Playback of Sine Wave in NAudio

Great post Mark.

do you happen to know if this would be possible for mpeg2 audio?

no to display a cine wave of mpeg2 audio file (actually a mplayer audiodump of a mpeg2-ts file)

Hi Mark,

Thanks for the run down, it's really helpful.

My problem is that I need to be able to create complex noises comprised of many waves.

Any idea how I could achieve this?

Thanks,

Tom

Hello Mark,
I'm really excited about this source, but I haven't used a third party library in a very long time and am not sure how to set your library up in Visual Stuido, without just altering the provided solution. Do you think you could show me how to do this?

Thanks
Malik

Hello Mark,
That was great, thanks! I've added a slider to the form in an attempt to update the frequency in realtime, realizing that the sine wave's settings are set when the startStop button is toggled on.
Is there a way to update things in realtime?
------------added-------------
private int Frequency;
private void trackBar1_Scroll(object sender, EventArgs e)
{
Frequency = trackBar1.Value;
}
------------------------------

VERY COOL! i just made the SineWaveProvider object viewable to whole class and updated it in the slider's scroll method. It works great, but is there a way to update it faster? seems to change values kind of choppy? would that be the fact that i'm no using Soundcard settings made for low latency response?

Thanks!!
Malik

I have the example working in a form. Clicking the button starts the tone, and clicking it again stops it. However, when I try to get it to run from Main() itself, the sound only lasts a fraction of a second, then stops. If I put a breakpoint in the Read() routine, I see it only gets called 3 times.

Here is my code in Main:

Worker worker = new Worker();
worker.StartStopSineWave();
System.Threading.Thread.Sleep(5000);
worker.StartStopSineWave();
worker = null;

And here is the class:

public class Worker
{
WaveOut waveOut;

public void StartStopSineWave()
{

if (waveOut == null)
{
SineWaveProvider32 sineWaveProvider = new SineWaveProvider32();
sineWaveProvider.SetWaveFormat(16000, 1); // 16kHz mono
sineWaveProvider.Frequency = 1000;
sineWaveProvider.Amplitude = 0.10f;
waveOut = new WaveOut();
waveOut.Init(sineWaveProvider);
waveOut.DesiredLatency = 1000;
waveOut.Play();
}
else
{
waveOut.Stop();
waveOut.Dispose();
waveOut = null;
}
}

}

I suspect this problem has something to do with threading. Any ideas on how to solve it?

Thanks Mark for the reply.

Actually I need it in a batch/console process. The override I am doing (in place of the sine generator) will continually get input from an external device, process it, and pass the resulting audio to the sound card for the user to hear. I guess I could create a dummy form but would prefer this to work in the background instead.

That works. Thanks!

Hi Mark!
I have implemented the sine wave generator. I would like it to play for a certain time so I added a timer. During that time I would like all other activites to stop, how do I manage that?
What I am after is:
Call StartSine
"While playing do nothing"
Call next action

Coach, can't you just set some type of flag when the timer starts, clear it when the timer is done, and then check for the flag in the other activities and don't proceed (say sleep and then check again) as long as its set?

Frank

Hi Mark!

Fantastic API - I've been struggling with the old Managed DirectX and this is much better.

I created the SineWaveProvider example you have. I'm using Windows 7 (64-bit) and Visual Studio C# Express. When I try to run the app, I get a "BadImageFormatException" error trying to load NAudio. The exception message is:

Could not load file or assembly 'NAudio, Version=1.3.8.0, Culture=neutral, PublicKeyToken=null' or one of its dependencies. An attempt was made to load a program with an incorrect format.

Is this a Windows 7 thing? Or a 64-bit thing? Or something else? Just wondering - I'll give it a go on my XP machine and see how that works out...

-Scott

I can't get this to work, i'm not familiar with C#. How do you structure this code, in separate classes? Is there an example I can look at?

Thanks

Richard

Interestingly enough, when I built the project in SharpDevelop 3.2, with the build options set to "Build for 32-bit Intel", it worked fine on my 64-bit system. Go figure. :)

Hi Mark
I try to use the naudio lib in c# with VS2005 .
Is there a way to select which channel (Left or right) with the Wavein Function.
I don't information about that.
Thanks

What would be the best way to do this in stereo? I'm currently using this for a soft-synth and have been struggling trying to get this to work with the mixer; i can't create a channel like this

WaveChannel32 = new WaveChannel32( new MyWaveProvider32) because MyWaveProvider32 cannot be cast to WaveStream and I can't figure out why this is the case...

Thanks for responding! I am so thankful for the time you've spent on this project. Kudos!!

Hello Mark,
I was just wondering, How fast could i use this example if I separate the start and stop code and put them in KeyDown and KeyUp methods?
If I press the keys too fast, the debugger says that waveOut.Stop(); is referencing an object that doesn't exist anymore. "Object reference not set to an instance of an object."

My code is below, do you have any Ideas why this might happen?

Thanks for your library :)

private void Button_KeyDown(object sender, KeyEventArgs e)
{
...
startSine(1000, amp);
...
}

private void startSine(int freq, float amp)
{
if (waveOut == null)
{
sineWaveProvider = new SineWaveProvider32();
waveOut = new WaveOut();
sineWaveProvider.SetWaveFormat(16000, 1);
sineWaveProvider.Frequency = freq;
sineWaveProvider.Amplitude = amp;
waveOut.Init(sineWaveProvider);
waveOut.Play();
}

#endregion

#region Key Up Event Handler
private void Button_KeyUp(object sender, KeyEventArgs e)
{
...
stopSine();
...
}
private void stopSine()
{
waveOut.Stop();
waveOut.Dispose();
waveOut = null;
}
#endregion

Oh, it seems that my problem is multiple key presses. I'll have to write code to fix this. Please delete my last post, if you see it fit to do so.
Thanks!!

Thats really wonderful lib but I want to know is there any option available to know the frequency and amplitude of the laoded wave file ultimately to modify it???

Hi Guys,How i can read the stereo channel using NAudio(Left Speaker and Right Speaker) values of a video while it's playing.Please reply for this.I need to use this for a project.

Hi Mark,

I was wondering, will the above code still work with the latest beta version of NAudio that is on codeplex, or have you included some of the functionality that you suggested you would in the update?

Hey Mark-

I have been playing with the NAudio library for a couple days. I got the sine wave to play. Very cool. But now I want to play back my own arbitrary float data. What's the best way to stuff in float data to a WaveOut object? My guess is to use BufferedWaveProvider. I'm a little uncertain how to get that class to handle float data though.

I tried instantiating BufferedWaveProvider a couple of ways. My first shot was using WaveForm(fs,1). I also tried WaveFormat.CreateIeeeFloatWaveFormat(fs,1).

But then I was shaky on how to pack my float data into a byte array to be used by the AddSamples() method of BufferedWaveProvider.

What do you recommend?

HI Mark

I was wondering maybe if there was a way to save the data directly to a .wav file ?

thanks a lot Mark
great work

Hi there Mark,

I've converted this into a derived WaveStream to allow mixing using WaveMixerStream.

I've added:

public override long Length
{
get { return long.MaxValue; }
}

However, I'm not sure how to get/set position.

Could you help me out?

Thanks.

Hello,

How could I use this to create other waveform types such as a triangle or sawtooth?

Also is there any way to avoid a click when changing frequencies during playback?

If I want to Stop the Play automatically, after finishing the buffer, what can I do?

It seems that the read method will be executed by offset=0 each time!

Hey Mark,

I'm a student and have the assignment to create a recording application which then records microphone input and has to determine the frequency and then display which note it is related to said frequency.

Will this article be the good way to follow in order to achieve this (retrieval of frequency)?

Thanks!

Thanks for the Post. I have noticed that it work wrong with Stereo....
To fix it i add a line in the SineWaveProviderClass to proof if byte is even Number.

for (int n = 0; n < sampleCount; n++)
{
buffer[n + offset] = (float)(Amplitude * Math.Sin((2 * Math.PI * sample * Frequency) / sampleRate));
if (this.WaveFormat.Channels == 1 || (n + offset) % 2 == 0)
sample++;
if (sample >= sampleRate) sample = 0;
}

I'm sorry for my bad english and hope it ist helpflull.

Tahnk you for this great Post. I have noticed that it is working wrong in stereo-mode. To fix it i have added a line to proof if floatarray is even number:

for (int n = 0; n < sampleCount; n++)
{
buffer[n + offset] = (float)(Amplitude * Math.Sin((2 * Math.PI * sample * Frequency) / sampleRate));
if (this.WaveFormat.Channels == 1 || (n + offset) % 2 == 0)
sample++;
if (sample >= sampleRate) sample = 0;
}

I'm sorry for my bad English and hope my Coment ist helpfully.

Hi!

Thanks a lot for your work on NAudio, it's much appreciated!

The last few days I have been working with additive synthesis based on this example. At first I used WaveOut (as in your code), but then I started moving my code to a Windows Store App. I know that NAudio isn't officially available for such apps, but I found some post stating how to use NuGet to get a pre-release. I changed WaveOut to WasapiOutRT like this:

_sampleMixer = new MixingSampleProvider(_voices[0].SampleProviders);
_sampleToWaveProvider = new SampleToWaveProvider(_sampleMixer);
_waveOut = new WasapiOutRT(AudioClientShareMode.Shared, 100);
await _waveOut.Init(_sampleToWaveProvider);
_waveOut.Play();

It works, but there's a horrible latency. I'm not sure, but I don't think it lagged as much when using WaveOut. I am a total beginner in programming audio like this, but when I stepped through the code I realized that the Read() method is called just once every second when using WasapiOutRT, with a buffer size of 44100, as opposed to every ~150ms or so on WaveOut (buffer size 6615), and to me that sounds like a source of latency. Or do I miss something?

Best regards,
Håkan Eriksson

Hi!

Thanks for your swift reply!

I stepped through the NAudio code. The reported latency from the AudioClient is 11 ms.

However, when stepping through the code I noticed that in MediaFoundationTransform you explicitly read one second from the provider:

private IMFSample ReadFromSource()
{
// we always read a full second
int bytesRead = sourceProvider.Read(sourceBuffer, 0, sourceBuffer.Length);

Maybe there's something I don't understand well enough, but by reading once a second, doesn't that imply that changes I make in the audio generation can be delayed up to one second until they are heard?

Thanks a lot for your help, and a happy New Year as well! :-)

Hi!

Thanks for your swift reply!

I stepped through the NAudio code. The reported latency from the AudioClient is 11 ms.

However, when stepping through the code I noticed that in MediaFoundationTransform you explicitly read one second from the provider:

private IMFSample ReadFromSource()
{
// we always read a full second
int bytesRead = sourceProvider.Read(sourceBuffer, 0, sourceBuffer.Length);

Maybe there's something I don't understand well enough, but by reading once a second, doesn't that imply that changes I make in the audio generation can be delayed up to one second until they are heard?

Thanks a lot for your help, and I wish you a happy New Year! :-)

I haven't given much thought to where the sampling rate comes from, so I might be resampling without intending to. You've seen my initialization code (in my first post), and the code producing the sine wave is pretty much as your example. If I understand your code correctly, that would imply that the sampling frequency is set by the default constructor of WaveProvider32, i.e. 44100 Hz in mono?

And I see that it is possible to use other constructors to set other sample rates, but how would I know what sample rate the soundcard has?

Actually, by stepping through the NAudio initialization, I learned that my sound card's sample rate indeed is different from the one I got from the default constructor. When setting the sample rate correctly there's no lag (an no glitches, which also occurred before).

Is there a way to get the sound card's sample rate from NAudio?