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author | Rostislav Pehlivanov <atomnuker@gmail.com> | 2017-02-11 00:25:06 +0000 |
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committer | Rostislav Pehlivanov <atomnuker@gmail.com> | 2017-02-14 06:15:36 +0000 |
commit | e538108c219d7b3628a9ec33d85bf252ee70c957 (patch) | |
tree | 796c62422dbc5f3e555d84ce57557729c7a8c900 /libavcodec/opus_pvq.c | |
parent | d2119f624d392f53f80c3d36ffaadca23aef8a10 (diff) | |
download | ffmpeg-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.c | 729 |
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; +} |