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https://github.com/nyanmisaka/ffmpeg-rockchip.git
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268 lines
6.9 KiB
C
268 lines
6.9 KiB
C
/*
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* G.723.1 common header and data tables
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* Copyright (c) 2006 Benjamin Larsson
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* Copyright (c) 2010 Mohamed Naufal Basheer
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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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/**
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* @file
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* G.723.1 types, functions and data tables
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*/
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#ifndef AVCODEC_G723_1_H
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#define AVCODEC_G723_1_H
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#include <stdint.h>
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#include "libavutil/log.h"
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#define SUBFRAMES 4
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#define SUBFRAME_LEN 60
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#define FRAME_LEN (SUBFRAME_LEN << 2)
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#define HALF_FRAME_LEN (FRAME_LEN / 2)
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#define LPC_FRAME (HALF_FRAME_LEN + SUBFRAME_LEN)
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#define LPC_ORDER 10
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#define LSP_BANDS 3
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#define LSP_CB_SIZE 256
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#define PITCH_MIN 18
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#define PITCH_MAX (PITCH_MIN + 127)
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#define PITCH_ORDER 5
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#define GRID_SIZE 2
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#define PULSE_MAX 6
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#define GAIN_LEVELS 24
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#define COS_TBL_SIZE 512
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/**
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* Bitexact implementation of 2ab scaled by 1/2^16.
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*
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* @param a 32 bit multiplicand
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* @param b 16 bit multiplier
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*/
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#define MULL2(a, b) \
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((((a) >> 16) * (b) * 2) + (((a) & 0xffff) * (b) >> 15))
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/**
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* G723.1 frame types
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*/
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enum FrameType {
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ACTIVE_FRAME, ///< Active speech
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SID_FRAME, ///< Silence Insertion Descriptor frame
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UNTRANSMITTED_FRAME
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};
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/**
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* G723.1 rate values
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*/
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enum Rate {
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RATE_6300,
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RATE_5300
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};
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/**
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* G723.1 unpacked data subframe
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*/
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typedef struct G723_1_Subframe {
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int ad_cb_lag; ///< adaptive codebook lag
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int ad_cb_gain;
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int dirac_train;
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int pulse_sign;
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int grid_index;
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int amp_index;
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int pulse_pos;
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} G723_1_Subframe;
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/**
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* Pitch postfilter parameters
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*/
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typedef struct PPFParam {
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int index; ///< postfilter backward/forward lag
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int16_t opt_gain; ///< optimal gain
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int16_t sc_gain; ///< scaling gain
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} PPFParam;
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/**
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* Harmonic filter parameters
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*/
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typedef struct HFParam {
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int index;
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int gain;
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} HFParam;
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/**
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* Optimized fixed codebook excitation parameters
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*/
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typedef struct FCBParam {
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int min_err;
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int amp_index;
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int grid_index;
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int dirac_train;
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int pulse_pos[PULSE_MAX];
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int pulse_sign[PULSE_MAX];
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} FCBParam;
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typedef struct G723_1_ChannelContext {
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G723_1_Subframe subframe[4];
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enum FrameType cur_frame_type;
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enum FrameType past_frame_type;
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enum Rate cur_rate;
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uint8_t lsp_index[LSP_BANDS];
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int pitch_lag[2];
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int erased_frames;
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int16_t prev_lsp[LPC_ORDER];
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int16_t sid_lsp[LPC_ORDER];
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int16_t prev_excitation[PITCH_MAX];
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int16_t excitation[PITCH_MAX + FRAME_LEN + 4];
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int16_t synth_mem[LPC_ORDER];
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int16_t fir_mem[LPC_ORDER];
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int iir_mem[LPC_ORDER];
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int random_seed;
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int cng_random_seed;
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int interp_index;
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int interp_gain;
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int sid_gain;
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int cur_gain;
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int reflection_coef;
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int pf_gain; ///< formant postfilter
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///< gain scaling unit memory
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int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
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/* encoder */
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int16_t prev_data[HALF_FRAME_LEN];
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int16_t prev_weight_sig[PITCH_MAX];
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int16_t hpf_fir_mem; ///< highpass filter fir
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int hpf_iir_mem; ///< and iir memories
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int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir
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int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories
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int16_t harmonic_mem[PITCH_MAX];
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} G723_1_ChannelContext;
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typedef struct G723_1_Context {
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AVClass *class;
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int postfilter;
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G723_1_ChannelContext ch[2];
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} G723_1_Context;
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/**
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* Scale vector contents based on the largest of their absolutes.
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*/
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int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length);
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/**
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* Calculate the number of left-shifts required for normalizing the input.
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*
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* @param num input number
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* @param width width of the input, 16 bits(0) / 32 bits(1)
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*/
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int ff_g723_1_normalize_bits(int num, int width);
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int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length);
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/**
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* Get delayed contribution from the previous excitation vector.
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*/
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void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation,
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int lag);
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/**
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* Generate a train of dirac functions with period as pitch lag.
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*/
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void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag);
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/**
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* Generate adaptive codebook excitation.
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*/
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void ff_g723_1_gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
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int pitch_lag, G723_1_Subframe *subfrm,
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enum Rate cur_rate);
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/**
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* Quantize LSP frequencies by interpolation and convert them to
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* the corresponding LPC coefficients.
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*
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* @param lpc buffer for LPC coefficients
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* @param cur_lsp the current LSP vector
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* @param prev_lsp the previous LSP vector
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*/
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void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp,
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int16_t *prev_lsp);
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/**
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* Perform inverse quantization of LSP frequencies.
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*
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* @param cur_lsp the current LSP vector
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* @param prev_lsp the previous LSP vector
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* @param lsp_index VQ indices
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* @param bad_frame bad frame flag
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*/
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void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
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uint8_t *lsp_index, int bad_frame);
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static const uint8_t frame_size[4] = { 24, 20, 4, 1 };
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/**
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* LSP DC component
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*/
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static const int16_t dc_lsp[LPC_ORDER] = {
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0x0c3b,
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0x1271,
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0x1e0a,
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0x2a36,
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0x3630,
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0x406f,
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0x4d28,
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0x56f4,
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0x638c,
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0x6c46
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};
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/* Cosine table scaled by 2^14 */
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extern const int16_t ff_g723_1_cos_tab[COS_TBL_SIZE + 1];
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#define G723_1_COS_TAB_FIRST_ELEMENT 16384
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/**
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* LSP VQ tables
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*/
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extern const int16_t ff_g723_1_lsp_band0[LSP_CB_SIZE][3];
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extern const int16_t ff_g723_1_lsp_band1[LSP_CB_SIZE][3];
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extern const int16_t ff_g723_1_lsp_band2[LSP_CB_SIZE][4];
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/**
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* Used for the coding/decoding of the pulses positions
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* for the MP-MLQ codebook
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*/
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extern const int32_t ff_g723_1_combinatorial_table[PULSE_MAX][SUBFRAME_LEN/GRID_SIZE];
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/**
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* Number of non-zero pulses in the MP-MLQ excitation
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*/
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static const int8_t pulses[4] = {6, 5, 6, 5};
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extern const int16_t ff_g723_1_fixed_cb_gain[GAIN_LEVELS];
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extern const int16_t ff_g723_1_adaptive_cb_gain85 [ 85 * 20];
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extern const int16_t ff_g723_1_adaptive_cb_gain170[170 * 20];
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#endif /* AVCODEC_G723_1_H */
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