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  • How to Join N live MP3 streams into one using FFMPEG?

    - by Ole Jak
    How to Join N live MP3 streams (radio streams like such live KCDX mp3 stream http://mp3.kcdx.com:8000/stream ) into 1 using FFMPEG? (I have N incoming live mp3 streams I want to join them and stream out 1 live mp3 stream) I mean I want to mix sounds like thay N speakers speak at the same time (btw N stereo to 1 mono), please help. BTW: My problem is mainly how to make FFMPEG read from stream not from file... Would you mind giving some code examples, please.

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  • How to Join N live MP3 streams into 1 using FFMPEG?

    - by Ole Jak
    How to Join N live MP3 streams (radio streams like such live KCDX mp3 stream http://mp3.kcdx.com:8000/stream ) into 1 using FFMPEG? (I have N incoming live mp3 streams I vant to join them and stream out 1 live mp3 stream) I mean I wanna to mix sounds like thay N speakers speak at the same time (btw N stereo to 1 mono), please help BTW: My problem is mainly how to make FFMPEG read from stream not from file... Would you mind giving some code examples, please...

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  • Sample code from apple iPhoneMultichannelMixerTest can't run mp3 file

    - by Usman
    Hi i want to control channel volume from my app and iPhoneMultichannelMixerTest (http://developer.apple.com/iphone/library/samplecode/iPhoneMultichannelMixerTest/)sample code form apple looks perfect for me. but i m heaving problem in playing mp3 files. code crashes in MultichannelMixerController.m at static OSStatus renderInput(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { SoundBufferPtr sndbuf = (SoundBufferPtr)inRefCon; UInt32 bufSamples = sndbuf[inBusNumber].numFrames; AudioUnitSampleType *in = sndbuf[inBusNumber].data; AudioUnitSampleType *outA = (AudioUnitSampleType *)ioData->mBuffers[0].mData; AudioUnitSampleType *outB = (AudioUnitSampleType *)ioData->mBuffers[1].mData; UInt32 sample = sndbuf[inBusNumber].sampleNum; for (UInt32 i = 0; i < inNumberFrames; ++i) { if (1 == inBusNumber) { outA[i] = 0; outB[i] = in[sample++]; //crash on this line. } else { outA[i] = in[sample++]; outB[i] = 0; } if (sample >= bufSamples) sample = 0; } sndbuf[inBusNumber].sampleNum = sample; // printf("bus %d sample %d\n", inBusNumber, sample); return noErr; } can any one tell me why this is happening. or is there any limitation to play mp3 on iPhone??? Thankz in Adv. Usman

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  • Trouble converting an MP3 file to a WAV file using Naudio

    - by WebDevHobo
    Naudio Library: http://naudio.codeplex.com/ I'm trying to convert an MP3 file to a WAV file, but I've run in to a small error. I know what's going wrong, but I don't really know how to go about fixing it. Here's the piece of code I'm running: private void button1_Click(object sender, EventArgs e) { using(Mp3FileReader reader = new Mp3FileReader(@"path\to\MP3")) { using(WaveFileWriter writer = new WaveFileWriter(@"C:\test.wav", new WaveFormat())) { int counter = 0; while(reader.Read(test, counter, test.Length + counter) != 0) { writer.WriteData(test, counter, test.Length + counter); counter += 512; } } } } reader.Read() goes into the Mp3FileReader class, and the method looks like this: public override int Read(byte[] sampleBuffer, int offset, int numBytes) { if (numBytes % waveFormat.BlockAlign != 0) //throw new ApplicationException("Must read complete blocks"); numBytes -= (numBytes % waveFormat.BlockAlign); return mp3Stream.Read(sampleBuffer, offset, numBytes); } mp3Stream is an object of the Stream class. The problem is: I'm getting an ArgumentException. MSDN says that this is because the sum of offset and numBytes is greater than the length of sampleBuffer. Documentation: http://msdn.microsoft.com/en-us/library/system.io.stream.read.aspx This happens because I increase the counter every time, but the size of the byte array test remains the same. What I've been wondering is: do I need to increase the size of the array dynamically, or do I need to find out the needed size at the beginning and set it right away? And also, instead of 512, the method in Mp3FileReader returns 365 the first time. Which is the size of a whole block. But I'm writing the full 512. I'm basically just using the read to check if I'm not at the end of the file yet. Do I need to catch the return value and do something with that, or am I good here?

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  • How to play an .mp3 using QBuffer with Phonon in a PyQt Qt environment

    - by thedax
    With the code I have right now I CAN play .mp3 data from files succesfully. However I need to play the same data using a QtCore.QBuffer (NOT from a file). When I use the example of the docs it errors an unexpected type of QBuffer! However...... that is what it SHOULD see, according to the docs. But............... it throws: TypeError: Phonon.MediaObject.setCurrentSource(Phonon.MediaSource): argument 1 has unexpected type 'QBuffer' The code I use is (1): someBuffer = QtCore.QBuffer() someBuffer.writeData(TrackData) mediaObject.setCurrentSource(someBuffer) I also tried (2): someBuffer = QtCore.QBuffer() mediaObject.setCurrentSource(someBuffer) someBuffer.writeData(TrackData) and (3): someBuffer = QtCore.QBuffer() someBuffer.writeData(TrackData) mediaObject.setCurrentSource(Phonon.MediaSource(someBuffer)) The last example (3) throws an different ERROR and wipes my Gui off screen ;-) ASSERT: "d-connected" in file /builddir/build/BUILD/phonon-4.5.1/phonon/streaminterface.cpp, line xxxx Notes: TrackData contains the mp3 data and IS PLAYING OK when I write it to a File and use that as a resource to mediaObject.setCurrentSource(Phonon.MediaSource())I also experimented with a QByteArray but that leads to the same "unexpected QBuffer" error. To be more precise everything I feed setCurrentSource is not accepted. Tried a string (errors an unexpected type of String), tried a QBuffer (errors an unexpected type of QBuffer), tried a QByteArray (errors an unexpected type of QByteArray). BTW: I run Qt, PyQt on Linux. Any ideas??

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  • How to play a MP3 file using NAudio

    - by Rookian
    WaveStream waveStream = new Mp3FileReader(mp3FileToPlay); var waveOut = new WaveOut(); waveOut.Init(waveStream); waveOut.Play(); This throws an exception: WaveBadFormat calling waveOutOpen How can I play a mp3 file with NAudio?

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  • MP3 Encoding in Java

    - by Mohit Nanda
    I need an OpenSource API in Java, which can encode *.wav and *.au formats to MP3, and vice-versa. Have evaluated Java Sound API, and LameOnJ, but they dont meet the requirements and aint stable, respectively. Please suggest one that is free, and platform independent.

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  • Reading ID3 tags of a remote mp3 file ?

    - by kps
    http://stackoverflow.com/questions/1477835/read-mp3-tags-with-silverlight got me started with reading id3 tags, but i realize that taglib# online deals with local file paths ? Is there a way of reading this info from a remote file ?

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  • Modify MP3 File

    - by arik-so
    Hello, I have an MP3 file uploader. I want to add an additional audio track to the file upon upload - via PHP. Is that possible? Thanks in advance!

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  • Merging MP3 files in Linux Debian using PHP

    - by pako
    What's the easiest way to merge the contents of several MP3 files into one using PHP 5.2 on Linux Debian system? I found some scripts that are supposed to do in PHP only, but they seem to be buggy. Perhaps there is a way to accomplish this task using command line programs, that I could install on my Linux Debian machine?

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  • Fetch MP3 ID3 tag under linux

    - by exic
    Hi, I have a few mp3 files which are not tagged. Winamp has a nice feature which I think is called "autotag" and which is very good at finding out artist and title for files. I'd like something like this for unix, so that I could possibly get artists and titles for my untagged files. Do you know some program which does this? Thanks.

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  • Using FFmpeg or wrapper to get mp3 from mp4 in C#

    - by Tom Allen
    I'm trying to extract an mp3 from a flash compatible mp4 file and have so far found FFMpeg and a bunch of different wrappers that all claim to be able to do the job. Ideally, I'd like to not have to rely on shelling to the FFMpeg exe, but none of the wrappers I've tried seem to work.... Has anyone got any code or advice for how to go about this? Thanks!

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  • Javascript Mp3 Player, NONE FLASH

    - by Yo Momma
    I currently have a simple flash Mp3 player on my site which works lovely. The issue is, most cell phones do not support flash and a large portion of my visitors come through via a mobile device. The alternative to this is to replace the flash player with a javascript player. I found a JQuery one but it only works on HTML5 compatible browsers. Does anyone know of a good alternative that would work on mobile devices as well as most desktop web browsers?

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  • Where to get pure C++ Lame MP3 encoder - PCM to MP3 example?

    - by Ole Jak
    So all I need is a simple function that sets it up (eating incoming PCM RATE (for example: rate near to 44100) It's channels (for example: 2) and -bits (for example: 16) and desirable 128 kb\s rate) and another one that takes PCM data and encodes it into pure MP3 frames. I know it looks like a silly homework task but I assure you - it is not. I hope it will be of help to all C++ developers starting with MP3s. So can anybody please help me with that?

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  • How I can get PUID from mp3-file?

    - by volume
    I want to streamline my library on FreeBsd using PUID from MusicBrainz. I need a command line utility for this, but I have not been able to find. It would be nice to give input mp3-file and receive PUID as result.

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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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