mirror of
https://github.com/hrydgard/ppsspp.git
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The ffmpeg submodule held prebuilt libraries for every platform next to the sources (about 1.5 GB checked out), and nothing uses those any more. Only the needed sources are copied in, from the submodule's commit, along with every header and template they include. The path is new, so checking out commits on either side of this works. It's also cut down to what PPSSPP uses: the H.264, AAC, MP3, ATRAC3 and ATRAC3+ decoders, the H.264 parser, the demuxers as before, and the FFV1 encoder, AVI muxer and file protocol for AVI dumping. config.h takes its CONFIG_ values from FFmpeg 3.0.2's own configure run with just those enabled. The raw H.264 demuxer stays even though nothing opens raw H.264: probing PSMF data uses it to identify the video stream's codec, which otherwise comes out as none. On top of that: - The files nothing links (configure would build them, but they're never pulled out of the static library, going by the GCC link map) are left out: mostly libavutil's crypto and hashing and the alternative DCT/IDCT/FFT versions. - libswresample isn't built, now that nothing uses it. - The H.264 decoder only does 8-bit video, which is all the PSP has. The SPS parser rejects any other bit depth, and the DSP files no longer build their 9- to 16-bit variants, which the linker kept because the init functions pick one by bit depth at run time. That's 160 files to compile instead of 277, and 391 of FFmpeg's vendored, about 5.8 MB. The README in ext/ffmpeg has what was done, the PPSSPP patches in the copy, how to add a component, and the tradeoffs. Co-Authored-By: Claude Opus 5.5 (1M context) <noreply@anthropic.com>
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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