forked from KolibriOS/kolibrios
431 lines
12 KiB
C
431 lines
12 KiB
C
/*
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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 "get_bits.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_ilog(i) (av_log2(i) + !!(i))
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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 RawBitsContext {
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const uint8_t *position;
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unsigned int bytes;
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unsigned int cachelen;
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unsigned int cacheval;
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} RawBitsContext;
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typedef struct OpusRangeCoder {
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GetBitContext gb;
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RawBitsContext rb;
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unsigned int range;
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unsigned int value;
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unsigned int total_read_bits;
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} OpusRangeCoder;
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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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static av_always_inline void opus_rc_normalize(OpusRangeCoder *rc)
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{
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while (rc->range <= 1<<23) {
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rc->value = ((rc->value << 8) | (get_bits(&rc->gb, 8) ^ 0xFF)) & ((1u << 31) - 1);
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rc->range <<= 8;
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rc->total_read_bits += 8;
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}
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}
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static av_always_inline void opus_rc_update(OpusRangeCoder *rc, unsigned int scale,
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unsigned int low, unsigned int high,
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unsigned int total)
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{
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rc->value -= scale * (total - high);
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rc->range = low ? scale * (high - low)
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: rc->range - scale * (total - high);
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opus_rc_normalize(rc);
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}
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static av_always_inline unsigned int opus_rc_getsymbol(OpusRangeCoder *rc, const uint16_t *cdf)
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{
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unsigned int k, scale, total, symbol, low, high;
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total = *cdf++;
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scale = rc->range / total;
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symbol = rc->value / scale + 1;
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symbol = total - FFMIN(symbol, total);
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for (k = 0; cdf[k] <= symbol; k++);
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high = cdf[k];
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low = k ? cdf[k-1] : 0;
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opus_rc_update(rc, scale, low, high, total);
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return k;
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}
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static av_always_inline unsigned int opus_rc_p2model(OpusRangeCoder *rc, unsigned int bits)
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{
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unsigned int k, scale;
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scale = rc->range >> bits; // in this case, scale = symbol
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if (rc->value >= scale) {
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rc->value -= scale;
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rc->range -= scale;
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k = 0;
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} else {
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rc->range = scale;
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k = 1;
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}
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opus_rc_normalize(rc);
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return k;
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}
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/**
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* CELT: estimate bits of entropy that have thus far been consumed for the
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* current CELT frame, to integer and fractional (1/8th bit) precision
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*/
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static av_always_inline unsigned int opus_rc_tell(const OpusRangeCoder *rc)
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{
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return rc->total_read_bits - av_log2(rc->range) - 1;
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}
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static av_always_inline unsigned int opus_rc_tell_frac(const OpusRangeCoder *rc)
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{
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unsigned int i, total_bits, rcbuffer, range;
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total_bits = rc->total_read_bits << 3;
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rcbuffer = av_log2(rc->range) + 1;
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range = rc->range >> (rcbuffer-16);
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for (i = 0; i < 3; i++) {
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int bit;
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range = range * range >> 15;
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bit = range >> 16;
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rcbuffer = rcbuffer << 1 | bit;
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range >>= bit;
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}
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return total_bits - rcbuffer;
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}
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/**
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* CELT: read 1-25 raw bits at the end of the frame, backwards byte-wise
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*/
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static av_always_inline unsigned int opus_getrawbits(OpusRangeCoder *rc, unsigned int count)
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{
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unsigned int value = 0;
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while (rc->rb.bytes && rc->rb.cachelen < count) {
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rc->rb.cacheval |= *--rc->rb.position << rc->rb.cachelen;
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rc->rb.cachelen += 8;
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rc->rb.bytes--;
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}
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value = av_mod_uintp2(rc->rb.cacheval, count);
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rc->rb.cacheval >>= count;
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rc->rb.cachelen -= count;
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rc->total_read_bits += count;
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return value;
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}
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/**
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* CELT: read a uniform distribution
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*/
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static av_always_inline unsigned int opus_rc_unimodel(OpusRangeCoder *rc, unsigned int size)
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{
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unsigned int bits, k, scale, total;
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bits = opus_ilog(size - 1);
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total = (bits > 8) ? ((size - 1) >> (bits - 8)) + 1 : size;
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scale = rc->range / total;
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k = rc->value / scale + 1;
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k = total - FFMIN(k, total);
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opus_rc_update(rc, scale, k, k + 1, total);
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if (bits > 8) {
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k = k << (bits - 8) | opus_getrawbits(rc, bits - 8);
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return FFMIN(k, size - 1);
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} else
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return k;
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}
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static av_always_inline int opus_rc_laplace(OpusRangeCoder *rc, unsigned int symbol, int decay)
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{
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/* extends the range coder to model a Laplace distribution */
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int value = 0;
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unsigned int scale, low = 0, center;
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scale = rc->range >> 15;
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center = rc->value / scale + 1;
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center = (1 << 15) - FFMIN(center, 1 << 15);
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if (center >= symbol) {
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value++;
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low = symbol;
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symbol = 1 + ((32768 - 32 - symbol) * (16384-decay) >> 15);
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while (symbol > 1 && center >= low + 2 * symbol) {
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value++;
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symbol *= 2;
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low += symbol;
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symbol = (((symbol - 2) * decay) >> 15) + 1;
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}
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if (symbol <= 1) {
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int distance = (center - low) >> 1;
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value += distance;
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low += 2 * distance;
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}
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if (center < low + symbol)
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value *= -1;
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else
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low += symbol;
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}
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opus_rc_update(rc, scale, low, FFMIN(low + symbol, 32768), 32768);
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return value;
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}
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static av_always_inline unsigned int opus_rc_stepmodel(OpusRangeCoder *rc, int k0)
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{
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/* Use a probability of 3 up to itheta=8192 and then use 1 after */
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unsigned int k, scale, symbol, total = (k0+1)*3 + k0;
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scale = rc->range / total;
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symbol = rc->value / scale + 1;
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symbol = total - FFMIN(symbol, total);
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k = (symbol < (k0+1)*3) ? symbol/3 : symbol - (k0+1)*2;
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opus_rc_update(rc, scale, (k <= k0) ? 3*(k+0) : (k-1-k0) + 3*(k0+1),
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(k <= k0) ? 3*(k+1) : (k-0-k0) + 3*(k0+1), total);
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return k;
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}
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static av_always_inline unsigned int opus_rc_trimodel(OpusRangeCoder *rc, int qn)
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{
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unsigned int k, scale, symbol, total, low, center;
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total = ((qn>>1) + 1) * ((qn>>1) + 1);
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scale = rc->range / total;
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center = rc->value / scale + 1;
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center = total - FFMIN(center, total);
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if (center < total >> 1) {
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k = (ff_sqrt(8 * center + 1) - 1) >> 1;
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low = k * (k + 1) >> 1;
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symbol = k + 1;
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} else {
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k = (2*(qn + 1) - ff_sqrt(8*(total - center - 1) + 1)) >> 1;
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low = total - ((qn + 1 - k) * (qn + 2 - k) >> 1);
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symbol = qn + 1 - k;
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}
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opus_rc_update(rc, scale, low, low + symbol, total);
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return k;
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}
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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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