From b70d7a4ac72d23f3448f3b08b770fdf5f57de222 Mon Sep 17 00:00:00 2001 From: Anton Khirnov Date: Thu, 17 Apr 2014 12:51:03 +0200 Subject: lavc: add a native Opus decoder. Initial implementation by Andrew D'Addesio during GSoC 2012. Completion by Anton Khirnov , sponsored by the Mozilla Corporation. Further contributions by: Christophe Gisquet Janne Grunau Luca Barbato --- libavcodec/opusdec.c | 674 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 674 insertions(+) create mode 100644 libavcodec/opusdec.c (limited to 'libavcodec/opusdec.c') diff --git a/libavcodec/opusdec.c b/libavcodec/opusdec.c new file mode 100644 index 0000000000..bf3a54b16b --- /dev/null +++ b/libavcodec/opusdec.c @@ -0,0 +1,674 @@ +/* + * Opus decoder + * Copyright (c) 2012 Andrew D'Addesio + * Copyright (c) 2013-2014 Mozilla Corporation + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; 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 + +#include "libavutil/attributes.h" +#include "libavutil/audio_fifo.h" +#include "libavutil/channel_layout.h" +#include "libavutil/opt.h" + +#include "libavresample/avresample.h" + +#include "avcodec.h" +#include "celp_filters.h" +#include "fft.h" +#include "get_bits.h" +#include "internal.h" +#include "mathops.h" +#include "opus.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 +}; + +static const uint8_t celt_band_end[] = { 13, 17, 17, 19, 21 }; + +static int get_silk_samplerate(int config) +{ + if (config < 4) + return 8000; + else if (config < 8) + return 12000; + return 16000; +} + +/** + * Range decoder + */ +static int opus_rc_init(OpusRangeCoder *rc, const uint8_t *data, int size) +{ + int ret = init_get_bits8(&rc->gb, data, size); + if (ret < 0) + return ret; + + rc->range = 128; + rc->value = 127 - get_bits(&rc->gb, 7); + rc->total_read_bits = 9; + opus_rc_normalize(rc); + + return 0; +} + +static void opus_raw_init(OpusRangeCoder *rc, const uint8_t *rightend, + unsigned int bytes) +{ + rc->rb.position = rightend; + rc->rb.bytes = bytes; + rc->rb.cachelen = 0; + rc->rb.cacheval = 0; +} + +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 = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples, + NULL, 0, 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->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->out[i], s->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->out[0] += nb_samples; + s->out[1] += nb_samples; + s->out_size -= nb_samples * sizeof(float); + + return 0; +} + +static int opus_init_resample(OpusStreamContext *s) +{ + float delay[16] = { 0.0 }; + uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay }; + int ret; + + av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0); + ret = avresample_open(s->avr); + if (ret < 0) { + av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n"); + return ret; + } + + ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay), + 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; + enum OpusBandwidth bw = s->packet.bandwidth; + + if (s->packet.mode == OPUS_MODE_SILK && + bw == OPUS_BANDWIDTH_MEDIUMBAND) + bw = OPUS_BANDWIDTH_WIDEBAND; + + ret = opus_rc_init(&s->redundancy_rc, data, size); + if (ret < 0) + goto fail; + opus_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, 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 = opus_rc_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 (!avresample_is_open(s->avr)) { + 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 = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, + s->packet.frame_duration, + (uint8_t**)s->silk_output, + sizeof(s->silk_buf[0]), + samples); + if (samples < 0) { + av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n"); + return samples; + } + 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 = opus_rc_p2model(&s->rc, 12); + else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8) + redundancy = 1; + + if (redundancy) { + redundancy_pos = opus_rc_p2model(&s->rc, 1); + + if (s->packet.mode == OPUS_MODE_HYBRID) + redundancy_size = opus_rc_unimodel(&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->out[0], s->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; + } + } + + opus_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, + 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->out[i], s->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->out[i] + samples - 120 + delayed_samples, + s->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->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float)); + opus_fade(s->out[i] + 120 + delayed_samples, + s->redundancy_output[i] + 120, + s->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; + + /* check if we need to flush the resampler */ + if (avresample_is_open(s->avr)) { + if (buf) { + int64_t cur_samplerate; + av_opt_get_int(s->avr, "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->out[0] || + (s->output_channels == 2 && !s->out[1])) { + av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size); + if (!s->out_dummy) + return AVERROR(ENOMEM); + if (!s->out[0]) + s->out[0] = s->out_dummy; + if (!s->out[1]) + s->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; + } + avresample_close(s->avr); + 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->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->out[j] += samples; + s->out_size -= samples * sizeof(float); + } + +finish: + s->out[0] = s->out[1] = NULL; + s->out_size = 0; + + return output_samples; +} + +static int opus_decode_packet(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ + OpusContext *c = avctx->priv_data; + AVFrame *frame = data; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; + int coded_samples = 0; + int decoded_samples = 0; + int i, ret; + + /* 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->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 + c->streams[0].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) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + frame->nb_samples = 0; + + for (i = 0; i < avctx->channels; i++) { + ChannelMap *map = &c->channel_maps[i]; + if (!map->copy) + c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i]; + } + + for (i = 0; i < c->nb_streams; i++) + c->streams[i].out_size = frame->linesize[0]; + + /* decode each sub-packet */ + for (i = 0; i < c->nb_streams; i++) { + OpusStreamContext *s = &c->streams[i]; + + if (i && buf) { + ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n"); + return ret; + } + 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; + if (decoded_samples && ret != decoded_samples) { + av_log(avctx, AV_LOG_ERROR, "Different numbers of decoded samples " + "in a multi-channel stream\n"); + return AVERROR_INVALIDDATA; + } + decoded_samples = ret; + buf += s->packet.packet_size; + buf_size -= s->packet.packet_size; + } + + for (i = 0; i < avctx->channels; i++) { + ChannelMap *map = &c->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->gain_i) { + 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; + int i; + + for (i = 0; i < c->nb_streams; i++) { + OpusStreamContext *s = &c->streams[i]; + + memset(&s->packet, 0, sizeof(s->packet)); + s->delayed_samples = 0; + + if (s->celt_delay) + av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay)); + avresample_close(s->avr); + + ff_silk_flush(s->silk); + ff_celt_flush(s->celt); + } +} + +static av_cold int opus_decode_close(AVCodecContext *avctx) +{ + OpusContext *c = avctx->priv_data; + int i; + + for (i = 0; i < c->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->celt_delay); + avresample_free(&s->avr); + } + + av_freep(&c->streams); + c->nb_streams = 0; + + av_freep(&c->channel_maps); + + return 0; +} + +static av_cold int opus_decode_init(AVCodecContext *avctx) +{ + OpusContext *c = avctx->priv_data; + int ret, i, j; + + avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; + avctx->sample_rate = 48000; + + avpriv_float_dsp_init(&c->fdsp, 0); + + /* find out the channel configuration */ + ret = ff_opus_parse_extradata(avctx, c); + if (ret < 0) + return ret; + + /* allocate and init each independent decoder */ + c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams)); + if (!c->streams) { + c->nb_streams = 0; + ret = AVERROR(ENOMEM); + goto fail; + } + + for (i = 0; i < c->nb_streams; i++) { + OpusStreamContext *s = &c->streams[i]; + uint64_t layout; + + s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1; + + s->avctx = avctx; + + for (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->avr = avresample_alloc_context(); + if (!s->avr) + goto fail; + + layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO; + av_opt_set_int(s->avr, "in_sample_fmt", avctx->sample_fmt, 0); + av_opt_set_int(s->avr, "out_sample_fmt", avctx->sample_fmt, 0); + av_opt_set_int(s->avr, "in_channel_layout", layout, 0); + av_opt_set_int(s->avr, "out_channel_layout", layout, 0); + av_opt_set_int(s->avr, "out_sample_rate", avctx->sample_rate, 0); + + ret = ff_silk_init(avctx, &s->silk, s->output_channels); + if (ret < 0) + goto fail; + + ret = ff_celt_init(avctx, &s->celt, s->output_channels); + if (ret < 0) + goto fail; + + s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt, + s->output_channels, 1024); + if (!s->celt_delay) { + ret = AVERROR(ENOMEM); + goto fail; + } + } + + return 0; +fail: + opus_decode_close(avctx); + return ret; +} + +AVCodec ff_opus_decoder = { + .name = "opus", + .long_name = NULL_IF_CONFIG_SMALL("Opus"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_OPUS, + .priv_data_size = sizeof(OpusContext), + .init = opus_decode_init, + .close = opus_decode_close, + .decode = opus_decode_packet, + .flush = opus_decode_flush, + .capabilities = CODEC_CAP_DR1 | CODEC_CAP_DELAY, +}; -- cgit v1.2.3