/// <summary>
/// Returns the fully mixed audio of the Flipnote, Including its Sound Effects.
/// Returns Null if no audio exists.
/// </summary>
/// <param name="flip"></param>
/// <param name="sampleRate"></param>
/// <returns>Signed 16-bit PCM audio</returns>
public byte[] GetWavBGM(PPMFile flip, int sampleRate = 32768)
{
// start decoding
AdpcmDecoder encoder = new AdpcmDecoder(flip);
var decoded = encoder.getAudioMasterPcm(sampleRate);
if (decoded.Length > 0)
{
byte[] output = new byte[decoded.Length];
// thank you https://github.com/meemo
for (int i = 0; i < decoded.Length; i += 2)
{
try
{
output[i] = (byte)(decoded[i + 1] & 0xff);
output[i + 1] = (byte)(decoded[i] >> 8);
}
catch (Exception)
{
}
}
var a = new WavePcmFormat(output, 1, (uint)(sampleRate / 2), 16);
return(a.ToBytesArray());
}
return(null);
}
private void UpdateBuffer(AMemory Memory)
{
//TODO: Implement conversion for formats other
//than interleaved stereo (2 channels).
//As of now, it assumes that HostChannelsCount == 2.
WaveBuffer Wb = WaveBuffers[BufferIndex];
if (SampleFormat == SampleFormat.PcmInt16)
{
int SamplesCount = (int)(Wb.Size / (sizeof(short) * ChannelsCount));
Samples = new int[SamplesCount * AudioConsts.HostChannelsCount];
if (ChannelsCount == 1)
{
for (int Index = 0; Index < SamplesCount; Index++)
{
short Sample = Memory.ReadInt16(Wb.Position + Index * 2);
Samples[Index * 2 + 0] = Sample;
Samples[Index * 2 + 1] = Sample;
}
}
else
{
for (int Index = 0; Index < SamplesCount * 2; Index++)
{
Samples[Index] = Memory.ReadInt16(Wb.Position + Index * 2);
}
}
}
else if (SampleFormat == SampleFormat.Adpcm)
{
byte[] Buffer = Memory.ReadBytes(Wb.Position, Wb.Size);
Samples = AdpcmDecoder.Decode(Buffer, AdpcmCtx);
}
else
{
throw new InvalidOperationException();
}
if (SampleRate != AudioConsts.HostSampleRate)
{
//TODO: We should keep the frames being discarded (see the 4 below)
//on a buffer and include it on the next samples buffer, to allow
//the resampler to do seamless interpolation between wave buffers.
int SamplesCount = Samples.Length / AudioConsts.HostChannelsCount;
SamplesCount = Math.Max(SamplesCount - 4, 0);
Samples = Resampler.Resample2Ch(
Samples,
SampleRate,
AudioConsts.HostSampleRate,
SamplesCount,
ref ResamplerFracPart);
}
}
private void UpdateBuffer(MemoryManager memory)
{
// TODO: Implement conversion for formats other
// than interleaved stereo (2 channels).
// As of now, it assumes that HostChannelsCount == 2.
WaveBuffer wb = WaveBuffers[_bufferIndex];
if (wb.Position == 0)
{
_samples = new int[0];
return;
}
if (SampleFormat == SampleFormat.PcmInt16)
{
int samplesCount = (int)(wb.Size / (sizeof(short) * ChannelsCount));
_samples = new int[samplesCount * AudioConsts.HostChannelsCount];
if (ChannelsCount == 1)
{
for (int index = 0; index < samplesCount; index++)
{
short sample = memory.ReadInt16(wb.Position + index * 2);
_samples[index * 2 + 0] = sample;
_samples[index * 2 + 1] = sample;
}
}
else
{
for (int index = 0; index < samplesCount * 2; index++)
{
_samples[index] = memory.ReadInt16(wb.Position + index * 2);
}
}
}
else if (SampleFormat == SampleFormat.Adpcm)
{
byte[] buffer = memory.ReadBytes(wb.Position, wb.Size);
_samples = AdpcmDecoder.Decode(buffer, AdpcmCtx);
}
else
{
throw new InvalidOperationException();
}
if (SampleRate != AudioConsts.HostSampleRate)
{
// TODO: We should keep the frames being discarded (see the 4 below)
// on a buffer and include it on the next samples buffer, to allow
// the resampler to do seamless interpolation between wave buffers.
int samplesCount = _samples.Length / AudioConsts.HostChannelsCount;
samplesCount = Math.Max(samplesCount - 4, 0);
_samples = Resampler.Resample2Ch(
_samples,
SampleRate,
AudioConsts.HostSampleRate,
samplesCount,
ref _resamplerFracPart);
}
}
