/// <summary>
/// Initializes a new instance of the <see cref="WaveFileStreamReader"/> class.
/// </summary>
/// <param name="name">Name of the WAVE file.</param>
/// <param name="path">Path of the WAVE file.</param>
/// <param name="startTime">Starting time for streams of data..</param>
/// <param name="audioBufferSizeMs">The size of each data buffer in milliseconds.</param>
internal WaveFileStreamReader(string name, string path, DateTime startTime, int audioBufferSizeMs = DefaultAudioBufferSizeMs)
{
this.Name = name;
this.Path = path;
this.startTime = startTime;
var file = System.IO.Path.Combine(path, name);
this.Size = file.Length;
this.waveFileReader = new BinaryReader(new FileStream(file, FileMode.Open, FileAccess.Read, FileShare.Read));
this.waveFormat = WaveFileHelper.ReadWaveFileHeader(this.waveFileReader);
this.dataLength = WaveFileHelper.ReadWaveDataLength(this.waveFileReader);
this.dataStart = this.waveFileReader.BaseStream.Position;
var bufferSize = (int)(this.waveFormat.AvgBytesPerSec * audioBufferSizeMs / 1000);
this.buffer = new byte[bufferSize];
// Compute originating times based on audio chunk start time + duration
var endTime = this.startTime.AddSeconds((double)this.dataLength / (double)this.waveFormat.AvgBytesPerSec);
this.MessageOriginatingTimeInterval = this.MessageCreationTimeInterval = this.StreamTimeInterval = new TimeInterval(this.startTime, endTime);
var messageCount = (long)Math.Ceiling((double)this.dataLength / bufferSize);
this.audioStreamMetadata = new WaveAudioStreamMetadata(AudioStreamName, typeof(AudioBuffer).AssemblyQualifiedName, name, path, this.startTime, endTime, messageCount, (double)this.dataLength / messageCount, audioBufferSizeMs);
}
private static IEnumerator <ValueTuple <AudioBuffer, DateTime> > EnumerateWaveFile(
Pipeline pipeline,
string filename,
DateTime?audioStartTime = null,
int targetLatencyMs = 20)
{
using (FileStream stream = new FileStream(filename, FileMode.Open, FileAccess.Read, FileShare.Read))
{
BinaryReader br = new BinaryReader(stream);
WaveFormat format = WaveFileHelper.ReadWaveFileHeader(br);
// Compute originating times based on audio chunk start time + duration
var startTime = audioStartTime ?? pipeline.GetCurrentTime();
// Buffer size based on target latency
int bufferSize = (int)(format.AvgBytesPerSec * targetLatencyMs / 1000);
// Get total length in bytes of audio data
long bytesRemaining = (int)WaveFileHelper.ReadWaveDataLength(br);
byte[] buffer = null;
while (bytesRemaining > 0)
{
int nextBytesToRead = (int)Math.Min(bufferSize, bytesRemaining);
// Re-allocate buffer if necessary
if ((buffer == null) || (buffer.Length != nextBytesToRead))
{
buffer = new byte[nextBytesToRead];
}
// Read next audio chunk
int bytesRead = br.Read(buffer, 0, (int)nextBytesToRead);
if (bytesRead == 0)
{
// Break on end of file
break;
}
// Bytes remaining
bytesRemaining -= bytesRead;
// Truncate buffer if necessary
if (bytesRead < nextBytesToRead)
{
byte[] buffer2 = new byte[bytesRead];
Array.Copy(buffer, 0, buffer2, 0, bytesRead);
buffer = buffer2;
}
// Add duration to get originating time
DateTime originatingTime = startTime.AddSeconds((double)bytesRead / (double)format.AvgBytesPerSec);
// Update for next audio chunk
startTime = originatingTime;
yield return(ValueTuple.Create(new AudioBuffer(buffer, format), originatingTime));
}
}
}
/// <summary>
/// Initializes a new instance of the <see cref="WaveStreamSampleSource"/> class.
/// </summary>
/// <param name="pipeline">The pipeline to add the component to.</param>
/// <param name="stream">Audio stream in WAVE format (48KHz, 1-channel, IEEE Float).</param>
/// <param name="name">An optional name for this component.</param>
public WaveStreamSampleSource(Pipeline pipeline, Stream stream, string name = nameof(WaveStreamSampleSource))
{
this.pipeline = pipeline;
this.name = name;
this.In = pipeline.CreateReceiver <bool>(this, this.Play, nameof(this.In));
this.Out = pipeline.CreateEmitter <AudioBuffer>(this, nameof(this.Out));
var reader = new BinaryReader(stream);
var inputFormat = WaveFileHelper.ReadWaveFileHeader(reader);
// we don't do resampling or conversion (must be 1-channel, 48kHz, float32).
// convert offline if needed: ffmpeg -i foo.wav -f wav -acodec pcm_f32le -ar 48000 -ac 1 bar.wav
if (inputFormat.Channels != 1 ||
inputFormat.SamplesPerSec != 48000 ||
(inputFormat.FormatTag != WaveFormatTag.WAVE_FORMAT_IEEE_FLOAT &&
inputFormat.FormatTag != WaveFormatTag.WAVE_FORMAT_EXTENSIBLE) ||
inputFormat.BitsPerSample != 32)
{
throw new ArgumentException("Expected 1-channel, 48kHz, float32 audio format.");
}
// break into 1 second audio buffers
var outputFormat = WaveFormat.CreateIeeeFloat(48000, 1);
var dataLength = WaveFileHelper.ReadWaveDataLength(reader);
// stepping over this line computing frames (e.g. F10) in the debugger will throw - still trying to understand why
var frames = (int)Math.Ceiling((double)dataLength / (double)outputFormat.AvgBytesPerSec);
this.audioData = new AudioBuffer[frames];
for (var i = 0; dataLength > 0; i++)
{
var count = (int)Math.Min(dataLength, outputFormat.AvgBytesPerSec);
var bytes = reader.ReadBytes(count);
this.audioData[i] = new AudioBuffer(bytes, outputFormat);
dataLength -= count;
}
}
