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FFmpeg/libavcodec/opusenc.c

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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 "encode.h"
#include "opusenc.h"
#include "opus_pvq.h"
#include "opusenc_psy.h"
#include "opustab.h"
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mem_internal.h"
#include "libavutil/opt.h"
#include "bytestream.h"
#include "audio_frame_queue.h"
#include "codec_internal.h"
typedef struct OpusEncContext {
AVClass *av_class;
OpusEncOptions options;
OpusPsyContext psyctx;
AVCodecContext *avctx;
AudioFrameQueue afq;
AVFloatDSPContext *dsp;
MDCT15Context *mdct[CELT_BLOCK_NB];
CeltPVQ *pvq;
struct FFBufQueue bufqueue;
uint8_t enc_id[64];
int enc_id_bits;
OpusPacketInfo packet;
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->ch_layout.nb_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 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->packet.framesize][s->packet.mode][s->packet.bandwidth];
*fsize_needed = 0;
if (!cfg)
return 1;
if (s->packet.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->packet.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 (int i = 0; i < (s->packet.frames - 1); i++)
*fsize_needed |= (s->frame[i].framebits != s->frame[i + 1].framebits);
tmp = (*fsize_needed) << 7; /* vbr flag */
tmp |= (0) << 6; /* padding flag */
tmp |= s->packet.frames;
*toc++ = tmp;
}
*size = 1 + extended_toc;
return 0;
}
static void celt_frame_setup_input(OpusEncContext *s, CeltFrame *f)
{
AVFrame *cur = NULL;
const int subframesize = s->avctx->frame_size;
int subframes = OPUS_BLOCK_SIZE(s->packet.framesize) / subframesize;
cur = ff_bufqueue_get(&s->bufqueue);
for (int 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 (int sf = 0; sf < subframes; sf++) {
if (sf != (subframes - 1))
cur = ff_bufqueue_get(&s->bufqueue);
else
cur = ff_bufqueue_peek(&s->bufqueue, 0);
for (int 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)
{
const int subframesize = s->avctx->frame_size;
const int subframes = OPUS_BLOCK_SIZE(s->packet.framesize) / subframesize;
/* Filter overlap */
for (int ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
float m = b->emph_coeff;
for (int 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 (int sf = 0; sf < subframes; sf++) {
for (int ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
float m = b->emph_coeff;
for (int 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)
{
float *win = s->scratch, *temp = s->scratch + 1920;
if (f->transient) {
for (int ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
float *src1 = b->overlap;
for (int t = 0; t < f->blocks; t++) {
float *src2 = &b->samples[CELT_OVERLAP*t];
s->dsp->vector_fmul(win, src1, ff_celt_window, 128);
s->dsp->vector_fmul_reverse(&win[CELT_OVERLAP], src2,
ff_celt_window - 8, 128);
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;
memset(win, 0, wlen*sizeof(float));
for (int ch = 0; ch < f->channels; ch++) {
CeltBlock *b = &f->block[ch];
/* Overlap */
s->dsp->vector_fmul(temp, b->overlap, ff_celt_window, 128);
memcpy(win + lap_dst, temp, CELT_OVERLAP*sizeof(float));
/* Samples, flat top window */
memcpy(&win[lap_dst + CELT_OVERLAP], b->samples, rwin*sizeof(float));
/* Samples, windowed */
s->dsp->vector_fmul_reverse(temp, b->samples + rwin,
ff_celt_window - 8, 128);
memcpy(win + lap_dst + blk_len, temp, CELT_OVERLAP*sizeof(float));
s->mdct[f->size]->mdct(s->mdct[f->size], b->coeffs, win, 1);
}
}
for (int ch = 0; ch < f->channels; ch++) {
CeltBlock *block = &f->block[ch];
for (int i = 0; i < CELT_MAX_BANDS; i++) {
float ener = 0.0f;
int band_offset = ff_celt_freq_bands[i] << f->size;
int band_size = ff_celt_freq_range[i] << f->size;
float *coeffs = &block->coeffs[band_offset];
for (int j = 0; j < band_size; j++)
ener += coeffs[j]*coeffs[j];
block->lin_energy[i] = sqrtf(ener) + FLT_EPSILON;
ener = 1.0f/block->lin_energy[i];
for (int j = 0; j < band_size; j++)
coeffs[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);
}
}
}
static void celt_enc_tf(CeltFrame *f, OpusRangeCoder *rc)
{
int 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 (int 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 (int 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_enc_quant_pfilter(OpusRangeCoder *rc, CeltFrame *f)
{
float gain = f->pf_gain;
int txval, octave = f->pf_octave, period = f->pf_period, tapset = f->pf_tapset;
ff_opus_rc_enc_log(rc, f->pfilter, 1);
if (!f->pfilter)
return;
/* Octave */
txval = FFMIN(octave, 6);
ff_opus_rc_enc_uint(rc, txval, 6);
octave = txval;
/* Period */
txval = av_clip(period - (16 << octave) + 1, 0, (1 << (4 + octave)) - 1);
ff_opus_rc_put_raw(rc, period, 4 + octave);
period = txval + (16 << octave) - 1;
/* Gain */
txval = FFMIN(((int)(gain / 0.09375f)) - 1, 7);
ff_opus_rc_put_raw(rc, txval, 3);
gain = 0.09375f * (txval + 1);
/* Tapset */
if ((opus_rc_tell(rc) + 2) <= f->framebits)
ff_opus_rc_enc_cdf(rc, tapset, ff_celt_model_tapset);
else
tapset = 0;
/* Finally create the coeffs */
for (int i = 0; i < 2; i++) {
CeltBlock *block = &f->block[i];
block->pf_period_new = FFMAX(period, CELT_POSTFILTER_MINPERIOD);
block->pf_gains_new[0] = gain * ff_celt_postfilter_taps[tapset][0];
block->pf_gains_new[1] = gain * ff_celt_postfilter_taps[tapset][1];
block->pf_gains_new[2] = gain * ff_celt_postfilter_taps[tapset][2];
}
}
static void exp_quant_coarse(OpusRangeCoder *rc, CeltFrame *f,
float last_energy[][CELT_MAX_BANDS], int intra)
{
float alpha, beta, prev[2] = { 0, 0 };
const uint8_t *pmod = ff_celt_coarse_energy_dist[f->size][intra];
/* Inter is really just differential coding */
if (opus_rc_tell(rc) + 3 <= f->framebits)
ff_opus_rc_enc_log(rc, intra, 3);
else
intra = 0;
if (intra) {
alpha = 0.0f;
beta = 1.0f - (4915.0f/32768.0f);
} else {
alpha = ff_celt_alpha_coef[f->size];
beta = ff_celt_beta_coef[f->size];
}
for (int i = f->start_band; i < f->end_band; i++) {
for (int 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, last_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_coarse(CeltFrame *f, OpusRangeCoder *rc,
float last_energy[][CELT_MAX_BANDS])
{
uint32_t inter, intra;
OPUS_RC_CHECKPOINT_SPAWN(rc);
exp_quant_coarse(rc, f, last_energy, 1);
intra = OPUS_RC_CHECKPOINT_BITS(rc);
OPUS_RC_CHECKPOINT_ROLLBACK(rc);
exp_quant_coarse(rc, f, last_energy, 0);
inter = OPUS_RC_CHECKPOINT_BITS(rc);
if (inter > intra) { /* Unlikely */
OPUS_RC_CHECKPOINT_ROLLBACK(rc);
exp_quant_coarse(rc, f, last_energy, 1);
}
}
static void celt_quant_fine(CeltFrame *f, OpusRangeCoder *rc)
{
for (int i = f->start_band; i < f->end_band; i++) {
if (!f->fine_bits[i])
continue;
for (int 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)
{
for (int priority = 0; priority < 2; priority++) {
for (int 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 (int 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_encode_frame(OpusEncContext *s, OpusRangeCoder *rc,
CeltFrame *f, int index)
{
ff_opus_rc_enc_init(rc);
ff_opus_psy_celt_frame_init(&s->psyctx, f, index);
celt_frame_setup_input(s, f);
if (f->silence) {
if (f->framebits >= 16)
ff_opus_rc_enc_log(rc, 1, 15); /* Silence (if using explicit singalling) */
for (int ch = 0; ch < s->channels; ch++)
memset(s->last_quantized_energy[ch], 0.0f, sizeof(float)*CELT_MAX_BANDS);
return;
}
/* Filters */
celt_apply_preemph_filter(s, f);
if (f->pfilter) {
ff_opus_rc_enc_log(rc, 0, 15);
celt_enc_quant_pfilter(rc, f);
}
/* Transform */
celt_frame_mdct(s, f);
/* Need to handle transient/non-transient switches at any point during analysis */
while (ff_opus_psy_celt_frame_process(&s->psyctx, f, index))
celt_frame_mdct(s, f);
ff_opus_rc_enc_init(rc);
/* Silence */
ff_opus_rc_enc_log(rc, 0, 15);
/* Pitch filter */
if (!f->start_band && opus_rc_tell(rc) + 16 <= f->framebits)
celt_enc_quant_pfilter(rc, f);
/* Transient flag */
if (f->size && opus_rc_tell(rc) + 3 <= f->framebits)
ff_opus_rc_enc_log(rc, f->transient, 3);
/* Main encoding */
celt_quant_coarse (f, rc, s->last_quantized_energy);
celt_enc_tf (f, rc);
ff_celt_bitalloc (f, rc, 1);
celt_quant_fine (f, rc);
ff_celt_quant_bands(f, rc);
/* Anticollapse bit */
if (f->anticollapse_needed)
ff_opus_rc_put_raw(rc, f->anticollapse, 1);
/* Final per-band energy adjustments from leftover bits */
celt_quant_final(s, rc, f);
for (int ch = 0; ch < f->channels; ch++) {
CeltBlock *block = &f->block[ch];
for (int i = 0; i < CELT_MAX_BANDS; i++)
s->last_quantized_energy[ch][i] = block->energy[i] + block->error_energy[i];
}
}
static inline int write_opuslacing(uint8_t *dst, int v)
{
dst[0] = FFMIN(v - FFALIGN(v - 255, 4), v);
dst[1] = v - dst[0] >> 2;
return 1 + (v >= 252);
}
static void opus_packet_assembler(OpusEncContext *s, AVPacket *avpkt)
{
int offset, fsize_needed;
/* Write toc */
opus_gen_toc(s, avpkt->data, &offset, &fsize_needed);
/* Frame sizes if needed */
if (fsize_needed) {
for (int i = 0; i < s->packet.frames - 1; i++) {
offset += write_opuslacing(avpkt->data + offset,
s->frame[i].framebits >> 3);
}
}
/* Packets */
for (int i = 0; i < s->packet.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)
{
AVFrame *f = av_frame_alloc();
int ret;
if (!f)
return NULL;
f->format = s->avctx->sample_fmt;
f->nb_samples = s->avctx->frame_size;
ret = av_channel_layout_copy(&f->ch_layout, &s->avctx->ch_layout);
if (ret < 0) {
av_frame_free(&f);
return NULL;
}
if (av_frame_get_buffer(f, 4)) {
av_frame_free(&f);
return NULL;
}
for (int 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 ret, frame_size, 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 {
ff_opus_psy_signal_eof(&s->psyctx);
if (!s->afq.remaining_samples || !avctx->frame_number)
return 0; /* We've been flushed and there's nothing left to encode */
}
/* Run the psychoacoustic system */
if (ff_opus_psy_process(&s->psyctx, &s->packet))
return 0;
frame_size = OPUS_BLOCK_SIZE(s->packet.framesize);
if (!frame) {
/* This can go negative, that's not a problem, we only pad if positive */
int pad_empty = s->packet.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 (int 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 (int i = 0; i < s->packet.frames; i++) {
celt_encode_frame(s, &s->rc[i], &s->frame[i], i);
alloc_size += s->frame[i].framebits >> 3;
}
/* Worst case toc + the frame lengths if needed */
alloc_size += 2 + s->packet.frames*2;
if ((ret = ff_alloc_packet(avctx, avpkt, alloc_size)) < 0)
return ret;
/* Assemble packet */
opus_packet_assembler(s, avpkt);
/* Update the psychoacoustic system */
ff_opus_psy_postencode_update(&s->psyctx, s->frame, s->rc);
/* Remove samples from queue and skip if needed */
ff_af_queue_remove(&s->afq, s->packet.frames*frame_size, &avpkt->pts, &avpkt->duration);
if (s->packet.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->packet.frames*frame_size - avpkt->duration + 120);
}
*got_packet_ptr = 1;
return 0;
}
static av_cold int opus_encode_end(AVCodecContext *avctx)
{
OpusEncContext *s = avctx->priv_data;
for (int i = 0; i < CELT_BLOCK_NB; i++)
ff_mdct15_uninit(&s->mdct[i]);
ff_celt_pvq_uninit(&s->pvq);
av_freep(&s->dsp);
av_freep(&s->frame);
av_freep(&s->rc);
ff_af_queue_close(&s->afq);
ff_opus_psy_end(&s->psyctx);
ff_bufqueue_discard_all(&s->bufqueue);
return 0;
}
static av_cold int opus_encode_init(AVCodecContext *avctx)
{
int ret, max_frames;
OpusEncContext *s = avctx->priv_data;
s->avctx = avctx;
s->channels = avctx->ch_layout.nb_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;
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 %"PRId64" kbps, clipping to %"PRId64" kbps\n",
avctx->bit_rate/1000, clipped_rate/1000);
avctx->bit_rate = clipped_rate;
}
/* 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 ((ret = ff_celt_pvq_init(&s->pvq, 1)) < 0)
return ret;
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 (int 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);
/* Zero out previous energy (matters for inter first frame) */
for (int ch = 0; ch < s->channels; ch++)
memset(s->last_quantized_energy[ch], 0.0f, sizeof(float)*CELT_MAX_BANDS);
/* 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);
if ((ret = ff_opus_psy_init(&s->psyctx, s->avctx, &s->bufqueue, &s->options)))
return ret;
/* Frame structs and range coder buffers */
max_frames = ceilf(FFMIN(s->options.max_delay_ms, 120.0f)/2.5f);
s->frame = av_malloc(max_frames*sizeof(CeltFrame));
if (!s->frame)
return AVERROR(ENOMEM);
s->rc = av_malloc(max_frames*sizeof(OpusRangeCoder));
if (!s->rc)
return AVERROR(ENOMEM);
for (int i = 0; i < max_frames; i++) {
s->frame[i].dsp = s->dsp;
s->frame[i].avctx = s->avctx;
s->frame[i].seed = 0;
s->frame[i].pvq = s->pvq;
s->frame[i].apply_phase_inv = s->options.apply_phase_inv;
s->frame[i].block[0].emph_coeff = s->frame[i].block[1].emph_coeff = 0.0f;
}
return 0;
}
#define OPUSENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
static const AVOption opusenc_options[] = {
{ "opus_delay", "Maximum delay in milliseconds", offsetof(OpusEncContext, options.max_delay_ms), AV_OPT_TYPE_FLOAT, { .dbl = OPUS_MAX_LOOKAHEAD }, 2.5f, OPUS_MAX_LOOKAHEAD, OPUSENC_FLAGS, "max_delay_ms" },
{ "apply_phase_inv", "Apply intensity stereo phase inversion", offsetof(OpusEncContext, options.apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, OPUSENC_FLAGS, "apply_phase_inv" },
{ 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 FFCodecDefault opusenc_defaults[] = {
{ "b", "0" },
{ "compression_level", "10" },
{ NULL },
};
const FFCodec ff_opus_encoder = {
.p.name = "opus",
.p.long_name = NULL_IF_CONFIG_SMALL("Opus"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_OPUS,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_EXPERIMENTAL,
.defaults = opusenc_defaults,
.p.priv_class = &opusenc_class,
.priv_data_size = sizeof(OpusEncContext),
.init = opus_encode_init,
FF_CODEC_ENCODE_CB(opus_encode_frame),
.close = opus_encode_end,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
.p.supported_samplerates = (const int []){ 48000, 0 },
#if FF_API_OLD_CHANNEL_LAYOUT
.p.channel_layouts = (const uint64_t []){ AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO, 0 },
#endif
.p.ch_layouts = (const AVChannelLayout []){ AV_CHANNEL_LAYOUT_MONO,
AV_CHANNEL_LAYOUT_STEREO, { 0 } },
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};