Mercurial > libavcodec.hg
view g729dec.c @ 10674:f04468776158 libavcodec
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author | michael |
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date | Fri, 11 Dec 2009 21:50:08 +0000 |
parents | 62705926ba33 |
children | 25e209f9153a |
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/* * G.729 decoder * Copyright (c) 2008 Vladimir Voroshilov * * 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 */ #include <stdlib.h> #include <inttypes.h> #include <limits.h> #include <stdio.h> #include <string.h> #include <math.h> #include <assert.h> #include "avcodec.h" #include "libavutil/avutil.h" #include "get_bits.h" #include "g729.h" #include "lsp.h" #include "celp_math.h" #include "acelp_filters.h" #include "acelp_pitch_delay.h" #include "acelp_vectors.h" #include "g729data.h" /** * minimum quantized LSF value (3.2.4) * 0.005 in Q13 */ #define LSFQ_MIN 40 /** * maximum quantized LSF value (3.2.4) * 3.135 in Q13 */ #define LSFQ_MAX 25681 /** * minimum LSF distance (3.2.4) * 0.0391 in Q13 */ #define LSFQ_DIFF_MIN 321 /** * minimum gain pitch value (3.8, Equation 47) * 0.2 in (1.14) */ #define SHARP_MIN 3277 /** * maximum gain pitch value (3.8, Equation 47) * (EE) This does not comply with the specification. * Specification says about 0.8, which should be * 13107 in (1.14), but reference C code uses * 13017 (equals to 0.7945) instead of it. */ #define SHARP_MAX 13017 typedef struct { uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits) uint8_t parity_bit; ///< parity bit for pitch delay uint8_t gc_1st_index_bits; ///< gain codebook (first stage) index (size in bits) uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits) uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry } G729FormatDescription; typedef struct { int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3) /// (2.13) LSP quantizer outputs int16_t past_quantizer_output_buf[MA_NP + 1][10]; int16_t* past_quantizer_outputs[MA_NP + 1]; int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5) int16_t *lsp[2]; ///< pointers to lsp_buf } G729Context; static const G729FormatDescription format_g729_8k = { .ac_index_bits = {8,5}, .parity_bit = 1, .gc_1st_index_bits = GC_1ST_IDX_BITS_8K, .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K, .fc_signs_bits = 4, .fc_indexes_bits = 13, }; static const G729FormatDescription format_g729d_6k4 = { .ac_index_bits = {8,4}, .parity_bit = 0, .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4, .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4, .fc_signs_bits = 2, .fc_indexes_bits = 9, }; /** * \brief pseudo random number generator */ static inline uint16_t g729_prng(uint16_t value) { return 31821 * value + 13849; } /** * Get parity bit of bit 2..7 */ static inline int get_parity(uint8_t value) { return (0x6996966996696996ULL >> (value >> 2)) & 1; } static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1], int16_t ma_predictor, int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high) { int i,j; static const uint8_t min_distance[2]={10, 5}; //(2.13) int16_t* quantizer_output = past_quantizer_outputs[MA_NP]; for (i = 0; i < 5; i++) { quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ]; quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5]; } for (j = 0; j < 2; j++) { for (i = 1; i < 10; i++) { int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1; if (diff > 0) { quantizer_output[i - 1] -= diff; quantizer_output[i ] += diff; } } } for (i = 0; i < 10; i++) { int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i]; for (j = 0; j < MA_NP; j++) sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i]; lsfq[i] = sum >> 15; } /* Rotate past_quantizer_outputs. */ memmove(past_quantizer_outputs + 1, past_quantizer_outputs, MA_NP * sizeof(int16_t*)); past_quantizer_outputs[0] = quantizer_output; ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10); } static av_cold int decoder_init(AVCodecContext * avctx) { G729Context* ctx = avctx->priv_data; int i,k; if (avctx->channels != 1) { av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels); return AVERROR_NOFMT; } /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */ avctx->frame_size = SUBFRAME_SIZE << 1; for (k = 0; k < MA_NP + 1; k++) { ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k]; for (i = 1; i < 11; i++) ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3; } ctx->lsp[0] = ctx->lsp_buf[0]; ctx->lsp[1] = ctx->lsp_buf[1]; memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t)); return 0; } static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; int16_t *out_frame = data; GetBitContext gb; G729FormatDescription format; int frame_erasure = 0; ///< frame erasure detected during decoding int bad_pitch = 0; ///< parity check failed int i; G729Context *ctx = avctx->priv_data; int16_t lp[2][11]; // (3.12) uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer uint8_t quantizer_1st; ///< first stage vector of quantizer uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits) uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits) int pitch_delay_int; // pitch delay, integer part int pitch_delay_3x; // pitch delay, multiplied by 3 if (*data_size < SUBFRAME_SIZE << 2) { av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer too small\n"); return AVERROR(EIO); } if (buf_size == 10) { format = format_g729_8k; av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s"); } else if (buf_size == 8) { format = format_g729d_6k4; av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s"); } else { av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size); return (AVERROR_NOFMT); } for (i=0; i < buf_size; i++) frame_erasure |= buf[i]; frame_erasure = !frame_erasure; init_get_bits(&gb, buf, buf_size); ma_predictor = get_bits(&gb, 1); quantizer_1st = get_bits(&gb, VQ_1ST_BITS); quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS); quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS); lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs, ma_predictor, quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi); ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10); ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10); FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]); for (i = 0; i < 2; i++) { uint8_t ac_index; ///< adaptive codebook index uint8_t pulses_signs; ///< fixed-codebook vector pulse signs int fc_indexes; ///< fixed-codebook indexes uint8_t gc_1st_index; ///< gain codebook (first stage) index uint8_t gc_2nd_index; ///< gain codebook (second stage) index ac_index = get_bits(&gb, format.ac_index_bits[i]); if(!i && format.parity_bit) bad_pitch = get_parity(ac_index) == get_bits1(&gb); fc_indexes = get_bits(&gb, format.fc_indexes_bits); pulses_signs = get_bits(&gb, format.fc_signs_bits); gc_1st_index = get_bits(&gb, format.gc_1st_index_bits); gc_2nd_index = get_bits(&gb, format.gc_2nd_index_bits); if(!i) { if (bad_pitch) pitch_delay_3x = 3 * ctx->pitch_delay_int_prev; else pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index); } else { int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5, PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9); if(packet_type == FORMAT_G729D_6K4) pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min); else pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min); } /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */ pitch_delay_int = (pitch_delay_3x + 1) / 3; ff_acelp_weighted_vector_sum(fc + pitch_delay_int, fc + pitch_delay_int, fc, 1 << 14, av_clip(ctx->gain_pitch, SHARP_MIN, SHARP_MAX), 0, 14, SUBFRAME_SIZE - pitch_delay_int); if (frame_erasure) { ctx->gain_pitch = (29491 * ctx->gain_pitch) >> 15; // 0.90 (0.15) ctx->gain_code = ( 2007 * ctx->gain_code ) >> 11; // 0.98 (0.11) gain_corr_factor = 0; } else { ctx->gain_pitch = cb_gain_1st_8k[gc_1st_index][0] + cb_gain_2nd_8k[gc_2nd_index][0]; gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] + cb_gain_2nd_8k[gc_2nd_index][1]; ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE, ctx->exc + i * SUBFRAME_SIZE, fc, (!voicing && frame_erasure) ? 0 : ctx->gain_pitch, ( voicing && frame_erasure) ? 0 : ctx->gain_code, 1 << 13, 14, SUBFRAME_SIZE); ctx->pitch_delay_int_prev = pitch_delay_int; } *data_size = SUBFRAME_SIZE << 2; return buf_size; } AVCodec g729_decoder = { "g729", CODEC_TYPE_AUDIO, CODEC_ID_G729, sizeof(G729Context), decoder_init, NULL, NULL, decode_frame, .long_name = NULL_IF_CONFIG_SMALL("G.729"), };