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How to know if audio is going dead in Naudio C#`

I am trying to find out when the song is going dead (inaudible sound for few seconds). I am using Naudio library in C#. Till now i am able to get the PCM data and plot the amplitude of the audio. I am guessing the dead audio through this amplitude i am obtaining. But i am bit confused about audio channels. Following is the piece of code i wrote.
NAudio.Wave.WaveChannel32 wave = new NAudio.Wave.WaveChannel32(new NAudio.Wave.WaveFileReader(open.FileName));
int songLength = (int)wave.Length;
byte[] songPCM = new byte[songLength];
int sampleRate = (int)wave.WaveFormat.SampleRate;
int bitsPerSample = (int)wave.WaveFormat.BitsPerSample;
int numChannels = (int)wave.WaveFormat.Channels;
wave.Read(songPCM, 0, songLength);
double[] _waveLeft = new double[songLength / 8];
double[] _waveRight = new double[songLength / 8];
System.IO.StreamWriter fileoutLeft = new System.IO.StreamWriter("E:\\LOutputSongPCM.dat", true);
System.IO.StreamWriter fileoutRight = new System.IO.StreamWriter("E:\\ROutputSongPCM.dat", true);
int h = 0;
for (int i = 0; i < songLength; i += 8)
{
_waveLeft[h] = (double)BitConverter.ToSingle(songPCM, i);
_waveRight[h] = (double)BitConverter.ToSingle(songPCM, i + 4);
chart1.Series["wave"].Points.Add(_waveLeft[h]);
//chart1.Series["wave"].Points.Add(_waveRight[h]);
fileoutLeft.WriteLine(_waveLeft[h]);
fileoutRight.WriteLine(_waveRight[h]);
h++;
}
fileoutLeft.Close();
fileoutRight.Close();
Now for this piece of code i know the audio is 2 channel. So i referred many links and threads and got confused if i am reading my pcm data for each channel correctly. However i compared the plots of each channel and they look good(Matching with original song) but i am not sure about their accuracy. Can you guide me to get the exact raw data for any channel. for mono, stereo and 5.1.
Thanks.

You'll find it easier to get at the samples by using the AudioFileReader class, whose Read method takes a float array. The samples are stored interleaved for multi-channel audio, so for stereo, you'll get left sample, then right sample, then another left and so on.

C# Real time waveform data plot using NAudio

I am new on processing wav file and C#.My goal is to real time data plotting in waveform of wavfile.I mean while recording sound(wav) file,i want to plot its graph simultaneously.I searched some sound libiraries and decide to use NAudio.(Dont know it is the best choice for me.I am open to any suggestions about choosing audio libirary). However i have no idea about real time data plotting using sound. Some people suggest GDI but as i said i am new and i think it will take too much time to use GDI efficiently.If i must learn GDI,pls share any article that can help a beginner like me. Actually i look like dont know where should i start. Need to be guided :)) And i have a question.
One of the tutorial of NAudio,he works with byte array to plot the waveform in Chart.It is fine if you know the size of wav file.However it works too slow and gives Out of Memory Exception for bigger wav files than 10mb.The code below refers to what i mean.
OpenFileDialog open = new OpenFileDialog();
open.Filter = "Wave File (*.wav)|*.wav;";
if (open.ShowDialog() != DialogResult.OK) return;
chart1.Series.Add("wave");
chart1.Series["wave"].ChartType = System.Windows.Forms.DataVisualization.Charting.SeriesChartType.FastLine;
chart1.Series["wave"].ChartArea = "ChartArea1";
NAudio.Wave.WaveChannel32 wave = new NAudio.Wave.WaveChannel32(new NAudio.Wave.WaveFileReader(open.FileName));
byte[] buffer = new byte[426565];
int read;
while (wave.Position < wave.Length)
{
read = wave.Read(buffer, 0, 426565);
for (int i = 0; i < read / 4; i++)
{
chart1.Series["wave"].Points.Add(BitConverter.ToSingle(buffer, i * 4));
}
}
Is there a way to perform this operation faster?

If you plot every single sample, you will end up with a waveform that is unmanageably large since audio usually contains many thousands of samples per second. A common way waveforms are drawn is by selecting the maximum value over a period of time, and then drawing a vertical line to represent it. For example, if you had a three minute song, and wanted a waveform around 600 pixels wide, each pixel would represent about a third of a second. So you'd find the largest sample value in that third of a second and use that to plot your waveform.
Also, in your sample code you are reading an odd number of bytes. But since this is floating point audio, you should always read in multiples of four bytes.

This worked for me
WaveChannel32 wave = new WaveChannel32(new WaveFileReader(txtWave.Text));
int sampleSize = 1024;
var bufferSize = 16384 * sampleSize;
var buffer = new byte[bufferSize];
int read = 0;
chart.Series.Add("wave");
chart.Series["wave"].ChartType = System.Windows.Forms.DataVisualization.Charting.SeriesChartType.FastLine;
chart.Series["wave"].ChartArea = "ChartArea1";
while (wave.Position < wave.Length)
{
read = wave.Read(buffer, 0, bufferSize);
for (int i = 0; i < read / sampleSize; i++)
{
var point = BitConverter.ToSingle(buffer, i * sampleSize);
chart.Series["wave"].Points.Add(point);
}
}

Resampling raw audio with NAudio

I'd like to resample audio: change its sample rate from 44k to 11k. The input I've got is raw audio in bytes. It really is raw, it has no headers - if I try loading it into a WaveFileReader, I get an exception saying "Not a WAVE file - no RIFF header".
How I'm currently trying to achieve it is something like this (just a really simplified piece of code):
WaveFormat ResampleInputFormat = new WaveFormat(44100, 1);
WaveFormat ResampleOutputFormat = new WaveFormat(11025, 1);
MemoryStream ResampleInputMemoryStream = new MemoryStream();
foreach (var b in InputListOfBytes)
{
ResampleInputMemoryStream.Write(new byte[]{b}, 0, 1);
}
RawSourceWaveStream ResampleInputWaveStream =
new RawSourceWaveStream(ResampleInputMemoryStream, ResampleInputFormat);
WaveFormatConversionStream ResampleOutputStream =
new WaveFormatConversionStream(ResampleOutputFormat, ResampleInputWaveStream);
byte[] bytes = new byte[2];
while (ResampleOutputStream.Read(bytes, 0, 2) > 0)
{
OutputListOfBytes.Add(bytes[0]);
OutputListOfBytes.Add(bytes[1]);
}
My problem is: the last loop is an infinite loop. The Read() always gets the same values and never advances in the Stream. I even tried Seek()-ing to the right position after each Read(), but that doesn't seem to work either, I still always get the same values.
What am I doing wrong? And is this even the right way to resample raw audio? Thanks in advance!

First, you need to reset ResampleInputMemoryStream's position to the start. It may actually have been easier to create the memory stream based on the array:
new MemoryStream(InputListOfBytes)
Second, when reading out of the resampler, you need to read in larger chunks than two bytes at a time. Try at least a second's worth of audio (use ResampleOutputStream.WaveFormat.AverageBytesPerSecond).

Play dynamically-created simple sounds in C# without external libraries

I need to be able to generate dynamically a waveform and play it, using C#, without any external libraries, and without having to store sound files on the hard disk. Latency isn't an issue; the sounds would be generated well before they are needed by the application.
Actually the Console.Beep() method might meet my needs if it weren't for the fact that Microsoft says it isn't supported in 64-bit versions of Windows.
If I generate my own sound dynamically I can get a more fancy than a simple beep. For example, I could make a waveform from a triangle wave that increases in frequency from 2 KHz to 4 KHz while decaying in volume. I don't need fancy 16-bit stereo, just 8-bit mono is fine. I don't need dynamic control over volume and pitch, just basically generate a soundfile in memory and play it without storing it.
Last time I needed to generate sounds was many years ago, on Apple II, on HP workstations, and on my old Amiga computer. Haven't needed to do it since then, and it seems that something simple that I describe has gotten a lot more complicated. I am having trouble believing that something so simple seems so hard. Most of the answers I see refer to NAudio or similar libraries, and that isn't an option for this project (aside from the fact that pulling in an entire library just to play a tone seems like a waste).

Based on one of the links in the answers I received, and some other pages I found about .wav header formats, here is my working code for a little class that generates an 8-bit "ding!" sound with a user-specified frequency and duration. It's basically a beep that decays linearly to zero in amplitude during the specified duration.
public class AlertDing {
private SoundPlayer player = null;
private BinaryWriter writer = null;
/// <summary>
/// Dynamically generate a "ding" sound and save it to a memory stream
/// </summary>
/// <param name="freq">Frequency in Hertz, e.g. 880</param>
/// <param name="tenthseconds">Duration in multiple of 1/10 second</param>
public AlertDing(double freq, uint tenthseconds) {
string header_GroupID = "RIFF"; // RIFF
uint header_FileLength = 0; // total file length minus 8, which is taken up by RIFF
string header_RiffType = "WAVE"; // always WAVE
string fmt_ChunkID = "fmt "; // Four bytes: "fmt "
uint fmt_ChunkSize = 16; // Length of header in bytes
ushort fmt_FormatTag = 1; // 1 for PCM
ushort fmt_Channels = 1; // Number of channels, 2=stereo
uint fmt_SamplesPerSec = 14000; // sample rate, e.g. CD=44100
ushort fmt_BitsPerSample = 8; // bits per sample
ushort fmt_BlockAlign =
(ushort)(fmt_Channels * (fmt_BitsPerSample / 8)); // sample frame size, in bytes
uint fmt_AvgBytesPerSec =
fmt_SamplesPerSec * fmt_BlockAlign; // for estimating RAM allocation
string data_ChunkID = "data"; // "data"
uint data_ChunkSize; // Length of header in bytes
byte [] data_ByteArray;
// Fill the data array with sample data
// Number of samples = sample rate * channels * bytes per sample * duration in seconds
uint numSamples = fmt_SamplesPerSec * fmt_Channels * tenthseconds / 10;
data_ByteArray = new byte[numSamples];
//int amplitude = 32760, offset=0; // for 16-bit audio
int amplitude = 127, offset = 128; // for 8-audio
double period = (2.0*Math.PI*freq) / (fmt_SamplesPerSec * fmt_Channels);
double amp;
for (uint i = 0; i < numSamples - 1; i += fmt_Channels) {
amp = amplitude * (double)(numSamples - i) / numSamples; // amplitude decay
// Fill with a waveform on each channel with amplitude decay
for (int channel = 0; channel < fmt_Channels; channel++) {
data_ByteArray[i+channel] = Convert.ToByte(amp * Math.Sin(i*period) + offset);
}
}
// Calculate file and data chunk size in bytes
data_ChunkSize = (uint)(data_ByteArray.Length * (fmt_BitsPerSample / 8));
header_FileLength = 4 + (8 + fmt_ChunkSize) + (8 + data_ChunkSize);
// write data to a MemoryStream with BinaryWriter
MemoryStream audioStream = new MemoryStream();
BinaryWriter writer = new BinaryWriter(audioStream);
// Write the header
writer.Write(header_GroupID.ToCharArray());
writer.Write(header_FileLength);
writer.Write(header_RiffType.ToCharArray());
// Write the format chunk
writer.Write(fmt_ChunkID.ToCharArray());
writer.Write(fmt_ChunkSize);
writer.Write(fmt_FormatTag);
writer.Write(fmt_Channels);
writer.Write(fmt_SamplesPerSec);
writer.Write(fmt_AvgBytesPerSec);
writer.Write(fmt_BlockAlign);
writer.Write(fmt_BitsPerSample);
// Write the data chunk
writer.Write(data_ChunkID.ToCharArray());
writer.Write(data_ChunkSize);
foreach (byte dataPoint in data_ByteArray) {
writer.Write(dataPoint);
}
player = new SoundPlayer(audioStream);
}
/// <summary>
/// Call this to clean up when program is done using this sound
/// </summary>
public void Dispose() {
if (writer != null) writer.Close();
if (player != null) player.Dispose();
writer = null;
player = null;
}
/// <summary>
/// Play "ding" sound
/// </summary>
public void Play() {
if (player != null) {
player.Stream.Seek(0, SeekOrigin.Begin); // rewind stream
player.Play();
}
}
}
Hopefully this should help others who are trying to produce a simple alert sound dynamically without needing a sound file.

The following article explains how *.wav file can be generated and played using SoundPlayer. Be aware that SoundPlayer can take a stream as an argument, so you can generate wav-file contents in a MemoryStream and avoid saving to a file.
http://blogs.msdn.com/b/dawate/archive/2009/06/24/intro-to-audio-programming-part-3-synthesizing-simple-wave-audio-using-c.aspx

I tried out the code-snipped from Anachronist (2012-10) - and it is working for me.
biggest hurdle for me:
get rid of the systematic "clicking-noise" at the end of "AlertDing" wav.
This is caused by a "soft-bug" in the code snipped:
for (uint i = 0; i < numSamples - 1; i += fmt_Channels)
needs to change to
for (uint i = 0; i < numSamples; i += fmt_Channels)
if not changed, a systematic "zero" will be generated at the end of each "play", causing a sharp clicking noise. (= amplitude jumps 0->min->0)
the original question implies "without clicking noise" of course :)

Use NAudio to get Ulaw samples for RTP

I've been looking over the NAudio examples trying to work out how I can get ulaw samples suitable for packaging up as an RTP payload. I'm attempting to generate the samples from an mp3 file using the code below. Not surprisingly, since I don't really have a clue what I'm doing with NAudio, when I transmit the samples across the network to a softphone all I get is static.
Can anyone provide any direction on how I should be getting 160 bytes (8Khz # 20ms) ULAW samples from an MP3 file using NAudio?
private void GetAudioSamples()
{
var pcmStream = WaveFormatConversionStream.CreatePcmStream(new Mp3FileReader("whitelight.mp3"));
byte[] buffer = new byte[2];
byte[] sampleBuffer = new byte[160];
int sampleIndex = 0;
int bytesRead = pcmStream.Read(buffer, 0, 2);
while (bytesRead > 0)
{
var ulawByte = MuLawEncoder.LinearToMuLawSample(BitConverter.ToInt16(buffer, 0));
sampleBuffer[sampleIndex++] = ulawByte;
if (sampleIndex == 160)
{
m_rtpChannel.AddSample(sampleBuffer);
sampleBuffer = new byte[160];
sampleIndex = 0;
}
bytesRead = pcmStream.Read(buffer, 0, 2);
}
logger.Debug("Finished adding audio samples.");
}

Here's a few pointers. First of all, as long as you are using NAudio 1.5, no need for the additional WaveFormatConversionStream - Mp3FileReader's Read method returns PCM.
However, you will not be getting 8kHz out, so you need to resample it first. WaveFormatConversionStream can do this, although it uses the built-in Windows ACM sample rate conversion, which doesn't seem to filter the incoming audio well, so there could be aliasing artefacts.
Also, you usually read bigger blocks than just two bytes at a time as the MP3 decoder needs to decode frames one at a time (the resampler also will want to deal with bigger block sizes). I would try reading at least 20ms worth of bytes at a time.
Your use of BitConverter.ToInt16 is correct for getting the 16 bit sample value, but bear in mind that an MP3 is likely stereo, with left, right samples. Are you sure your phone expects stereo.
Finally, I recommend making a mu-law WAV file as a first step, using WaveFileWriter. Then you can easily listen to it in Windows Media Player and check that what you are sending to your softphone is what you intended.

Below is the way I eventually got it working. I do lose one of the channels from the mp3, and I guess there's some way to combine the channels as part of a conversion, but that doesn't matter for my situation.
The 160 byte buffer size gives me 20ms ulaw samples which work perfectly with the SIP softphone I'm testing with.
var pcmFormat = new WaveFormat(8000, 16, 1);
var ulawFormat = WaveFormat.CreateMuLawFormat(8000, 1);
using (WaveFormatConversionStream pcmStm = new WaveFormatConversionStream(pcmFormat, new Mp3FileReader("whitelight.mp3")))
{
using (WaveFormatConversionStream ulawStm = new WaveFormatConversionStream(ulawFormat, pcmStm))
{
byte[] buffer = new byte[160];
int bytesRead = ulawStm.Read(buffer, 0, 160);
while (bytesRead > 0)
{
byte[] sample = new byte[bytesRead];
Array.Copy(buffer, sample, bytesRead);
m_rtpChannel.AddSample(sample);
bytesRead = ulawStm.Read(buffer, 0, 160);
}
}
}