/* * Opus decoder * Copyright (c) 2012 Andrew D'Addesio * Copyright (c) 2013-2014 Mozilla Corporation * * 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 */ /** * @file * Opus decoder * @author Andrew D'Addesio, Anton Khirnov * * Codec homepage: http://opus-codec.org/ * Specification: http://tools.ietf.org/html/rfc6716 * Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03 * * Ogg-contained .opus files can be produced with opus-tools: * http://git.xiph.org/?p=opus-tools.git */ #include <stdint.h> #include "libavutil/attributes.h" #include "libavutil/audio_fifo.h" #include "libavutil/channel_layout.h" #include "libavutil/ffmath.h" #include "libavutil/float_dsp.h" #include "libavutil/frame.h" #include "libavutil/mem_internal.h" #include "libavutil/opt.h" #include "libswresample/swresample.h" #include "avcodec.h" #include "codec_internal.h" #include "decode.h" #include "opus.h" #include "opustab.h" #include "opus_celt.h" #include "opus_parse.h" #include "opus_rc.h" #include "opus_silk.h" static const uint16_t silk_frame_duration_ms[16] = { 10, 20, 40, 60, 10, 20, 40, 60, 10, 20, 40, 60, 10, 20, 10, 20, }; /* number of samples of silence to feed to the resampler * at the beginning */ static const int silk_resample_delay[] = { 4, 8, 11, 11, 11 }; typedef struct OpusStreamContext { AVCodecContext *avctx; int output_channels; /* number of decoded samples for this stream */ int decoded_samples; /* current output buffers for this stream */ float *out[2]; int out_size; /* Buffer with samples from this stream for synchronizing * the streams when they have different resampling delays */ AVAudioFifo *sync_buffer; OpusRangeCoder rc; OpusRangeCoder redundancy_rc; SilkContext *silk; CeltFrame *celt; AVFloatDSPContext *fdsp; float silk_buf[2][960]; float *silk_output[2]; DECLARE_ALIGNED(32, float, celt_buf)[2][960]; float *celt_output[2]; DECLARE_ALIGNED(32, float, redundancy_buf)[2][960]; float *redundancy_output[2]; /* buffers for the next samples to be decoded */ float *cur_out[2]; int remaining_out_size; float *out_dummy; int out_dummy_allocated_size; SwrContext *swr; AVAudioFifo *celt_delay; int silk_samplerate; /* number of samples we still want to get from the resampler */ int delayed_samples; OpusPacket packet; int redundancy_idx; } OpusStreamContext; typedef struct OpusContext { AVClass *av_class; struct OpusStreamContext *streams; int apply_phase_inv; AVFloatDSPContext *fdsp; float gain; OpusParseContext p; } OpusContext; static int get_silk_samplerate(int config) { if (config < 4) return 8000; else if (config < 8) return 12000; return 16000; } static void opus_fade(float *out, const float *in1, const float *in2, const float *window, int len) { int i; for (i = 0; i < len; i++) out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]); } static int opus_flush_resample(OpusStreamContext *s, int nb_samples) { int celt_size = av_audio_fifo_size(s->celt_delay); int ret, i; ret = swr_convert(s->swr, (uint8_t**)s->cur_out, nb_samples, NULL, 0); if (ret < 0) return ret; else if (ret != nb_samples) { av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n", ret); return AVERROR_BUG; } if (celt_size) { if (celt_size != nb_samples) { av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n"); return AVERROR_BUG; } av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples); for (i = 0; i < s->output_channels; i++) { s->fdsp->vector_fmac_scalar(s->cur_out[i], s->celt_output[i], 1.0, nb_samples); } } if (s->redundancy_idx) { for (i = 0; i < s->output_channels; i++) opus_fade(s->cur_out[i], s->cur_out[i], s->redundancy_output[i] + 120 + s->redundancy_idx, ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx); s->redundancy_idx = 0; } s->cur_out[0] += nb_samples; s->cur_out[1] += nb_samples; s->remaining_out_size -= nb_samples * sizeof(float); return 0; } static int opus_init_resample(OpusStreamContext *s) { static const float delay[16] = { 0.0 }; const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay }; int ret; av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0); ret = swr_init(s->swr); if (ret < 0) { av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n"); return ret; } ret = swr_convert(s->swr, NULL, 0, delayptr, silk_resample_delay[s->packet.bandwidth]); if (ret < 0) { av_log(s->avctx, AV_LOG_ERROR, "Error feeding initial silence to the resampler.\n"); return ret; } return 0; } static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size) { int ret = ff_opus_rc_dec_init(&s->redundancy_rc, data, size); if (ret < 0) goto fail; ff_opus_rc_dec_raw_init(&s->redundancy_rc, data + size, size); ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc, s->redundancy_output, s->packet.stereo + 1, 240, 0, ff_celt_band_end[s->packet.bandwidth]); if (ret < 0) goto fail; return 0; fail: av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n"); return ret; } static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size) { int samples = s->packet.frame_duration; int redundancy = 0; int redundancy_size, redundancy_pos; int ret, i, consumed; int delayed_samples = s->delayed_samples; ret = ff_opus_rc_dec_init(&s->rc, data, size); if (ret < 0) return ret; /* decode the silk frame */ if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) { if (!swr_is_initialized(s->swr)) { ret = opus_init_resample(s); if (ret < 0) return ret; } samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output, FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND), s->packet.stereo + 1, silk_frame_duration_ms[s->packet.config]); if (samples < 0) { av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n"); return samples; } samples = swr_convert(s->swr, (uint8_t**)s->cur_out, s->packet.frame_duration, (const uint8_t**)s->silk_output, samples); if (samples < 0) { av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n"); return samples; } av_assert2((samples & 7) == 0); s->delayed_samples += s->packet.frame_duration - samples; } else ff_silk_flush(s->silk); // decode redundancy information consumed = opus_rc_tell(&s->rc); if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8) redundancy = ff_opus_rc_dec_log(&s->rc, 12); else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8) redundancy = 1; if (redundancy) { redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1); if (s->packet.mode == OPUS_MODE_HYBRID) redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2; else redundancy_size = size - (consumed + 7) / 8; size -= redundancy_size; if (size < 0) { av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n"); return AVERROR_INVALIDDATA; } if (redundancy_pos) { ret = opus_decode_redundancy(s, data + size, redundancy_size); if (ret < 0) return ret; ff_celt_flush(s->celt); } } /* decode the CELT frame */ if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) { float *out_tmp[2] = { s->cur_out[0], s->cur_out[1] }; float **dst = (s->packet.mode == OPUS_MODE_CELT) ? out_tmp : s->celt_output; int celt_output_samples = samples; int delay_samples = av_audio_fifo_size(s->celt_delay); if (delay_samples) { if (s->packet.mode == OPUS_MODE_HYBRID) { av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples); for (i = 0; i < s->output_channels; i++) { s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0, delay_samples); out_tmp[i] += delay_samples; } celt_output_samples -= delay_samples; } else { av_log(s->avctx, AV_LOG_WARNING, "Spurious CELT delay samples present.\n"); av_audio_fifo_drain(s->celt_delay, delay_samples); if (s->avctx->err_recognition & AV_EF_EXPLODE) return AVERROR_BUG; } } ff_opus_rc_dec_raw_init(&s->rc, data + size, size); ret = ff_celt_decode_frame(s->celt, &s->rc, dst, s->packet.stereo + 1, s->packet.frame_duration, (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0, ff_celt_band_end[s->packet.bandwidth]); if (ret < 0) return ret; if (s->packet.mode == OPUS_MODE_HYBRID) { int celt_delay = s->packet.frame_duration - celt_output_samples; void *delaybuf[2] = { s->celt_output[0] + celt_output_samples, s->celt_output[1] + celt_output_samples }; for (i = 0; i < s->output_channels; i++) { s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0, celt_output_samples); } ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay); if (ret < 0) return ret; } } else ff_celt_flush(s->celt); if (s->redundancy_idx) { for (i = 0; i < s->output_channels; i++) opus_fade(s->cur_out[i], s->cur_out[i], s->redundancy_output[i] + 120 + s->redundancy_idx, ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx); s->redundancy_idx = 0; } if (redundancy) { if (!redundancy_pos) { ff_celt_flush(s->celt); ret = opus_decode_redundancy(s, data + size, redundancy_size); if (ret < 0) return ret; for (i = 0; i < s->output_channels; i++) { opus_fade(s->cur_out[i] + samples - 120 + delayed_samples, s->cur_out[i] + samples - 120 + delayed_samples, s->redundancy_output[i] + 120, ff_celt_window2, 120 - delayed_samples); if (delayed_samples) s->redundancy_idx = 120 - delayed_samples; } } else { for (i = 0; i < s->output_channels; i++) { memcpy(s->cur_out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float)); opus_fade(s->cur_out[i] + 120 + delayed_samples, s->redundancy_output[i] + 120, s->cur_out[i] + 120 + delayed_samples, ff_celt_window2, 120); } } } return samples; } static int opus_decode_subpacket(OpusStreamContext *s, const uint8_t *buf, int buf_size, int nb_samples) { int output_samples = 0; int flush_needed = 0; int i, j, ret; s->cur_out[0] = s->out[0]; s->cur_out[1] = s->out[1]; s->remaining_out_size = s->out_size; /* check if we need to flush the resampler */ if (swr_is_initialized(s->swr)) { if (buf) { int64_t cur_samplerate; av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate); flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate); } else { flush_needed = !!s->delayed_samples; } } if (!buf && !flush_needed) return 0; /* use dummy output buffers if the channel is not mapped to anything */ if (!s->cur_out[0] || (s->output_channels == 2 && !s->cur_out[1])) { av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->remaining_out_size); if (!s->out_dummy) return AVERROR(ENOMEM); if (!s->cur_out[0]) s->cur_out[0] = s->out_dummy; if (!s->cur_out[1]) s->cur_out[1] = s->out_dummy; } /* flush the resampler if necessary */ if (flush_needed) { ret = opus_flush_resample(s, s->delayed_samples); if (ret < 0) { av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n"); return ret; } swr_close(s->swr); output_samples += s->delayed_samples; s->delayed_samples = 0; if (!buf) goto finish; } /* decode all the frames in the packet */ for (i = 0; i < s->packet.frame_count; i++) { int size = s->packet.frame_size[i]; int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size); if (samples < 0) { av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n"); if (s->avctx->err_recognition & AV_EF_EXPLODE) return samples; for (j = 0; j < s->output_channels; j++) memset(s->cur_out[j], 0, s->packet.frame_duration * sizeof(float)); samples = s->packet.frame_duration; } output_samples += samples; for (j = 0; j < s->output_channels; j++) s->cur_out[j] += samples; s->remaining_out_size -= samples * sizeof(float); } finish: s->cur_out[0] = s->cur_out[1] = NULL; s->remaining_out_size = 0; return output_samples; } static int opus_decode_packet(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt) { OpusContext *c = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; int coded_samples = 0; int decoded_samples = INT_MAX; int delayed_samples = 0; int i, ret; /* calculate the number of delayed samples */ for (int i = 0; i < c->p.nb_streams; i++) { OpusStreamContext *s = &c->streams[i]; s->out[0] = s->out[1] = NULL; delayed_samples = FFMAX(delayed_samples, s->delayed_samples + av_audio_fifo_size(s->sync_buffer)); } /* decode the header of the first sub-packet to find out the sample count */ if (buf) { OpusPacket *pkt = &c->streams[0].packet; ret = ff_opus_parse_packet(pkt, buf, buf_size, c->p.nb_streams > 1); if (ret < 0) { av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n"); return ret; } coded_samples += pkt->frame_count * pkt->frame_duration; c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config); } frame->nb_samples = coded_samples + delayed_samples; /* no input or buffered data => nothing to do */ if (!frame->nb_samples) { *got_frame_ptr = 0; return 0; } /* setup the data buffers */ ret = ff_get_buffer(avctx, frame, 0); if (ret < 0) return ret; frame->nb_samples = 0; for (i = 0; i < avctx->ch_layout.nb_channels; i++) { ChannelMap *map = &c->p.channel_maps[i]; if (!map->copy) c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i]; } /* read the data from the sync buffers */ for (int i = 0; i < c->p.nb_streams; i++) { OpusStreamContext *s = &c->streams[i]; float **out = s->out; int sync_size = av_audio_fifo_size(s->sync_buffer); float sync_dummy[32]; int out_dummy = (!out[0]) | ((!out[1]) << 1); if (!out[0]) out[0] = sync_dummy; if (!out[1]) out[1] = sync_dummy; if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy)) return AVERROR_BUG; ret = av_audio_fifo_read(s->sync_buffer, (void**)out, sync_size); if (ret < 0) return ret; if (out_dummy & 1) out[0] = NULL; else out[0] += ret; if (out_dummy & 2) out[1] = NULL; else out[1] += ret; s->out_size = frame->linesize[0] - ret * sizeof(float); } /* decode each sub-packet */ for (int i = 0; i < c->p.nb_streams; i++) { OpusStreamContext *s = &c->streams[i]; if (i && buf) { ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->p.nb_streams - 1); if (ret < 0) { av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n"); return ret; } if (coded_samples != s->packet.frame_count * s->packet.frame_duration) { av_log(avctx, AV_LOG_ERROR, "Mismatching coded sample count in substream %d.\n", i); return AVERROR_INVALIDDATA; } s->silk_samplerate = get_silk_samplerate(s->packet.config); } ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size, coded_samples); if (ret < 0) return ret; s->decoded_samples = ret; decoded_samples = FFMIN(decoded_samples, ret); buf += s->packet.packet_size; buf_size -= s->packet.packet_size; } /* buffer the extra samples */ for (int i = 0; i < c->p.nb_streams; i++) { OpusStreamContext *s = &c->streams[i]; int buffer_samples = s->decoded_samples - decoded_samples; if (buffer_samples) { float *buf[2] = { s->out[0] ? s->out[0] : (float*)frame->extended_data[0], s->out[1] ? s->out[1] : (float*)frame->extended_data[0] }; buf[0] += decoded_samples; buf[1] += decoded_samples; ret = av_audio_fifo_write(s->sync_buffer, (void**)buf, buffer_samples); if (ret < 0) return ret; } } for (i = 0; i < avctx->ch_layout.nb_channels; i++) { ChannelMap *map = &c->p.channel_maps[i]; /* handle copied channels */ if (map->copy) { memcpy(frame->extended_data[i], frame->extended_data[map->copy_idx], frame->linesize[0]); } else if (map->silence) { memset(frame->extended_data[i], 0, frame->linesize[0]); } if (c->p.gain_i && decoded_samples > 0) { c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i], (float*)frame->extended_data[i], c->gain, FFALIGN(decoded_samples, 8)); } } frame->nb_samples = decoded_samples; *got_frame_ptr = !!decoded_samples; return avpkt->size; } static av_cold void opus_decode_flush(AVCodecContext *ctx) { OpusContext *c = ctx->priv_data; for (int i = 0; i < c->p.nb_streams; i++) { OpusStreamContext *s = &c->streams[i]; memset(&s->packet, 0, sizeof(s->packet)); s->delayed_samples = 0; av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay)); swr_close(s->swr); av_audio_fifo_drain(s->sync_buffer, av_audio_fifo_size(s->sync_buffer)); ff_silk_flush(s->silk); ff_celt_flush(s->celt); } } static av_cold int opus_decode_close(AVCodecContext *avctx) { OpusContext *c = avctx->priv_data; for (int i = 0; i < c->p.nb_streams; i++) { OpusStreamContext *s = &c->streams[i]; ff_silk_free(&s->silk); ff_celt_free(&s->celt); av_freep(&s->out_dummy); s->out_dummy_allocated_size = 0; av_audio_fifo_free(s->sync_buffer); av_audio_fifo_free(s->celt_delay); swr_free(&s->swr); } av_freep(&c->streams); c->p.nb_streams = 0; av_freep(&c->p.channel_maps); av_freep(&c->fdsp); return 0; } static av_cold int opus_decode_init(AVCodecContext *avctx) { OpusContext *c = avctx->priv_data; int ret; avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; avctx->sample_rate = 48000; c->fdsp = avpriv_float_dsp_alloc(0); if (!c->fdsp) return AVERROR(ENOMEM); /* find out the channel configuration */ ret = ff_opus_parse_extradata(avctx, &c->p); if (ret < 0) return ret; if (c->p.gain_i) c->gain = ff_exp10(c->p.gain_i / (20.0 * 256)); /* allocate and init each independent decoder */ c->streams = av_calloc(c->p.nb_streams, sizeof(*c->streams)); if (!c->streams) { c->p.nb_streams = 0; return AVERROR(ENOMEM); } for (int i = 0; i < c->p.nb_streams; i++) { OpusStreamContext *s = &c->streams[i]; AVChannelLayout layout; s->output_channels = (i < c->p.nb_stereo_streams) ? 2 : 1; s->avctx = avctx; for (int j = 0; j < s->output_channels; j++) { s->silk_output[j] = s->silk_buf[j]; s->celt_output[j] = s->celt_buf[j]; s->redundancy_output[j] = s->redundancy_buf[j]; } s->fdsp = c->fdsp; s->swr =swr_alloc(); if (!s->swr) return AVERROR(ENOMEM); layout = (s->output_channels == 1) ? (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO : (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO; av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0); av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0); av_opt_set_chlayout(s->swr, "in_chlayout", &layout, 0); av_opt_set_chlayout(s->swr, "out_chlayout", &layout, 0); av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0); av_opt_set_int(s->swr, "filter_size", 16, 0); ret = ff_silk_init(avctx, &s->silk, s->output_channels); if (ret < 0) return ret; ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv); if (ret < 0) return ret; s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt, s->output_channels, 1024); if (!s->celt_delay) return AVERROR(ENOMEM); s->sync_buffer = av_audio_fifo_alloc(avctx->sample_fmt, s->output_channels, 32); if (!s->sync_buffer) return AVERROR(ENOMEM); } return 0; } #define OFFSET(x) offsetof(OpusContext, x) #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM static const AVOption opus_options[] = { { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AD }, { NULL }, }; static const AVClass opus_class = { .class_name = "Opus Decoder", .item_name = av_default_item_name, .option = opus_options, .version = LIBAVUTIL_VERSION_INT, }; const FFCodec ff_opus_decoder = { .p.name = "opus", CODEC_LONG_NAME("Opus"), .p.priv_class = &opus_class, .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_OPUS, .priv_data_size = sizeof(OpusContext), .init = opus_decode_init, .close = opus_decode_close, FF_CODEC_DECODE_CB(opus_decode_packet), .flush = opus_decode_flush, .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY | AV_CODEC_CAP_CHANNEL_CONF, .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, };