/* blocksize 0 is guaranteed to be short, 1 is guaranteed to be long. They may be equal, but short will never ge greater than long */
static public int vorbis_info_blocksize(ref vorbis_info vi, int zo)
{
codec_setup_info ci = vi.codec_setup as codec_setup_info;
return((ci != null) ? ci.blocksizes[zo] : -1);
}
static int mapping0_inverse(ref vorbis_block vb, vorbis_info_mapping l)
{
vorbis_dsp_state vd = vb.vd;
vorbis_info vi = vd.vi;
codec_setup_info ci = vi.codec_setup as codec_setup_info;
private_state b = vd.backend_state as private_state;
vorbis_info_mapping0 info = l as vorbis_info_mapping0;
int i, j;
int n = vb.pcmend = ci.blocksizes[vb.W];
float **pcmbundle = stackalloc float *[vi.channels];
int * zerobundle = stackalloc int[vi.channels];
int * nonzero = stackalloc int[vi.channels];
void **floormemo = stackalloc void *[vi.channels];
/* recover the spectral envelope; store it in the PCM vector for now */
for (i = 0; i < vi.channels; i++)
{
int submap = info.chmuxlist[i];
floormemo[i] = _floor_P[ci.floor_type[info.floorsubmap[submap]]].inverse1(ref vb, b.flr[info.floorsubmap[submap]]);
if (floormemo[i] != null)
{
nonzero[i] = 1;
}
else
{
nonzero[i] = 0;
}
ZeroMemory(vb.pcm[i], sizeof(float) * n / 2);
}
/* channel coupling can 'dirty' the nonzero listing */
for (i = 0; i < info.coupling_steps; i++)
{
if (nonzero[info.coupling_mag[i]] != 0 || nonzero[info.coupling_ang[i]] != 0)
{
nonzero[info.coupling_mag[i]] = 1;
nonzero[info.coupling_ang[i]] = 1;
}
}
/* recover the residue into our working vectors */
for (i = 0; i < info.submaps; i++)
{
int ch_in_bundle = 0;
for (j = 0; j < vi.channels; j++)
{
if (info.chmuxlist[j] == i)
{
if (nonzero[j] != 0)
{
zerobundle[ch_in_bundle] = 1;
}
else
{
zerobundle[ch_in_bundle] = 0;
}
pcmbundle[ch_in_bundle++] = vb.pcm[j];
}
}
_residue_P[ci.residue_type[info.residuesubmap[i]]].inverse(ref vb, b.residue[info.residuesubmap[i]], pcmbundle, zerobundle, ch_in_bundle);
}
/* channel coupling */
for (i = info.coupling_steps - 1; i >= 0; i--)
{
float *pcmM = vb.pcm[info.coupling_mag[i]];
float *pcmA = vb.pcm[info.coupling_ang[i]];
for (j = 0; j < n / 2; j++)
{
float mag = pcmM[j];
float ang = pcmA[j];
if (mag > 0)
{
if (ang > 0)
{
pcmM[j] = mag;
pcmA[j] = mag - ang;
}
else
{
pcmA[j] = mag;
pcmM[j] = mag + ang;
}
}
else
{
if (ang > 0)
{
pcmM[j] = mag;
pcmA[j] = mag + ang;
}
else
{
pcmA[j] = mag;
pcmM[j] = mag - ang;
}
}
}
}
/* compute and apply spectral envelope */
for (i = 0; i < vi.channels; i++)
{
float *pcm = vb.pcm[i];
int submap = info.chmuxlist[i];
_floor_P[ci.floor_type[info.floorsubmap[submap]]].inverse2(ref vb, b.flr[info.floorsubmap[submap]], floormemo[i], pcm);
}
/* transform the PCM data; takes PCM vector, vb; modifies PCM vector */
/* only MDCT right now.... */
for (i = 0; i < vi.channels; i++)
{
float *pcm = vb.pcm[i];
mdct_backward(b.transform[vb.W][0] as mdct_lookup, pcm, pcm);
}
/* all done! */
return(0);
}
/* also responsible for range checking */
static vorbis_info_mapping mapping0_unpack(ref vorbis_info vi, ref Ogg.oggpack_buffer opb)
{
int i, b;
vorbis_info_mapping _info = new vorbis_info_mapping0();
vorbis_info_mapping0 info = _info as vorbis_info_mapping0;
codec_setup_info ci = vi.codec_setup as codec_setup_info;
b = Ogg.oggpack_read(ref opb, 1);
if (b < 0)
{
goto err_out;
}
if (b != 0)
{
info.submaps = Ogg.oggpack_read(ref opb, 4) + 1;
if (info.submaps <= 0)
{
goto err_out;
}
}
else
{
info.submaps = 1;
}
b = Ogg.oggpack_read(ref opb, 1);
if (b < 0)
{
goto err_out;
}
if (b != 0)
{
info.coupling_steps = Ogg.oggpack_read(ref opb, 8) + 1;
if (info.coupling_steps <= 0)
{
goto err_out;
}
for (i = 0; i < info.coupling_steps; i++)
{
int testM = info.coupling_mag[i] = Ogg.oggpack_read(ref opb, ilog((uint)vi.channels));
int testA = info.coupling_ang[i] = Ogg.oggpack_read(ref opb, ilog((uint)vi.channels));
if (testM < 0 || testA < 0 || testM == testA || testM >= vi.channels || testA >= vi.channels)
{
goto err_out;
}
}
}
if (Ogg.oggpack_read(ref opb, 2) != 0) /* 2,3:reserved */
{
goto err_out;
}
if (info.submaps > 1)
{
for (i = 0; i < vi.channels; i++)
{
info.chmuxlist[i] = Ogg.oggpack_read(ref opb, 4);
if (info.chmuxlist[i] >= info.submaps || info.chmuxlist[i] < 0)
{
goto err_out;
}
}
}
for (i = 0; i < info.submaps; i++)
{
Ogg.oggpack_read(ref opb, 8); /* time submap unused */
info.floorsubmap[i] = Ogg.oggpack_read(ref opb, 8);
if (info.floorsubmap[i] >= ci.floors || info.floorsubmap[i] < 0)
{
goto err_out;
}
info.residuesubmap[i] = Ogg.oggpack_read(ref opb, 8);
if (info.residuesubmap[i] >= ci.residues || info.residuesubmap[i] < 0)
{
goto err_out;
}
}
return(info);
err_out:
mapping0_free_info(ref _info);
return(null);
}
static int mapping0_forward(ref vorbis_block vb)
{
vorbis_dsp_state vd = vb.vd;
vorbis_info vi = vd.vi;
codec_setup_info ci = vi.codec_setup as codec_setup_info;
private_state b = vb.vd.backend_state as private_state;
vorbis_block_internal vbi = vb._internal as vorbis_block_internal;
int n = vb.pcmend;
int i, j, k;
int * nonzero = stackalloc int[vi.channels];
float **gmdct = (float **)_vorbis_block_alloc(ref vb, vi.channels * sizeof(float *));
int ** iwork = (int **)_vorbis_block_alloc(ref vb, vi.channels * sizeof(int *));
int *** floor_posts = (int ***)_vorbis_block_alloc(ref vb, vi.channels * sizeof(int **));
float global_ampmax = vbi.ampmax;
float *local_ampmax = stackalloc float[vi.channels];
int blocktype = vbi.blocktype;
int modenumber = vb.W;
vorbis_info_mapping0 info = ci.map_param[modenumber] as vorbis_info_mapping0;
vorbis_look_psy psy_look = b.psy[blocktype + (vb.W != 0 ? 2 : 0)];
vb.mode = modenumber;
for (i = 0; i < vi.channels; i++)
{
float scale = 4.0f / n;
float scale_dB;
float *pcm = vb.pcm[i];
float *logfft = pcm;
iwork[i] = (int *)_vorbis_block_alloc(ref vb, (n / 2) * sizeof(int));
gmdct[i] = (float *)_vorbis_block_alloc(ref vb, (n / 2) * sizeof(float));
/* + .345 is a hack; the original todB estimation used on IEEE 754 compliant machines had a bug that
* returned dB values about a third of a decibel too high. The bug was harmless because tunings
* implicitly took that into account. However, fixing the bug in the estimator requires changing all the tunings as well.
* For now, it's easier to sync things back up here, and recalibrate the tunings in the next major model upgrade. */
scale_dB = todB(scale) + 0.345f;
/* window the PCM data */
_vorbis_apply_window(pcm, ref b.window, ref ci.blocksizes, vb.lW, vb.W, vb.nW);
/* transform the PCM data */
/* only MDCT right now.... */
mdct_forward(b.transform[vb.W][0] as mdct_lookup, pcm, gmdct[i]);
/* FFT yields more accurate tonal estimation (not phase sensitive) */
drft_forward(ref b.fft_look[vb.W], pcm);
/* + .345 is a hack; the original todB estimation used on IEEE 754 compliant machines had a bug that
* returned dB values about a third of a decibel too high. The bug was harmless because tunings
* implicitly took that into account. However, fixing the bug in the estimator requires changing all the tunings as well.
* For now, it's easier to sync things back up here, and recalibrate the tunings in the next major model upgrade. */
logfft[0] = scale_dB + todB(*pcm) + 0.345f;
local_ampmax[i] = logfft[0];
for (j = 1; j < n - 1; j += 2)
{
float temp = pcm[j] * pcm[j] + pcm[j + 1] * pcm[j + 1];
/* + .345 is a hack; the original todB estimation used on IEEE 754 compliant machines had a bug that
* returned dB values about a third of a decibel too high. The bug was harmless because tunings
* implicitly took that into account. However, fixing the bug in the estimator requires changing all the tunings as well.
* For now, it's easier to sync things back up here, and recalibrate the tunings in the next major model upgrade. */
temp = logfft[(j + 1) >> 1] = scale_dB + 0.5f * todB(temp) + 0.345f;
if (temp > local_ampmax[i])
{
local_ampmax[i] = temp;
}
}
if (local_ampmax[i] > 0.0f)
{
local_ampmax[i] = 0.0f;
}
if (local_ampmax[i] > global_ampmax)
{
global_ampmax = local_ampmax[i];
}
}
{
float *noise = (float *)_vorbis_block_alloc(ref vb, n / 2 * sizeof(float));
float *tone = (float *)_vorbis_block_alloc(ref vb, n / 2 * sizeof(float));
for (i = 0; i < vi.channels; i++)
{
/* the encoder setup assumes that all the modes used by any
* specific bitrate tweaking use the same floor */
int submap = info.chmuxlist[i];
/* the following makes things clearer to *me* anyway */
float *mdct = gmdct[i];
float *logfft = vb.pcm[i];
float *logmdct = logfft + n / 2;
float *logmask = logfft;
vb.mode = modenumber;
floor_posts[i] = (int **)_vorbis_block_alloc(ref vb, PACKETBLOBS * sizeof(int *));
ZeroMemory(floor_posts[i], sizeof(int *) * PACKETBLOBS);
for (j = 0; j < n / 2; j++)
{
/* + .345 is a hack; the original todB estimation used on IEEE 754 compliant machines had a bug that
* returned dB values about a third of a decibel too high. The bug was harmless because tunings
* implicitly took that into account. However, fixing the bug in the estimator requires changing all the tunings as well.
* For now, it's easier to sync things back up here, and recalibrate the tunings in the next major model upgrade. */
logmdct[j] = todB(mdct[j]) + 0.345f;
/* first step; noise masking. Not only does 'noise masking' give us curves from which we can decide how much resolution
* to give noise parts of the spectrum, it also implicitly hands us a tonality estimate (the larger the value in the
* 'noise_depth' vector, the more tonal that area is) */
_vp_noisemask(ref psy_look, logmdct, noise); /* noise does not have by-frequency offset bias applied yet */
/* second step: 'all the other crap'; all the stuff that isn't computed/fit for bitrate management goes in the second psy
* vector. This includes tone masking, peak limiting and ATH */
_vp_tonemask(ref psy_look, logfft, tone, global_ampmax, local_ampmax[i]);
/* third step; we offset the noise vectors, overlay tone masking. We then do a floor1-specific line fit. If we're
* performing bitrate management, the line fit is performed multiple times for up/down tweakage on demand. */
_vp_offset_and_mix(ref psy_look, noise, tone, 1, logmask, mdct, logmdct);
/* this algorithm is hardwired to floor 1 for now; abort out if we're *not* floor1. This won't happen unless someone has
* broken the encode setup lib. Guard it anyway. */
if (ci.floor_type[info.floorsubmap[submap]] != 1)
{
return(-1);
}
floor_posts[i][PACKETBLOBS / 2] = floor1_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, logmdct, logmask);
/* are we managing bitrate? If so, perform two more fits for later rate tweaking (fits represent hi/lo) */
if (vorbis_bitrate_managed(ref vb) != 0 && floor_posts[i][PACKETBLOBS / 2] != null)
{
/* higher rate by way of lower noise curve */
_vp_offset_and_mix(ref psy_look, noise, tone, 2, logmask, mdct, logmdct);
floor_posts[i][PACKETBLOBS - 1] = floor1_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, logmdct, logmask);
/* lower rate by way of higher noise curve */
_vp_offset_and_mix(ref psy_look, noise, tone, 0, logmask, mdct, logmdct);
floor_posts[i][0] = floor1_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, logmdct, logmask);
/* we also interpolate a range of intermediate curves for
* intermediate rates */
for (k = 1; k < PACKETBLOBS / 2; k++)
{
floor_posts[i][k] = floor1_interpolate_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, floor_posts[i][0], floor_posts[i][PACKETBLOBS / 2], k * 65536 / (PACKETBLOBS / 2));
}
for (k = PACKETBLOBS / 2 + 1; k < PACKETBLOBS - 1; k++)
{
floor_posts[i][k] = floor1_interpolate_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, floor_posts[i][PACKETBLOBS / 2], floor_posts[i][PACKETBLOBS - 1], (k - PACKETBLOBS / 2) * 65536 / (PACKETBLOBS / 2));
}
}
}
}
vbi.ampmax = global_ampmax;
/*
* the next phases are performed once for vbr-only and PACKETBLOB
* times for bitrate managed modes.
*
* 1) encode actual mode being used
* 2) encode the floor for each channel, compute coded mask curve/res
* 3) normalize and couple.
* 4) encode residue
* 5) save packet bytes to the packetblob vector
*/
/* iterate over the many masking curve fits we've created */
{
int **couple_bundle = stackalloc int *[vi.channels];
int * zerobundle = stackalloc int[vi.channels];
for (k = (vorbis_bitrate_managed(ref vb) != 0 ? 0 : PACKETBLOBS / 2); k <= (vorbis_bitrate_managed(ref vb) != 0 ? PACKETBLOBS - 1 : PACKETBLOBS / 2); k++)
{
Ogg.oggpack_buffer opb = vbi.packetblob[k];
/* start out our new packet blob with packet type and mode */
/* Encode the packet type */
Ogg.oggpack_write(ref opb, 0, 1);
/* Encode the modenumber */
/* Encode frame mode, pre,post windowsize, then dispatch */
Ogg.oggpack_write(ref opb, (uint)modenumber, b.modebits);
if (vb.W != 0)
{
Ogg.oggpack_write(ref opb, (uint)vb.lW, 1);
Ogg.oggpack_write(ref opb, (uint)vb.nW, 1);
}
/* encode floor, compute masking curve, sep out residue */
for (i = 0; i < vi.channels; i++)
{
int submap = info.chmuxlist[i];
int *ilogmask = iwork[i];
nonzero[i] = floor1_encode(ref opb, ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, floor_posts[i][k], ilogmask);
}
/* our iteration is now based on masking curve, not prequant and coupling. Only one prequant/coupling step */
/* quantize/couple */
/* incomplete implementation that assumes the tree is all depth one, or no tree at all */
_vp_couple_quantize_normalize(k, ci.psy_g_param, ref psy_look, info, gmdct, iwork, nonzero, ci.psy_g_param.sliding_lowpass[vb.W, k], vi.channels);
/* classify and encode by submap */
for (i = 0; i < info.submaps; i++)
{
int ch_in_bundle = 0;
int **classifications;
int resnum = info.residuesubmap[i];
for (j = 0; j < vi.channels; j++)
{
if (info.chmuxlist[j] == i)
{
zerobundle[ch_in_bundle] = 0;
if (nonzero[j] != 0)
{
zerobundle[ch_in_bundle] = 1;
}
couple_bundle[ch_in_bundle++] = iwork[j];
}
}
classifications = _residue_P[ci.residue_type[resnum]]._class(ref vb, b.residue[resnum], couple_bundle, zerobundle, ch_in_bundle);
ch_in_bundle = 0;
for (j = 0; j < vi.channels; j++)
{
if (info.chmuxlist[j] == i)
{
couple_bundle[ch_in_bundle++] = iwork[j];
}
}
_residue_P[ci.residue_type[resnum]].forward(ref opb, ref vb, b.residue[resnum], couple_bundle, zerobundle, ch_in_bundle, classifications, i);
}
/* ok, done encoding. Next protopacket. */
}
}
return(0);
}
}