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FFmpeg/libavcodec/opusenc.c
Rostislav Pehlivanov b71f62a9c8 opusenc: initialize the emphasis coefficients on init 
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2017-02-18 01:13:13 +00:00

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/*
* Opus encoder
* 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 "opus_celt.h"
#include "opus_pvq.h"
#include "opustab.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "internal.h"
#include "bytestream.h"
#include "audio_frame_queue.h"
/* Determines the maximum delay the psychoacoustic system will use for lookahead */
#define FF_BUFQUEUE_SIZE 145
#include "libavfilter/bufferqueue.h"
#define OPUS_MAX_LOOKAHEAD ((FF_BUFQUEUE_SIZE - 1)*2.5f)
#define OPUS_MAX_CHANNELS 2
/* 120 ms / 2.5 ms = 48 frames (extremely improbable, but the encoder'll work) */
#define OPUS_MAX_FRAMES_PER_PACKET 48
#define OPUS_BLOCK_SIZE(x) (2 * 15 * (1 << ((x) + 2)))
#define OPUS_SAMPLES_TO_BLOCK_SIZE(x) (ff_log2((x) / (2 * 15)) - 2)
typedef struct OpusEncOptions {
float max_delay_ms;
} OpusEncOptions;
typedef struct OpusEncContext {
AVClass *av_class;
OpusEncOptions options;
AVCodecContext *avctx;
AudioFrameQueue afq;
AVFloatDSPContext *dsp;
MDCT15Context *mdct[CELT_BLOCK_NB];
struct FFBufQueue bufqueue;
enum OpusMode mode;
enum OpusBandwidth bandwidth;
int pkt_framesize;
int pkt_frames;
int channels;
CeltFrame *frame;
OpusRangeCoder *rc;
/* Actual energy the decoder will have */
float last_quantized_energy[OPUS_MAX_CHANNELS][CELT_MAX_BANDS];
DECLARE_ALIGNED(32, float, scratch)[2048];
} OpusEncContext;
static void opus_write_extradata(AVCodecContext *avctx)
{
uint8_t *bs = avctx->extradata;
bytestream_put_buffer(&bs, "OpusHead", 8);
bytestream_put_byte (&bs, 0x1);
bytestream_put_byte (&bs, avctx->channels);
bytestream_put_le16 (&bs, avctx->initial_padding);
bytestream_put_le32 (&bs, avctx->sample_rate);
bytestream_put_le16 (&bs, 0x0);
bytestream_put_byte (&bs, 0x0); /* Default layout */
}
static int opus_gen_toc(OpusEncContext *s, uint8_t *toc, int *size, int *fsize_needed)
{
int i, tmp = 0x0, extended_toc = 0;
static const int toc_cfg[][OPUS_MODE_NB][OPUS_BANDWITH_NB] = {
/* Silk Hybrid Celt Layer */
/* NB MB WB SWB FB NB MB WB SWB FB NB MB WB SWB FB Bandwidth */
{ { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 }, { 17, 0, 21, 25, 29 } }, /* 2.5 ms */
{ { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 }, { 18, 0, 22, 26, 30 } }, /* 5 ms */
{ { 1, 5, 9, 0, 0 }, { 0, 0, 0, 13, 15 }, { 19, 0, 23, 27, 31 } }, /* 10 ms */
{ { 2, 6, 10, 0, 0 }, { 0, 0, 0, 14, 16 }, { 20, 0, 24, 28, 32 } }, /* 20 ms */
{ { 3, 7, 11, 0, 0 }, { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 } }, /* 40 ms */
{ { 4, 8, 12, 0, 0 }, { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 } }, /* 60 ms */
};
int cfg = toc_cfg[s->pkt_framesize][s->mode][s->bandwidth];
*fsize_needed = 0;
if (!cfg)
return 1;
if (s->pkt_frames == 2) { /* 2 packets */
if (s->frame[0].framebits == s->frame[1].framebits) { /* same size */
tmp = 0x1;
} else { /* different size */
tmp = 0x2;
*fsize_needed = 1; /* put frame sizes in the packet */
}
} else if (s->pkt_frames > 2) {
tmp = 0x3;
extended_toc = 1;
}
tmp |= (s->channels > 1) << 2; /* Stereo or mono */
tmp |= (cfg - 1) << 3; /* codec configuration */
*toc++ = tmp;
if (extended_toc) {
for (i = 0; i < (s->pkt_frames - 1); i++)
*fsize_needed |= (s->frame[i].framebits != s->frame[i + 1].framebits);
tmp = (*fsize_needed) << 7; /* vbr flag */
tmp |= s->pkt_frames; /* frame number - can be 0 as well */
*toc++ = tmp;
}
*size = 1 + extended_toc;
return 0;
}
static void celt_frame_setup_input(OpusEncContext *s, CeltFrame *f)
{
int sf, ch;
AVFrame *cur = NULL;
const int subframesize = s->avctx->frame_size;
int subframes = OPUS_BLOCK_SIZE(s->pkt_framesize) / subframesize;
cur = ff_bufqueue_get(&s->bufqueue);
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
const void *input = cur->extended_data[ch];
size_t bps = av_get_bytes_per_sample(cur->format);
memcpy(b->overlap, input, bps*cur->nb_samples);
}
av_frame_free(&cur);
for (sf = 0; sf < subframes; sf++) {
if (sf != (subframes - 1))
cur = ff_bufqueue_get(&s->bufqueue);
else
cur = ff_bufqueue_peek(&s->bufqueue, 0);
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
const void *input = cur->extended_data[ch];
const size_t bps = av_get_bytes_per_sample(cur->format);
const size_t left = (subframesize - cur->nb_samples)*bps;
const size_t len = FFMIN(subframesize, cur->nb_samples)*bps;
memcpy(&b->samples[sf*subframesize], input, len);
memset(&b->samples[cur->nb_samples], 0, left);
}
/* Last frame isn't popped off and freed yet - we need it for overlap */
if (sf != (subframes - 1))
av_frame_free(&cur);
}
}
/* Apply the pre emphasis filter */
static void celt_apply_preemph_filter(OpusEncContext *s, CeltFrame *f)
{
int i, sf, ch;
const int subframesize = s->avctx->frame_size;
const int subframes = OPUS_BLOCK_SIZE(s->pkt_framesize) / subframesize;
/* Filter overlap */
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
float m = b->emph_coeff;
for (i = 0; i < CELT_OVERLAP; i++) {
float sample = b->overlap[i];
b->overlap[i] = sample - m;
m = sample * CELT_EMPH_COEFF;
}
b->emph_coeff = m;
}
/* Filter the samples but do not update the last subframe's coeff - overlap ^^^ */
for (sf = 0; sf < subframes; sf++) {
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
float m = b->emph_coeff;
for (i = 0; i < subframesize; i++) {
float sample = b->samples[sf*subframesize + i];
b->samples[sf*subframesize + i] = sample - m;
m = sample * CELT_EMPH_COEFF;
}
if (sf != (subframes - 1))
b->emph_coeff = m;
}
}
}
/* Create the window and do the mdct */
static void celt_frame_mdct(OpusEncContext *s, CeltFrame *f)
{
int i, t, ch;
float *win = s->scratch;
/* I think I can use s->dsp->vector_fmul_window for transients at least */
if (f->transient) {
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
float *src1 = b->overlap;
for (t = 0; t < f->blocks; t++) {
float *src2 = &b->samples[CELT_OVERLAP*t];
for (i = 0; i < CELT_OVERLAP; i++) {
win[ i] = src1[i]*ff_celt_window[i];
win[CELT_OVERLAP + i] = src2[i]*ff_celt_window[CELT_OVERLAP - i - 1];
}
src1 = src2;
s->mdct[0]->mdct(s->mdct[0], b->coeffs + t, win, f->blocks);
}
}
} else {
int blk_len = OPUS_BLOCK_SIZE(f->size), wlen = OPUS_BLOCK_SIZE(f->size + 1);
int rwin = blk_len - CELT_OVERLAP, lap_dst = (wlen - blk_len - CELT_OVERLAP) >> 1;
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
memset(win, 0, wlen*sizeof(float));
memcpy(&win[lap_dst + CELT_OVERLAP], b->samples, rwin*sizeof(float));
/* Alignment fucks me over */
//s->dsp->vector_fmul(&dst[lap_dst], b->overlap, ff_celt_window, CELT_OVERLAP);
//s->dsp->vector_fmul_reverse(&dst[lap_dst + blk_len - CELT_OVERLAP], b->samples, ff_celt_window, CELT_OVERLAP);
for (i = 0; i < CELT_OVERLAP; i++) {
win[lap_dst + i] = b->overlap[i] *ff_celt_window[i];
win[lap_dst + blk_len + i] = b->samples[rwin + i]*ff_celt_window[CELT_OVERLAP - i - 1];
}
s->mdct[f->size]->mdct(s->mdct[f->size], b->coeffs, win, 1);
}
}
}
/* Fills the bands and normalizes them */
static int celt_frame_map_norm_bands(OpusEncContext *s, CeltFrame *f)
{
int i, j, ch, noise = 0;
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *block = &f->block[ch];
float *start = block->coeffs;
for (i = 0; i < CELT_MAX_BANDS; i++) {
float ener = 0.0f;
/* Calculate band bins */
block->band_bins[i] = ff_celt_freq_range[i] << f->size;
block->band_coeffs[i] = start;
start += block->band_bins[i];
/* Normalize band energy */
for (j = 0; j < block->band_bins[i]; j++)
ener += block->band_coeffs[i][j]*block->band_coeffs[i][j];
block->lin_energy[i] = sqrtf(ener) + FLT_EPSILON;
ener = 1.0f/block->lin_energy[i];
for (j = 0; j < block->band_bins[i]; j++)
block->band_coeffs[i][j] *= ener;
block->energy[i] = log2f(block->lin_energy[i]) - ff_celt_mean_energy[i];
/* CELT_ENERGY_SILENCE is what the decoder uses and its not -infinity */
block->energy[i] = FFMAX(block->energy[i], CELT_ENERGY_SILENCE);
noise |= block->energy[i] > CELT_ENERGY_SILENCE;
}
}
return !noise;
}
static void celt_enc_tf(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
{
int i, tf_select = 0, diff = 0, tf_changed = 0, tf_select_needed;
int bits = f->transient ? 2 : 4;
tf_select_needed = ((f->size && (opus_rc_tell(rc) + bits + 1) <= f->framebits));
for (i = f->start_band; i < f->end_band; i++) {
if ((opus_rc_tell(rc) + bits + tf_select_needed) <= f->framebits) {
const int tbit = (diff ^ 1) == f->tf_change[i];
ff_opus_rc_enc_log(rc, tbit, bits);
diff ^= tbit;
tf_changed |= diff;
}
bits = f->transient ? 4 : 5;
}
if (tf_select_needed && ff_celt_tf_select[f->size][f->transient][0][tf_changed] !=
ff_celt_tf_select[f->size][f->transient][1][tf_changed]) {
ff_opus_rc_enc_log(rc, f->tf_select, 1);
tf_select = f->tf_select;
}
for (i = f->start_band; i < f->end_band; i++)
f->tf_change[i] = ff_celt_tf_select[f->size][f->transient][tf_select][f->tf_change[i]];
}
static void celt_bitalloc(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
{
int i, j, low, high, total, done, bandbits, remaining, tbits_8ths;
int skip_startband = f->start_band;
int skip_bit = 0;
int intensitystereo_bit = 0;
int dualstereo_bit = 0;
int dynalloc = 6;
int extrabits = 0;
int *cap = f->caps;
int boost[CELT_MAX_BANDS];
int trim_offset[CELT_MAX_BANDS];
int threshold[CELT_MAX_BANDS];
int bits1[CELT_MAX_BANDS];
int bits2[CELT_MAX_BANDS];
/* Tell the spread to the decoder */
if (opus_rc_tell(rc) + 4 <= f->framebits)
ff_opus_rc_enc_cdf(rc, f->spread, ff_celt_model_spread);
/* Generate static allocation caps */
for (i = 0; i < CELT_MAX_BANDS; i++) {
cap[i] = (ff_celt_static_caps[f->size][f->channels - 1][i] + 64)
* ff_celt_freq_range[i] << (f->channels - 1) << f->size >> 2;
}
/* Band boosts */
tbits_8ths = f->framebits << 3;
for (i = f->start_band; i < f->end_band; i++) {
int quanta, b_dynalloc, boost_amount = f->alloc_boost[i];
boost[i] = 0;
quanta = ff_celt_freq_range[i] << (f->channels - 1) << f->size;
quanta = FFMIN(quanta << 3, FFMAX(6 << 3, quanta));
b_dynalloc = dynalloc;
while (opus_rc_tell_frac(rc) + (b_dynalloc << 3) < tbits_8ths && boost[i] < cap[i]) {
int is_boost = boost_amount--;
ff_opus_rc_enc_log(rc, is_boost, b_dynalloc);
if (!is_boost)
break;
boost[i] += quanta;
tbits_8ths -= quanta;
b_dynalloc = 1;
}
if (boost[i])
dynalloc = FFMAX(2, dynalloc - 1);
}
/* Put allocation trim */
if (opus_rc_tell_frac(rc) + (6 << 3) <= tbits_8ths)
ff_opus_rc_enc_cdf(rc, f->alloc_trim, ff_celt_model_alloc_trim);
/* Anti-collapse bit reservation */
tbits_8ths = (f->framebits << 3) - opus_rc_tell_frac(rc) - 1;
f->anticollapse_needed = 0;
if (f->transient && f->size >= 2 && tbits_8ths >= ((f->size + 2) << 3))
f->anticollapse_needed = 1 << 3;
tbits_8ths -= f->anticollapse_needed;
/* Band skip bit reservation */
if (tbits_8ths >= 1 << 3)
skip_bit = 1 << 3;
tbits_8ths -= skip_bit;
/* Intensity/dual stereo bit reservation */
if (f->channels == 2) {
intensitystereo_bit = ff_celt_log2_frac[f->end_band - f->start_band];
if (intensitystereo_bit <= tbits_8ths) {
tbits_8ths -= intensitystereo_bit;
if (tbits_8ths >= 1 << 3) {
dualstereo_bit = 1 << 3;
tbits_8ths -= 1 << 3;
}
} else {
intensitystereo_bit = 0;
}
}
/* Trim offsets */
for (i = f->start_band; i < f->end_band; i++) {
int trim = f->alloc_trim - 5 - f->size;
int band = ff_celt_freq_range[i] * (f->end_band - i - 1);
int duration = f->size + 3;
int scale = duration + f->channels - 1;
/* PVQ minimum allocation threshold, below this value the band is
* skipped */
threshold[i] = FFMAX(3 * ff_celt_freq_range[i] << duration >> 4,
f->channels << 3);
trim_offset[i] = trim * (band << scale) >> 6;
if (ff_celt_freq_range[i] << f->size == 1)
trim_offset[i] -= f->channels << 3;
}
/* Bisection */
low = 1;
high = CELT_VECTORS - 1;
while (low <= high) {
int center = (low + high) >> 1;
done = total = 0;
for (i = f->end_band - 1; i >= f->start_band; i--) {
bandbits = ff_celt_freq_range[i] * ff_celt_static_alloc[center][i]
<< (f->channels - 1) << f->size >> 2;
if (bandbits)
bandbits = FFMAX(0, bandbits + trim_offset[i]);
bandbits += boost[i];
if (bandbits >= threshold[i] || done) {
done = 1;
total += FFMIN(bandbits, cap[i]);
} else if (bandbits >= f->channels << 3)
total += f->channels << 3;
}
if (total > tbits_8ths)
high = center - 1;
else
low = center + 1;
}
high = low--;
/* Bisection */
for (i = f->start_band; i < f->end_band; i++) {
bits1[i] = ff_celt_freq_range[i] * ff_celt_static_alloc[low][i]
<< (f->channels - 1) << f->size >> 2;
bits2[i] = high >= CELT_VECTORS ? cap[i] :
ff_celt_freq_range[i] * ff_celt_static_alloc[high][i]
<< (f->channels - 1) << f->size >> 2;
if (bits1[i])
bits1[i] = FFMAX(0, bits1[i] + trim_offset[i]);
if (bits2[i])
bits2[i] = FFMAX(0, bits2[i] + trim_offset[i]);
if (low)
bits1[i] += boost[i];
bits2[i] += boost[i];
if (boost[i])
skip_startband = i;
bits2[i] = FFMAX(0, bits2[i] - bits1[i]);
}
/* Bisection */
low = 0;
high = 1 << CELT_ALLOC_STEPS;
for (i = 0; i < CELT_ALLOC_STEPS; i++) {
int center = (low + high) >> 1;
done = total = 0;
for (j = f->end_band - 1; j >= f->start_band; j--) {
bandbits = bits1[j] + (center * bits2[j] >> CELT_ALLOC_STEPS);
if (bandbits >= threshold[j] || done) {
done = 1;
total += FFMIN(bandbits, cap[j]);
} else if (bandbits >= f->channels << 3)
total += f->channels << 3;
}
if (total > tbits_8ths)
high = center;
else
low = center;
}
/* Bisection */
done = total = 0;
for (i = f->end_band - 1; i >= f->start_band; i--) {
bandbits = bits1[i] + (low * bits2[i] >> CELT_ALLOC_STEPS);
if (bandbits >= threshold[i] || done)
done = 1;
else
bandbits = (bandbits >= f->channels << 3) ?
f->channels << 3 : 0;
bandbits = FFMIN(bandbits, cap[i]);
f->pulses[i] = bandbits;
total += bandbits;
}
/* Band skipping */
for (f->coded_bands = f->end_band; ; f->coded_bands--) {
int allocation;
j = f->coded_bands - 1;
if (j == skip_startband) {
/* all remaining bands are not skipped */
tbits_8ths += skip_bit;
break;
}
/* determine the number of bits available for coding "do not skip" markers */
remaining = tbits_8ths - total;
bandbits = remaining / (ff_celt_freq_bands[j+1] - ff_celt_freq_bands[f->start_band]);
remaining -= bandbits * (ff_celt_freq_bands[j+1] - ff_celt_freq_bands[f->start_band]);
allocation = f->pulses[j] + bandbits * ff_celt_freq_range[j]
+ FFMAX(0, remaining - (ff_celt_freq_bands[j] - ff_celt_freq_bands[f->start_band]));
/* a "do not skip" marker is only coded if the allocation is
above the chosen threshold */
if (allocation >= FFMAX(threshold[j], (f->channels + 1) << 3)) {
const int do_not_skip = f->coded_bands <= f->skip_band_floor;
ff_opus_rc_enc_log(rc, do_not_skip, 1);
if (do_not_skip)
break;
total += 1 << 3;
allocation -= 1 << 3;
}
/* the band is skipped, so reclaim its bits */
total -= f->pulses[j];
if (intensitystereo_bit) {
total -= intensitystereo_bit;
intensitystereo_bit = ff_celt_log2_frac[j - f->start_band];
total += intensitystereo_bit;
}
total += f->pulses[j] = (allocation >= f->channels << 3) ? f->channels << 3 : 0;
}
/* Encode stereo flags */
if (intensitystereo_bit) {
f->intensity_stereo = FFMIN(f->intensity_stereo, f->coded_bands);
ff_opus_rc_enc_uint(rc, f->intensity_stereo, f->coded_bands + 1 - f->start_band);
}
if (f->intensity_stereo <= f->start_band)
tbits_8ths += dualstereo_bit; /* no intensity stereo means no dual stereo */
else if (dualstereo_bit)
ff_opus_rc_enc_log(rc, f->dual_stereo, 1);
/* Supply the remaining bits in this frame to lower bands */
remaining = tbits_8ths - total;
bandbits = remaining / (ff_celt_freq_bands[f->coded_bands] - ff_celt_freq_bands[f->start_band]);
remaining -= bandbits * (ff_celt_freq_bands[f->coded_bands] - ff_celt_freq_bands[f->start_band]);
for (i = f->start_band; i < f->coded_bands; i++) {
int bits = FFMIN(remaining, ff_celt_freq_range[i]);
f->pulses[i] += bits + bandbits * ff_celt_freq_range[i];
remaining -= bits;
}
/* Finally determine the allocation */
for (i = f->start_band; i < f->coded_bands; i++) {
int N = ff_celt_freq_range[i] << f->size;
int prev_extra = extrabits;
f->pulses[i] += extrabits;
if (N > 1) {
int dof; // degrees of freedom
int temp; // dof * channels * log(dof)
int offset; // fine energy quantization offset, i.e.
// extra bits assigned over the standard
// totalbits/dof
int fine_bits, max_bits;
extrabits = FFMAX(0, f->pulses[i] - cap[i]);
f->pulses[i] -= extrabits;
/* intensity stereo makes use of an extra degree of freedom */
dof = N * f->channels + (f->channels == 2 && N > 2 && !f->dual_stereo && i < f->intensity_stereo);
temp = dof * (ff_celt_log_freq_range[i] + (f->size << 3));
offset = (temp >> 1) - dof * CELT_FINE_OFFSET;
if (N == 2) /* dof=2 is the only case that doesn't fit the model */
offset += dof << 1;
/* grant an additional bias for the first and second pulses */
if (f->pulses[i] + offset < 2 * (dof << 3))
offset += temp >> 2;
else if (f->pulses[i] + offset < 3 * (dof << 3))
offset += temp >> 3;
fine_bits = (f->pulses[i] + offset + (dof << 2)) / (dof << 3);
max_bits = FFMIN((f->pulses[i] >> 3) >> (f->channels - 1), CELT_MAX_FINE_BITS);
max_bits = FFMAX(max_bits, 0);
f->fine_bits[i] = av_clip(fine_bits, 0, max_bits);
/* if fine_bits was rounded down or capped,
give priority for the final fine energy pass */
f->fine_priority[i] = (f->fine_bits[i] * (dof << 3) >= f->pulses[i] + offset);
/* the remaining bits are assigned to PVQ */
f->pulses[i] -= f->fine_bits[i] << (f->channels - 1) << 3;
} else {
/* all bits go to fine energy except for the sign bit */
extrabits = FFMAX(0, f->pulses[i] - (f->channels << 3));
f->pulses[i] -= extrabits;
f->fine_bits[i] = 0;
f->fine_priority[i] = 1;
}
/* hand back a limited number of extra fine energy bits to this band */
if (extrabits > 0) {
int fineextra = FFMIN(extrabits >> (f->channels + 2),
CELT_MAX_FINE_BITS - f->fine_bits[i]);
f->fine_bits[i] += fineextra;
fineextra <<= f->channels + 2;
f->fine_priority[i] = (fineextra >= extrabits - prev_extra);
extrabits -= fineextra;
}
}
f->remaining = extrabits;
/* skipped bands dedicate all of their bits for fine energy */
for (; i < f->end_band; i++) {
f->fine_bits[i] = f->pulses[i] >> (f->channels - 1) >> 3;
f->pulses[i] = 0;
f->fine_priority[i] = f->fine_bits[i] < 1;
}
}
static void celt_quant_coarse(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
{
int i, ch;
float alpha, beta, prev[2] = { 0, 0 };
const uint8_t *pmod = ff_celt_coarse_energy_dist[f->size][f->intra];
/* Inter is really just differential coding */
if (opus_rc_tell(rc) + 3 <= f->framebits)
ff_opus_rc_enc_log(rc, f->intra, 3);
else
f->intra = 0;
if (f->intra) {
alpha = 0.0f;
beta = 1.0f - 4915.0f/32768.0f;
} else {
alpha = ff_celt_alpha_coef[f->size];
beta = 1.0f - ff_celt_beta_coef[f->size];
}
for (i = f->start_band; i < f->end_band; i++) {
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *block = &f->block[ch];
const int left = f->framebits - opus_rc_tell(rc);
const float last = FFMAX(-9.0f, s->last_quantized_energy[ch][i]);
float diff = block->energy[i] - prev[ch] - last*alpha;
int q_en = lrintf(diff);
if (left >= 15) {
ff_opus_rc_enc_laplace(rc, &q_en, pmod[i << 1] << 7, pmod[(i << 1) + 1] << 6);
} else if (left >= 2) {
q_en = av_clip(q_en, -1, 1);
ff_opus_rc_enc_cdf(rc, 2*q_en + 3*(q_en < 0), ff_celt_model_energy_small);
} else if (left >= 1) {
q_en = av_clip(q_en, -1, 0);
ff_opus_rc_enc_log(rc, (q_en & 1), 1);
} else q_en = -1;
block->error_energy[i] = q_en - diff;
prev[ch] += beta * q_en;
}
}
}
static void celt_quant_fine(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
{
int i, ch;
for (i = f->start_band; i < f->end_band; i++) {
if (!f->fine_bits[i])
continue;
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *block = &f->block[ch];
int quant, lim = (1 << f->fine_bits[i]);
float offset, diff = 0.5f - block->error_energy[i];
quant = av_clip(floor(diff*lim), 0, lim - 1);
ff_opus_rc_put_raw(rc, quant, f->fine_bits[i]);
offset = 0.5f - ((quant + 0.5f) * (1 << (14 - f->fine_bits[i])) / 16384.0f);
block->error_energy[i] -= offset;
}
}
}
static void celt_quant_final(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
{
int i, ch, priority;
for (priority = 0; priority < 2; priority++) {
for (i = f->start_band; i < f->end_band && (f->framebits - opus_rc_tell(rc)) >= f->channels; i++) {
if (f->fine_priority[i] != priority || f->fine_bits[i] >= CELT_MAX_FINE_BITS)
continue;
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *block = &f->block[ch];
const float err = block->error_energy[i];
const float offset = 0.5f * (1 << (14 - f->fine_bits[i] - 1)) / 16384.0f;
const int sign = FFABS(err + offset) < FFABS(err - offset);
ff_opus_rc_put_raw(rc, sign, 1);
block->error_energy[i] -= offset*(1 - 2*sign);
}
}
}
}
static void celt_quant_bands(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
{
float lowband_scratch[8 * 22];
float norm[2 * 8 * 100];
int totalbits = (f->framebits << 3) - f->anticollapse_needed;
int update_lowband = 1;
int lowband_offset = 0;
int i, j;
for (i = f->start_band; i < f->end_band; i++) {
int band_offset = ff_celt_freq_bands[i] << f->size;
int band_size = ff_celt_freq_range[i] << f->size;
float *X = f->block[0].coeffs + band_offset;
float *Y = (f->channels == 2) ? f->block[1].coeffs + band_offset : NULL;
int consumed = opus_rc_tell_frac(rc);
float *norm2 = norm + 8 * 100;
int effective_lowband = -1;
unsigned int cm[2];
int b;
/* Compute how many bits we want to allocate to this band */
if (i != f->start_band)
f->remaining -= consumed;
f->remaining2 = totalbits - consumed - 1;
if (i <= f->coded_bands - 1) {
int curr_balance = f->remaining / FFMIN(3, f->coded_bands-i);
b = av_clip_uintp2(FFMIN(f->remaining2 + 1, f->pulses[i] + curr_balance), 14);
} else
b = 0;
if (ff_celt_freq_bands[i] - ff_celt_freq_range[i] >= ff_celt_freq_bands[f->start_band] &&
(update_lowband || lowband_offset == 0))
lowband_offset = i;
/* Get a conservative estimate of the collapse_mask's for the bands we're
going to be folding from. */
if (lowband_offset != 0 && (f->spread != CELT_SPREAD_AGGRESSIVE ||
f->blocks > 1 || f->tf_change[i] < 0)) {
int foldstart, foldend;
/* This ensures we never repeat spectral content within one band */
effective_lowband = FFMAX(ff_celt_freq_bands[f->start_band],
ff_celt_freq_bands[lowband_offset] - ff_celt_freq_range[i]);
foldstart = lowband_offset;
while (ff_celt_freq_bands[--foldstart] > effective_lowband);
foldend = lowband_offset - 1;
while (ff_celt_freq_bands[++foldend] < effective_lowband + ff_celt_freq_range[i]);
cm[0] = cm[1] = 0;
for (j = foldstart; j < foldend; j++) {
cm[0] |= f->block[0].collapse_masks[j];
cm[1] |= f->block[f->channels - 1].collapse_masks[j];
}
} else
/* Otherwise, we'll be using the LCG to fold, so all blocks will (almost
always) be non-zero.*/
cm[0] = cm[1] = (1 << f->blocks) - 1;
if (f->dual_stereo && i == f->intensity_stereo) {
/* Switch off dual stereo to do intensity */
f->dual_stereo = 0;
for (j = ff_celt_freq_bands[f->start_band] << f->size; j < band_offset; j++)
norm[j] = (norm[j] + norm2[j]) / 2;
}
if (f->dual_stereo) {
cm[0] = ff_celt_encode_band(f, rc, i, X, NULL, band_size, b / 2, f->blocks,
effective_lowband != -1 ? norm + (effective_lowband << f->size) : NULL, f->size,
norm + band_offset, 0, 1.0f, lowband_scratch, cm[0]);
cm[1] = ff_celt_encode_band(f, rc, i, Y, NULL, band_size, b/2, f->blocks,
effective_lowband != -1 ? norm2 + (effective_lowband << f->size) : NULL, f->size,
norm2 + band_offset, 0, 1.0f, lowband_scratch, cm[1]);
} else {
cm[0] = ff_celt_encode_band(f, rc, i, X, Y, band_size, b, f->blocks,
effective_lowband != -1 ? norm + (effective_lowband << f->size) : NULL, f->size,
norm + band_offset, 0, 1.0f, lowband_scratch, cm[0]|cm[1]);
cm[1] = cm[0];
}
f->block[0].collapse_masks[i] = (uint8_t)cm[0];
f->block[f->channels - 1].collapse_masks[i] = (uint8_t)cm[1];
f->remaining += f->pulses[i] + consumed;
/* Update the folding position only as long as we have 1 bit/sample depth */
update_lowband = (b > band_size << 3);
}
}
static void celt_encode_frame(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
{
int i, ch;
celt_frame_setup_input(s, f);
celt_apply_preemph_filter(s, f);
if (f->pfilter) {
/* Not implemented */
}
celt_frame_mdct(s, f);
f->silence = celt_frame_map_norm_bands(s, f);
if (f->silence) {
f->framebits = 1;
return;
}
ff_opus_rc_enc_log(rc, f->silence, 15);
if (!f->start_band && opus_rc_tell(rc) + 16 <= f->framebits)
ff_opus_rc_enc_log(rc, f->pfilter, 1);
if (f->pfilter) {
/* Not implemented */
}
if (f->size && opus_rc_tell(rc) + 3 <= f->framebits)
ff_opus_rc_enc_log(rc, f->transient, 3);
celt_quant_coarse (s, rc, f);
celt_enc_tf (s, rc, f);
celt_bitalloc (s, rc, f);
celt_quant_fine (s, rc, f);
celt_quant_bands (s, rc, f);
if (f->anticollapse_needed)
ff_opus_rc_put_raw(rc, f->anticollapse, 1);
celt_quant_final(s, rc, f);
for (ch = 0; ch < f->channels; ch++) {
CeltBlock *block = &f->block[ch];
for (i = 0; i < CELT_MAX_BANDS; i++)
s->last_quantized_energy[ch][i] = block->energy[i] + block->error_energy[i];
}
}
static void ff_opus_psy_process(OpusEncContext *s, int end, int *need_more)
{
int max_delay_samples = (s->options.max_delay_ms*s->avctx->sample_rate)/1000;
int max_bsize = FFMIN(OPUS_SAMPLES_TO_BLOCK_SIZE(max_delay_samples), CELT_BLOCK_960);
s->pkt_frames = 1;
s->pkt_framesize = max_bsize;
s->mode = OPUS_MODE_CELT;
s->bandwidth = OPUS_BANDWIDTH_FULLBAND;
*need_more = s->bufqueue.available*s->avctx->frame_size < (max_delay_samples + CELT_OVERLAP);
/* Don't request more if we start being flushed with NULL frames */
*need_more = !end && *need_more;
}
static void ff_opus_psy_celt_frame_setup(OpusEncContext *s, CeltFrame *f, int index)
{
int frame_size = OPUS_BLOCK_SIZE(s->pkt_framesize);
f->avctx = s->avctx;
f->dsp = s->dsp;
f->start_band = (s->mode == OPUS_MODE_HYBRID) ? 17 : 0;
f->end_band = ff_celt_band_end[s->bandwidth];
f->channels = s->channels;
f->size = s->pkt_framesize;
/* Decisions */
f->silence = 0;
f->pfilter = 0;
f->transient = 0;
f->intra = 1;
f->tf_select = 0;
f->anticollapse = 0;
f->alloc_trim = 5;
f->skip_band_floor = f->end_band;
f->intensity_stereo = f->end_band;
f->dual_stereo = 0;
f->spread = CELT_SPREAD_NORMAL;
memset(f->tf_change, 0, sizeof(int)*CELT_MAX_BANDS);
memset(f->alloc_boost, 0, sizeof(int)*CELT_MAX_BANDS);
f->blocks = f->transient ? frame_size/CELT_OVERLAP : 1;
f->framebits = FFALIGN(lrintf((double)s->avctx->bit_rate/(s->avctx->sample_rate/frame_size)), 8);
}
static void opus_packet_assembler(OpusEncContext *s, AVPacket *avpkt)
{
int i, offset, fsize_needed;
/* Write toc */
opus_gen_toc(s, avpkt->data, &offset, &fsize_needed);
for (i = 0; i < s->pkt_frames; i++) {
ff_opus_rc_enc_end(&s->rc[i], avpkt->data + offset, s->frame[i].framebits >> 3);
offset += s->frame[i].framebits >> 3;
}
avpkt->size = offset;
}
/* Used as overlap for the first frame and padding for the last encoded packet */
static AVFrame *spawn_empty_frame(OpusEncContext *s)
{
int i;
AVFrame *f = av_frame_alloc();
if (!f)
return NULL;
f->format = s->avctx->sample_fmt;
f->nb_samples = s->avctx->frame_size;
f->channel_layout = s->avctx->channel_layout;
if (av_frame_get_buffer(f, 4)) {
av_frame_free(&f);
return NULL;
}
for (i = 0; i < s->channels; i++) {
size_t bps = av_get_bytes_per_sample(f->format);
memset(f->extended_data[i], 0, bps*f->nb_samples);
}
return f;
}
static int opus_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
OpusEncContext *s = avctx->priv_data;
int i, ret, frame_size, need_more, alloc_size = 0;
if (frame) { /* Add new frame to queue */
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
return ret;
ff_bufqueue_add(avctx, &s->bufqueue, av_frame_clone(frame));
} else {
if (!s->afq.remaining_samples)
return 0; /* We've been flushed and there's nothing left to encode */
}
/* Run the psychoacoustic system */
ff_opus_psy_process(s, !frame, &need_more);
/* Get more samples for lookahead/encoding */
if (need_more)
return 0;
frame_size = OPUS_BLOCK_SIZE(s->pkt_framesize);
if (!frame) {
/* This can go negative, that's not a problem, we only pad if positive */
int pad_empty = s->pkt_frames*(frame_size/s->avctx->frame_size) - s->bufqueue.available + 1;
/* Pad with empty 2.5 ms frames to whatever framesize was decided,
* this should only happen at the very last flush frame. The frames
* allocated here will be freed (because they have no other references)
* after they get used by celt_frame_setup_input() */
for (i = 0; i < pad_empty; i++) {
AVFrame *empty = spawn_empty_frame(s);
if (!empty)
return AVERROR(ENOMEM);
ff_bufqueue_add(avctx, &s->bufqueue, empty);
}
}
for (i = 0; i < s->pkt_frames; i++) {
ff_opus_rc_enc_init(&s->rc[i]);
ff_opus_psy_celt_frame_setup(s, &s->frame[i], i);
celt_encode_frame(s, &s->rc[i], &s->frame[i]);
alloc_size += s->frame[i].framebits >> 3;
}
/* Worst case toc + the frame lengths if needed */
alloc_size += 2 + s->pkt_frames*2;
if ((ret = ff_alloc_packet2(avctx, avpkt, alloc_size, 0)) < 0)
return ret;
/* Assemble packet */
opus_packet_assembler(s, avpkt);
/* Remove samples from queue and skip if needed */
ff_af_queue_remove(&s->afq, s->pkt_frames*frame_size, &avpkt->pts, &avpkt->duration);
if (s->pkt_frames*frame_size > avpkt->duration) {
uint8_t *side = av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10);
if (!side)
return AVERROR(ENOMEM);
AV_WL32(&side[4], s->pkt_frames*frame_size - avpkt->duration + 120);
}
*got_packet_ptr = 1;
return 0;
}
static av_cold int opus_encode_end(AVCodecContext *avctx)
{
int i;
OpusEncContext *s = avctx->priv_data;
for (i = 0; i < CELT_BLOCK_NB; i++)
ff_mdct15_uninit(&s->mdct[i]);
av_freep(&s->dsp);
av_freep(&s->frame);
av_freep(&s->rc);
ff_af_queue_close(&s->afq);
ff_bufqueue_discard_all(&s->bufqueue);
av_freep(&avctx->extradata);
return 0;
}
static av_cold int opus_encode_init(AVCodecContext *avctx)
{
int i, ch, ret;
OpusEncContext *s = avctx->priv_data;
s->avctx = avctx;
s->channels = avctx->channels;
/* Opus allows us to change the framesize on each packet (and each packet may
* have multiple frames in it) but we can't change the codec's frame size on
* runtime, so fix it to the lowest possible number of samples and use a queue
* to accumulate AVFrames until we have enough to encode whatever the encoder
* decides is the best */
avctx->frame_size = 120;
/* Initial padding will change if SILK is ever supported */
avctx->initial_padding = 120;
avctx->cutoff = !avctx->cutoff ? 20000 : avctx->cutoff;
if (!avctx->bit_rate) {
int coupled = ff_opus_default_coupled_streams[s->channels - 1];
avctx->bit_rate = coupled*(96000) + (s->channels - coupled*2)*(48000);
} else if (avctx->bit_rate < 6000 || avctx->bit_rate > 255000 * s->channels) {
int64_t clipped_rate = av_clip(avctx->bit_rate, 6000, 255000 * s->channels);
av_log(avctx, AV_LOG_ERROR, "Unsupported bitrate %li kbps, clipping to %li kbps\n",
avctx->bit_rate/1000, clipped_rate/1000);
avctx->bit_rate = clipped_rate;
}
/* Frame structs and range coder buffers */
s->frame = av_malloc(OPUS_MAX_FRAMES_PER_PACKET*sizeof(CeltFrame));
if (!s->frame)
return AVERROR(ENOMEM);
s->rc = av_malloc(OPUS_MAX_FRAMES_PER_PACKET*sizeof(OpusRangeCoder));
if (!s->rc)
return AVERROR(ENOMEM);
/* Extradata */
avctx->extradata_size = 19;
avctx->extradata = av_malloc(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
return AVERROR(ENOMEM);
opus_write_extradata(avctx);
ff_af_queue_init(avctx, &s->afq);
if (!(s->dsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT)))
return AVERROR(ENOMEM);
/* I have no idea why a base scaling factor of 68 works, could be the twiddles */
for (i = 0; i < CELT_BLOCK_NB; i++)
if ((ret = ff_mdct15_init(&s->mdct[i], 0, i + 3, 68 << (CELT_BLOCK_NB - 1 - i))))
return AVERROR(ENOMEM);
for (i = 0; i < OPUS_MAX_FRAMES_PER_PACKET; i++)
s->frame[i].block[0].emph_coeff = s->frame[i].block[1].emph_coeff = 0.0f;
/* Zero out previous energy (matters for inter first frame) */
for (ch = 0; ch < s->channels; ch++)
for (i = 0; i < CELT_MAX_BANDS; i++)
s->last_quantized_energy[ch][i] = 0.0f;
/* Allocate an empty frame to use as overlap for the first frame of audio */
ff_bufqueue_add(avctx, &s->bufqueue, spawn_empty_frame(s));
if (!ff_bufqueue_peek(&s->bufqueue, 0))
return AVERROR(ENOMEM);
return 0;
}
#define OPUSENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
static const AVOption opusenc_options[] = {
{ "opus_delay", "Maximum delay (and lookahead) in milliseconds", offsetof(OpusEncContext, options.max_delay_ms), AV_OPT_TYPE_FLOAT, { .dbl = OPUS_MAX_LOOKAHEAD }, 2.5f, OPUS_MAX_LOOKAHEAD, OPUSENC_FLAGS },
{ NULL },
};
static const AVClass opusenc_class = {
.class_name = "Opus encoder",
.item_name = av_default_item_name,
.option = opusenc_options,
.version = LIBAVUTIL_VERSION_INT,
};
static const AVCodecDefault opusenc_defaults[] = {
{ "b", "0" },
{ "compression_level", "10" },
{ NULL },
};
AVCodec ff_opus_encoder = {
.name = "opus",
.long_name = NULL_IF_CONFIG_SMALL("Opus"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_OPUS,
.defaults = opusenc_defaults,
.priv_class = &opusenc_class,
.priv_data_size = sizeof(OpusEncContext),
.init = opus_encode_init,
.encode2 = opus_encode_frame,
.close = opus_encode_end,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
.capabilities = AV_CODEC_CAP_EXPERIMENTAL | AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
.supported_samplerates = (const int []){ 48000, 0 },
.channel_layouts = (const uint64_t []){ AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO, 0 },
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
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