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- /*
- * 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 CELT_VECTORS 11
- #define CELT_ALLOC_STEPS 6
- #define CELT_FINE_OFFSET 21
- #define CELT_MAX_FINE_BITS 8
- #define CELT_NORM_SCALE 16384
- #define CELT_QTHETA_OFFSET 4
- #define CELT_QTHETA_OFFSET_TWOPHASE 16
- #define CELT_DEEMPH_COEFF 0.85000610f
- #define CELT_POSTFILTER_MINPERIOD 15
- #define CELT_ENERGY_SILENCE (-28.0f)
-
- #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
- };
-
- enum OpusBandwidth {
- OPUS_BANDWIDTH_NARROWBAND,
- OPUS_BANDWIDTH_MEDIUMBAND,
- OPUS_BANDWIDTH_WIDEBAND,
- OPUS_BANDWIDTH_SUPERWIDEBAND,
- OPUS_BANDWIDTH_FULLBAND
- };
-
- typedef struct SilkContext SilkContext;
-
- typedef struct CeltContext CeltContext;
-
- 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;
- CeltContext *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 {
- OpusStreamContext *streams;
-
- /* 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);
-
- int ff_celt_init(AVCodecContext *avctx, CeltContext **s, int output_channels);
-
- void ff_celt_free(CeltContext **s);
-
- void ff_celt_flush(CeltContext *s);
-
- int ff_celt_decode_frame(CeltContext *s, OpusRangeCoder *rc,
- float **output, int coded_channels, int frame_size,
- int startband, int endband);
-
- #endif /* AVCODEC_OPUS_H */
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