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210 lines
6.5 KiB
210 lines
6.5 KiB
/* |
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* Opus decoder/demuxer common functions |
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* Copyright (c) 2012 Andrew D'Addesio |
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* Copyright (c) 2013-2014 Mozilla Corporation |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#ifndef AVCODEC_OPUS_H |
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#define AVCODEC_OPUS_H |
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#include <stdint.h> |
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#include "libavutil/audio_fifo.h" |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/frame.h" |
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#include "libswresample/swresample.h" |
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#include "avcodec.h" |
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#include "opus_rc.h" |
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#define MAX_FRAME_SIZE 1275 |
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#define MAX_FRAMES 48 |
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#define MAX_PACKET_DUR 5760 |
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#define CELT_SHORT_BLOCKSIZE 120 |
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#define CELT_OVERLAP CELT_SHORT_BLOCKSIZE |
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#define CELT_MAX_LOG_BLOCKS 3 |
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#define CELT_MAX_FRAME_SIZE (CELT_SHORT_BLOCKSIZE * (1 << CELT_MAX_LOG_BLOCKS)) |
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#define CELT_MAX_BANDS 21 |
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#define CELT_VECTORS 11 |
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#define CELT_ALLOC_STEPS 6 |
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#define CELT_FINE_OFFSET 21 |
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#define CELT_MAX_FINE_BITS 8 |
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#define CELT_NORM_SCALE 16384 |
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#define CELT_QTHETA_OFFSET 4 |
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#define CELT_QTHETA_OFFSET_TWOPHASE 16 |
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#define CELT_DEEMPH_COEFF 0.85000610f |
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#define CELT_POSTFILTER_MINPERIOD 15 |
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#define CELT_ENERGY_SILENCE (-28.0f) |
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#define SILK_HISTORY 322 |
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#define SILK_MAX_LPC 16 |
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#define ROUND_MULL(a,b,s) (((MUL64(a, b) >> ((s) - 1)) + 1) >> 1) |
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#define ROUND_MUL16(a,b) ((MUL16(a, b) + 16384) >> 15) |
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#define OPUS_TS_HEADER 0x7FE0 // 0x3ff (11 bits) |
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#define OPUS_TS_MASK 0xFFE0 // top 11 bits |
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static const uint8_t opus_default_extradata[30] = { |
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'O', 'p', 'u', 's', 'H', 'e', 'a', 'd', |
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1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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}; |
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enum OpusMode { |
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OPUS_MODE_SILK, |
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OPUS_MODE_HYBRID, |
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OPUS_MODE_CELT |
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}; |
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enum OpusBandwidth { |
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OPUS_BANDWIDTH_NARROWBAND, |
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OPUS_BANDWIDTH_MEDIUMBAND, |
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OPUS_BANDWIDTH_WIDEBAND, |
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OPUS_BANDWIDTH_SUPERWIDEBAND, |
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OPUS_BANDWIDTH_FULLBAND |
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}; |
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typedef struct SilkContext SilkContext; |
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typedef struct CeltContext CeltContext; |
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typedef struct OpusPacket { |
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int packet_size; /**< packet size */ |
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int data_size; /**< size of the useful data -- packet size - padding */ |
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int code; /**< packet code: specifies the frame layout */ |
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int stereo; /**< whether this packet is mono or stereo */ |
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int vbr; /**< vbr flag */ |
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int config; /**< configuration: tells the audio mode, |
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** bandwidth, and frame duration */ |
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int frame_count; /**< frame count */ |
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int frame_offset[MAX_FRAMES]; /**< frame offsets */ |
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int frame_size[MAX_FRAMES]; /**< frame sizes */ |
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int frame_duration; /**< frame duration, in samples @ 48kHz */ |
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enum OpusMode mode; /**< mode */ |
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enum OpusBandwidth bandwidth; /**< bandwidth */ |
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} OpusPacket; |
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typedef struct OpusStreamContext { |
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AVCodecContext *avctx; |
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int output_channels; |
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OpusRangeCoder rc; |
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OpusRangeCoder redundancy_rc; |
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SilkContext *silk; |
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CeltContext *celt; |
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AVFloatDSPContext *fdsp; |
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float silk_buf[2][960]; |
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float *silk_output[2]; |
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DECLARE_ALIGNED(32, float, celt_buf)[2][960]; |
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float *celt_output[2]; |
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float redundancy_buf[2][960]; |
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float *redundancy_output[2]; |
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/* data buffers for the final output data */ |
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float *out[2]; |
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int out_size; |
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float *out_dummy; |
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int out_dummy_allocated_size; |
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SwrContext *swr; |
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AVAudioFifo *celt_delay; |
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int silk_samplerate; |
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/* number of samples we still want to get from the resampler */ |
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int delayed_samples; |
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OpusPacket packet; |
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int redundancy_idx; |
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} OpusStreamContext; |
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// a mapping between an opus stream and an output channel |
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typedef struct ChannelMap { |
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int stream_idx; |
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int channel_idx; |
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// when a single decoded channel is mapped to multiple output channels, we |
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// write to the first output directly and copy from it to the others |
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// this field is set to 1 for those copied output channels |
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int copy; |
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// this is the index of the output channel to copy from |
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int copy_idx; |
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// this channel is silent |
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int silence; |
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} ChannelMap; |
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typedef struct OpusContext { |
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OpusStreamContext *streams; |
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/* current output buffers for each streams */ |
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float **out; |
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int *out_size; |
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/* Buffers for synchronizing the streams when they have different |
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* resampling delays */ |
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AVAudioFifo **sync_buffers; |
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/* number of decoded samples for each stream */ |
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int *decoded_samples; |
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int nb_streams; |
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int nb_stereo_streams; |
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AVFloatDSPContext *fdsp; |
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int16_t gain_i; |
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float gain; |
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ChannelMap *channel_maps; |
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} OpusContext; |
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int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size, |
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int self_delimited); |
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int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s); |
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int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels); |
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void ff_silk_free(SilkContext **ps); |
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void ff_silk_flush(SilkContext *s); |
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/** |
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* Decode the LP layer of one Opus frame (which may correspond to several SILK |
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* frames). |
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*/ |
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int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc, |
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float *output[2], |
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enum OpusBandwidth bandwidth, int coded_channels, |
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int duration_ms); |
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int ff_celt_init(AVCodecContext *avctx, CeltContext **s, int output_channels); |
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void ff_celt_free(CeltContext **s); |
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void ff_celt_flush(CeltContext *s); |
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int ff_celt_decode_frame(CeltContext *s, OpusRangeCoder *rc, |
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float **output, int coded_channels, int frame_size, |
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int startband, int endband); |
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extern const float ff_celt_window2[120]; |
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#endif /* AVCODEC_OPUS_H */
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