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5f47c85e5c
FFmpeg/libavcodec/opus_pvq.c
Rostislav Pehlivanov 5f47c85e5c opus: add a native Opus encoder 
This marks the first time anyone has written an Opus encoder without
using any libopus code. The aim of the encoder is to prove how far
the format can go by writing the craziest encoder for it.

Right now the encoder's basic, it only supports CBR encoding, however
internally every single feature the CELT layer has is implemented
(except the pitch pre-filter which needs to work well with the rest of
whatever gets implemented). Psychoacoustic and rate control systems are
under development.

The encoder takes in frames of 120 samples and depending on the value of
opus_delay the plan is to use the extra buffered frames as lookahead.
Right now the encoder will pick the nearest largest legal frame size and
won't use the lookahead, but that'll change once there's a
psychoacoustic system.

Even though its a pretty basic encoder its already outperforming
any other native encoder FFmpeg has by a huge amount.

The PVQ search algorithm is faster and more accurate than libopus's
algorithm so the encoder's performance is close to that of libopus
at zero complexity (libopus has more SIMD).
The algorithm might be ported to libopus or other codecs using PVQ in
the future.

The encoder still has a few minor bugs, like desyncs at ultra low
bitrates (below 9kbps with 20ms frames).

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2017-02-14 06:15:36 +00:00

1158 lines
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/*
* Copyright (c) 2012 Andrew D'Addesio
* Copyright (c) 2013-2014 Mozilla Corporation
* Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "opustab.h"
#include "opus_pvq.h"
#define CELT_PVQ_U(n, k) (ff_celt_pvq_u_row[FFMIN(n, k)][FFMAX(n, k)])
#define CELT_PVQ_V(n, k) (CELT_PVQ_U(n, k) + CELT_PVQ_U(n, (k) + 1))
static inline int16_t celt_cos(int16_t x)
{
x = (MUL16(x, x) + 4096) >> 13;
x = (32767-x) + ROUND_MUL16(x, (-7651 + ROUND_MUL16(x, (8277 + ROUND_MUL16(-626, x)))));
return 1+x;
}
static inline int celt_log2tan(int isin, int icos)
{
int lc, ls;
lc = opus_ilog(icos);
ls = opus_ilog(isin);
icos <<= 15 - lc;
isin <<= 15 - ls;
return (ls << 11) - (lc << 11) +
ROUND_MUL16(isin, ROUND_MUL16(isin, -2597) + 7932) -
ROUND_MUL16(icos, ROUND_MUL16(icos, -2597) + 7932);
}
static inline int celt_bits2pulses(const uint8_t *cache, int bits)
{
// TODO: Find the size of cache and make it into an array in the parameters list
int i, low = 0, high;
high = cache[0];
bits--;
for (i = 0; i < 6; i++) {
int center = (low + high + 1) >> 1;
if (cache[center] >= bits)
high = center;
else
low = center;
}
return (bits - (low == 0 ? -1 : cache[low]) <= cache[high] - bits) ? low : high;
}
static inline int celt_pulses2bits(const uint8_t *cache, int pulses)
{
// TODO: Find the size of cache and make it into an array in the parameters list
return (pulses == 0) ? 0 : cache[pulses] + 1;
}
static inline void celt_normalize_residual(const int * av_restrict iy, float * av_restrict X,
int N, float g)
{
int i;
for (i = 0; i < N; i++)
X[i] = g * iy[i];
}
static void celt_exp_rotation_impl(float *X, uint32_t len, uint32_t stride,
float c, float s)
{
float *Xptr;
int i;
Xptr = X;
for (i = 0; i < len - stride; i++) {
float x1, x2;
x1 = Xptr[0];
x2 = Xptr[stride];
Xptr[stride] = c * x2 + s * x1;
*Xptr++ = c * x1 - s * x2;
}
Xptr = &X[len - 2 * stride - 1];
for (i = len - 2 * stride - 1; i >= 0; i--) {
float x1, x2;
x1 = Xptr[0];
x2 = Xptr[stride];
Xptr[stride] = c * x2 + s * x1;
*Xptr-- = c * x1 - s * x2;
}
}
static inline void celt_exp_rotation(float *X, uint32_t len,
uint32_t stride, uint32_t K,
enum CeltSpread spread, const int encode)
{
uint32_t stride2 = 0;
float c, s;
float gain, theta;
int i;
if (2*K >= len || spread == CELT_SPREAD_NONE)
return;
gain = (float)len / (len + (20 - 5*spread) * K);
theta = M_PI * gain * gain / 4;
c = cosf(theta);
s = sinf(theta);
if (len >= stride << 3) {
stride2 = 1;
/* This is just a simple (equivalent) way of computing sqrt(len/stride) with rounding.
It's basically incrementing long as (stride2+0.5)^2 < len/stride. */
while ((stride2 * stride2 + stride2) * stride + (stride >> 2) < len)
stride2++;
}
/*NOTE: As a minor optimization, we could be passing around log2(B), not B, for both this and for
extract_collapse_mask().*/
len /= stride;
for (i = 0; i < stride; i++) {
if (encode) {
celt_exp_rotation_impl(X + i * len, len, 1, c, -s);
if (stride2)
celt_exp_rotation_impl(X + i * len, len, stride2, s, -c);
} else {
if (stride2)
celt_exp_rotation_impl(X + i * len, len, stride2, s, c);
celt_exp_rotation_impl(X + i * len, len, 1, c, s);
}
}
}
static inline uint32_t celt_extract_collapse_mask(const int *iy, uint32_t N, uint32_t B)
{
uint32_t collapse_mask;
int N0;
int i, j;
if (B <= 1)
return 1;
/*NOTE: As a minor optimization, we could be passing around log2(B), not B, for both this and for
exp_rotation().*/
N0 = N/B;
collapse_mask = 0;
for (i = 0; i < B; i++)
for (j = 0; j < N0; j++)
collapse_mask |= (iy[i*N0+j]!=0)<<i;
return collapse_mask;
}
static inline void celt_stereo_merge(float *X, float *Y, float mid, int N)
{
int i;
float xp = 0, side = 0;
float E[2];
float mid2;
float t, gain[2];
/* Compute the norm of X+Y and X-Y as |X|^2 + |Y|^2 +/- sum(xy) */
for (i = 0; i < N; i++) {
xp += X[i] * Y[i];
side += Y[i] * Y[i];
}
/* Compensating for the mid normalization */
xp *= mid;
mid2 = mid;
E[0] = mid2 * mid2 + side - 2 * xp;
E[1] = mid2 * mid2 + side + 2 * xp;
if (E[0] < 6e-4f || E[1] < 6e-4f) {
for (i = 0; i < N; i++)
Y[i] = X[i];
return;
}
t = E[0];
gain[0] = 1.0f / sqrtf(t);
t = E[1];
gain[1] = 1.0f / sqrtf(t);
for (i = 0; i < N; i++) {
float value[2];
/* Apply mid scaling (side is already scaled) */
value[0] = mid * X[i];
value[1] = Y[i];
X[i] = gain[0] * (value[0] - value[1]);
Y[i] = gain[1] * (value[0] + value[1]);
}
}
static void celt_interleave_hadamard(float *tmp, float *X, int N0,
int stride, int hadamard)
{
int i, j;
int N = N0*stride;
if (hadamard) {
const uint8_t *ordery = ff_celt_hadamard_ordery + stride - 2;
for (i = 0; i < stride; i++)
for (j = 0; j < N0; j++)
tmp[j*stride+i] = X[ordery[i]*N0+j];
} else {
for (i = 0; i < stride; i++)
for (j = 0; j < N0; j++)
tmp[j*stride+i] = X[i*N0+j];
}
for (i = 0; i < N; i++)
X[i] = tmp[i];
}
static void celt_deinterleave_hadamard(float *tmp, float *X, int N0,
int stride, int hadamard)
{
int i, j;
int N = N0*stride;
if (hadamard) {
const uint8_t *ordery = ff_celt_hadamard_ordery + stride - 2;
for (i = 0; i < stride; i++)
for (j = 0; j < N0; j++)
tmp[ordery[i]*N0+j] = X[j*stride+i];
} else {
for (i = 0; i < stride; i++)
for (j = 0; j < N0; j++)
tmp[i*N0+j] = X[j*stride+i];
}
for (i = 0; i < N; i++)
X[i] = tmp[i];
}
static void celt_haar1(float *X, int N0, int stride)
{
int i, j;
N0 >>= 1;
for (i = 0; i < stride; i++) {
for (j = 0; j < N0; j++) {
float x0 = X[stride * (2 * j + 0) + i];
float x1 = X[stride * (2 * j + 1) + i];
X[stride * (2 * j + 0) + i] = (x0 + x1) * M_SQRT1_2;
X[stride * (2 * j + 1) + i] = (x0 - x1) * M_SQRT1_2;
}
}
}
static inline int celt_compute_qn(int N, int b, int offset, int pulse_cap,
int dualstereo)
{
int qn, qb;
int N2 = 2 * N - 1;
if (dualstereo && N == 2)
N2--;
/* The upper limit ensures that in a stereo split with itheta==16384, we'll
* always have enough bits left over to code at least one pulse in the
* side; otherwise it would collapse, since it doesn't get folded. */
qb = FFMIN3(b - pulse_cap - (4 << 3), (b + N2 * offset) / N2, 8 << 3);
qn = (qb < (1 << 3 >> 1)) ? 1 : ((ff_celt_qn_exp2[qb & 0x7] >> (14 - (qb >> 3))) + 1) >> 1 << 1;
return qn;
}
/* Convert the quantized vector to an index */
static inline uint32_t celt_icwrsi(uint32_t N, const int *y)
{
int i, idx = 0, sum = 0;
for (i = N - 1; i >= 0; i--) {
const uint32_t i_s = CELT_PVQ_U(N - i, sum + FFABS(y[i]) + 1);
idx += CELT_PVQ_U(N - i, sum) + (y[i] < 0)*i_s;
sum += FFABS(y[i]);
}
return idx;
}
// this code was adapted from libopus
static inline uint64_t celt_cwrsi(uint32_t N, uint32_t K, uint32_t i, int *y)
{
uint64_t norm = 0;
uint32_t p;
int s, val;
int k0;
while (N > 2) {
uint32_t q;
/*Lots of pulses case:*/
if (K >= N) {
const uint32_t *row = ff_celt_pvq_u_row[N];
/* Are the pulses in this dimension negative? */
p = row[K + 1];
s = -(i >= p);
i -= p & s;
/*Count how many pulses were placed in this dimension.*/
k0 = K;
q = row[N];
if (q > i) {
K = N;
do {
p = ff_celt_pvq_u_row[--K][N];
} while (p > i);
} else
for (p = row[K]; p > i; p = row[K])
K--;
i -= p;
val = (k0 - K + s) ^ s;
norm += val * val;
*y++ = val;
} else { /*Lots of dimensions case:*/
/*Are there any pulses in this dimension at all?*/
p = ff_celt_pvq_u_row[K ][N];
q = ff_celt_pvq_u_row[K + 1][N];
if (p <= i && i < q) {
i -= p;
*y++ = 0;
} else {
/*Are the pulses in this dimension negative?*/
s = -(i >= q);
i -= q & s;
/*Count how many pulses were placed in this dimension.*/
k0 = K;
do p = ff_celt_pvq_u_row[--K][N];
while (p > i);
i -= p;
val = (k0 - K + s) ^ s;
norm += val * val;
*y++ = val;
}
}
N--;
}
/* N == 2 */
p = 2 * K + 1;
s = -(i >= p);
i -= p & s;
k0 = K;
K = (i + 1) / 2;
if (K)
i -= 2 * K - 1;
val = (k0 - K + s) ^ s;
norm += val * val;
*y++ = val;
/* N==1 */
s = -i;
val = (K + s) ^ s;
norm += val * val;
*y = val;
return norm;
}
static inline void celt_encode_pulses(OpusRangeCoder *rc, int *y, uint32_t N, uint32_t K)
{
ff_opus_rc_enc_uint(rc, celt_icwrsi(N, y), CELT_PVQ_V(N, K));
}
static inline float celt_decode_pulses(OpusRangeCoder *rc, int *y, uint32_t N, uint32_t K)
{
const uint32_t idx = ff_opus_rc_dec_uint(rc, CELT_PVQ_V(N, K));
return celt_cwrsi(N, K, idx, y);
}
/*
* Faster than libopus's search, operates entirely in the signed domain.
* Slightly worse/better depending on N, K and the input vector.
*/
static void celt_pvq_search(float *X, int *y, int K, int N)
{
int i;
float res = 0.0f, y_norm = 0.0f, xy_norm = 0.0f;
for (i = 0; i < N; i++)
res += FFABS(X[i]);
res = K/res;
for (i = 0; i < N; i++) {
y[i] = lrintf(res*X[i]);
y_norm += y[i]*y[i];
xy_norm += y[i]*X[i];
K -= FFABS(y[i]);
}
while (K) {
int max_idx = 0, phase = FFSIGN(K);
float max_den = 1.0f, max_num = 0.0f;
y_norm += 1.0f;
for (i = 0; i < N; i++) {
float xy_new = xy_norm + 1*phase*FFABS(X[i]);
float y_new = y_norm + 2*phase*FFABS(y[i]);
xy_new = xy_new * xy_new;
if ((max_den*xy_new) > (y_new*max_num)) {
max_den = y_new;
max_num = xy_new;
max_idx = i;
}
}
K -= phase;
phase *= FFSIGN(X[max_idx]);
xy_norm += 1*phase*X[max_idx];
y_norm += 2*phase*y[max_idx];
y[max_idx] += phase;
}
}
static uint32_t celt_alg_quant(OpusRangeCoder *rc, float *X, uint32_t N, uint32_t K,
enum CeltSpread spread, uint32_t blocks, float gain)
{
int y[176];
celt_exp_rotation(X, N, blocks, K, spread, 1);
celt_pvq_search(X, y, K, N);
celt_encode_pulses(rc, y, N, K);
return celt_extract_collapse_mask(y, N, blocks);
}
/** Decode pulse vector and combine the result with the pitch vector to produce
the final normalised signal in the current band. */
static uint32_t celt_alg_unquant(OpusRangeCoder *rc, float *X, uint32_t N, uint32_t K,
enum CeltSpread spread, uint32_t blocks, float gain)
{
int y[176];
gain /= sqrtf(celt_decode_pulses(rc, y, N, K));
celt_normalize_residual(y, X, N, gain);
celt_exp_rotation(X, N, blocks, K, spread, 0);
return celt_extract_collapse_mask(y, N, blocks);
}
uint32_t ff_celt_decode_band(CeltFrame *f, OpusRangeCoder *rc, const int band,
float *X, float *Y, int N, int b, uint32_t blocks,
float *lowband, int duration, float *lowband_out, int level,
float gain, float *lowband_scratch, int fill)
{
const uint8_t *cache;
int dualstereo, split;
int imid = 0, iside = 0;
uint32_t N0 = N;
int N_B;
int N_B0;
int B0 = blocks;
int time_divide = 0;
int recombine = 0;
int inv = 0;
float mid = 0, side = 0;
int longblocks = (B0 == 1);
uint32_t cm = 0;
N_B0 = N_B = N / blocks;
split = dualstereo = (Y != NULL);
if (N == 1) {
/* special case for one sample */
int i;
float *x = X;
for (i = 0; i <= dualstereo; i++) {
int sign = 0;
if (f->remaining2 >= 1<<3) {
sign = ff_opus_rc_get_raw(rc, 1);
f->remaining2 -= 1 << 3;
b -= 1 << 3;
}
x[0] = sign ? -1.0f : 1.0f;
x = Y;
}
if (lowband_out)
lowband_out[0] = X[0];
return 1;
}
if (!dualstereo && level == 0) {
int tf_change = f->tf_change[band];
int k;
if (tf_change > 0)
recombine = tf_change;
/* Band recombining to increase frequency resolution */
if (lowband &&
(recombine || ((N_B & 1) == 0 && tf_change < 0) || B0 > 1)) {
int j;
for (j = 0; j < N; j++)
lowband_scratch[j] = lowband[j];
lowband = lowband_scratch;
}
for (k = 0; k < recombine; k++) {
if (lowband)
celt_haar1(lowband, N >> k, 1 << k);
fill = ff_celt_bit_interleave[fill & 0xF] | ff_celt_bit_interleave[fill >> 4] << 2;
}
blocks >>= recombine;
N_B <<= recombine;
/* Increasing the time resolution */
while ((N_B & 1) == 0 && tf_change < 0) {
if (lowband)
celt_haar1(lowband, N_B, blocks);
fill |= fill << blocks;
blocks <<= 1;
N_B >>= 1;
time_divide++;
tf_change++;
}
B0 = blocks;
N_B0 = N_B;
/* Reorganize the samples in time order instead of frequency order */
if (B0 > 1 && lowband)
celt_deinterleave_hadamard(f->scratch, lowband, N_B >> recombine,
B0 << recombine, longblocks);
}
/* If we need 1.5 more bit than we can produce, split the band in two. */
cache = ff_celt_cache_bits +
ff_celt_cache_index[(duration + 1) * CELT_MAX_BANDS + band];
if (!dualstereo && duration >= 0 && b > cache[cache[0]] + 12 && N > 2) {
N >>= 1;
Y = X + N;
split = 1;
duration -= 1;
if (blocks == 1)
fill = (fill & 1) | (fill << 1);
blocks = (blocks + 1) >> 1;
}
if (split) {
int qn;
int itheta = 0;
int mbits, sbits, delta;
int qalloc;
int pulse_cap;
int offset;
int orig_fill;
int tell;
/* Decide on the resolution to give to the split parameter theta */
pulse_cap = ff_celt_log_freq_range[band] + duration * 8;
offset = (pulse_cap >> 1) - (dualstereo && N == 2 ? CELT_QTHETA_OFFSET_TWOPHASE :
CELT_QTHETA_OFFSET);
qn = (dualstereo && band >= f->intensity_stereo) ? 1 :
celt_compute_qn(N, b, offset, pulse_cap, dualstereo);
tell = opus_rc_tell_frac(rc);
if (qn != 1) {
/* Entropy coding of the angle. We use a uniform pdf for the
time split, a step for stereo, and a triangular one for the rest. */
if (dualstereo && N > 2)
itheta = ff_opus_rc_dec_uint_step(rc, qn/2);
else if (dualstereo || B0 > 1)
itheta = ff_opus_rc_dec_uint(rc, qn+1);
else
itheta = ff_opus_rc_dec_uint_tri(rc, qn);
itheta = itheta * 16384 / qn;
/* NOTE: Renormalising X and Y *may* help fixed-point a bit at very high rate.
Let's do that at higher complexity */
} else if (dualstereo) {
inv = (b > 2 << 3 && f->remaining2 > 2 << 3) ? ff_opus_rc_dec_log(rc, 2) : 0;
itheta = 0;
}
qalloc = opus_rc_tell_frac(rc) - tell;
b -= qalloc;
orig_fill = fill;
if (itheta == 0) {
imid = 32767;
iside = 0;
fill = av_mod_uintp2(fill, blocks);
delta = -16384;
} else if (itheta == 16384) {
imid = 0;
iside = 32767;
fill &= ((1 << blocks) - 1) << blocks;
delta = 16384;
} else {
imid = celt_cos(itheta);
iside = celt_cos(16384-itheta);
/* This is the mid vs side allocation that minimizes squared error
in that band. */
delta = ROUND_MUL16((N - 1) << 7, celt_log2tan(iside, imid));
}
mid = imid / 32768.0f;
side = iside / 32768.0f;
/* This is a special case for N=2 that only works for stereo and takes
advantage of the fact that mid and side are orthogonal to encode
the side with just one bit. */
if (N == 2 && dualstereo) {
int c;
int sign = 0;
float tmp;
float *x2, *y2;
mbits = b;
/* Only need one bit for the side */
sbits = (itheta != 0 && itheta != 16384) ? 1 << 3 : 0;
mbits -= sbits;
c = (itheta > 8192);
f->remaining2 -= qalloc+sbits;
x2 = c ? Y : X;
y2 = c ? X : Y;
if (sbits)
sign = ff_opus_rc_get_raw(rc, 1);
sign = 1 - 2 * sign;
/* We use orig_fill here because we want to fold the side, but if
itheta==16384, we'll have cleared the low bits of fill. */
cm = ff_celt_decode_band(f, rc, band, x2, NULL, N, mbits, blocks,
lowband, duration, lowband_out, level, gain,
lowband_scratch, orig_fill);
/* We don't split N=2 bands, so cm is either 1 or 0 (for a fold-collapse),
and there's no need to worry about mixing with the other channel. */
y2[0] = -sign * x2[1];
y2[1] = sign * x2[0];
X[0] *= mid;
X[1] *= mid;
Y[0] *= side;
Y[1] *= side;
tmp = X[0];
X[0] = tmp - Y[0];
Y[0] = tmp + Y[0];
tmp = X[1];
X[1] = tmp - Y[1];
Y[1] = tmp + Y[1];
} else {
/* "Normal" split code */
float *next_lowband2 = NULL;
float *next_lowband_out1 = NULL;
int next_level = 0;
int rebalance;
/* Give more bits to low-energy MDCTs than they would
* otherwise deserve */
if (B0 > 1 && !dualstereo && (itheta & 0x3fff)) {
if (itheta > 8192)
/* Rough approximation for pre-echo masking */
delta -= delta >> (4 - duration);
else
/* Corresponds to a forward-masking slope of
* 1.5 dB per 10 ms */
delta = FFMIN(0, delta + (N << 3 >> (5 - duration)));
}
mbits = av_clip((b - delta) / 2, 0, b);
sbits = b - mbits;
f->remaining2 -= qalloc;
if (lowband && !dualstereo)
next_lowband2 = lowband + N; /* >32-bit split case */
/* Only stereo needs to pass on lowband_out.
* Otherwise, it's handled at the end */
if (dualstereo)
next_lowband_out1 = lowband_out;
else
next_level = level + 1;
rebalance = f->remaining2;
if (mbits >= sbits) {
/* In stereo mode, we do not apply a scaling to the mid
* because we need the normalized mid for folding later */
cm = ff_celt_decode_band(f, rc, band, X, NULL, N, mbits, blocks,
lowband, duration, next_lowband_out1,
next_level, dualstereo ? 1.0f : (gain * mid),
lowband_scratch, fill);
rebalance = mbits - (rebalance - f->remaining2);
if (rebalance > 3 << 3 && itheta != 0)
sbits += rebalance - (3 << 3);
/* For a stereo split, the high bits of fill are always zero,
* so no folding will be done to the side. */
cm |= ff_celt_decode_band(f, rc, band, Y, NULL, N, sbits, blocks,
next_lowband2, duration, NULL,
next_level, gain * side, NULL,
fill >> blocks) << ((B0 >> 1) & (dualstereo - 1));
} else {
/* For a stereo split, the high bits of fill are always zero,
* so no folding will be done to the side. */
cm = ff_celt_decode_band(f, rc, band, Y, NULL, N, sbits, blocks,
next_lowband2, duration, NULL,
next_level, gain * side, NULL,
fill >> blocks) << ((B0 >> 1) & (dualstereo - 1));
rebalance = sbits - (rebalance - f->remaining2);
if (rebalance > 3 << 3 && itheta != 16384)
mbits += rebalance - (3 << 3);
/* In stereo mode, we do not apply a scaling to the mid because
* we need the normalized mid for folding later */
cm |= ff_celt_decode_band(f, rc, band, X, NULL, N, mbits, blocks,
lowband, duration, next_lowband_out1,
next_level, dualstereo ? 1.0f : (gain * mid),
lowband_scratch, fill);
}
}
} else {
/* This is the basic no-split case */
uint32_t q = celt_bits2pulses(cache, b);
uint32_t curr_bits = celt_pulses2bits(cache, q);
f->remaining2 -= curr_bits;
/* Ensures we can never bust the budget */
while (f->remaining2 < 0 && q > 0) {
f->remaining2 += curr_bits;
curr_bits = celt_pulses2bits(cache, --q);
f->remaining2 -= curr_bits;
}
if (q != 0) {
/* Finally do the actual quantization */
cm = celt_alg_unquant(rc, X, N, (q < 8) ? q : (8 + (q & 7)) << ((q >> 3) - 1),
f->spread, blocks, gain);
} else {
/* If there's no pulse, fill the band anyway */
int j;
uint32_t cm_mask = (1 << blocks) - 1;
fill &= cm_mask;
if (!fill) {
for (j = 0; j < N; j++)
X[j] = 0.0f;
} else {
if (!lowband) {
/* Noise */
for (j = 0; j < N; j++)
X[j] = (((int32_t)celt_rng(f)) >> 20);
cm = cm_mask;
} else {
/* Folded spectrum */
for (j = 0; j < N; j++) {
/* About 48 dB below the "normal" folding level */
X[j] = lowband[j] + (((celt_rng(f)) & 0x8000) ? 1.0f / 256 : -1.0f / 256);
}
cm = fill;
}
celt_renormalize_vector(X, N, gain);
}
}
}
/* This code is used by the decoder and by the resynthesis-enabled encoder */
if (dualstereo) {
int j;
if (N != 2)
celt_stereo_merge(X, Y, mid, N);
if (inv) {
for (j = 0; j < N; j++)
Y[j] *= -1;
}
} else if (level == 0) {
int k;
/* Undo the sample reorganization going from time order to frequency order */
if (B0 > 1)
celt_interleave_hadamard(f->scratch, X, N_B>>recombine,
B0<<recombine, longblocks);
/* Undo time-freq changes that we did earlier */
N_B = N_B0;
blocks = B0;
for (k = 0; k < time_divide; k++) {
blocks >>= 1;
N_B <<= 1;
cm |= cm >> blocks;
celt_haar1(X, N_B, blocks);
}
for (k = 0; k < recombine; k++) {
cm = ff_celt_bit_deinterleave[cm];
celt_haar1(X, N0>>k, 1<<k);
}
blocks <<= recombine;
/* Scale output for later folding */
if (lowband_out) {
int j;
float n = sqrtf(N0);
for (j = 0; j < N0; j++)
lowband_out[j] = n * X[j];
}
cm = av_mod_uintp2(cm, blocks);
}
return cm;
}
/* This has to be, AND MUST BE done by the psychoacoustic system, this has a very
* big impact on the entire quantization and especially huge on transients */
static int celt_calc_theta(const float *X, const float *Y, int coupling, int N)
{
int j;
float e[2] = { 0.0f, 0.0f };
for (j = 0; j < N; j++) {
if (coupling) { /* Coupling case */
e[0] += (X[j] + Y[j])*(X[j] + Y[j]);
e[1] += (X[j] - Y[j])*(X[j] - Y[j]);
} else {
e[0] += X[j]*X[j];
e[1] += Y[j]*Y[j];
}
}
return lrintf(32768.0f*atan2f(sqrtf(e[1]), sqrtf(e[0]))/M_PI);
}
static void celt_stereo_is_decouple(float *X, float *Y, float e_l, float e_r, int N)
{
int i;
const float energy_n = 1.0f/(sqrtf(e_l*e_l + e_r*e_r) + FLT_EPSILON);
e_l *= energy_n;
e_r *= energy_n;
for (i = 0; i < N; i++)
X[i] = e_l*X[i] + e_r*Y[i];
}
static void celt_stereo_ms_decouple(float *X, float *Y, int N)
{
int i;
const float decouple_norm = 1.0f/sqrtf(2.0f);
for (i = 0; i < N; i++) {
const float Xret = X[i];
X[i] = (X[i] + Y[i])*decouple_norm;
Y[i] = (Y[i] - Xret)*decouple_norm;
}
}
uint32_t ff_celt_encode_band(CeltFrame *f, OpusRangeCoder *rc, const int band,
float *X, float *Y, int N, int b, uint32_t blocks,
float *lowband, int duration, float *lowband_out, int level,
float gain, float *lowband_scratch, int fill)
{
const uint8_t *cache;
int dualstereo, split;
int imid = 0, iside = 0;
//uint32_t N0 = N;
int N_B;
//int N_B0;
int B0 = blocks;
int time_divide = 0;
int recombine = 0;
int inv = 0;
float mid = 0, side = 0;
int longblocks = (B0 == 1);
uint32_t cm = 0;
//N_B0 = N_B = N / blocks;
split = dualstereo = (Y != NULL);
if (N == 1) {
/* special case for one sample - the decoder's output will be +- 1.0f!!! */
int i;
float *x = X;
for (i = 0; i <= dualstereo; i++) {
if (f->remaining2 >= 1<<3) {
ff_opus_rc_put_raw(rc, x[0] < 0, 1);
f->remaining2 -= 1 << 3;
b -= 1 << 3;
}
x = Y;
}
if (lowband_out)
lowband_out[0] = X[0];
return 1;
}
if (!dualstereo && level == 0) {
int tf_change = f->tf_change[band];
int k;
if (tf_change > 0)
recombine = tf_change;
/* Band recombining to increase frequency resolution */
if (lowband &&
(recombine || ((N_B & 1) == 0 && tf_change < 0) || B0 > 1)) {
int j;
for (j = 0; j < N; j++)
lowband_scratch[j] = lowband[j];
lowband = lowband_scratch;
}
for (k = 0; k < recombine; k++) {
celt_haar1(X, N >> k, 1 << k);
fill = ff_celt_bit_interleave[fill & 0xF] | ff_celt_bit_interleave[fill >> 4] << 2;
}
blocks >>= recombine;
N_B <<= recombine;
/* Increasing the time resolution */
while ((N_B & 1) == 0 && tf_change < 0) {
celt_haar1(X, N_B, blocks);
fill |= fill << blocks;
blocks <<= 1;
N_B >>= 1;
time_divide++;
tf_change++;
}
B0 = blocks;
//N_B0 = N_B;
/* Reorganize the samples in time order instead of frequency order */
if (B0 > 1)
celt_deinterleave_hadamard(f->scratch, X, N_B >> recombine,
B0 << recombine, longblocks);
}
/* If we need 1.5 more bit than we can produce, split the band in two. */
cache = ff_celt_cache_bits +
ff_celt_cache_index[(duration + 1) * CELT_MAX_BANDS + band];
if (!dualstereo && duration >= 0 && b > cache[cache[0]] + 12 && N > 2) {
N >>= 1;
Y = X + N;
split = 1;
duration -= 1;
if (blocks == 1)
fill = (fill & 1) | (fill << 1);
blocks = (blocks + 1) >> 1;
}
if (split) {
int qn;
int itheta = celt_calc_theta(X, Y, dualstereo, N);
int mbits, sbits, delta;
int qalloc;
int pulse_cap;
int offset;
int orig_fill;
int tell;
/* Decide on the resolution to give to the split parameter theta */
pulse_cap = ff_celt_log_freq_range[band] + duration * 8;
offset = (pulse_cap >> 1) - (dualstereo && N == 2 ? CELT_QTHETA_OFFSET_TWOPHASE :
CELT_QTHETA_OFFSET);
qn = (dualstereo && band >= f->intensity_stereo) ? 1 :
celt_compute_qn(N, b, offset, pulse_cap, dualstereo);
tell = opus_rc_tell_frac(rc);
if (qn != 1) {
itheta = (itheta*qn + 8192) >> 14;
/* Entropy coding of the angle. We use a uniform pdf for the
* time split, a step for stereo, and a triangular one for the rest. */
if (dualstereo && N > 2)
ff_opus_rc_enc_uint_step(rc, itheta, qn / 2);
else if (dualstereo || B0 > 1)
ff_opus_rc_enc_uint(rc, itheta, qn + 1);
else
ff_opus_rc_enc_uint_tri(rc, itheta, qn);
itheta = itheta * 16384 / qn;
if (dualstereo) {
if (itheta == 0)
celt_stereo_is_decouple(X, Y, f->block[0].lin_energy[band], f->block[1].lin_energy[band], N);
else
celt_stereo_ms_decouple(X, Y, N);
}
} else if (dualstereo) {
inv = itheta > 8192;
if (inv)
{
int j;
for (j=0;j<N;j++)
Y[j] = -Y[j];
}
celt_stereo_is_decouple(X, Y, f->block[0].lin_energy[band], f->block[1].lin_energy[band], N);
if (b > 2 << 3 && f->remaining2 > 2 << 3) {
ff_opus_rc_enc_log(rc, inv, 2);
} else {
inv = 0;
}
itheta = 0;
}
qalloc = opus_rc_tell_frac(rc) - tell;
b -= qalloc;
orig_fill = fill;
if (itheta == 0) {
imid = 32767;
iside = 0;
fill = av_mod_uintp2(fill, blocks);
delta = -16384;
} else if (itheta == 16384) {
imid = 0;
iside = 32767;
fill &= ((1 << blocks) - 1) << blocks;
delta = 16384;
} else {
imid = celt_cos(itheta);
iside = celt_cos(16384-itheta);
/* This is the mid vs side allocation that minimizes squared error
in that band. */
delta = ROUND_MUL16((N - 1) << 7, celt_log2tan(iside, imid));
}
mid = imid / 32768.0f;
side = iside / 32768.0f;
/* This is a special case for N=2 that only works for stereo and takes
advantage of the fact that mid and side are orthogonal to encode
the side with just one bit. */
if (N == 2 && dualstereo) {
int c;
int sign = 0;
float tmp;
float *x2, *y2;
mbits = b;
/* Only need one bit for the side */
sbits = (itheta != 0 && itheta != 16384) ? 1 << 3 : 0;
mbits -= sbits;
c = (itheta > 8192);
f->remaining2 -= qalloc+sbits;
x2 = c ? Y : X;
y2 = c ? X : Y;
if (sbits) {
sign = x2[0]*y2[1] - x2[1]*y2[0] < 0;
ff_opus_rc_put_raw(rc, sign, 1);
}
sign = 1 - 2 * sign;
/* We use orig_fill here because we want to fold the side, but if
itheta==16384, we'll have cleared the low bits of fill. */
cm = ff_celt_encode_band(f, rc, band, x2, NULL, N, mbits, blocks,
lowband, duration, lowband_out, level, gain,
lowband_scratch, orig_fill);
/* We don't split N=2 bands, so cm is either 1 or 0 (for a fold-collapse),
and there's no need to worry about mixing with the other channel. */
y2[0] = -sign * x2[1];
y2[1] = sign * x2[0];
X[0] *= mid;
X[1] *= mid;
Y[0] *= side;
Y[1] *= side;
tmp = X[0];
X[0] = tmp - Y[0];
Y[0] = tmp + Y[0];
tmp = X[1];
X[1] = tmp - Y[1];
Y[1] = tmp + Y[1];
} else {
/* "Normal" split code */
float *next_lowband2 = NULL;
float *next_lowband_out1 = NULL;
int next_level = 0;
int rebalance;
/* Give more bits to low-energy MDCTs than they would
* otherwise deserve */
if (B0 > 1 && !dualstereo && (itheta & 0x3fff)) {
if (itheta > 8192)
/* Rough approximation for pre-echo masking */
delta -= delta >> (4 - duration);
else
/* Corresponds to a forward-masking slope of
* 1.5 dB per 10 ms */
delta = FFMIN(0, delta + (N << 3 >> (5 - duration)));
}
mbits = av_clip((b - delta) / 2, 0, b);
sbits = b - mbits;
f->remaining2 -= qalloc;
if (lowband && !dualstereo)
next_lowband2 = lowband + N; /* >32-bit split case */
/* Only stereo needs to pass on lowband_out.
* Otherwise, it's handled at the end */
if (dualstereo)
next_lowband_out1 = lowband_out;
else
next_level = level + 1;
rebalance = f->remaining2;
if (mbits >= sbits) {
/* In stereo mode, we do not apply a scaling to the mid
* because we need the normalized mid for folding later */
cm = ff_celt_encode_band(f, rc, band, X, NULL, N, mbits, blocks,
lowband, duration, next_lowband_out1,
next_level, dualstereo ? 1.0f : (gain * mid),
lowband_scratch, fill);
rebalance = mbits - (rebalance - f->remaining2);
if (rebalance > 3 << 3 && itheta != 0)
sbits += rebalance - (3 << 3);
/* For a stereo split, the high bits of fill are always zero,
* so no folding will be done to the side. */
cm |= ff_celt_encode_band(f, rc, band, Y, NULL, N, sbits, blocks,
next_lowband2, duration, NULL,
next_level, gain * side, NULL,
fill >> blocks) << ((B0 >> 1) & (dualstereo - 1));
} else {
/* For a stereo split, the high bits of fill are always zero,
* so no folding will be done to the side. */
cm = ff_celt_encode_band(f, rc, band, Y, NULL, N, sbits, blocks,
next_lowband2, duration, NULL,
next_level, gain * side, NULL,
fill >> blocks) << ((B0 >> 1) & (dualstereo - 1));
rebalance = sbits - (rebalance - f->remaining2);
if (rebalance > 3 << 3 && itheta != 16384)
mbits += rebalance - (3 << 3);
/* In stereo mode, we do not apply a scaling to the mid because
* we need the normalized mid for folding later */
cm |= ff_celt_encode_band(f, rc, band, X, NULL, N, mbits, blocks,
lowband, duration, next_lowband_out1,
next_level, dualstereo ? 1.0f : (gain * mid),
lowband_scratch, fill);
}
}
} else {
/* This is the basic no-split case */
uint32_t q = celt_bits2pulses(cache, b);
uint32_t curr_bits = celt_pulses2bits(cache, q);
f->remaining2 -= curr_bits;
/* Ensures we can never bust the budget */
while (f->remaining2 < 0 && q > 0) {
f->remaining2 += curr_bits;
curr_bits = celt_pulses2bits(cache, --q);
f->remaining2 -= curr_bits;
}
if (q != 0) {
/* Finally do the actual quantization */
cm = celt_alg_quant(rc, X, N, (q < 8) ? q : (8 + (q & 7)) << ((q >> 3) - 1),
f->spread, blocks, gain);
}
}
return cm;
}
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