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FFmpeg/libavcodec/opusenc.c
Andreas Rheinhardt 56e9e0273a avcodec/encode: Always use intermediate buffer in ff_alloc_packet2() 
Up until now, ff_alloc_packet2() has a min_size parameter:
It is supposed to be a lower bound on the final size of the packet
to allocate. If it is not too far from the upper bound (namely,
if it is at least half the upper bound), then ff_alloc_packet2()
already allocates the final, already refcounted packet; if it is
not, then the packet is not refcounted and its data only points to
a buffer owned by the AVCodecContext (in this case, the packet will
be made refcounted in encode_simple_internal() in libavcodec/encode.c).
The goal of this was to avoid data copies and intermediate buffers
if one has a precise lower bound.

Yet those encoders for which precise lower bounds exist have recently
been switched to ff_get_encode_buffer() (which automatically allocates
final buffers), leaving only two encoders to actually set the min_size
to something else than zero (namely aliaspixenc and hapenc). Both of
these encoders use a very low lower bound that is not helpful in any
nontrivial case.

This commit therefore removes the min_size parameter as well as the
codepath in ff_alloc_packet2() for the allocation of final buffers.
Furthermore, the function has been renamed to ff_alloc_packet() and
moved to encode.h alongside ff_get_encode_buffer().

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-06-08 12:52:50 +02:00

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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 "encode.h"
#include "opusenc.h"
#include "opus_pvq.h"
#include "opusenc_psy.h"
#include "opustab.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mem_internal.h"
#include "libavutil/opt.h"
#include "internal.h"
#include "bytestream.h"
#include "audio_frame_queue.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->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();
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 (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->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 AVCodecDefault opusenc_defaults[] = {
{ "b", "0" },
{ "compression_level", "10" },
{ NULL },
};
const 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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