mirror of https://github.com/FFmpeg/FFmpeg.git
Originally committed as revision 25086 to svn://svn.ffmpeg.org/ffmpeg/trunkoldabi
Martin Storsjö
15 years ago
parent
a3f0d2b967
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58e37eafce
6 changed files with 311 additions and 2 deletions
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/*
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* G.722 ADPCM audio decoder |
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* |
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* Copyright (c) CMU 1993 Computer Science, Speech Group |
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* Chengxiang Lu and Alex Hauptmann |
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* Copyright (c) 2005 Steve Underwood <steveu at coppice.org> |
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* Copyright (c) 2009 Kenan Gillet |
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* Copyright (c) 2010 Martin Storsjo |
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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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* |
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* G.722 ADPCM audio codec |
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* |
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* This G.722 decoder is a bit-exact implementation of the ITU G.722 |
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* specification for all three specified bitrates - 64000bps, 56000bps |
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* and 48000bps. It passes the ITU tests. |
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* |
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* @note For the 56000bps and 48000bps bitrates, the lowest 1 or 2 bits |
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* respectively of each byte are ignored. |
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*/ |
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#include "avcodec.h" |
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#include "mathops.h" |
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#include "get_bits.h" |
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#define PREV_SAMPLES_BUF_SIZE 1024 |
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typedef struct { |
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int16_t prev_samples[PREV_SAMPLES_BUF_SIZE]; ///< memory of past decoded samples
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int prev_samples_pos; ///< the number of values in prev_samples
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/**
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* The band[0] and band[1] correspond respectively to the lower band and higher band. |
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*/ |
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struct G722Band { |
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int16_t s_predictor; ///< predictor output value
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int32_t s_zero; ///< previous output signal from zero predictor
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int8_t part_reconst_mem[2]; ///< signs of previous partially reconstructed signals
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int16_t prev_qtzd_reconst; ///< previous quantized reconstructed signal (internal value, using low_inv_quant4)
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int16_t pole_mem[2]; ///< second-order pole section coefficient buffer
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int32_t diff_mem[6]; ///< quantizer difference signal memory
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int16_t zero_mem[6]; ///< Seventh-order zero section coefficient buffer
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int16_t log_factor; ///< delayed 2-logarithmic quantizer factor
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int16_t scale_factor; ///< delayed quantizer scale factor
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} band[2]; |
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} G722Context; |
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static const int8_t sign_lookup[2] = { -1, 1 }; |
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static const int16_t inv_log2_table[32] = { |
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2048, 2093, 2139, 2186, 2233, 2282, 2332, 2383, |
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2435, 2489, 2543, 2599, 2656, 2714, 2774, 2834, |
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2896, 2960, 3025, 3091, 3158, 3228, 3298, 3371, |
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3444, 3520, 3597, 3676, 3756, 3838, 3922, 4008 |
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}; |
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static const int16_t high_log_factor_step[2] = { 798, -214 }; |
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static const int16_t high_inv_quant[4] = { -926, -202, 926, 202 }; |
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/**
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* low_log_factor_step[index] == wl[rl42[index]] |
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*/ |
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static const int16_t low_log_factor_step[16] = { |
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-60, 3042, 1198, 538, 334, 172, 58, -30, |
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3042, 1198, 538, 334, 172, 58, -30, -60 |
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}; |
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static const int16_t low_inv_quant4[16] = { |
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0, -2557, -1612, -1121, -786, -530, -323, -150, |
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2557, 1612, 1121, 786, 530, 323, 150, 0 |
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}; |
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/**
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* quadrature mirror filter (QMF) coefficients |
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* |
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* ITU-T G.722 Table 11 |
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*/ |
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static const int16_t qmf_coeffs[12] = { |
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3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11, |
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}; |
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/**
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* adaptive predictor |
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* |
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* @param cur_diff the dequantized and scaled delta calculated from the |
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* current codeword |
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*/ |
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static void do_adaptive_prediction(struct G722Band *band, const int cur_diff) |
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{ |
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int sg[2], limit, i, cur_qtzd_reconst; |
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const int cur_part_reconst = band->s_zero + cur_diff < 0; |
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sg[0] = sign_lookup[cur_part_reconst != band->part_reconst_mem[0]]; |
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sg[1] = sign_lookup[cur_part_reconst == band->part_reconst_mem[1]]; |
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band->part_reconst_mem[1] = band->part_reconst_mem[0]; |
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band->part_reconst_mem[0] = cur_part_reconst; |
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band->pole_mem[1] = av_clip((sg[0] * av_clip(band->pole_mem[0], -8191, 8191) >> 5) + |
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(sg[1] << 7) + (band->pole_mem[1] * 127 >> 7), -12288, 12288); |
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limit = 15360 - band->pole_mem[1]; |
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band->pole_mem[0] = av_clip(-192 * sg[0] + (band->pole_mem[0] * 255 >> 8), -limit, limit); |
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if (cur_diff) { |
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for (i = 0; i < 6; i++) |
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band->zero_mem[i] = ((band->zero_mem[i]*255) >> 8) + |
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((band->diff_mem[i]^cur_diff) < 0 ? -128 : 128); |
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} else |
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for (i = 0; i < 6; i++) |
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band->zero_mem[i] = (band->zero_mem[i]*255) >> 8; |
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for (i = 5; i > 0; i--) |
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band->diff_mem[i] = band->diff_mem[i-1]; |
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band->diff_mem[0] = av_clip_int16(cur_diff << 1); |
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band->s_zero = 0; |
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for (i = 5; i >= 0; i--) |
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band->s_zero += (band->zero_mem[i]*band->diff_mem[i]) >> 15; |
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cur_qtzd_reconst = av_clip_int16((band->s_predictor + cur_diff) << 1); |
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band->s_predictor = av_clip_int16(band->s_zero + |
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(band->pole_mem[0] * cur_qtzd_reconst >> 15) + |
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(band->pole_mem[1] * band->prev_qtzd_reconst >> 15)); |
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band->prev_qtzd_reconst = cur_qtzd_reconst; |
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} |
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static int inline linear_scale_factor(const int log_factor) |
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{ |
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const int wd1 = inv_log2_table[(log_factor >> 6) & 31]; |
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const int shift = log_factor >> 11; |
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return shift < 0 ? wd1 >> -shift : wd1 << shift; |
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} |
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static void update_low_predictor(struct G722Band *band, const int ilow) |
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{ |
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do_adaptive_prediction(band, |
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band->scale_factor * low_inv_quant4[ilow] >> 10); |
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// quantizer adaptation
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band->log_factor = av_clip((band->log_factor * 127 >> 7) + |
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low_log_factor_step[ilow], 0, 18432); |
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band->scale_factor = linear_scale_factor(band->log_factor - (8 << 11)); |
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} |
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static void update_high_predictor(struct G722Band *band, const int dhigh, |
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const int ihigh) |
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{ |
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do_adaptive_prediction(band, dhigh); |
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// quantizer adaptation
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band->log_factor = av_clip((band->log_factor * 127 >> 7) + |
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high_log_factor_step[ihigh&1], 0, 22528); |
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band->scale_factor = linear_scale_factor(band->log_factor - (10 << 11)); |
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} |
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static void apply_qmf(const int16_t *prev_samples, int *xout1, int *xout2) |
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{ |
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int i; |
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*xout1 = 0; |
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*xout2 = 0; |
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for (i = 0; i < 12; i++) { |
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MAC16(*xout2, prev_samples[2*i ], qmf_coeffs[i ]); |
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MAC16(*xout1, prev_samples[2*i+1], qmf_coeffs[11-i]); |
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} |
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} |
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static av_cold int g722_init(AVCodecContext * avctx) |
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{ |
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G722Context *c = avctx->priv_data; |
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if (avctx->channels != 1) { |
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av_log(avctx, AV_LOG_ERROR, "Only mono tracks are allowed.\n"); |
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return AVERROR_INVALIDDATA; |
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} |
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avctx->sample_fmt = SAMPLE_FMT_S16; |
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switch (avctx->bits_per_coded_sample) { |
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case 8: |
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case 7: |
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case 6: |
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break; |
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default: |
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av_log(avctx, AV_LOG_WARNING, "Unsupported bits_per_coded_sample [%d], " |
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"assuming 8\n", |
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avctx->bits_per_coded_sample); |
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case 0: |
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avctx->bits_per_coded_sample = 8; |
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break; |
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} |
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c->band[0].scale_factor = 8; |
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c->band[1].scale_factor = 2; |
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c->prev_samples_pos = 22; |
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if (avctx->lowres) |
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avctx->sample_rate /= 2; |
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return 0; |
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} |
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static const int16_t low_inv_quant5[32] = { |
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-35, -35, -2919, -2195, -1765, -1458, -1219, -1023, |
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-858, -714, -587, -473, -370, -276, -190, -110, |
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2919, 2195, 1765, 1458, 1219, 1023, 858, 714, |
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587, 473, 370, 276, 190, 110, 35, -35 |
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}; |
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static const int16_t low_inv_quant6[64] = { |
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-17, -17, -17, -17, -3101, -2738, -2376, -2088, |
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-1873, -1689, -1535, -1399, -1279, -1170, -1072, -982, |
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-899, -822, -750, -682, -618, -558, -501, -447, |
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-396, -347, -300, -254, -211, -170, -130, -91, |
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3101, 2738, 2376, 2088, 1873, 1689, 1535, 1399, |
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1279, 1170, 1072, 982, 899, 822, 750, 682, |
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618, 558, 501, 447, 396, 347, 300, 254, |
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211, 170, 130, 91, 54, 17, -54, -17 |
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}; |
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static const int16_t *low_inv_quants[3] = { low_inv_quant6, low_inv_quant5, |
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low_inv_quant4 }; |
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static int g722_decode_frame(AVCodecContext *avctx, void *data, |
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int *data_size, AVPacket *avpkt) |
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{ |
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G722Context *c = avctx->priv_data; |
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int16_t *out_buf = data; |
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int j, out_len = 0; |
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const int skip = 8 - avctx->bits_per_coded_sample; |
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const int16_t *quantizer_table = low_inv_quants[skip]; |
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GetBitContext gb; |
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init_get_bits(&gb, avpkt->data, avpkt->size * 8); |
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for (j = 0; j < avpkt->size; j++) { |
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int ilow, ihigh, rlow; |
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ihigh = get_bits(&gb, 2); |
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ilow = get_bits(&gb, 6 - skip); |
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skip_bits(&gb, skip); |
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rlow = av_clip((c->band[0].scale_factor * quantizer_table[ilow] >> 10) |
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+ c->band[0].s_predictor, -16384, 16383); |
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update_low_predictor(&c->band[0], ilow >> (2 - skip)); |
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if (!avctx->lowres) { |
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const int dhigh = c->band[1].scale_factor * |
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high_inv_quant[ihigh] >> 10; |
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const int rhigh = av_clip(dhigh + c->band[1].s_predictor, |
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-16384, 16383); |
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int xout1, xout2; |
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update_high_predictor(&c->band[1], dhigh, ihigh); |
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c->prev_samples[c->prev_samples_pos++] = rlow + rhigh; |
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c->prev_samples[c->prev_samples_pos++] = rlow - rhigh; |
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apply_qmf(c->prev_samples + c->prev_samples_pos - 24, |
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&xout1, &xout2); |
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out_buf[out_len++] = av_clip_int16(xout1 >> 12); |
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out_buf[out_len++] = av_clip_int16(xout2 >> 12); |
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if (c->prev_samples_pos >= PREV_SAMPLES_BUF_SIZE) { |
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memmove(c->prev_samples, |
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c->prev_samples + c->prev_samples_pos - 22, |
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22 * sizeof(c->prev_samples[0])); |
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c->prev_samples_pos = 22; |
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} |
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} else |
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out_buf[out_len++] = rlow; |
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} |
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*data_size = out_len << 1; |
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return avpkt->size; |
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} |
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AVCodec adpcm_g722_decoder = { |
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.name = "g722", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = CODEC_ID_ADPCM_G722, |
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.priv_data_size = sizeof(G722Context), |
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.init = g722_init, |
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.decode = g722_decode_frame, |
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.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"), |
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.max_lowres = 1, |
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}; |
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