/* * Opus decoder/demuxer common functions * 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 */ #ifndef AVCODEC_OPUS_H #define AVCODEC_OPUS_H #include <stdint.h> #include "libavutil/audio_fifo.h" #include "libavutil/float_dsp.h" #include "libavutil/frame.h" #include "libswresample/swresample.h" #include "avcodec.h" #include "opus_rc.h" #define MAX_FRAME_SIZE 1275 #define MAX_FRAMES 48 #define MAX_PACKET_DUR 5760 #define CELT_SHORT_BLOCKSIZE 120 #define CELT_OVERLAP CELT_SHORT_BLOCKSIZE #define CELT_MAX_LOG_BLOCKS 3 #define CELT_MAX_FRAME_SIZE (CELT_SHORT_BLOCKSIZE * (1 << CELT_MAX_LOG_BLOCKS)) #define CELT_MAX_BANDS 21 #define SILK_HISTORY 322 #define SILK_MAX_LPC 16 #define ROUND_MULL(a,b,s) (((MUL64(a, b) >> ((s) - 1)) + 1) >> 1) #define ROUND_MUL16(a,b) ((MUL16(a, b) + 16384) >> 15) #define OPUS_TS_HEADER 0x7FE0 // 0x3ff (11 bits) #define OPUS_TS_MASK 0xFFE0 // top 11 bits static const uint8_t opus_default_extradata[30] = { 'O', 'p', 'u', 's', 'H', 'e', 'a', 'd', 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, }; enum OpusMode { OPUS_MODE_SILK, OPUS_MODE_HYBRID, OPUS_MODE_CELT, OPUS_MODE_NB }; enum OpusBandwidth { OPUS_BANDWIDTH_NARROWBAND, OPUS_BANDWIDTH_MEDIUMBAND, OPUS_BANDWIDTH_WIDEBAND, OPUS_BANDWIDTH_SUPERWIDEBAND, OPUS_BANDWIDTH_FULLBAND, OPUS_BANDWITH_NB }; typedef struct SilkContext SilkContext; typedef struct CeltFrame CeltFrame; typedef struct OpusPacket { int packet_size; /**< packet size */ int data_size; /**< size of the useful data -- packet size - padding */ int code; /**< packet code: specifies the frame layout */ int stereo; /**< whether this packet is mono or stereo */ int vbr; /**< vbr flag */ int config; /**< configuration: tells the audio mode, ** bandwidth, and frame duration */ int frame_count; /**< frame count */ int frame_offset[MAX_FRAMES]; /**< frame offsets */ int frame_size[MAX_FRAMES]; /**< frame sizes */ int frame_duration; /**< frame duration, in samples @ 48kHz */ enum OpusMode mode; /**< mode */ enum OpusBandwidth bandwidth; /**< bandwidth */ } OpusPacket; typedef struct OpusStreamContext { AVCodecContext *avctx; int output_channels; 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]; float redundancy_buf[2][960]; float *redundancy_output[2]; /* data buffers for the final output data */ float *out[2]; int 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; // a mapping between an opus stream and an output channel typedef struct ChannelMap { int stream_idx; int channel_idx; // when a single decoded channel is mapped to multiple output channels, we // write to the first output directly and copy from it to the others // this field is set to 1 for those copied output channels int copy; // this is the index of the output channel to copy from int copy_idx; // this channel is silent int silence; } ChannelMap; typedef struct OpusContext { AVClass *av_class; OpusStreamContext *streams; int apply_phase_inv; /* current output buffers for each streams */ float **out; int *out_size; /* Buffers for synchronizing the streams when they have different * resampling delays */ AVAudioFifo **sync_buffers; /* number of decoded samples for each stream */ int *decoded_samples; int nb_streams; int nb_stereo_streams; AVFloatDSPContext *fdsp; int16_t gain_i; float gain; ChannelMap *channel_maps; } OpusContext; int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size, int self_delimited); int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s); int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels); void ff_silk_free(SilkContext **ps); void ff_silk_flush(SilkContext *s); /** * Decode the LP layer of one Opus frame (which may correspond to several SILK * frames). */ int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc, float *output[2], enum OpusBandwidth bandwidth, int coded_channels, int duration_ms); /* Encode or decode CELT bands */ void ff_celt_quant_bands(CeltFrame *f, OpusRangeCoder *rc); /* Encode or decode CELT bitallocation */ void ff_celt_bitalloc(CeltFrame *f, OpusRangeCoder *rc, int encode); #endif /* AVCODEC_OPUS_H */