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New Post: Moving source code on GitHub

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I'm doing that already. I've forked the NAudio source code and it's ok. Nonetheless i think that Github, which is more than just a git server control, it offers more of a social coding network and brings more contributors to interesting projects, like this one. It offers a greater visualisation of changes occuring on all the differents forks with links to your commits hierarchy. The same for the code browsing ... The web interface of github is way faster / better then CodePlex.

But in the end, I'm not saying that CodePlex is not good enough but I'm just proposing you to take a look (whenever you have some free time) on GitHub. You might like it.

Here's a url to one of my projects source code browser : https://github.com/lvaleriu/bltoolkit

, all forks graph : https://github.com/igor-tkachev/bltoolkit/network

Valeriu


New Post: Noise in playback start

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Wow! Great, this works! :)

Do you need some more information to track why simple approach doesn't work?

And what does this mean that we need to create CustomMp3FrameDecompressor -- does this has any drawbacks?

New Post: Noise in playback start

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Well the fix I made before passed a flag into the codec that is supposed to tell the codec that we have repositioned and it should flush its internal state. However, it seems that your codec was ignoring that flag, so this solution closes the codec and reopens it when you reposition. Ideally I'd like to find a fix inside AcmMp3FrameDecompressor itself as this might not perform too well if you are repositioning during playback.

Mark

New Post: Moving source code on GitHub

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Yes, github is really nice and I have an account there and have contibuted to some projects.

There are a few reasons why I have no plans at the moment to move there though:

- There is a long history of discussions and issues for NAudio on Codeplex that would not easily move across. Also existing forks would need to be moved.

- I prefer Mercurial to Git and github forces all projects to use Git.

- My place of work blocks the ports git uses, making it very hard for me to do anything with git at work.

My hope is that codeplex keeps evolving to be competitive with github. There have been some steps in the right direction recently.

New Post: Noise in playback start

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I tried this workaround on big MP3 file (187 MB, 4 hours) -- no noticable delay -- so this works well for me.

Write if you'll need any more information or testing from my side, I will be glad to help you fix this problem completely, if you'll decide to do this.

You do a great job -- NAudio is very easy to use and it calculates duration of MP3 correctly (something that Windows Media Player control does wrong). Thanks!

New Post: Noise in playback start

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thanks SergeyPo,

The way I'm trying to do this with the AcmMp3FrameConverter is calling into acmStreamConvert with the ACM_STREAMCONVERTF_START flag set. As far as I can see, that is how you are supposed to do a reposition, and I'm surprised the codec on your machine doesn't seem to be responding to it correctly.

Mark

ACM_STREAMCONVERTF_START

New Post: Not Sure How To Use The SimpleCompressorStream

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It's because the SimpleCompressorStream never stops returning audio. It is a very old class, and I have plans to obsolete it and replace it with something better.

Your best fix for now is simply not to use the CreateWaveFile method. Instead, write a loop that reads from the SimpleCompressorStream (e.g. 16kb at a time) and writes to a WaveFileWriter. You can check where you are up to in your source file and stop writing when you go beyond the end.

Source code checked in, #eb267fc71acb

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WaveFileWriter.CreateFile protects against filling up the disk when you pass in a never-ending source provider

Source code checked in, #dd1b4de48c46

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creating a NuGet package for WMA

Commented Issue: Generator [16367]

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Here is an audio generator to test the installation (Pink noise, white noise, Sweep, Sin, Square, Triangle, Sawtooth) with PhaseReverse.
Frequency Fixed or variable.
Waveforms and frequencies were correct with external software (SmaartLive7) and an oscilloscope.


There is nothing unusual in the calculation of waveforms.

Integration into NAudio a IWaveProvider or ISampleProvider.

Files Attachment: Class WaveGenerator.cs and complement for demo.

I suspect that this class will not be used much but it just still be useful
Comments: Bonjour, Mark, please change the "Type Issue" to Feature I have not respect the hierarchy. Sorry Merci

Commented Issue: AutoCue [16366]

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I created a class on the basis of AudioFileReader to integrate an Autocue option (as with CD Player) to readjust the beginning of the song without white or 'click' the beginning of song.


Enclosed Class AudioFileReaderAutoCue.cs and a small demo based on your PlugIn (AutoCue.cs, AutoCue.Designer.cs, AutoCuePlugin.cs)
Files are compiled in VS 2012 Express Desktop
Comments: Bonjour, Mark, please change the "Type Issue" to Feature I have not respect the hierarchy. Sorry Merci

Edited Issue: wrong speed for the different rate [16368]

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when i records audio with the sample WaveIn class, it's get wrong speed, if i shoose rate different from 8000. Only in 8 kHz i've got correct speed - on the 16000 audio got two times longer and lower tone, at 44000 it almost nothing to hear, sound terrify. Please check, what i've done wrong.

public static void StartRecording(string path, int device)
{
waveInStream = new WaveIn();

writer = new WaveFileWriter(path, waveInStream.WaveFormat);

waveInStream.DeviceNumber = device;

waveInStream.WaveFormat = new WaveFormat(8000, 32, 1);
waveInStream.BufferMilliseconds = 200;
waveInStream.NumberOfBuffers = 4;

waveInStream.DataAvailable += waveInStream_DataAvailable;
waveInStream.StartRecording();
}

static void waveInStream_DataAvailable(object sender, WaveInEventArgs e)
{
writer.Write(e.Buffer, 0, e.BytesRecorded);
}


Commented Issue: wrong speed for the different rate [16368]

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when i records audio with the sample WaveIn class, it's get wrong speed, if i shoose rate different from 8000. Only in 8 kHz i've got correct speed - on the 16000 audio got two times longer and lower tone, at 44000 it almost nothing to hear, sound terrify. Please check, what i've done wrong.

public static void StartRecording(string path, int device)
{
waveInStream = new WaveIn();

writer = new WaveFileWriter(path, waveInStream.WaveFormat);

waveInStream.DeviceNumber = device;

waveInStream.WaveFormat = new WaveFormat(8000, 32, 1);
waveInStream.BufferMilliseconds = 200;
waveInStream.NumberOfBuffers = 4;

waveInStream.DataAvailable += waveInStream_DataAvailable;
waveInStream.StartRecording();
}

static void waveInStream_DataAvailable(object sender, WaveInEventArgs e)
{
writer.Write(e.Buffer, 0, e.BytesRecorded);
}


Comments: oh. what an akward moment=) Thanks!

New Post: Play Pause Play

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Bonjour,

Is it possible to empty the buffer before a play, to avoid hearing before the samples after a Play / Pause / Change Position / Play.


I wonder if there is a solution to the problem

Merci, Manu

New Post: Play Pause Play

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Stop clears the buffers, Pause doesn't, so call Stop instead of Pause if you need to reposition.


New Post: Play Pause Play

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yes of course

except that liberates the Stop Audio driver and depending on the audio cards play some more or less long.

New Post: Play Pause Play

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If you use an AES or MADI connection, the slave may take 1 to 1.5 seconds before being synchronized to a first Play

Source code checked in, #8d0fd69b7212

Source code checked in, #272f3fb3426c

Updated Release: NAudio 1.6 (Oct 26, 2012)

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