private static ssamp crossfade(ssamp lhs, ssamp rhs, int cur, int start, int end)
{
if (cur <= start)
{
return(lhs);
}
if (cur >= end)
{
return(rhs);
}
// in case we want sine wave interpolation instead of linear here
//float ang = 3.14159f * (float)(cur - start) / (float)(end - start);
//cur = start + (int)((1-cosf(ang))*0.5f * (end - start));
int inNum = cur - start;
int outNum = end - cur;
int denom = end - start;
int lrv = ((int)lhs.l * outNum + (int)rhs.l * inNum) / denom;
int rrv = ((int)lhs.r * outNum + (int)rhs.r * inNum) / denom;
return(new ssamp((short)lrv, (short)rrv));
}
public int OutputSamples(short[] buf, int samples_requested)
{
Console.WriteLine("{0} {1}", samples_requested, sampleQueue.Count); //add this line
int bufcursor = 0;
int audiosize = samples_requested;
int queued = sampleQueue.Count;
// I am too lazy to deal with odd numbers
audiosize &= ~1;
queued &= ~1;
if (queued > 0x200 && audiosize > 0) // is there any work to do?
{
// are we going at normal speed?
// or more precisely, are the input and output queues/buffers of similar size?
if (queued > 900 || audiosize > queued * 2)
{
// not normal speed. we have to resample it somehow in this case.
if (audiosize <= queued)
{
// fast forward speed
// this is the easy case, just crossfade it and it sounds ok
for (int i = 0; i < audiosize; i++)
{
int j = i + queued - audiosize;
ssamp outsamp = crossfade(sampleQueue[i], sampleQueue[j], i, 0, audiosize);
emit_sample(buf, ref bufcursor, outsamp);
}
}
else
{
// slow motion speed
// here we take a very different approach,
// instead of crossfading it, we select a single sample from the queue
// and make sure that the index we use to select a sample is constantly moving
// and that it starts at the first sample in the queue and ends on the last one.
//
// hopefully the index doesn't move discontinuously or we'll get slight crackling
// (there might still be a minor bug here that causes this occasionally)
//
// here's a diagram of how the index we sample from moves:
//
// queued (this axis represents the index we sample from. the top means the end of the queue)
// ^
// | --> audiosize (this axis represents the output index we write to, right meaning forward in output time/position)
// | A C C end
// A A B C C C
// A A A B C C C
// A A A B C C
// A A C
// start
//
// yes, this means we are spending some stretches of time playing the sound backwards,
// but the stretches are short enough that this doesn't sound weird.
// this lets us avoid most crackling problems due to the endpoints matching up.
// first calculate a shorter-than-full window
// that has minimal slope at the endpoints
// (to further reduce crackling, especially in sine waves)
int beststart = 0, extraAtEnd = 0;
{
int bestend = queued;
const int worstdiff = 99999999;
int beststartdiff = worstdiff;
int bestenddiff = worstdiff;
for (int i = 0; i < 128; i += 2)
{
int diff = abs(sampleQueue[i].l - sampleQueue[i + 1].l) + abs(sampleQueue[i].r - sampleQueue[i + 1].r);
if (diff < beststartdiff)
{
beststartdiff = diff;
beststart = i;
}
}
for (int i = queued - 3; i > queued - 3 - 128; i -= 2)
{
int diff = abs(sampleQueue[i].l - sampleQueue[i + 1].l) + abs(sampleQueue[i].r - sampleQueue[i + 1].r);
if (diff < bestenddiff)
{
bestenddiff = diff;
bestend = i + 1;
}
}
extraAtEnd = queued - bestend;
queued = bestend - beststart;
int oksize = queued;
while (oksize + queued * 2 + beststart + extraAtEnd <= samples_requested)
{
oksize += queued * 2;
}
audiosize = oksize;
for (int x = 0; x < beststart; x++)
{
emit_sample(buf, ref bufcursor, sampleQueue[x]);
}
// sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin() + beststart);
sampleQueue.RemoveRange(0, beststart);
// zero 08-nov-2010: did i do this right?
}
int midpointX = audiosize >> 1;
int midpointY = queued >> 1;
// all we need to do here is calculate the X position of the leftmost "B" in the above diagram.
// TODO: we should calculate it with a simple equation like
// midpointXOffset = min(something,somethingElse);
// but it's a little difficult to work it out exactly
// so here's a stupid search for the value for now:
int prevA = 999999;
int midpointXOffset = queued / 2;
while (true)
{
int a = abs(pingpong(midpointX - midpointXOffset, queued) - midpointY) - midpointXOffset;
if (((a > 0) != (prevA > 0) || (a < 0) != (prevA < 0)) && prevA != 999999)
{
if (((a + prevA) & 1) != 0) // there's some sort of off-by-one problem with this search since we're moving diagonally...
{
midpointXOffset++; // but this fixes it most of the time...
}
break; // found it
}
prevA = a;
midpointXOffset--;
if (midpointXOffset < 0)
{
midpointXOffset = 0;
break; // failed to find it. the two sides probably meet exactly in the center.
}
}
int leftMidpointX = midpointX - midpointXOffset;
int rightMidpointX = midpointX + midpointXOffset;
int leftMidpointY = pingpong(leftMidpointX, queued);
int rightMidpointY = (queued - 1) - pingpong((int)audiosize - 1 - rightMidpointX + queued * 2, queued);
// output the left almost-half of the sound (section "A")
for (int x = 0; x < leftMidpointX; x++)
{
int i = pingpong(x, queued);
emit_sample(buf, ref bufcursor, sampleQueue[i]);
}
// output the middle stretch (section "B")
int y = leftMidpointY;
int dyMidLeft = (leftMidpointY < midpointY) ? 1 : -1;
int dyMidRight = (rightMidpointY > midpointY) ? 1 : -1;
for (int x = leftMidpointX; x < midpointX; x++, y += dyMidLeft)
{
emit_sample(buf, ref bufcursor, sampleQueue[y]);
}
for (int x = midpointX; x < rightMidpointX; x++, y += dyMidRight)
{
emit_sample(buf, ref bufcursor, sampleQueue[y]);
}
// output the end of the queued sound (section "C")
for (int x = rightMidpointX; x < audiosize; x++)
{
int i = (queued - 1) - pingpong((int)audiosize - 1 - x + queued * 2, queued);
emit_sample(buf, ref bufcursor, sampleQueue[i]);
}
for (int x = 0; x < extraAtEnd; x++)
{
int i = queued + x;
emit_sample(buf, ref bufcursor, sampleQueue[i]);
}
queued += extraAtEnd;
audiosize += beststart + extraAtEnd;
} // end else
// sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin() + queued);
sampleQueue.RemoveRange(0, queued);
// zero 08-nov-2010: did i do this right?
return(audiosize);
}
else
{
// normal speed
// just output the samples straightforwardly.
//
// at almost-full speeds (like 50/60 FPS)
// what will happen is that we rapidly fluctuate between entering this branch
// and entering the "slow motion speed" branch above.
// but that's ok! because all of these branches sound similar enough that we can get away with it.
// so the two cases actually complement each other.
if (audiosize >= queued)
{
emit_samples(buf, ref bufcursor, sampleQueue.ToArray(), 0, queued);
// sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin() + queued);
sampleQueue.RemoveRange(0, queued);
// zero 08-nov-2010: did i do this right?
return(queued);
}
else
{
emit_samples(buf, ref bufcursor, sampleQueue.ToArray(), 0, audiosize);
// sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin()+audiosize);
sampleQueue.RemoveRange(0, audiosize);
// zero 08-nov-2010: did i do this right?
return(audiosize);
}
} // end normal speed
} // end if there is any work to do
else
{
return(0);
}
} // output_samples
private static void emit_sample(short[] outbuf, ref int cursor, ssamp sample)
{
outbuf[cursor++] = sample.l;
outbuf[cursor++] = sample.r;
}
static void emit_samples(short[] outbuf, ref int outcursor, ssamp[] samplebuf, int incursor, int samples)
{
for (int i = 0; i < samples; i++)
emit_sample(outbuf, ref outcursor, samplebuf[i + incursor]);
}
static void emit_sample(short[] outbuf, ref int cursor, ssamp sample)
{
outbuf[cursor++] = sample.l;
outbuf[cursor++] = sample.r;
}
static ssamp crossfade(ssamp lhs, ssamp rhs, int cur, int start, int end)
{
if (cur <= start)
return lhs;
if (cur >= end)
return rhs;
// in case we want sine wave interpolation instead of linear here
//float ang = 3.14159f * (float)(cur - start) / (float)(end - start);
//cur = start + (int)((1-cosf(ang))*0.5f * (end - start));
int inNum = cur - start;
int outNum = end - cur;
int denom = end - start;
int lrv = ((int)lhs.l * outNum + (int)rhs.l * inNum) / denom;
int rrv = ((int)lhs.r * outNum + (int)rhs.r * inNum) / denom;
return new ssamp((short)lrv, (short)rrv);
}
