public Mixer()
{
BuildAUGraph();
_converter = AudioConverter.Create(MixerNode.GetAudioFormat(AudioUnitScopeType.Output), AudioStreamBasicDescription.CreateLinearPCM());
Metronome.Instance.TempoChanged += TempoChanged;
_countOff = new PitchStream(StreamInfoProvider.GetDefault(), null);
_countOff.IntervalLoop = new SampleIntervalLoop(_countOff, new double[] { 1 });
_countOff.AddFrequency("A4");
}
public static bool Convert(string input, string output, AudioFormatType targetFormat, AudioFileType containerType, Microsoft.Xna.Framework.Content.Pipeline.Audio.ConversionQuality quality)
{
CFUrl source = CFUrl.FromFile(input);
CFUrl dest = CFUrl.FromFile(output);
var dstFormat = new AudioStreamBasicDescription();
var sourceFile = AudioFile.Open(source, AudioFilePermission.Read);
AudioFormatType outputFormat = targetFormat;
// get the source data format
var srcFormat = (AudioStreamBasicDescription)sourceFile.DataFormat;
var outputSampleRate = 0;
switch (quality)
{
case Microsoft.Xna.Framework.Content.Pipeline.Audio.ConversionQuality.Low:
outputSampleRate = (int)Math.Max(8000, srcFormat.SampleRate / 2);
break;
default:
outputSampleRate = (int)Math.Max(8000, srcFormat.SampleRate);
break;
}
dstFormat.SampleRate = (outputSampleRate == 0 ? srcFormat.SampleRate : outputSampleRate); // set sample rate
if (outputFormat == AudioFormatType.LinearPCM)
{
// if the output format is PC create a 16-bit int PCM file format description as an example
dstFormat.Format = outputFormat;
dstFormat.ChannelsPerFrame = srcFormat.ChannelsPerFrame;
dstFormat.BitsPerChannel = 16;
dstFormat.BytesPerPacket = dstFormat.BytesPerFrame = 2 * dstFormat.ChannelsPerFrame;
dstFormat.FramesPerPacket = 1;
dstFormat.FormatFlags = AudioFormatFlags.LinearPCMIsPacked | AudioFormatFlags.LinearPCMIsSignedInteger;
}
else
{
// compressed format - need to set at least format, sample rate and channel fields for kAudioFormatProperty_FormatInfo
dstFormat.Format = outputFormat;
dstFormat.ChannelsPerFrame = (outputFormat == AudioFormatType.iLBC ? 1 : srcFormat.ChannelsPerFrame); // for iLBC num channels must be 1
// use AudioFormat API to fill out the rest of the description
var fie = AudioStreamBasicDescription.GetFormatInfo(ref dstFormat);
if (fie != AudioFormatError.None)
{
return(false);
}
}
var converter = AudioConverter.Create(srcFormat, dstFormat);
converter.InputData += HandleInputData;
// if the source has a cookie, get it and set it on the Audio Converter
ReadCookie(sourceFile, converter);
// get the actual formats back from the Audio Converter
srcFormat = converter.CurrentInputStreamDescription;
dstFormat = converter.CurrentOutputStreamDescription;
// if encoding to AAC set the bitrate to 192k which is a nice value for this demo
// kAudioConverterEncodeBitRate is a UInt32 value containing the number of bits per second to aim for when encoding data
if (dstFormat.Format == AudioFormatType.MPEG4AAC)
{
uint outputBitRate = 192000; // 192k
// ignore errors as setting may be invalid depending on format specifics such as samplerate
try {
converter.EncodeBitRate = outputBitRate;
} catch {
}
// get it back and print it out
outputBitRate = converter.EncodeBitRate;
}
// create the destination file
var destinationFile = AudioFile.Create(dest, containerType, dstFormat, AudioFileFlags.EraseFlags);
// set up source buffers and data proc info struct
afio = new AudioFileIO(32768);
afio.SourceFile = sourceFile;
afio.SrcFormat = srcFormat;
if (srcFormat.BytesPerPacket == 0)
{
// if the source format is VBR, we need to get the maximum packet size
// use kAudioFilePropertyPacketSizeUpperBound which returns the theoretical maximum packet size
// in the file (without actually scanning the whole file to find the largest packet,
// as may happen with kAudioFilePropertyMaximumPacketSize)
afio.SrcSizePerPacket = sourceFile.PacketSizeUpperBound;
// how many packets can we read for our buffer size?
afio.NumPacketsPerRead = afio.SrcBufferSize / afio.SrcSizePerPacket;
// allocate memory for the PacketDescription structures describing the layout of each packet
afio.PacketDescriptions = new AudioStreamPacketDescription [afio.NumPacketsPerRead];
}
else
{
// CBR source format
afio.SrcSizePerPacket = srcFormat.BytesPerPacket;
afio.NumPacketsPerRead = afio.SrcBufferSize / afio.SrcSizePerPacket;
// allocate memory for the PacketDescription structures describing the layout of each packet
afio.PacketDescriptions = new AudioStreamPacketDescription [afio.NumPacketsPerRead];
}
// set up output buffers
int outputSizePerPacket = dstFormat.BytesPerPacket; // this will be non-zero if the format is CBR
const int theOutputBufSize = 32768;
var outputBuffer = Marshal.AllocHGlobal(theOutputBufSize);
AudioStreamPacketDescription[] outputPacketDescriptions = null;
if (outputSizePerPacket == 0)
{
// if the destination format is VBR, we need to get max size per packet from the converter
outputSizePerPacket = (int)converter.MaximumOutputPacketSize;
}
// allocate memory for the PacketDescription structures describing the layout of each packet
outputPacketDescriptions = new AudioStreamPacketDescription [theOutputBufSize / outputSizePerPacket];
int numOutputPackets = theOutputBufSize / outputSizePerPacket;
// if the destination format has a cookie, get it and set it on the output file
WriteCookie(converter, destinationFile);
// write destination channel layout
if (srcFormat.ChannelsPerFrame > 2)
{
WriteDestinationChannelLayout(converter, sourceFile, destinationFile);
}
long totalOutputFrames = 0; // used for debugging
long outputFilePos = 0;
AudioBuffers fillBufList = new AudioBuffers(1);
bool error = false;
// loop to convert data
while (true)
{
// set up output buffer list
fillBufList [0] = new AudioBuffer()
{
NumberChannels = dstFormat.ChannelsPerFrame,
DataByteSize = theOutputBufSize,
Data = outputBuffer
};
// convert data
int ioOutputDataPackets = numOutputPackets;
var fe = converter.FillComplexBuffer(ref ioOutputDataPackets, fillBufList, outputPacketDescriptions);
// if interrupted in the process of the conversion call, we must handle the error appropriately
if (fe != AudioConverterError.None)
{
error = true;
break;
}
if (ioOutputDataPackets == 0)
{
// this is the EOF conditon
break;
}
// write to output file
var inNumBytes = fillBufList [0].DataByteSize;
var we = destinationFile.WritePackets(false, inNumBytes, outputPacketDescriptions, outputFilePos, ref ioOutputDataPackets, outputBuffer);
if (we != 0)
{
error = true;
break;
}
// advance output file packet position
outputFilePos += ioOutputDataPackets;
if (dstFormat.FramesPerPacket != 0)
{
// the format has constant frames per packet
totalOutputFrames += (ioOutputDataPackets * dstFormat.FramesPerPacket);
}
else
{
// variable frames per packet require doing this for each packet (adding up the number of sample frames of data in each packet)
for (var i = 0; i < ioOutputDataPackets; ++i)
{
totalOutputFrames += outputPacketDescriptions [i].VariableFramesInPacket;
}
}
}
Marshal.FreeHGlobal(outputBuffer);
if (!error)
{
// write out any of the leading and trailing frames for compressed formats only
if (dstFormat.BitsPerChannel == 0)
{
// our output frame count should jive with
WritePacketTableInfo(converter, destinationFile);
}
// write the cookie again - sometimes codecs will update cookies at the end of a conversion
WriteCookie(converter, destinationFile);
}
converter.Dispose();
destinationFile.Dispose();
sourceFile.Dispose();
return(true);
}
bool DoConvertFile(CFUrl sourceURL, NSUrl destinationURL, AudioFormatType outputFormat, double outputSampleRate)
{
AudioStreamBasicDescription dstFormat = new AudioStreamBasicDescription();
// in this sample we should never be on the main thread here
Debug.Assert(!NSThread.IsMain);
// transition thread state to State::Running before continuing
AppDelegate.ThreadStateSetRunning();
Debug.WriteLine("DoConvertFile");
// get the source file
var sourceFile = AudioFile.Open(sourceURL, AudioFilePermission.Read);
// get the source data format
var srcFormat = (AudioStreamBasicDescription)sourceFile.DataFormat;
// setup the output file format
dstFormat.SampleRate = (outputSampleRate == 0 ? srcFormat.SampleRate : outputSampleRate); // set sample rate
if (outputFormat == AudioFormatType.LinearPCM)
{
// if the output format is PC create a 16-bit int PCM file format description as an example
dstFormat.Format = outputFormat;
dstFormat.ChannelsPerFrame = srcFormat.ChannelsPerFrame;
dstFormat.BitsPerChannel = 16;
dstFormat.BytesPerPacket = dstFormat.BytesPerFrame = 2 * dstFormat.ChannelsPerFrame;
dstFormat.FramesPerPacket = 1;
dstFormat.FormatFlags = AudioFormatFlags.LinearPCMIsPacked | AudioFormatFlags.LinearPCMIsSignedInteger;
}
else
{
// compressed format - need to set at least format, sample rate and channel fields for kAudioFormatProperty_FormatInfo
dstFormat.Format = outputFormat;
dstFormat.ChannelsPerFrame = (outputFormat == AudioFormatType.iLBC ? 1 : srcFormat.ChannelsPerFrame); // for iLBC num channels must be 1
// use AudioFormat API to fill out the rest of the description
var fie = AudioStreamBasicDescription.GetFormatInfo(ref dstFormat);
if (fie != AudioFormatError.None)
{
Debug.Print("Cannot create destination format {0:x}", fie);
AppDelegate.ThreadStateSetDone();
return(false);
}
}
// create the AudioConverter
AudioConverterError ce;
var converter = AudioConverter.Create(srcFormat, dstFormat, out ce);
Debug.Assert(ce == AudioConverterError.None);
converter.InputData += EncoderDataProc;
// if the source has a cookie, get it and set it on the Audio Converter
ReadCookie(sourceFile, converter);
// get the actual formats back from the Audio Converter
srcFormat = converter.CurrentInputStreamDescription;
dstFormat = converter.CurrentOutputStreamDescription;
// if encoding to AAC set the bitrate to 192k which is a nice value for this demo
// kAudioConverterEncodeBitRate is a UInt32 value containing the number of bits per second to aim for when encoding data
if (dstFormat.Format == AudioFormatType.MPEG4AAC)
{
uint outputBitRate = 192000; // 192k
// ignore errors as setting may be invalid depending on format specifics such as samplerate
try {
converter.EncodeBitRate = outputBitRate;
} catch {
}
// get it back and print it out
outputBitRate = converter.EncodeBitRate;
Debug.Print("AAC Encode Bitrate: {0}", outputBitRate);
}
// can the Audio Converter resume conversion after an interruption?
// this property may be queried at any time after construction of the Audio Converter after setting its output format
// there's no clear reason to prefer construction time, interruption time, or potential resumption time but we prefer
// construction time since it means less code to execute during or after interruption time
bool canResumeFromInterruption;
try {
canResumeFromInterruption = converter.CanResumeFromInterruption;
Debug.Print("Audio Converter {0} continue after interruption!", canResumeFromInterruption ? "CAN" : "CANNOT");
} catch (Exception e) {
// if the property is unimplemented (kAudioConverterErr_PropertyNotSupported, or paramErr returned in the case of PCM),
// then the codec being used is not a hardware codec so we're not concerned about codec state
// we are always going to be able to resume conversion after an interruption
canResumeFromInterruption = false;
Debug.Print("CanResumeFromInterruption: {0}", e.Message);
}
// create the destination file
var destinationFile = AudioFile.Create(destinationURL, AudioFileType.CAF, dstFormat, AudioFileFlags.EraseFlags);
// set up source buffers and data proc info struct
afio = new AudioFileIO(32768);
afio.SourceFile = sourceFile;
afio.SrcFormat = srcFormat;
if (srcFormat.BytesPerPacket == 0)
{
// if the source format is VBR, we need to get the maximum packet size
// use kAudioFilePropertyPacketSizeUpperBound which returns the theoretical maximum packet size
// in the file (without actually scanning the whole file to find the largest packet,
// as may happen with kAudioFilePropertyMaximumPacketSize)
afio.SrcSizePerPacket = sourceFile.PacketSizeUpperBound;
// how many packets can we read for our buffer size?
afio.NumPacketsPerRead = afio.SrcBufferSize / afio.SrcSizePerPacket;
// allocate memory for the PacketDescription structures describing the layout of each packet
afio.PacketDescriptions = new AudioStreamPacketDescription [afio.NumPacketsPerRead];
}
else
{
// CBR source format
afio.SrcSizePerPacket = srcFormat.BytesPerPacket;
afio.NumPacketsPerRead = afio.SrcBufferSize / afio.SrcSizePerPacket;
}
// set up output buffers
int outputSizePerPacket = dstFormat.BytesPerPacket; // this will be non-zero if the format is CBR
const int theOutputBufSize = 32768;
var outputBuffer = Marshal.AllocHGlobal(theOutputBufSize);
AudioStreamPacketDescription[] outputPacketDescriptions = null;
if (outputSizePerPacket == 0)
{
// if the destination format is VBR, we need to get max size per packet from the converter
outputSizePerPacket = (int)converter.MaximumOutputPacketSize;
// allocate memory for the PacketDescription structures describing the layout of each packet
outputPacketDescriptions = new AudioStreamPacketDescription [theOutputBufSize / outputSizePerPacket];
}
int numOutputPackets = theOutputBufSize / outputSizePerPacket;
// if the destination format has a cookie, get it and set it on the output file
WriteCookie(converter, destinationFile);
// write destination channel layout
if (srcFormat.ChannelsPerFrame > 2)
{
WriteDestinationChannelLayout(converter, sourceFile, destinationFile);
}
long totalOutputFrames = 0; // used for debugging
long outputFilePos = 0;
AudioBuffers fillBufList = new AudioBuffers(1);
bool error = false;
// loop to convert data
Debug.WriteLine("Converting...");
while (true)
{
// set up output buffer list
fillBufList [0] = new AudioBuffer()
{
NumberChannels = dstFormat.ChannelsPerFrame,
DataByteSize = theOutputBufSize,
Data = outputBuffer
};
// this will block if we're interrupted
var wasInterrupted = AppDelegate.ThreadStatePausedCheck();
if (wasInterrupted && !canResumeFromInterruption)
{
// this is our interruption termination condition
// an interruption has occured but the Audio Converter cannot continue
Debug.WriteLine("Cannot resume from interruption");
error = true;
break;
}
// convert data
int ioOutputDataPackets = numOutputPackets;
var fe = converter.FillComplexBuffer(ref ioOutputDataPackets, fillBufList, outputPacketDescriptions);
// if interrupted in the process of the conversion call, we must handle the error appropriately
if (fe != AudioConverterError.None)
{
Debug.Print("FillComplexBuffer: {0}", fe);
error = true;
break;
}
if (ioOutputDataPackets == 0)
{
// this is the EOF conditon
break;
}
// write to output file
var inNumBytes = fillBufList [0].DataByteSize;
var we = destinationFile.WritePackets(false, inNumBytes, outputPacketDescriptions, outputFilePos, ref ioOutputDataPackets, outputBuffer);
if (we != 0)
{
Debug.Print("WritePackets: {0}", we);
error = true;
break;
}
// advance output file packet position
outputFilePos += ioOutputDataPackets;
if (dstFormat.FramesPerPacket != 0)
{
// the format has constant frames per packet
totalOutputFrames += (ioOutputDataPackets * dstFormat.FramesPerPacket);
}
else
{
// variable frames per packet require doing this for each packet (adding up the number of sample frames of data in each packet)
for (var i = 0; i < ioOutputDataPackets; ++i)
{
totalOutputFrames += outputPacketDescriptions [i].VariableFramesInPacket;
}
}
}
Marshal.FreeHGlobal(outputBuffer);
if (!error)
{
// write out any of the leading and trailing frames for compressed formats only
if (dstFormat.BitsPerChannel == 0)
{
// our output frame count should jive with
Debug.Print("Total number of output frames counted: {0}", totalOutputFrames);
WritePacketTableInfo(converter, destinationFile);
}
// write the cookie again - sometimes codecs will update cookies at the end of a conversion
WriteCookie(converter, destinationFile);
}
converter.Dispose();
destinationFile.Dispose();
sourceFile.Dispose();
// transition thread state to State.Done before continuing
AppDelegate.ThreadStateSetDone();
return(!error);
}
void Convert(string sourceFilePath, string destinationFilePath, AudioFormatType outputFormatType, int?sampleRate = null)
{
var destinationUrl = NSUrl.FromFilename(destinationFilePath);
var sourceUrl = NSUrl.FromFilename(sourceFilePath);
// get the source file
var name = Path.GetFileNameWithoutExtension(destinationFilePath);
using var sourceFile = AudioFile.Open(sourceUrl, AudioFilePermission.Read);
var srcFormat = (AudioStreamBasicDescription)sourceFile.DataFormat;
var dstFormat = new AudioStreamBasicDescription();
// setup the output file format
dstFormat.SampleRate = sampleRate ?? srcFormat.SampleRate;
if (outputFormatType == AudioFormatType.LinearPCM)
{
// if the output format is PCM create a 16 - bit int PCM file format
dstFormat.Format = AudioFormatType.LinearPCM;
dstFormat.ChannelsPerFrame = srcFormat.ChannelsPerFrame;
dstFormat.BitsPerChannel = 16;
dstFormat.BytesPerPacket = dstFormat.BytesPerFrame = 2 * dstFormat.ChannelsPerFrame;
dstFormat.FramesPerPacket = 1;
dstFormat.FormatFlags = AudioFormatFlags.LinearPCMIsPacked | AudioFormatFlags.LinearPCMIsSignedInteger;
}
else
{
// compressed format - need to set at least format, sample rate and channel fields for kAudioFormatProperty_FormatInfo
dstFormat.Format = outputFormatType;
dstFormat.ChannelsPerFrame = srcFormat.ChannelsPerFrame; // for iLBC num channels must be 1
// use AudioFormat API to fill out the rest of the description
var afe = AudioStreamBasicDescription.GetFormatInfo(ref dstFormat);
Assert.AreEqual(AudioFormatError.None, afe, $"GetFormatInfo: {name}");
}
// create the AudioConverter
using var converter = AudioConverter.Create(srcFormat, dstFormat, out var ce);
Assert.AreEqual(AudioConverterError.None, ce, $"AudioConverterCreate: {name}");
// set up source buffers and data proc info struct
var afio = new AudioFileIO(32 * 1024); // 32Kb
converter.InputData += (ref int numberDataPackets, AudioBuffers data, ref AudioStreamPacketDescription [] dataPacketDescription) => {
return(EncoderDataProc(afio, ref numberDataPackets, data, ref dataPacketDescription));
};
// Some audio formats have a magic cookie associated with them which is required to decompress audio data
// When converting audio data you must check to see if the format of the data has a magic cookie
// If the audio data format has a magic cookie associated with it, you must add this information to anAudio Converter
// using AudioConverterSetProperty and kAudioConverterDecompressionMagicCookie to appropriately decompress the data
// http://developer.apple.com/mac/library/qa/qa2001/qa1318.html
var cookie = sourceFile.MagicCookie;
// if there is an error here, then the format doesn't have a cookie - this is perfectly fine as some formats do not
if (cookie?.Length > 0)
{
converter.DecompressionMagicCookie = cookie;
}
// get the actual formats back from the Audio Converter
srcFormat = converter.CurrentInputStreamDescription;
dstFormat = converter.CurrentOutputStreamDescription;
// create the destination file
using var destinationFile = AudioFile.Create(destinationUrl, AudioFileType.CAF, dstFormat, AudioFileFlags.EraseFlags);
// set up source buffers and data proc info struct
afio.SourceFile = sourceFile;
afio.SrcFormat = srcFormat;
if (srcFormat.BytesPerPacket == 0)
{
// if the source format is VBR, we need to get the maximum packet size
// use kAudioFilePropertyPacketSizeUpperBound which returns the theoretical maximum packet size
// in the file (without actually scanning the whole file to find the largest packet,
// as may happen with kAudioFilePropertyMaximumPacketSize)
afio.SrcSizePerPacket = sourceFile.PacketSizeUpperBound;
// how many packets can we read for our buffer size?
afio.NumPacketsPerRead = afio.SrcBufferSize / afio.SrcSizePerPacket;
// allocate memory for the PacketDescription structures describing the layout of each packet
afio.PacketDescriptions = new AudioStreamPacketDescription [afio.NumPacketsPerRead];
}
else
{
// CBR source format
afio.SrcSizePerPacket = srcFormat.BytesPerPacket;
afio.NumPacketsPerRead = afio.SrcBufferSize / afio.SrcSizePerPacket;
}
// set up output buffers
int outputSizePerPacket = dstFormat.BytesPerPacket; // this will be non-zero if the format is CBR
const int theOutputBufSize = 32 * 1024; // 32Kb
var outputBuffer = Marshal.AllocHGlobal(theOutputBufSize);
AudioStreamPacketDescription [] outputPacketDescriptions = null;
if (outputSizePerPacket == 0)
{
// if the destination format is VBR, we need to get max size per packet from the converter
outputSizePerPacket = (int)converter.MaximumOutputPacketSize;
// allocate memory for the PacketDescription structures describing the layout of each packet
outputPacketDescriptions = new AudioStreamPacketDescription [theOutputBufSize / outputSizePerPacket];
}
int numOutputPackets = theOutputBufSize / outputSizePerPacket;
// if the destination format has a cookie, get it and set it on the output file
WriteCookie(converter, destinationFile);
long totalOutputFrames = 0; // used for debugging
long outputFilePos = 0;
AudioBuffers fillBufList = new AudioBuffers(1);
// loop to convert data
while (true)
{
// set up output buffer list
fillBufList [0] = new AudioBuffer()
{
NumberChannels = dstFormat.ChannelsPerFrame,
DataByteSize = theOutputBufSize,
Data = outputBuffer
};
// convert data
int ioOutputDataPackets = numOutputPackets;
var fe = converter.FillComplexBuffer(ref ioOutputDataPackets, fillBufList, outputPacketDescriptions);
// if interrupted in the process of the conversion call, we must handle the error appropriately
Assert.AreEqual(AudioConverterError.None, fe, $"FillComplexBuffer: {name}");
if (ioOutputDataPackets == 0)
{
// this is the EOF conditon
break;
}
// write to output file
var inNumBytes = fillBufList [0].DataByteSize;
var we = destinationFile.WritePackets(false, inNumBytes, outputPacketDescriptions, outputFilePos, ref ioOutputDataPackets, outputBuffer);
Assert.AreEqual(AudioFileError.Success, we, $"WritePackets: {name}");
// advance output file packet position
outputFilePos += ioOutputDataPackets;
// the format has constant frames per packet
totalOutputFrames += (ioOutputDataPackets * dstFormat.FramesPerPacket);
}
Marshal.FreeHGlobal(outputBuffer);
// write out any of the leading and trailing frames for compressed formats only
if (dstFormat.BitsPerChannel == 0)
{
WritePacketTableInfo(converter, destinationFile);
}
// write the cookie again - sometimes codecs will update cookies at the end of a conversion
WriteCookie(converter, destinationFile);
}
/// <summary>
/// Renders the given number of seconds to the given wav file
/// </summary>
/// <param name="fileName">File name.</param>
/// <param name="seconds">Seconds.</param>
public void RenderToFile(string fileName, double seconds)
{
long samples = (long)(seconds * Metronome.SampleRate);
var inputStream = MixerNode.GetAudioFormat(AudioUnitScopeType.Output);
var outputStream = AudioStreamBasicDescription.CreateLinearPCM(44100, 2);
AudioConverter converter = AudioConverter.Create(inputStream, outputStream);
var file = ExtAudioFile.CreateWithUrl(
new Foundation.NSUrl(fileName, false),
AudioFileType.WAVE,
outputStream,
AudioFileFlags.EraseFlags,
out ExtAudioFileError e
);
long samplesRead = 0;
// initialize the buffers
var buffers = new AudioBuffers(2);
buffers[0] = new AudioBuffer()
{
DataByteSize = BufferSize * 4,
NumberChannels = 1,
Data = Marshal.AllocHGlobal(sizeof(float) * BufferSize)
};
buffers[1] = new AudioBuffer()
{
DataByteSize = BufferSize * 4,
NumberChannels = 1,
Data = Marshal.AllocHGlobal(sizeof(float) * BufferSize)
};
var convBuffers = new AudioBuffers(1);
convBuffers[0] = new AudioBuffer()
{
DataByteSize = BufferSize * 4,
NumberChannels = 2,
Data = Marshal.AllocHGlobal(sizeof(float) * BufferSize)
};
while (samples > 0)
{
int numSamples = (int)(Math.Min(BufferSize, samples));
// get samples from the mixer
Render((uint)numSamples, buffers, samplesRead);
// conver to the file's format
converter.ConvertComplexBuffer(numSamples, buffers, convBuffers);
// write samples to the file
var error = file.Write((uint)numSamples, convBuffers);
if (error != ExtAudioFileError.OK)
{
throw new ApplicationException();
}
samples -= BufferSize;
samplesRead += numSamples;
}
buffers.Dispose();
convBuffers.Dispose();
converter.Dispose();
file.Dispose();
}
