private ElfSymbol GetSymbol(IMemoryManager memory, long address, long strTblAddr)
{
int nameIndex = memory.ReadInt32(address + 0);
int info = memory.ReadByte(address + 4);
int other = memory.ReadByte(address + 5);
int shIdx = memory.ReadInt16(address + 6);
long value = memory.ReadInt64(address + 8);
long size = memory.ReadInt64(address + 16);
string name = string.Empty;
for (int chr; (chr = memory.ReadByte(strTblAddr + nameIndex++)) != 0;)
{
name += (char)chr;
}
return(new ElfSymbol(name, info, other, shIdx, value, size));
}
private AdpcmDecoderContext GetAdpcmDecoderContext(long position, long size)
{
if (size == 0)
{
return(null);
}
AdpcmDecoderContext context = new AdpcmDecoderContext();
context.Coefficients = new short[size >> 1];
for (int offset = 0; offset < size; offset += 2)
{
context.Coefficients[offset >> 1] = _memory.ReadInt16(position + offset);
}
return(context);
}
private void UpdateBuffer(IMemoryManager 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 * AudioRendererConsts.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 != AudioRendererConsts.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 / AudioRendererConsts.HostChannelsCount;
samplesCount = Math.Max(samplesCount - 4, 0);
_samples = Resampler.Resample2Ch(
_samples,
SampleRate,
AudioRendererConsts.HostSampleRate,
samplesCount,
ref _resamplerFracPart);
}
}
