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authorRostislav Pehlivanov <atomnuker@gmail.com>2017-02-11 00:25:06 +0000
committerRostislav Pehlivanov <atomnuker@gmail.com>2017-02-14 06:15:36 +0000
commite538108c219d7b3628a9ec33d85bf252ee70c957 (patch)
tree796c62422dbc5f3e555d84ce57557729c7a8c900 /libavcodec/opus_pvq.c
parentd2119f624d392f53f80c3d36ffaadca23aef8a10 (diff)
downloadffmpeg-e538108c219d7b3628a9ec33d85bf252ee70c957.tar.gz
opus_celt: move quantization and band decoding to opus_pvq.c
A huge amount can be reused by the encoder, as the only thing which needs to be done would be to add a 10 line celt_icwrsi, a wrapper around it (celt_alg_quant) and templating the ff_celt_decode_band to replace entropy decoding functions with entropy encoding. There is no performance loss but in fact a performance gain of around 6% which is caused by the compiler being able to optimize the decoding more efficiently. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Diffstat (limited to 'libavcodec/opus_pvq.c')
-rw-r--r--libavcodec/opus_pvq.c729
1 files changed, 729 insertions, 0 deletions
diff --git a/libavcodec/opus_pvq.c b/libavcodec/opus_pvq.c
new file mode 100644
index 0000000000..b4e23c86b8
--- /dev/null
+++ b/libavcodec/opus_pvq.c
@@ -0,0 +1,729 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2016 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_rotation1(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)
+{
+ 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 (stride2)
+ celt_exp_rotation1(X + i * len, len, stride2, s, c);
+ celt_exp_rotation1(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;
+}
+
+// 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 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);
+}
+
+/** 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);
+ return celt_extract_collapse_mask(y, N, blocks);
+}
+
+uint32_t ff_celt_decode_band(CeltContext *s, 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 (s->remaining2 >= 1<<3) {
+ sign = ff_opus_rc_get_raw(rc, 1);
+ s->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 = s->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(s->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 >= s->intensitystereo) ? 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 && s->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);
+ s->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(s, 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;
+ s->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 = s->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(s, 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 - s->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(s, 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(s, 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 - s->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(s, 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);
+ s->remaining2 -= curr_bits;
+
+ /* Ensures we can never bust the budget */
+ while (s->remaining2 < 0 && q > 0) {
+ s->remaining2 += curr_bits;
+ curr_bits = celt_pulses2bits(cache, --q);
+ s->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),
+ s->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(s)) >> 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(s)) & 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(s->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;
+}