Mercurial > libavcodec.hg
view g722.c @ 12478:5942c42c44e5 libavcodec
Add G.722 ADPCM audio decoder
author | mstorsjo |
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date | Thu, 09 Sep 2010 19:21:16 +0000 |
parents | |
children | 750ff18b7394 |
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/* * G.722 ADPCM audio decoder * * Copyright (c) CMU 1993 Computer Science, Speech Group * Chengxiang Lu and Alex Hauptmann * Copyright (c) 2005 Steve Underwood <steveu at coppice.org> * Copyright (c) 2009 Kenan Gillet * Copyright (c) 2010 Martin Storsjo * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * * G.722 ADPCM audio codec * * This G.722 decoder is a bit-exact implementation of the ITU G.722 * specification for all three specified bitrates - 64000bps, 56000bps * and 48000bps. It passes the ITU tests. * * @note For the 56000bps and 48000bps bitrates, the lowest 1 or 2 bits * respectively of each byte are ignored. */ #include "avcodec.h" #include "mathops.h" #include "get_bits.h" #define PREV_SAMPLES_BUF_SIZE 1024 typedef struct { int16_t prev_samples[PREV_SAMPLES_BUF_SIZE]; ///< memory of past decoded samples int prev_samples_pos; ///< the number of values in prev_samples /** * The band[0] and band[1] correspond respectively to the lower band and higher band. */ struct G722Band { int16_t s_predictor; ///< predictor output value int32_t s_zero; ///< previous output signal from zero predictor int8_t part_reconst_mem[2]; ///< signs of previous partially reconstructed signals int16_t prev_qtzd_reconst; ///< previous quantized reconstructed signal (internal value, using low_inv_quant4) int16_t pole_mem[2]; ///< second-order pole section coefficient buffer int32_t diff_mem[6]; ///< quantizer difference signal memory int16_t zero_mem[6]; ///< Seventh-order zero section coefficient buffer int16_t log_factor; ///< delayed 2-logarithmic quantizer factor int16_t scale_factor; ///< delayed quantizer scale factor } band[2]; } G722Context; static const int8_t sign_lookup[2] = { -1, 1 }; static const int16_t inv_log2_table[32] = { 2048, 2093, 2139, 2186, 2233, 2282, 2332, 2383, 2435, 2489, 2543, 2599, 2656, 2714, 2774, 2834, 2896, 2960, 3025, 3091, 3158, 3228, 3298, 3371, 3444, 3520, 3597, 3676, 3756, 3838, 3922, 4008 }; static const int16_t high_log_factor_step[2] = { 798, -214 }; static const int16_t high_inv_quant[4] = { -926, -202, 926, 202 }; /** * low_log_factor_step[index] == wl[rl42[index]] */ static const int16_t low_log_factor_step[16] = { -60, 3042, 1198, 538, 334, 172, 58, -30, 3042, 1198, 538, 334, 172, 58, -30, -60 }; static const int16_t low_inv_quant4[16] = { 0, -2557, -1612, -1121, -786, -530, -323, -150, 2557, 1612, 1121, 786, 530, 323, 150, 0 }; /** * quadrature mirror filter (QMF) coefficients * * ITU-T G.722 Table 11 */ static const int16_t qmf_coeffs[12] = { 3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11, }; /** * adaptive predictor * * @param cur_diff the dequantized and scaled delta calculated from the * current codeword */ static void do_adaptive_prediction(struct G722Band *band, const int cur_diff) { int sg[2], limit, i, cur_qtzd_reconst; const int cur_part_reconst = band->s_zero + cur_diff < 0; sg[0] = sign_lookup[cur_part_reconst != band->part_reconst_mem[0]]; sg[1] = sign_lookup[cur_part_reconst == band->part_reconst_mem[1]]; band->part_reconst_mem[1] = band->part_reconst_mem[0]; band->part_reconst_mem[0] = cur_part_reconst; band->pole_mem[1] = av_clip((sg[0] * av_clip(band->pole_mem[0], -8191, 8191) >> 5) + (sg[1] << 7) + (band->pole_mem[1] * 127 >> 7), -12288, 12288); limit = 15360 - band->pole_mem[1]; band->pole_mem[0] = av_clip(-192 * sg[0] + (band->pole_mem[0] * 255 >> 8), -limit, limit); if (cur_diff) { for (i = 0; i < 6; i++) band->zero_mem[i] = ((band->zero_mem[i]*255) >> 8) + ((band->diff_mem[i]^cur_diff) < 0 ? -128 : 128); } else for (i = 0; i < 6; i++) band->zero_mem[i] = (band->zero_mem[i]*255) >> 8; for (i = 5; i > 0; i--) band->diff_mem[i] = band->diff_mem[i-1]; band->diff_mem[0] = av_clip_int16(cur_diff << 1); band->s_zero = 0; for (i = 5; i >= 0; i--) band->s_zero += (band->zero_mem[i]*band->diff_mem[i]) >> 15; cur_qtzd_reconst = av_clip_int16((band->s_predictor + cur_diff) << 1); band->s_predictor = av_clip_int16(band->s_zero + (band->pole_mem[0] * cur_qtzd_reconst >> 15) + (band->pole_mem[1] * band->prev_qtzd_reconst >> 15)); band->prev_qtzd_reconst = cur_qtzd_reconst; } static int inline linear_scale_factor(const int log_factor) { const int wd1 = inv_log2_table[(log_factor >> 6) & 31]; const int shift = log_factor >> 11; return shift < 0 ? wd1 >> -shift : wd1 << shift; } static void update_low_predictor(struct G722Band *band, const int ilow) { do_adaptive_prediction(band, band->scale_factor * low_inv_quant4[ilow] >> 10); // quantizer adaptation band->log_factor = av_clip((band->log_factor * 127 >> 7) + low_log_factor_step[ilow], 0, 18432); band->scale_factor = linear_scale_factor(band->log_factor - (8 << 11)); } static void update_high_predictor(struct G722Band *band, const int dhigh, const int ihigh) { do_adaptive_prediction(band, dhigh); // quantizer adaptation band->log_factor = av_clip((band->log_factor * 127 >> 7) + high_log_factor_step[ihigh&1], 0, 22528); band->scale_factor = linear_scale_factor(band->log_factor - (10 << 11)); } static void apply_qmf(const int16_t *prev_samples, int *xout1, int *xout2) { int i; *xout1 = 0; *xout2 = 0; for (i = 0; i < 12; i++) { MAC16(*xout2, prev_samples[2*i ], qmf_coeffs[i ]); MAC16(*xout1, prev_samples[2*i+1], qmf_coeffs[11-i]); } } static av_cold int g722_init(AVCodecContext * avctx) { G722Context *c = avctx->priv_data; if (avctx->channels != 1) { av_log(avctx, AV_LOG_ERROR, "Only mono tracks are allowed.\n"); return AVERROR_INVALIDDATA; } avctx->sample_fmt = SAMPLE_FMT_S16; switch (avctx->bits_per_coded_sample) { case 8: case 7: case 6: break; default: av_log(avctx, AV_LOG_WARNING, "Unsupported bits_per_coded_sample [%d], " "assuming 8\n", avctx->bits_per_coded_sample); case 0: avctx->bits_per_coded_sample = 8; break; } c->band[0].scale_factor = 8; c->band[1].scale_factor = 2; c->prev_samples_pos = 22; if (avctx->lowres) avctx->sample_rate /= 2; return 0; } static const int16_t low_inv_quant5[32] = { -35, -35, -2919, -2195, -1765, -1458, -1219, -1023, -858, -714, -587, -473, -370, -276, -190, -110, 2919, 2195, 1765, 1458, 1219, 1023, 858, 714, 587, 473, 370, 276, 190, 110, 35, -35 }; static const int16_t low_inv_quant6[64] = { -17, -17, -17, -17, -3101, -2738, -2376, -2088, -1873, -1689, -1535, -1399, -1279, -1170, -1072, -982, -899, -822, -750, -682, -618, -558, -501, -447, -396, -347, -300, -254, -211, -170, -130, -91, 3101, 2738, 2376, 2088, 1873, 1689, 1535, 1399, 1279, 1170, 1072, 982, 899, 822, 750, 682, 618, 558, 501, 447, 396, 347, 300, 254, 211, 170, 130, 91, 54, 17, -54, -17 }; static const int16_t *low_inv_quants[3] = { low_inv_quant6, low_inv_quant5, low_inv_quant4 }; static int g722_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { G722Context *c = avctx->priv_data; int16_t *out_buf = data; int j, out_len = 0; const int skip = 8 - avctx->bits_per_coded_sample; const int16_t *quantizer_table = low_inv_quants[skip]; GetBitContext gb; init_get_bits(&gb, avpkt->data, avpkt->size * 8); for (j = 0; j < avpkt->size; j++) { int ilow, ihigh, rlow; ihigh = get_bits(&gb, 2); ilow = get_bits(&gb, 6 - skip); skip_bits(&gb, skip); rlow = av_clip((c->band[0].scale_factor * quantizer_table[ilow] >> 10) + c->band[0].s_predictor, -16384, 16383); update_low_predictor(&c->band[0], ilow >> (2 - skip)); if (!avctx->lowres) { const int dhigh = c->band[1].scale_factor * high_inv_quant[ihigh] >> 10; const int rhigh = av_clip(dhigh + c->band[1].s_predictor, -16384, 16383); int xout1, xout2; update_high_predictor(&c->band[1], dhigh, ihigh); c->prev_samples[c->prev_samples_pos++] = rlow + rhigh; c->prev_samples[c->prev_samples_pos++] = rlow - rhigh; apply_qmf(c->prev_samples + c->prev_samples_pos - 24, &xout1, &xout2); out_buf[out_len++] = av_clip_int16(xout1 >> 12); out_buf[out_len++] = av_clip_int16(xout2 >> 12); if (c->prev_samples_pos >= PREV_SAMPLES_BUF_SIZE) { memmove(c->prev_samples, c->prev_samples + c->prev_samples_pos - 22, 22 * sizeof(c->prev_samples[0])); c->prev_samples_pos = 22; } } else out_buf[out_len++] = rlow; } *data_size = out_len << 1; return avpkt->size; } AVCodec adpcm_g722_decoder = { .name = "g722", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_ADPCM_G722, .priv_data_size = sizeof(G722Context), .init = g722_init, .decode = g722_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"), .max_lowres = 1, };