/* * Opus encoder * Copyright (c) 2017 Rostislav Pehlivanov * * 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; AVTXContext *tx[CELT_BLOCK_NB]; av_tx_fn tx_fn[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->tx_fn[0](s->tx[0], b->coeffs + t, win, sizeof(float)*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->tx_fn[f->size](s->tx[f->size], b->coeffs, win, sizeof(float)); } } 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++) av_tx_uninit(&s->tx[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++) { const float scale = 68 << (CELT_BLOCK_NB - 1 - i); if ((ret = av_tx_init(&s->tx[i], &s->tx_fn[i], AV_TX_FLOAT_MDCT, 0, 15 << (i + 3), &scale, 0))) 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", CODEC_LONG_NAME("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 }, };