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NAudio Get Audio Samples

I am using NAudio to get the sample from the song that is currently playing and draw the waveform as the song plays. I am using AudioFileReader and ToSampleProvider to get all samples as float and then I plot them into an InkCanvas while the song is playing. My problem is that the samples don't seem to match the sound. I also have verify this by using this same song in the WPF example that is located in the source code of NAudio. In the example the waveform matches the sound, but in my application it doesn't. So I was wondering if someone could help me find out what I am doing (or reading) wrong or if maybe my drawing logic is wrong.
Here is my current code:
public partial class MainWindow : Window, INotifyPropertyChanged
{
private ISampleProvider provider;
private DispatcherTimer timer;
private AudioFileReader reader;
private WaveOut waveOut;
private StylusPointCollection topPoints, bottomPoints;
private DrawingAttributes attr;
private double canvasHeight, canvasWidth;
private int samplesGroupSize;
private double drawPos = 0;
private StrokeCollection _WaveformLines;
public StrokeCollection WaveformLines
{
get { return _WaveformLines; }
set
{
_WaveformLines = value;
OnPropertyChanged("WaveformLines");
}
}
public MainWindow()
{
InitializeComponent();
this.DataContext = this;
}
private void Window_Loaded(object sender, RoutedEventArgs e)
{
reader = new AudioFileReader("C:\\Users\\Agustin\\Desktop\\DragonRider.mp3");
waveOut = new WaveOut();
waveOut.Init(reader);
provider = reader.ToSampleProvider(); //Here I get the samples
reader.Position = 0; //Go to the position 0 after reading the samples
canvasHeight = Waveform.ActualHeight;
canvasWidth = Waveform.ActualWidth;
WaveformLines = new StrokeCollection();
topPoints = new StylusPointCollection();
topPoints.Add(new StylusPoint(0, (canvasHeight / 2)));
topPoints.Changed += topPoints_Changed;
bottomPoints = new StylusPointCollection();
bottomPoints.Add(new StylusPoint(0, (canvasHeight / 2)));
bottomPoints.Changed += topPoints_Changed;
WaveformLines.Add(new Stroke(topPoints));
WaveformLines.Add(new Stroke(bottomPoints));
attr = new DrawingAttributes();
attr.Color = Colors.Green;
attr.Width = 1.5;
attr.Height = 1;
timer = new DispatcherTimer();
timer.Interval = TimeSpan.FromMilliseconds(1);
timer.Tick += timer_Tick;
timer.Start();
samplesGroupSize = (int)(timer.Interval.TotalSeconds * reader.WaveFormat.SampleRate); //The value for this is 44.
}
private void PlayButton_Click(object sender, RoutedEventArgs e)
{
waveOut.Play();
}
private void PauseButton_Click(object sender, RoutedEventArgs e)
{
waveOut.Pause();
}
private void timer_Tick(object sender, EventArgs e)
{
if (waveOut.PlaybackState == PlaybackState.Playing)
{
TimeLabel.Content = string.Format("Time: {0}", reader.CurrentTime.ToString(#"mm\:ss\:ff")); //NEED TO KEEP WORKING
float[] samps = new float[samplesGroupSize];
provider.Read(samps, 0, samps.Length);
float max = Max(samps);
float min = Min(samps);
topPoints.Add(new StylusPoint(drawPos, (canvasHeight / 2) - ((canvasHeight / 2) * max)));
bottomPoints.Add(new StylusPoint(drawPos, (canvasHeight / 2) - ((canvasHeight / 2) * min)));
drawPos += 2;
if (drawPos > canvasWidth)
{
WaveformLines.Clear();
topPoints = new StylusPointCollection();
topPoints.Add(new StylusPoint(0, (canvasHeight / 2)));
bottomPoints = new StylusPointCollection();
bottomPoints.Add(new StylusPoint(0, (canvasHeight / 2)));
WaveformLines.Add(new Stroke(topPoints));
WaveformLines.Add(new Stroke(bottomPoints));
drawPos = 0;
}
}
}
private float Min(float[] samps)
{
float max = samps[0];
foreach (float s in samps)
{
if (s > max)
max = s;
}
return max;
}
private float Max(float[] samps)
{
float min = samps[0];
foreach (float s in samps)
{
if (s < min)
min = s;
}
return min;
}
//I excluded the INotifyPropertyChanged implementation, but in the
//actual code is located here
}
I know this drawing algorithm is not very good but I have tried others and they also seem to not follow the audio.
Thanks.
NOTE: I know that there are similar question, but the other questions suggest to use things like AudioFileReader or ToSampleProvider which I am already using. My error is probably more into how I am reading the samples, maybe I am missing some bytes or have to skip some byte, or maybe some missing property that I am not setting.

You should seriously consider using the portions of the code from the WFP example that handles read/play and the min / max calculation work.
It will take you a little bit of effort but I promise it will be worth it.
You then will be working with a dataset that you know is in the correct form and you can focus on the drawing part. Look closely at AudioPlayback.cs and its relationship to SampleAggregator.cs.
You'll also find that getting automatic callbacks while the samples are being read (and played) is a much better way to refresh your waveform drawing than trying to use a DispatchTimer. It will also get you out of re-reading the wave buffer as well -- you really want to avoid that if you can.
EDIT:
I tested your conversion code and the resulting float values appear to be correct (in the range of -1 through 1). So I think the problem is with the way you are plotting the waveform in WPF.

C# Frequency Retrieval

What I need to do is calculate the frequency of the microphone input. I'm using IWaveProvider for this and its implemented Read(). The buffer always has a size of 8820 elements and something seems to be going wrong with the conversion from byte array to float array as well (the FloatBuffer property part).
Here are some of the important bits...
This is where I start my recording:
private void InitializeSoundRecording()
{
WaveIn waveIn = new WaveIn();
waveIn.DeviceNumber = 0;
waveIn.DataAvailable += (s, e) => this.waveIn_DataAvailable(s, e);
waveIn.RecordingStopped += (s, e) => this.waveIn_RecordingStopped(s, e);
waveIn.WaveFormat = new WaveFormat(44100, 1);
waveIn.StartRecording();
}
When the DataAvailable event handler is called, the following is executed:
private void waveIn_DataAvailable(object sender, WaveInEventArgs e)
{
WaveBuffer wb = new WaveBuffer(e.Buffer.Length);
IWaveProvider iWaveProvider = new PitchDetector(new WaveInProvider(sender as WaveIn), new WaveBuffer(e.Buffer));
iWaveProvider.Read(wb, 0, e.Buffer.Length);
PitchDetector pd = iWaveProvider as PitchDetector;
this.ShowPitch(pd.Pitch);
}
And lastly, this is the "actual" important bit:
private const int FLOAT_BUFFER_SIZE = 8820;
private IWaveProvider source;
private WaveBuffer waveBuffer;
private int sampleRate;
private float[] fftBuffer;
private float[] prevBuffer;
public float Pitch { get; private set; }
public WaveFormat WaveFormat { get { return this.source.WaveFormat; } }
internal PitchDetector(IWaveProvider waveProvider, WaveBuffer waveBuffer = null)
{
this.source = waveProvider;
this.sampleRate = waveProvider.WaveFormat.SampleRate;
this.waveBuffer = waveBuffer;
}
/// <summary>
/// UNSAFE METHOD!
/// </summary>
/// <param name="input"></param>
/// <returns></returns>
private unsafe float[] ByteArrayToFloatArray(byte[] input)
{
float[] fb = new float[FLOAT_BUFFER_SIZE];
unsafe
{
fixed (byte* ptrBuffer = input)
{
float* ptrFloatBuffer = (float*)ptrBuffer;
for (int i = 0; i < FLOAT_BUFFER_SIZE; i++)
{
fb[i] = *ptrFloatBuffer;
ptrFloatBuffer++;
}
}
}
return fb;
}
public int Read(byte[] buffer, int offset = 0, int count = 0)
{
if (this.waveBuffer == null || this.waveBuffer.MaxSize < count)
this.waveBuffer = new WaveBuffer(count);
int readBytes = this.source.Read(this.waveBuffer, 0, count);
if (readBytes > 0) readBytes = count;
int frames = readBytes / sizeof(float);
this.Pitch = this.DeterminePitch(this.waveBuffer.FloatBuffer, frames);
return frames * 4;
}
Strangely enough, when it enters the constructor, waveBuffer contains some data (255, 1, 0, etc.), but when I check the "buffer" parameter of Read(), it's entirely 0. Every element.
Out of curiosity also, why does Read() have a buffer parameter, but isn't actually used in the method at all (I got that piece of code from one of your articles)?
Any help to resolve this issue would be greatly appreciated! I've been at this for quite a while already, but can make no sense out of it.
Thanks,
Alain

It is not clear what article you are referring to and I am not familiar with this library. However, the Read method is clearly reading in your 'time-series'/or other data. From this, the buffer parameter you speak of is likely to be the padding length that you want to place on either end of your data set.
This padding is known as 'Zero Padding' and it pads your recorded signal with zeros (places n zeros on either end of the signal, where n is set according to the radix used). This allows one to use a longer FFT, which will produce a longer FFT resulting vector.
A longer FFT result has more frequency bins that are more closely spaced in frequency. But they will be essentially providing the same result as a high quality Sinc interpolation of a shorter non-zero-padded FFT of the original data.
This might result in a smoother looking spectrum when plotted without further interpolation.
For further in formation see
https://dsp.stackexchange.com/questions/741/why-should-i-zero-pad-a-signal-before-taking-the-fourier-transform
I hope this helps.

This is not really an answer to your question but I wrote a safe generic alternative to your array conversion function.
using System;
using System.Runtime.InteropServices;
public static class Extensions
{
public staitc TDestination[] Transform<TSource, TDestination>(
this TSource[] source)
where TSource : struct
where TDestination : struct
{
if (source.Length == 0)
{
return new TDestination[0];
}
var sourceSize = Marshal.SizeOf(typeof(TSource));
var destinationSize = Marshal.SizeOf(typeof(TDestination));
var byteLength = source.Length * sourceSize;
int remainder;
var destinationLength = Math.DivRem(
byteLength,
destinationSize,
out remainder);
if (remainder > 0)
{
destinationLength++;
}
var destination = new TDestination[destinationLength];
Buffer.BlockCopy(source, 0, destination, 0, byteLength);
return destination;
}
}
Which obviously, you could use like
var bytes = new byte[] { 1, 1, 2, 3, 5, 8, 13, 21 };
var floats = bytes.Transform<byte, float>();

Recording with AudioQueue and Monotouch static sound

I have written a small program in MonoTouch to record sound from the mic of my iPhone 4s using an InputAudioQueue.
I save the recorded data in an array and feed this buffer to the my audio player for playback (using OutputAudioQueue).
When playing back it's just some stuttering garbage / static sound. I have tried filling the buffer with sin waves before playback and then it sounds good, so I guess the problem is in the recording, not the playback. Can anyone help me see what is wrong? (Code below)
public class AQRecorder
{
private const int CountAudioBuffers = 3;
private const int AudioBufferLength = 22050;
private const int SampleRate = 44100;
private const int BitsPerChannel = 16;
private const int Channels = 1;
private const int MaxRecordingTime = 5;
private AudioStreamBasicDescription audioStreamDescription;
private InputAudioQueue inputQueue;
private short[] rawData;
private int indexNextRawData;
public AQRecorder ()
{
this.audioStreamDescription.Format = AudioFormatType.LinearPCM;
this.audioStreamDescription.FormatFlags = AudioFormatFlags.LinearPCMIsSignedInteger |
AudioFormatFlags.LinearPCMIsPacked;
this.audioStreamDescription.SampleRate = AQRecorder.SampleRate;
this.audioStreamDescription.BitsPerChannel = AQRecorder.BitsPerChannel;
this.audioStreamDescription.ChannelsPerFrame = AQRecorder.Channels;
this.audioStreamDescription.BytesPerFrame = (AQRecorder.BitsPerChannel / 8) * AQRecorder.Channels;
this.audioStreamDescription.FramesPerPacket = 1;
this.audioStreamDescription.BytesPerPacket = audioStreamDescription.BytesPerFrame * audioStreamDescription.FramesPerPacket;
this.audioStreamDescription.Reserved = 0;
}
public void Start ()
{
int totalBytesToRecord = this.audioStreamDescription.BytesPerFrame * AQRecorder.SampleRate * AQRecorder.MaxRecordingTime;
this.rawData = new short[totalBytesToRecord / sizeof(short)];
this.indexNextRawData = 0;
this.inputQueue = SetupInputQueue (this.audioStreamDescription);
this.inputQueue.Start ();
}
public void Stop ()
{
if (this.inputQueue.IsRunning)
{
this.inputQueue.Stop (true);
}
}
public short[] GetData ()
{
return this.rawData;;
}
private InputAudioQueue SetupInputQueue (AudioStreamBasicDescription audioStreamDescription)
{
InputAudioQueue inputQueue = new InputAudioQueue (audioStreamDescription);
for (int count = 0; count < AQRecorder.CountAudioBuffers; count++)
{
IntPtr bufferPointer;
inputQueue.AllocateBuffer(AQRecorder.AudioBufferLength, out bufferPointer);
inputQueue.EnqueueBuffer(bufferPointer, AQRecorder.AudioBufferLength, null);
}
inputQueue.InputCompleted += HandleInputCompleted;
return inputQueue;
}
private void HandleInputCompleted (object sender, InputCompletedEventArgs e)
{
unsafe
{
short* shortPtr = (short*)e.IntPtrBuffer;
for (int count = 0; count < AQRecorder.AudioBufferLength; count += sizeof(short))
{
if (indexNextRawData >= this.rawData.Length)
{
this.inputQueue.Stop (true);
return;
}
this.rawData [indexNextRawData] = *shortPtr;
indexNextRawData++;
shortPtr++;
}
}
this.inputQueue.EnqueueBuffer(e.IntPtrBuffer, AQRecorder.AudioBufferLength, null);
}
}

ok, this might be too late, but I had the same problem with hearing garbage sound only and found the solution.
You cannot read the audio data directly from e.IntPtrBuffer. This pointer is a pointer to a AudioQueueBuffer object and not to the audio data itself. So to read the audio data you can make use of the e.UnsafeBuffer which gives you the access to this object and use its AudioData pointer. This is a IntPtr which you can cast (in unsafe context) to a byte* or short* and you have your audio data.
Best regards
Alex

Playing sine wave for unknown time

Whole day I was looking for some tutorial or piece of code, "just" to play simple sin wave for "infinity" time. I know it sounds a little crazy.
But I want to be able to change frequency of tone in time, for instance - increase it.
Imagine that I want to play tone A, and increase it to C in "+5" frequency steps each 3ms (it's really just example), don't want to have free places, stop the tone.
Is it possible? Or can you help me?

Use NAudio library for audio output.
Make notes wave provider:
class NotesWaveProvider : WaveProvider32
{
public NotesWaveProvider(Queue<Note> notes)
{
this.Notes = notes;
}
public readonly Queue<Note> Notes;
int sample = 0;
Note NextNote()
{
for (; ; )
{
if (Notes.Count == 0)
return null;
var note = Notes.Peek();
if (sample < note.Duration.TotalSeconds * WaveFormat.SampleRate)
return note;
Notes.Dequeue();
sample = 0;
}
}
public override int Read(float[] buffer, int offset, int sampleCount)
{
int sampleRate = WaveFormat.SampleRate;
for (int n = 0; n < sampleCount; n++)
{
var note = NextNote();
if (note == null)
buffer[n + offset] = 0;
else
buffer[n + offset] = (float)(note.Amplitude * Math.Sin((2 * Math.PI * sample * note.Frequency) / sampleRate));
sample++;
}
return sampleCount;
}
}
class Note
{
public float Frequency;
public float Amplitude = 1.0f;
public TimeSpan Duration = TimeSpan.FromMilliseconds(50);
}
start play:
WaveOut waveOut;
this.Notes = new Queue<Note>(new[] { new Note { Frequency = 1000 }, new Note { Frequency = 1100 } });
var waveProvider = new NotesWaveProvider(Notes);
waveProvider.SetWaveFormat(16000, 1); // 16kHz mono
waveOut = new WaveOut();
waveOut.Init(waveProvider);
waveOut.Play();
add new notes:
void Timer_Tick(...)
{
if (Notes.Count < 10)
Notes.Add(new Note{Frecuency = 900});
}
ps this code is idea only. for real using add mt-locking etc

use NAudio and SineWaveProvider32: http://mark-dot-net.blogspot.com/2009/10/playback-of-sine-wave-in-naudio.html
private WaveOut waveOut;
private void button1_Click(object sender, EventArgs e)
{
StartStopSineWave();
}
private void StartStopSineWave()
{
if (waveOut == null)
{
var sineWaveProvider = new SineWaveProvider32();
sineWaveProvider.SetWaveFormat(16000, 1); // 16kHz mono
sineWaveProvider.Frequency = 1000;
sineWaveProvider.Amplitude = 0.25f;
waveOut = new WaveOut();
waveOut.Init(sineWaveProvider);
waveOut.Play();
}
else
{
waveOut.Stop();
waveOut.Dispose();
waveOut = null;
}
}

Playing WAVE file in C# using DirectX and threading?

at the moment im trying to figure out how i can manage to play a wave file in C# by filling up the secondary buffer with data from the wave file through threading and then play the wave file.
Any help or sample coding i can use?
thanks
sample code being used:
public delegate void PullAudio(short[] buffer, int length);
public class SoundPlayer : IDisposable
{
private Device soundDevice;
private SecondaryBuffer soundBuffer;
private int samplesPerUpdate;
private AutoResetEvent[] fillEvent = new AutoResetEvent[2];
private Thread thread;
private PullAudio pullAudio;
private short channels;
private bool halted;
private bool running;
public SoundPlayer(Control owner, PullAudio pullAudio, short channels)
{
this.channels = channels;
this.pullAudio = pullAudio;
this.soundDevice = new Device();
this.soundDevice.SetCooperativeLevel(owner, CooperativeLevel.Priority);
// Set up our wave format to 44,100Hz, with 16 bit resolution
WaveFormat wf = new WaveFormat();
wf.FormatTag = WaveFormatTag.Pcm;
wf.SamplesPerSecond = 44100;
wf.BitsPerSample = 16;
wf.Channels = channels;
wf.BlockAlign = (short)(wf.Channels * wf.BitsPerSample / 8);
wf.AverageBytesPerSecond = wf.SamplesPerSecond * wf.BlockAlign;
this.samplesPerUpdate = 512;
// Create a buffer with 2 seconds of sample data
BufferDescription bufferDesc = new BufferDescription(wf);
bufferDesc.BufferBytes = this.samplesPerUpdate * wf.BlockAlign * 2;
bufferDesc.ControlPositionNotify = true;
bufferDesc.GlobalFocus = true;
this.soundBuffer = new SecondaryBuffer(bufferDesc, this.soundDevice);
Notify notify = new Notify(this.soundBuffer);
fillEvent[0] = new AutoResetEvent(false);
fillEvent[1] = new AutoResetEvent(false);
// Set up two notification events, one at halfway, and one at the end of the buffer
BufferPositionNotify[] posNotify = new BufferPositionNotify[2];
posNotify[0] = new BufferPositionNotify();
posNotify[0].Offset = bufferDesc.BufferBytes / 2 - 1;
posNotify[0].EventNotifyHandle = fillEvent[0].Handle;
posNotify[1] = new BufferPositionNotify();
posNotify[1].Offset = bufferDesc.BufferBytes - 1;
posNotify[1].EventNotifyHandle = fillEvent[1].Handle;
notify.SetNotificationPositions(posNotify);
this.thread = new Thread(new ThreadStart(SoundPlayback));
this.thread.Priority = ThreadPriority.Highest;
this.Pause();
this.running = true;
this.thread.Start();
}
public void Pause()
{
if (this.halted) return;
this.halted = true;
Monitor.Enter(this.thread);
}
public void Resume()
{
if (!this.halted) return;
this.halted = false;
Monitor.Pulse(this.thread);
Monitor.Exit(this.thread);
}
private void SoundPlayback()
{
lock (this.thread)
{
if (!this.running) return;
// Set up the initial sound buffer to be the full length
int bufferLength = this.samplesPerUpdate * 2 * this.channels;
short[] soundData = new short[bufferLength];
// Prime it with the first x seconds of data
this.pullAudio(soundData, soundData.Length);
this.soundBuffer.Write(0, soundData, LockFlag.None);
// Start it playing
this.soundBuffer.Play(0, BufferPlayFlags.Looping);
int lastWritten = 0;
while (this.running)
{
if (this.halted)
{
Monitor.Pulse(this.thread);
Monitor.Wait(this.thread);
}
// Wait on one of the notification events
WaitHandle.WaitAny(this.fillEvent, 3, true);
// Get the current play position (divide by two because we are using 16 bit samples)
int tmp = this.soundBuffer.PlayPosition / 2;
// Generate new sounds from lastWritten to tmp in the sound buffer
if (tmp == lastWritten)
{
continue;
}
else
{
soundData = new short[(tmp - lastWritten + bufferLength) % bufferLength];
}
this.pullAudio(soundData, soundData.Length);
// Write in the generated data
soundBuffer.Write(lastWritten * 2, soundData, LockFlag.None);
// Save the position we were at
lastWritten = tmp;
}
}
}
public void Dispose()
{
this.running = false;
this.Resume();
if (this.soundBuffer != null)
{
this.soundBuffer.Dispose();
}
if (this.soundDevice != null)
{
this.soundDevice.Dispose();
}
}
}
}
The concept is the same that im using but i can't manage to get a set on wave byte [] data to play

I have not done this.
But the first place i would look is XNA.
I know that the c# managed directx project was ditched in favor of XNA and i have found it to be good for graphics - i prefer using it to directx.

what is the reason that you decided not to just use soundplayer, as per this msdn entry below?
private SoundPlayer Player = new SoundPlayer();
private void loadSoundAsync()
{
// Note: You may need to change the location specified based on
// the location of the sound to be played.
this.Player.SoundLocation = http://www.tailspintoys.com/sounds/stop.wav";
this.Player.LoadAsync();
}
private void Player_LoadCompleted (
object sender,
System.ComponentModel.AsyncCompletedEventArgs e)
{
if (this.Player.IsLoadCompleted)
{
this.Player.PlaySync();
}
}
usually i just load them all up in a thread, or asynch delegate, then play or playsynch them when needed.

You can use the DirectSound support in SlimDX: http://slimdx.org/ :-)

You can use nBASS or better FMOD both are great audio libraries and can work nicely together with .NET.

DirectSound is where you want to go. It's a piece of cake to use, but I'm not sure what formats it can play besides .wav
http://msdn.microsoft.com/en-us/library/windows/desktop/ee416960(v=vs.85).aspx