public static int a52_syncinfo(ReadOnlySpan <byte> span, out eFrameFlags flags, out int sampleRate, out int bitRate)
{
unsafe
{
fixed(byte *p = span)
return(a52_syncinfo(p, out flags, out sampleRate, out bitRate));
}
}
void iAudioDecoder.decodeFrame(ReadOnlySpan <byte> data, byte volume)
{
eFrameFlags flags = channelFlags;
liba52.a52_frame(ac3, data, flags, volume);
// Logger.logVerbose( "a52_frame" );
// The decoder makes 6, but since we double the blocks tell player 3 are available.
blocksLeft = 3;
}
public static void a52_frame(IntPtr state, ReadOnlySpan <byte> span, eFrameFlags flags, byte volume)
{
flags |= eFrameFlags.AdjustLevel;
// Set the bias so the complete range maps to [ 383.0f .. 385.0f ] interval.
// These particular magic numbers result in 16-bit samples in the lowest 16 bits of the float mantissa: https://www.h-schmidt.net/FloatConverter/IEEE754.html
// This way we only need a few integer SIMD instructions to convert, very fast.
// Specifically, AMD64 code uses _mm_sub_epi32 and _mm_packs_epi32, NEON version vsubq_s32, vqmovn_s32 and vcombine_s16.
// Both _mm_packs_epi32 and vqmovn_s32 instructions use saturation to clip values into the range of int16_t, exactly what we need for PCM audio.
float level = volume * (1.0f / 255.0f);
float bias = 384;
unsafe
{
fixed(byte *p = span)
if (0 != a52_frame(state, p, ref flags, &level, bias))
{
throw new ApplicationException("liba52.a52_frame failed");
}
}
}
static unsafe extern int a52_frame(IntPtr state, byte *buffer, ref eFrameFlags flags, float *level, float bias);
static unsafe extern int a52_syncinfo(byte *buffer, out eFrameFlags flags, out int sampleRate, out int bitRate);
