3 -f wav - | neroAacEnc -ignorelength -q 0.5 -if - -of output.m4aNow the output.m4a is a intermediate state that we now put a ACC wrapper on via FFMpeg STEP 2ffmpeg -i output.m4a -vn -acodec copy final.aacDone :) There are a couple of options with the FFMPEG library as in we can look at importing the librarys and manipulation the API for the direct result FFMPEG has this support. You can get the relevant librarys from HereThey even have the source if you are that keen :-)In this case I am going to wrap the command lines into c# external process threads.( For the app that i am building to convert the 10 million tracks there is a complex multithreaded app to support this novel code )//Arrange Metadata about Call Process myProcess = new Process();ProcessStartInfo p = new ProcessStartInfo();string sArgs = string.format(" -i {0} -f wav - | neroAacEnc -ignorelength -q 0.5 -if - -of {1}",inputfile,outputfil) ; p.FileName = "ffmpeg.exe" ; p.CreateNoWindow = true; p.RedirectStandardOutput = true; //p.WindowStyle = ProcessWindowStyle.Normal p.UseShellExecute = false;//Execute p.Arguments = sArgs; myProcess.StartInfo = p; myProcess.Start(); myProcess.WaitForExit();//Write details about call  myProcess.StandardOutput.ReadToEnd();Now in this case we would execute a 2nd call using the same code but with different sArgs to put the AAC wrapper on the m4a file. Thats it. So if you need to do some conversions of any kind for you ASP.net sites/apps this is a great start and super fast.. With conversion times of around 2-3 seconds all of this can be done on the fly:-)Justin Oehlmannref : StackOverflow.com

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  • Getting iTunes to play third party AAC files

    - by Redmastif
    I have a library filled with some old MP3 files and I'm in the process of changing them all to AAC for the better sound quality. Obviously I can't just create AAC versions of the files I already have because they would sound worse (lossy compression to converted to more lossy compression), so I'm going to their source and downloading them in a lossless form and using a third party to make them into AAC. Apparently iTunes will not handle AAC files that aren't made with iTunes. Is there a way around this? I've looked at third party programs and would be willing to use them, but since they all require the iTunes/iPod/iEverything driver, I don't know if they would still prevent my files or not. Also before you jump on my back about pirating, these files are from old CDs that I lost years ago. I paid for them. Thanks.

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  • Decoding AAC using MediaCodec API on Android

    - by Benjamin Reed
    I'm trying to used the MediaCodec API on Android to decode an AAC stream. (It's raw AAC.) I tried using the MediaFormat.createAudioFormat() to create the format object to pass to MediaCodec.configure(), but I kept getting errors when using AAC (audio/mp4a-latm). (It works with MP3 (audio/mpeg) though...) Finally I created a MediaExtractor for an AAC file and looked at the format object it was producing. I saw that it included the key "csd-0" for a ByteBuffer composed of two bytes both with the value 0x12. If I include that key and value in the format object that I used to configure the AAC codec, everything works. Does anyone have an idea what is going on? The documentation states that I shouldn't configure that key. Does anyone have a pointer to MediaCodec examples to decode AAC files without using MediaExtractor to generate the format object?

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  • iTunes: Burn an MP3 CD when I only have AAC files?

    - by Phillip Oldham
    Is there a way to burn an MP3 CD when iTunes only contains (unprotected) AAC files purchased from the iTunes store? There isn't an option directly in iTunes to convert AAC - MP3 like there is for MP3 - AAC, and having to convert the AAC files to MP3 and use an extrnal tool to burn the disk would be time-consuming. Are there any plugins for iTunes that can accomplish this? Or programs which can read an iTunes playlist, convert the tracks to MP3 "on the fly" while burning the CD?

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  • Flash/Flex: play embedded AAC audio?

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

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  • Ruby: play, pause, resume aac (audio) files

    - by rahul
    I need to play, pause and resume AAC (audio) files from a ruby console program (much like iTunes or any music player). After much searching, I've come across these libraries: mp3info metadata id3lib-ruby rvideo (uses ffmpeg) These seem to help me in getting track length and tags which i also need, but I need something to play AAC (at least) and if possible other formats. I also must be able to pause and resume (so shelling a program like mpg321 is out).

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  • AAC.js : le décodeur audio JavaScript open source supporte le profile Low Complexity

    AAC.js : le dernier décodeur audio JavaScript de Official.fm Labs qui supporte le profile Low Complexity [IMG]http://media.tumblr.com/tumblr_m6wpozHbxB1qbis4g.png[/IMG] L'équipe de Official.fm Labs vient de sortir un codec audio qui pourrait d'ailleurs être le prochain codec le plus utilisé après le MP3, voire le surpasser. AAC.js est entièrement codé en JavaScript avec le framework Aurora.js qui facilite l'écriture de codecs. AAC, qui signifie Advanced Audio Codec, est l'un des codecs les plus courants et des noms comm...

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  • Is there now any way to convert mp3 files to m4a or aac 192kbit?

    - by piedro
    Since about two years now I am trying to find a way to convert high quality mp3 files to m4a or aac files with a fixed bitrate 192k. Please don't suggest using another format - i thought this through as far as it goes. The problem here is: ffmpeg obvioulsy can't convert to a higher bitrate than 152k. Even when it says it does so the resulting files still have 152k instead of 192k. ffmpeg also has/had a bug not writing the bitrate into the audio file tags which means when testing you have to calculate the bitrate manually by dividing the filesize by the length of the audio in seconds (resulting in 152k - see above) choosing faac as converter gets me the same results other programs don't work reliably (see this thread Howto convert audio files to *.m4a? I know that this is not an original new problem but I am wondering if there is still no way to convert with ubuntu/kubuntu 12.04 after a lot time passed and I can't find some of the bug issues mentioned in the other thread anymore. So: Is there a solution after all?

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  • How to develop an AAC converter for Linux?

    - by Arnab
    Hi, I am a beginner developer want to develop a MP3 to AAC converter for Linux. But I don't know how to start as I havn't done any development in Linux before. I am using Ubuntu 10.04. Can anyone tell me how to start and from where? What libraries do I need? Thanks

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  • Encode audio to aac with libavcodec

    - by ryan
    I'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac (libfaac support enabled). I do this by calling avcodec_encode_audio repeatedly with codec_context-frame_size samples each time. The first four calls return successfully, but the fifth call never returns. When I use gdb to break, the stack is corrupt. If I use audacity to export the pcm data to a .wav file, then I can use command-line ffmpeg to convert to aac without any issues, so I'm sure it's something I'm doing wrong. I've written a small test program that duplicates my problem. It reads the test data from a file, which is available here: http://birdie.protoven.com/audio.pcm (~2 seconds of signed 16 bit LE pcm) I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I'm also encoding video, and writing both to an mp4. ffmpeg version info: FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 3 2010 15:40:46 with gcc 4.4.1 configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 Is there something I'm not setting, or setting incorrectly in my codec context, maybe? Any help is greatly appreciated! Here is my test code: #include <stdio.h> #include <libavcodec/avcodec.h> void EncodeTest(int sampleRate, int channels, int audioBitrate, uint8_t *audioData, size_t audioSize) { AVCodecContext *audioCodec; AVCodec *codec; uint8_t *buf; int bufSize, frameBytes; avcodec_register_all(); //Set up audio encoder codec = avcodec_find_encoder(CODEC_ID_AAC); if (codec == NULL) return; audioCodec = avcodec_alloc_context(); audioCodec->bit_rate = audioBitrate; audioCodec->sample_fmt = SAMPLE_FMT_S16; audioCodec->sample_rate = sampleRate; audioCodec->channels = channels; audioCodec->profile = FF_PROFILE_AAC_MAIN; audioCodec->time_base = (AVRational){1, sampleRate}; audioCodec->codec_type = CODEC_TYPE_AUDIO; if (avcodec_open(audioCodec, codec) < 0) return; bufSize = FF_MIN_BUFFER_SIZE * 10; buf = (uint8_t *)malloc(bufSize); if (buf == NULL) return; frameBytes = audioCodec->frame_size * audioCodec->channels * 2; while (audioSize >= frameBytes) { int packetSize; packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData); printf("encoder returned %d bytes of data\n", packetSize); audioData += frameBytes; audioSize -= frameBytes; } } int main() { FILE *stream = fopen("audio.pcm", "rb"); size_t size; uint8_t *buf; if (stream == NULL) { printf("Unable to open file\n"); return 1; } fseek(stream, 0, SEEK_END); size = ftell(stream); fseek(stream, 0, SEEK_SET); buf = (uint8_t *)malloc(size); fread(buf, sizeof(uint8_t), size, stream); fclose(stream); EncodeTest(32000, 2, 448000, buf, size); }

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  • Question About MP4(AAC) Binary Tree

    - by DeanMc
    I'm in the very early stages of working on a tag editor for mp4 files and more specifically iTunes AAC ones. After doing some snooping around it seems that the file's structure is not as complicated as I first thought and is built in a sort of tree like the following 4 Bytes [Atom Length] 4 Bytes [Atom Name] X Bytes [Atom Data] An atom's data is as large as the length and can contain either Data(information) or another atom. What I am trying to work out is how one determines if the data is information or an actual atom. Any insight would be much appreciated.

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  • Can FLV AAC stream be played in Android

    - by HariKJ
    Hi, I'm trying to build a radio player and the client is providing a stream which is a FLV container with the audio being AAC When I read the headers it shows up as audio/aacp. I have tried all possible ways such as using the 1) Streaming through mediaplayer (Does not work) 2) Use the NPR mode of using a proxy stream (I get a broken pipe exception) 3) Play it in chunks ( Plays but I need the SDCard and the playback is not very great) 4) Use the GPL'd FAAD2 Library but I would have to pay the royalty fee Can some one help me out on figuring this issue out. The last option that I have is to have my client change the stream to mp3 container (which I know that it works) Regards, Hari

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  • How do I programmatically convert mp3 to an itunes-playable aac/m4a file?

    - by kwork
    I've been looking for a way to convert an mp3 to aac programmatically or via the command line with no luck. Ideally, I'd have a snippet of code that I could call from my rails app that converts an mp3 to an aac. I installed ffmpeg and libfaac and was able to create an aac file with the following command: ffmpeg -i test.mp3 -acodec libfaac -ab 163840 dest.aac When i change the output file's name to dest.m4a, it doesn't play in iTunes. Thanks!

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  • Should I convert my AAC M4A files to MP3?

    - by j0rd4n
    Due to Apple, I have a large majority of my music files in the AAC M4A format. They do NOT have DRM so I don't have to worry about that. I'm getting tired of Apple products and really want to switch to a different brand player (and something more compatible with Linux). It appears most MP3 players support...well...MP3 and not AAC. Should I convert my library to be free of Apple and open to other players? Is this a lossless conversion? Can it be lossless? If I will lose quality, I'm not interested. Am I even doing the right thing? AAC is the better format, but I'm not seeing a lot of support for it yet. I'll be honest and say that I need some education in this department. Any helpful advice is most welcome.

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  • How to convert a mpeg4-aac video to mpeg-ts using MEncoder?

    - by Mahendra Liya
    Hello, I know this can be done using FFMpeg and I have done it. The only problem here was that FFMpeg actually tries to "encode" the file while converting to mpeg-ts which I don't want. i.e. I just want to change the container format to mpeg-ts without encoding the media. Is this type of conversion possible with FFMpeg? (I know about "copy" option, but it works with H264-aac and not with mpeg4-aac). Is it possible to change mpeg4-aac to mpeg-ts container format with MEncoder? I wish to know the personal opinion/advice of those who have already worked on such stuff. Thanks in advance.

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  • AAC Sample Rate and Bit Rate for High Quality Audio?

    - by marco.ragogna
    What are the AAC Sample Rate and Bit Rate settings to set in order to encode an audio track with a quality comparable to MP3 320kbps? I need to backup a DVD movie, the default settings for AAC are Bitrate (KB/s) 128 Sample Rate (HZ) 44100 should I set Bitrate (KB/s) 320 Sample Rate (HZ) 48000 or the default are already good?

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  • Can i use aac in an commercial app for free?

    - by Jason123
    I was wondering if i can use the aac codec in my commercial app for free (through lgpl ffmpeg). It says on the wiki: No licenses or payments are required to be able to stream or distribute content in AAC format.[36] This reason alone makes AAC a much more attractive format to distribute content than MP3, particularly for streaming content (such as Internet radio). However, a patent license is required for all manufacturers or developers of AAC codecs. For this reason free and open source software implementations such as FFmpeg and FAAC may be distributed in source form only, in order to avoid patent infringement. (See below under Products that support AAC, Software.) But the xSplit program had to cancel the AAC for free members because they have to pay royalties per person. Is this true (that you have to pay per each person that uses aac)? If you do have to pay, which company do you pay to and how does one apply?

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  • AAC 256kbit to MP3 320kbit conversion. I know it's lossy, but how?

    - by Fabian Zeindl
    Has anyone ever transcoded music from a high-quality aac to an mp3 (or vice-versa). The internet is full of people who say this should never be done, but apart from the theoretical standpoint that you can only lose information, does it matter in practise? is the difference perceivable, except on studio-equipment? does the re-encoding actually lose much information? If, p.e., high frequences are chopped away by the initial compression, those frequencies aren't there anymore, so this part of the compression-algorithm won't touch the data during the second compression. Am i wrong?

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  • Invalid ADTS sampling_frequency_index and channel_configuration why?

    - by Moto
    Hello all, I hope someone can direct me on the right path before I put a lot of time and effort on this. I'm currently trying to parse an AAC+ frame to get information such as number of channels and sample frequency. So it seems that we can simply get this information from the ADTS header but most of the time this information is inaccurate. So the question is: -Why is this data inaccurate? What is the meaning of the ADTS header channel and sample freq? Should I rely on it? -Should I parse further down the frame to get this information? FYI, the AAC+ raw data is coming from streaming servers... Thanks for the help! -Moto

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  • How can I transfer metadata from several flac files to aac (m4a) files?

    - by abckookooman
    Suppose I have two folders, dir1 and dir2, with deveral files in each of them, and all the files in dir1 are named like "ExampleFileName.flac" and all the files in dir2 are named as "ExampleFileName.m4a" - basically their names are the same except the extension. What I need to do is transfer all of the metadata for each of the files somehow - even though their codecs are different. It would be great if I can do this via command line, but anything is appreciated. Thank you.

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