| public GetSampleSizeInBytes ( System.TimeSpan duration ) : int | ||
| duration | System.TimeSpan | |
| return | int | |
public void PlaySound(Tone tone, TimeSpan duration)
{
Deployment.Current.Dispatcher.BeginInvoke(() =>
{
if (_timer == null)
{
_timer = new DispatcherTimer
{
Interval = TimeSpan.FromMilliseconds(33)
};
_timer.Tick += delegate { try { FrameworkDispatcher.Update(); } catch { } };
}
if (_timer.IsEnabled) _timer.Stop();
_timeLeft = duration;
FrameworkDispatcher.Update();
_frequency = tone;
_dynamicSound = new DynamicSoundEffectInstance(SampleRate, AudioChannels.Mono);
_dynamicSound.BufferNeeded += dynamicSound_BufferNeeded;
_dynamicSound.Play();
_bufferSize = _dynamicSound.GetSampleSizeInBytes(TimeSpan.FromSeconds(1));
_soundBuffer = new byte[_bufferSize];
_timer.Start();
});
}
public OggSong(string oggFile)
{
reader = new VorbisReader(oggFile);
effect = new DynamicSoundEffectInstance(reader.SampleRate, (AudioChannels)reader.Channels);
buffer = new byte[effect.GetSampleSizeInBytes(TimeSpan.FromMilliseconds(500))];
nvBuffer = new float[buffer.Length / 2];
// when a buffer is needed, set our handle so the helper thread will read in more data
effect.BufferNeeded += (s, e) => readNextBuffer();
}
public SoundReversibleInstance(SoundReversible sound, byte[] audioBytes, int sampleRate, AudioChannels channels, bool inReverse)
{
this.sound = sound;
this.sampleRate = sampleRate;
this.channels = channels;
reversed = inReverse;
baseAudioBytes = audioBytes;
dynamicSound = NewDynamicSoundEffectInstance();
count = dynamicSound.GetSampleSizeInBytes(TimeSpan.FromMilliseconds(BUFFER_CHUNK_SIZE));
}
/// <summary>
/// Constructor.
/// </summary>
public XnaAudio()
{
// Event handler for getting audio data when the buffer is full
microphone.BufferDuration = TimeSpan.FromMilliseconds(100);
microphone.BufferReady += new EventHandler<EventArgs>(microphone_BufferReady);
// initialize dynamic sound effect instance
playback = new DynamicSoundEffectInstance(microphone.SampleRate, AudioChannels.Mono);
playback.BufferNeeded += GetSamples;
sampleSize = playback.GetSampleSizeInBytes(TimeSpan.FromMilliseconds(100));
}
private static void PlayThings(Context ctx, Node srcNode)
{
FrameworkDispatcher.Update();
using (var aud = new DynamicSoundEffectInstance(48000, AudioChannels.Mono))
{
SampleSize = aud.GetSampleSizeInBytes(new TimeSpan(0, 0, 0, 0, SampleTimeMs));
var sub = ctx.Socket(SocketType.SUB);
sub.Subscribe(new byte[0]);
sub.Connect(srcNode.Url);
mBar.SignalAndWait();
bool bufferNeeded = true;
aud.BufferNeeded += (_, __) => bufferNeeded = true;
while (isRunning)
{
if (bufferNeeded)
{
foreach (var n in mNodes.Where(n => n is GeneratorNode).Cast<GeneratorNode>())
{
n.SignalForData();
}
bufferNeeded = false;
}
var bytes = sub.Recv(SampleTimeMs / 2);
if (bytes != null)
{
aud.SubmitBuffer(bytes);
if (aud.State == SoundState.Stopped)
aud.Play();
}
FrameworkDispatcher.Update();
//running = false;
}
aud.Stop();
sub.Dispose();
}
}
/// <summary>
/// インスタンスを生成します。
/// StreamingWave が存在する間、ストリーミングのために input はオープン状態が継続され、
/// Dispose メソッドでその Dispose メソッドが呼び出されます。
/// </summary>
/// <param name="input">Wave ファイルの Stream。</param>
/// <param name="bufferDuration">バッファリングする再生時間。</param>
public StreamingWave(Stream input, TimeSpan bufferDuration)
{
if (input == null) throw new ArgumentNullException("input");
this.input = input;
reader = new BinaryReader(input);
// 'data' chunk のデータ部の直前まで読み込みます。
riffChunk = RiffChunk.ReadFrom(reader);
formatChunk = WaveFormatChunk.ReadFrom(reader);
dataChunkHeader = ChunkHeader.ReadFrom(reader);
// 'data' chunk のデータ部の開始位置を記憶します。
dataOffset = input.Position;
int sampleRate = (int) formatChunk.SampleRate;
AudioChannels channels = (AudioChannels) formatChunk.Channels;
dynamicSound = new DynamicSoundEffectInstance(sampleRate, channels);
dynamicSound.BufferNeeded += new EventHandler<EventArgs>(OnDynamicSoundBufferNeeded);
bufferSize = dynamicSound.GetSampleSizeInBytes(bufferDuration);
buffer = new byte[bufferSize];
readDataAsyncCaller = new ReadDataAsyncCaller(ReadData);
}
