1 /* 2 * Opus decoder/demuxer common functions 3 * Copyright (c) 2012 Andrew D'Addesio 4 * Copyright (c) 2013-2014 Mozilla Corporation 5 * 6 * This file is part of FFmpeg. 7 * 8 * FFmpeg is free software; you can redistribute it and/or 9 * modify it under the terms of the GNU Lesser General Public 10 * License as published by the Free Software Foundation; either 11 * version 2.1 of the License, or (at your option) any later version. 12 * 13 * FFmpeg is distributed in the hope that it will be useful, 14 * but WITHOUT ANY WARRANTY; without even the implied warranty of 15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 16 * Lesser General Public License for more details. 17 * 18 * You should have received a copy of the GNU Lesser General Public 19 * License along with FFmpeg; if not, write to the Free Software 20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 21 */ 22 23 #ifndef AVCODEC_OPUS_H 24 #define AVCODEC_OPUS_H 25 26 #include <stdint.h> 27 28 #include "libavutil/audio_fifo.h" 29 #include "libavutil/float_dsp.h" 30 #include "libavutil/frame.h" 31 #include "libavutil/mem_internal.h" 32 33 #include "libswresample/swresample.h" 34 35 #include "avcodec.h" 36 #include "opus_rc.h" 37 38 #define MAX_FRAME_SIZE 1275 39 #define MAX_FRAMES 48 40 #define MAX_PACKET_DUR 5760 41 42 #define CELT_SHORT_BLOCKSIZE 120 43 #define CELT_OVERLAP CELT_SHORT_BLOCKSIZE 44 #define CELT_MAX_LOG_BLOCKS 3 45 #define CELT_MAX_FRAME_SIZE (CELT_SHORT_BLOCKSIZE * (1 << CELT_MAX_LOG_BLOCKS)) 46 #define CELT_MAX_BANDS 21 47 48 #define SILK_HISTORY 322 49 #define SILK_MAX_LPC 16 50 51 #define ROUND_MULL(a,b,s) (((MUL64(a, b) >> ((s) - 1)) + 1) >> 1) 52 #define ROUND_MUL16(a,b) ((MUL16(a, b) + 16384) >> 15) 53 54 #define OPUS_TS_HEADER 0x7FE0 // 0x3ff (11 bits) 55 #define OPUS_TS_MASK 0xFFE0 // top 11 bits 56 57 static const uint8_t opus_default_extradata[30] = { 58 'O', 'p', 'u', 's', 'H', 'e', 'a', 'd', 59 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 60 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 61 }; 62 63 enum OpusMode { 64 OPUS_MODE_SILK, 65 OPUS_MODE_HYBRID, 66 OPUS_MODE_CELT, 67 68 OPUS_MODE_NB 69 }; 70 71 enum OpusBandwidth { 72 OPUS_BANDWIDTH_NARROWBAND, 73 OPUS_BANDWIDTH_MEDIUMBAND, 74 OPUS_BANDWIDTH_WIDEBAND, 75 OPUS_BANDWIDTH_SUPERWIDEBAND, 76 OPUS_BANDWIDTH_FULLBAND, 77 78 OPUS_BANDWITH_NB 79 }; 80 81 typedef struct SilkContext SilkContext; 82 83 typedef struct CeltFrame CeltFrame; 84 85 typedef struct OpusPacket { 86 int packet_size; /**< packet size */ 87 int data_size; /**< size of the useful data -- packet size - padding */ 88 int code; /**< packet code: specifies the frame layout */ 89 int stereo; /**< whether this packet is mono or stereo */ 90 int vbr; /**< vbr flag */ 91 int config; /**< configuration: tells the audio mode, 92 ** bandwidth, and frame duration */ 93 int frame_count; /**< frame count */ 94 int frame_offset[MAX_FRAMES]; /**< frame offsets */ 95 int frame_size[MAX_FRAMES]; /**< frame sizes */ 96 int frame_duration; /**< frame duration, in samples @ 48kHz */ 97 enum OpusMode mode; /**< mode */ 98 enum OpusBandwidth bandwidth; /**< bandwidth */ 99 } OpusPacket; 100 101 typedef struct OpusStreamContext { 102 AVCodecContext *avctx; 103 int output_channels; 104 105 /* number of decoded samples for this stream */ 106 int decoded_samples; 107 /* current output buffers for this stream */ 108 float *out[2]; 109 int out_size; 110 /* Buffer with samples from this stream for synchronizing 111 * the streams when they have different resampling delays */ 112 AVAudioFifo *sync_buffer; 113 114 OpusRangeCoder rc; 115 OpusRangeCoder redundancy_rc; 116 SilkContext *silk; 117 CeltFrame *celt; 118 AVFloatDSPContext *fdsp; 119 120 float silk_buf[2][960]; 121 float *silk_output[2]; 122 DECLARE_ALIGNED(32, float, celt_buf)[2][960]; 123 float *celt_output[2]; 124 125 DECLARE_ALIGNED(32, float, redundancy_buf)[2][960]; 126 float *redundancy_output[2]; 127 128 /* buffers for the next samples to be decoded */ 129 float *cur_out[2]; 130 int remaining_out_size; 131 132 float *out_dummy; 133 int out_dummy_allocated_size; 134 135 SwrContext *swr; 136 AVAudioFifo *celt_delay; 137 int silk_samplerate; 138 /* number of samples we still want to get from the resampler */ 139 int delayed_samples; 140 141 OpusPacket packet; 142 143 int redundancy_idx; 144 } OpusStreamContext; 145 146 // a mapping between an opus stream and an output channel 147 typedef struct ChannelMap { 148 int stream_idx; 149 int channel_idx; 150 151 // when a single decoded channel is mapped to multiple output channels, we 152 // write to the first output directly and copy from it to the others 153 // this field is set to 1 for those copied output channels 154 int copy; 155 // this is the index of the output channel to copy from 156 int copy_idx; 157 158 // this channel is silent 159 int silence; 160 } ChannelMap; 161 162 typedef struct OpusContext { 163 AVClass *av_class; 164 OpusStreamContext *streams; 165 int apply_phase_inv; 166 167 int nb_streams; 168 int nb_stereo_streams; 169 170 AVFloatDSPContext *fdsp; 171 int16_t gain_i; 172 float gain; 173 174 ChannelMap *channel_maps; 175 } OpusContext; 176 177 int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size, 178 int self_delimited); 179 180 int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s); 181 182 int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels); 183 void ff_silk_free(SilkContext **ps); 184 void ff_silk_flush(SilkContext *s); 185 186 /** 187 * Decode the LP layer of one Opus frame (which may correspond to several SILK 188 * frames). 189 */ 190 int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc, 191 float *output[2], 192 enum OpusBandwidth bandwidth, int coded_channels, 193 int duration_ms); 194 195 /* Encode or decode CELT bands */ 196 void ff_celt_quant_bands(CeltFrame *f, OpusRangeCoder *rc); 197 198 /* Encode or decode CELT bitallocation */ 199 void ff_celt_bitalloc(CeltFrame *f, OpusRangeCoder *rc, int encode); 200 201 #endif /* AVCODEC_OPUS_H */ 202