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Null reference when trying to play audio using NAudio on ASIO output

I have 6 audio sources which I need to play on 6 separate channels using ASIO.
I managed to get this working using WaveOutEvent as the output, but when I switch to AsioOut I get a null reference error, and I can't figure out what I'm doing wrong.
I need to use ASIO since I require 6 output channels and because I have to broadcast the audio over the network using Dante protocol.
The output device is a Dante Virtual Soundcard.
The error is:
NullReferenceException: Object reference not set to an instance of an object
NAudio.Wave.AsioOut.driver_BufferUpdate (System.IntPtr[] inputChannels, System.IntPtr[] outputChannels) (at <7b1c1a8badc0497bac142a81b5ef5bcf>:0)
NAudio.Wave.Asio.AsioDriverExt.BufferSwitchCallBack (System.Int32 doubleBufferIndex, System.Boolean directProcess) (at <7b1c1a8badc0497bac142a81b5ef5bcf>:0)
UnityEngine.<>c:<RegisterUECatcher>b__0_0(Object, UnhandledExceptionEventArgs)
This is the (simplified) code that plays the audio. The buffers are filled by other methods in external classes.
using NAudio.Wave;
using System;
using System.Collections.Generic;
public class AudioMultiplexer
{
MultiplexingWaveProvider multiplexer;
AsioOut asioOut;
public List<BufferedWaveProvider> buffers;
public int outputChannels = 6;
public int waveFormatSampleRate = 48000;
public int waveFormatBitDepth = 24;
public int waveFormatChannels = 2;
public void Start()
{
buffers = new List<BufferedWaveProvider>();
var outputFormat = new WaveFormat(waveFormatSampleRate, waveFormatBitDepth, waveFormatChannels);
for (int i = 0; i < outputChannels; i++)
{
var buffer = new BufferedWaveProvider(outputFormat);
buffer.DiscardOnBufferOverflow = true;
// Make sure the buffers are big enough, just in case
buffer.BufferDuration = TimeSpan.FromMinutes(5);
buffers.Add(buffer);
}
multiplexer = new MultiplexingWaveProvider(buffers, outputChannels);
for (int i = 0; i < outputChannels; i++)
{
// Each input has 2 channels, left & right, take only one channel from each input source
multiplexer.ConnectInputToOutput(i * 2, i);
}
var driverName = GetDanteDriverName();
if (string.IsNullOrEmpty(driverName))
{
return;
}
asioOut = new AsioOut(driverName);
asioOut.AudioAvailable += AsioOut_AudioAvailable;
asioOut.Init(multiplexer);
asioOut.Play();
}
private void AsioOut_AudioAvailable(object sender, AsioAudioAvailableEventArgs e)
{
// Do nothing for now
Console.WriteLine("Audio available");
}
private string GetDanteDriverName()
{
foreach (var driverName in AsioOut.GetDriverNames())
{
if (driverName.Contains("Dante Virtual Soundcard"))
{
return driverName;
}
}
return null;
}
private void OnDestroy()
{
asioOut.Stop();
asioOut.Dispose();
asioOut = null;
}
}
I may have a misunderstanding of how AsioOut works but I'm not sure where to start on this or how to debug the error.

You can use the Low-latency Multichannel Audio asset for unity to play multi-channel audio with asio. It's made especially for unity, unlike naudio, and it works without problems!

Saving Kinect Frames as screenshots

I just started Kinect programming and I am quite happy to have been able to display RGB and IR images at the same time.
Now using the screenshot button I am able to save each frame when I want. (same procedure as in the sample SDKs)
So now if I want to continuously save those frames how can I go about doing that?
I am new to C# and Kinect programming general. So can anyone help me?
Thanks;

just try:
private unsafe void saveFrame(Object reference)
{
MultiSourceFrame mSF = (MultiSourceFrame)reference;
using (var frame = mSF.DepthFrameReference.AcquireFrame())
{
if (frame != null)
{
using (Microsoft.Kinect.KinectBuffer depthBuffer = frame.LockImageBuffer())
{
if ((frame.FrameDescription.Width * frame.FrameDescription.Height) == (depthBuffer.Size / frame.FrameDescription.BytesPerPixel))
{
ushort* frameData = (ushort*)depthBuffer.UnderlyingBuffer;
byte[] rawDataConverted = new byte[(int)(depthBuffer.Size / 2)];
for (int i = 0; i < (int)(depthBuffer.Size / 2); ++i)
{
ushort depth = frameData[i];
rawDataConverted[i] = (byte)(depth >= frame.DepthMinReliableDistance && depth <= frame.DepthMaxReliableDistance ? (depth) : 0);
}
String date = string.Format("{0:hh-mm-ss}", DateTime.Now);
String filePath = System.IO.Directory.GetCurrentDirectory() + "/test/" +date+".raw";
File.WriteAllBytes(filePath, rawDataConverted);
rawDataConverted = null;
}
}
}
}
}
You also take a look here:
Saving raw detph-data

Playing sinus through XAudio2

I'm making an audio player using XAudio2. We are streaming data in packets of 640 bytes, at a sample rate of 8000Hz and sample depth of 16 bytes. We are using SlimDX to access XAudio2.
But when playing sound, we are noticing that the sound quality is bad. This, for example, is a 3KHz sine curve, captured with Audacity.
I have condensed the audio player to the bare basics, but the audio quality is still bad. Is this a bug in XAudio2, SlimDX, or my code, or is this simply an artifact that occurs when one go from 8KHz to 44.1KHz? The last one seems unreasonable, as we also generate PCM wav files which are played perfectly by Windows Media Player.
The following is the basic implementation, which generates the broken Sine.
public partial class MainWindow : Window
{
private XAudio2 device = new XAudio2();
private WaveFormatExtensible format = new WaveFormatExtensible();
private SourceVoice sourceVoice = null;
private MasteringVoice masteringVoice = null;
private Guid KSDATAFORMAT_SUBTYPE_PCM = new Guid("00000001-0000-0010-8000-00aa00389b71");
private AutoResetEvent BufferReady = new AutoResetEvent(false);
private PlayBufferPool PlayBuffers = new PlayBufferPool();
public MainWindow()
{
InitializeComponent();
Closing += OnClosing;
format.Channels = 1;
format.BitsPerSample = 16;
format.FormatTag = WaveFormatTag.Extensible;
format.BlockAlignment = (short)(format.Channels * (format.BitsPerSample / 8));
format.SamplesPerSecond = 8000;
format.AverageBytesPerSecond = format.SamplesPerSecond * format.BlockAlignment;
format.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
}
private void OnClosing(object sender, CancelEventArgs cancelEventArgs)
{
sourceVoice.Stop();
sourceVoice.Dispose();
masteringVoice.Dispose();
PlayBuffers.Dispose();
}
private void button_Click(object sender, RoutedEventArgs e)
{
masteringVoice = new MasteringVoice(device);
PlayBuffer buffer = PlayBuffers.NextBuffer();
GenerateSine(buffer.Buffer);
buffer.AudioBuffer.AudioBytes = 640;
sourceVoice = new SourceVoice(device, format, VoiceFlags.None, 8);
sourceVoice.BufferStart += new EventHandler<ContextEventArgs>(sourceVoice_BufferStart);
sourceVoice.BufferEnd += new EventHandler<ContextEventArgs>(sourceVoice_BufferEnd);
sourceVoice.SubmitSourceBuffer(buffer.AudioBuffer);
sourceVoice.Start();
}
private void sourceVoice_BufferEnd(object sender, ContextEventArgs e)
{
BufferReady.Set();
}
private void sourceVoice_BufferStart(object sender, ContextEventArgs e)
{
BufferReady.WaitOne(1000);
PlayBuffer nextBuffer = PlayBuffers.NextBuffer();
nextBuffer.DataStream.Position = 0;
nextBuffer.AudioBuffer.AudioBytes = 640;
GenerateSine(nextBuffer.Buffer);
Result r = sourceVoice.SubmitSourceBuffer(nextBuffer.AudioBuffer);
}
private void GenerateSine(byte[] buffer)
{
double sampleRate = 8000.0;
double amplitude = 0.25 * short.MaxValue;
double frequency = 3000.0;
for (int n = 0; n < buffer.Length / 2; n++)
{
short[] s = { (short)(amplitude * Math.Sin((2 * Math.PI * n * frequency) / sampleRate)) };
Buffer.BlockCopy(s, 0, buffer, n * 2, 2);
}
}
}
public class PlayBuffer : IDisposable
{
#region Private variables
private IntPtr BufferPtr;
private GCHandle BufferHandle;
#endregion
#region Constructors
public PlayBuffer()
{
Index = 0;
Buffer = new byte[640 * 4]; // 640 = 30ms
BufferHandle = GCHandle.Alloc(this.Buffer, GCHandleType.Pinned);
BufferPtr = new IntPtr(BufferHandle.AddrOfPinnedObject().ToInt32());
DataStream = new DataStream(BufferPtr, 640 * 4, true, false);
AudioBuffer = new AudioBuffer();
AudioBuffer.AudioData = DataStream;
}
public PlayBuffer(int index)
: this()
{
Index = index;
}
#endregion
#region Destructor
~PlayBuffer()
{
Dispose();
}
#endregion
#region Properties
protected int Index { get; private set; }
public byte[] Buffer { get; private set; }
public DataStream DataStream { get; private set; }
public AudioBuffer AudioBuffer { get; private set; }
#endregion
#region Public functions
public void Dispose()
{
if (AudioBuffer != null)
{
AudioBuffer.Dispose();
AudioBuffer = null;
}
if (DataStream != null)
{
DataStream.Dispose();
DataStream = null;
}
}
#endregion
}
public class PlayBufferPool : IDisposable
{
#region Private variables
private int _currentIndex = -1;
private PlayBuffer[] _buffers = new PlayBuffer[2];
#endregion
#region Constructors
public PlayBufferPool()
{
for (int i = 0; i < 2; i++)
Buffers[i] = new PlayBuffer(i);
}
#endregion
#region Desctructor
~PlayBufferPool()
{
Dispose();
}
#endregion
#region Properties
protected int CurrentIndex
{
get { return _currentIndex; }
set { _currentIndex = value; }
}
protected PlayBuffer[] Buffers
{
get { return _buffers; }
set { _buffers = value; }
}
#endregion
#region Public functions
public void Dispose()
{
for (int i = 0; i < Buffers.Length; i++)
{
if (Buffers[i] == null)
continue;
Buffers[i].Dispose();
Buffers[i] = null;
}
}
public PlayBuffer NextBuffer()
{
CurrentIndex = (CurrentIndex + 1) % Buffers.Length;
return Buffers[CurrentIndex];
}
#endregion
}
Some extra details:
This is used to replay recorded voice with various compression such as ALAW, µLAW or TrueSpeech. The data is sent in small packets, decoded and sent to this player. This is the reason for why we're using so low sampling rate, and so small buffers.
There are no problems with our data, however, as generating a WAV file with the data results in perfect replay by WMP or VLC.
edit: We have now "solved" this by rewriting the player in NAudio.
I'd still be interested in any input as to what is happening here. Is it our approach in the PlayBuffers, or is it simply a bug/limitation in DirectX, or the wrappers? I tried using SharpDX instead of SlimDX, but that did not change the result anything.

It looks as if the upsampling is done without a proper anti-aliasing (reconstruction) filter. The cutoff frequency is far too high (above the original Nyquist frequency) and therefore a lot of the aliases are being preserved, resulting in output resembling piecewise-linear interpolation between the samples taken at 8000 Hz.
Although all your different options are doing an upconversion from 8kHz to 44.1kHz, the way in which they do that is important, and the fact that one library does it well is no proof that the upconversion is not the source of error in the other.

It's been a while since I worked with sound and frequencies, but here is what I remember: You have a sample rate of 8000Hz and want a sine frequency of 3000Hz. So for 1 second you have 8000 samples and in that second you want your sine to oscillate 3000 times. That is below the Nyquist-frequency (half your sample rate) but barely (see Nyquist–Shannon sampling theorem). So I would not expect a good quality here.
In fact: step through the GenerateSine-method and you'll see that s[0] will contain the values 0, 5792, -8191, 5792, 0, -5792, 8191, -5792, 0, 5792...
None the less this doesn't explain the odd sine you recorded back and I'm not sure how much samples the human ear need to hear a "good" sine wave.

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