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Diffstat (limited to 'ffmpeg1/libavcodec/g729dec.c')
| -rw-r--r-- | ffmpeg1/libavcodec/g729dec.c | 730 |
1 files changed, 0 insertions, 730 deletions
diff --git a/ffmpeg1/libavcodec/g729dec.c b/ffmpeg1/libavcodec/g729dec.c deleted file mode 100644 index 440bf80..0000000 --- a/ffmpeg1/libavcodec/g729dec.c +++ /dev/null @@ -1,730 +0,0 @@ -/* - * G.729, G729 Annex D decoders - * 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 <inttypes.h> -#include <string.h> - -#include "avcodec.h" -#include "libavutil/avutil.h" -#include "get_bits.h" -#include "dsputil.h" -#include "internal.h" - - -#include "g729.h" -#include "lsp.h" -#include "celp_math.h" -#include "celp_filters.h" -#include "acelp_filters.h" -#include "acelp_pitch_delay.h" -#include "acelp_vectors.h" -#include "g729data.h" -#include "g729postfilter.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 - -/// interpolation filter length -#define INTERPOL_LEN 11 - -/** - * 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 - -/** - * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7.13) - */ -#define MR_ENERGY 1018156 - -#define DECISION_NOISE 0 -#define DECISION_INTERMEDIATE 1 -#define DECISION_VOICE 2 - -typedef enum { - FORMAT_G729_8K = 0, - FORMAT_G729D_6K4, - FORMAT_COUNT, -} G729Formats; - -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 { - DSPContext dsp; - AVFrame frame; - - /// past excitation signal buffer - int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN]; - - int16_t* exc; ///< start of past excitation data in buffer - 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 - - int16_t quant_energy[4]; ///< (5.10) past quantized energy - - /// previous speech data for LP synthesis filter - int16_t syn_filter_data[10]; - - - /// residual signal buffer (used in long-term postfilter) - int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE]; - - /// previous speech data for residual calculation filter - int16_t res_filter_data[SUBFRAME_SIZE+10]; - - /// previous speech data for short-term postfilter - int16_t pos_filter_data[SUBFRAME_SIZE+10]; - - /// (1.14) pitch gain of current and five previous subframes - int16_t past_gain_pitch[6]; - - /// (14.1) gain code from current and previous subframe - int16_t past_gain_code[2]; - - /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D - int16_t voice_decision; - - int16_t onset; ///< detected onset level (0-2) - int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4) - int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86 - int gain_coeff; ///< (1.14) gain coefficient (4.2.4) - uint16_t rand_value; ///< random number generator value (4.4.4) - int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame - - /// (14.14) high-pass filter data (past input) - int hpf_f[2]; - - /// high-pass filter data (past output) - int16_t hpf_z[2]; -} 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; -} - -/** - * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4). - * @param[out] lsfq (2.13) quantized LSF coefficients - * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames - * @param ma_predictor switched MA predictor of LSP quantizer - * @param vq_1st first stage vector of quantizer - * @param vq_2nd_low second stage lower vector of LSP quantizer - * @param vq_2nd_high second stage higher vector of LSP quantizer - */ -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; - } - - ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10); -} - -/** - * Restores past LSP quantizer output using LSF from previous frame - * @param[in,out] lsfq (2.13) quantized LSF coefficients - * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames - * @param ma_predictor_prev MA predictor from previous frame - * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame - */ -static void lsf_restore_from_previous(int16_t* lsfq, - int16_t* past_quantizer_outputs[MA_NP + 1], - int ma_predictor_prev) -{ - int16_t* quantizer_output = past_quantizer_outputs[MA_NP]; - int i,k; - - for (i = 0; i < 10; i++) { - int tmp = lsfq[i] << 15; - - for (k = 0; k < MA_NP; k++) - tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i]; - - quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12; - } -} - -/** - * Constructs new excitation signal and applies phase filter to it - * @param[out] out constructed speech signal - * @param in original excitation signal - * @param fc_cur (2.13) original fixed-codebook vector - * @param gain_code (14.1) gain code - * @param subframe_size length of the subframe - */ -static void g729d_get_new_exc( - int16_t* out, - const int16_t* in, - const int16_t* fc_cur, - int dstate, - int gain_code, - int subframe_size) -{ - int i; - int16_t fc_new[SUBFRAME_SIZE]; - - ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size); - - for(i=0; i<subframe_size; i++) - { - out[i] = in[i]; - out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14; - out[i] += (gain_code * fc_new[i] + 0x2000) >> 14; - } -} - -/** - * Makes decision about onset in current subframe - * @param past_onset decision result of previous subframe - * @param past_gain_code gain code of current and previous subframe - * - * @return onset decision result for current subframe - */ -static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code) -{ - if((past_gain_code[0] >> 1) > past_gain_code[1]) - return 2; - else - return FFMAX(past_onset-1, 0); -} - -/** - * Makes decision about voice presence in current subframe - * @param onset onset level - * @param prev_voice_decision voice decision result from previous subframe - * @param past_gain_pitch pitch gain of current and previous subframes - * - * @return voice decision result for current subframe - */ -static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch) -{ - int i, low_gain_pitch_cnt, voice_decision; - - if(past_gain_pitch[0] >= 14745) // 0.9 - voice_decision = DECISION_VOICE; - else if (past_gain_pitch[0] <= 9830) // 0.6 - voice_decision = DECISION_NOISE; - else - voice_decision = DECISION_INTERMEDIATE; - - for(i=0, low_gain_pitch_cnt=0; i<6; i++) - if(past_gain_pitch[i] < 9830) - low_gain_pitch_cnt++; - - if(low_gain_pitch_cnt > 2 && !onset) - voice_decision = DECISION_NOISE; - - if(!onset && voice_decision > prev_voice_decision + 1) - voice_decision--; - - if(onset && voice_decision < DECISION_VOICE) - voice_decision++; - - return voice_decision; -} - -static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order) -{ - int res = 0; - - while (order--) - res += *v1++ * *v2++; - - return res; -} - -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(EINVAL); - } - avctx->sample_fmt = AV_SAMPLE_FMT_S16; - - /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */ - avctx->frame_size = SUBFRAME_SIZE << 1; - - ctx->gain_coeff = 16384; // 1.0 in (1.14) - - 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)); - - ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN]; - - ctx->pitch_delay_int_prev = PITCH_DELAY_MIN; - - /* random seed initialization */ - ctx->rand_value = 21845; - - /* quantized prediction error */ - for(i=0; i<4; i++) - ctx->quant_energy[i] = -14336; // -14 in (5.10) - - ff_dsputil_init(&ctx->dsp, avctx); - ctx->dsp.scalarproduct_int16 = scalarproduct_int16_c; - - avcodec_get_frame_defaults(&ctx->frame); - avctx->coded_frame = &ctx->frame; - - return 0; -} - -static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, - AVPacket *avpkt) -{ - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; - int16_t *out_frame; - GetBitContext gb; - const G729FormatDescription *format; - int frame_erasure = 0; ///< frame erasure detected during decoding - int bad_pitch = 0; ///< parity check failed - int i; - int16_t *tmp; - G729Formats packet_type; - 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[2]; // pitch delay, integer part - int pitch_delay_3x; // pitch delay, multiplied by 3 - int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector - int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector - int j, ret; - int gain_before, gain_after; - int is_periodic = 0; // whether one of the subframes is declared as periodic or not - - ctx->frame.nb_samples = SUBFRAME_SIZE<<1; - if ((ret = ff_get_buffer(avctx, &ctx->frame, 0)) < 0) - return ret; - out_frame = (int16_t*) ctx->frame.data[0]; - - if (buf_size == 10) { - packet_type = FORMAT_G729_8K; - format = &format_g729_8k; - //Reset voice decision - ctx->onset = 0; - ctx->voice_decision = DECISION_VOICE; - av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s"); - } else if (buf_size == 8) { - packet_type = FORMAT_G729D_6K4; - 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_INVALIDDATA; - } - - for (i=0; i < buf_size; i++) - frame_erasure |= buf[i]; - frame_erasure = !frame_erasure; - - init_get_bits(&gb, buf, 8*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); - - if(frame_erasure) - lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs, - ctx->ma_predictor_prev); - else { - lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs, - ma_predictor, - quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi); - ctx->ma_predictor_prev = ma_predictor; - } - - tmp = ctx->past_quantizer_outputs[MA_NP]; - memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs, - MA_NP * sizeof(int16_t*)); - ctx->past_quantizer_outputs[0] = tmp; - - 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++) { - int gain_corr_factor; - - 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 (frame_erasure) - pitch_delay_3x = 3 * ctx->pitch_delay_int_prev; - else 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[i] = (pitch_delay_3x + 1) / 3; - if (pitch_delay_int[i] > PITCH_DELAY_MAX) { - av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]); - pitch_delay_int[i] = PITCH_DELAY_MAX; - } - - if (frame_erasure) { - ctx->rand_value = g729_prng(ctx->rand_value); - fc_indexes = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1); - - ctx->rand_value = g729_prng(ctx->rand_value); - pulses_signs = ctx->rand_value; - } - - - memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE); - switch (packet_type) { - case FORMAT_G729_8K: - ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13, - ff_fc_4pulses_8bits_track_4, - fc_indexes, pulses_signs, 3, 3); - break; - case FORMAT_G729D_6K4: - ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray, - ff_fc_2pulses_9bits_track2_gray, - fc_indexes, pulses_signs, 1, 4); - break; - } - - /* - This filter enhances harmonic components of the fixed-codebook vector to - improve the quality of the reconstructed speech. - - / fc_v[i], i < pitch_delay - fc_v[i] = < - \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay - */ - ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i], - fc + pitch_delay_int[i], - fc, 1 << 14, - av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX), - 0, 14, - SUBFRAME_SIZE - pitch_delay_int[i]); - - memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t)); - ctx->past_gain_code[1] = ctx->past_gain_code[0]; - - if (frame_erasure) { - ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15) - ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11) - - gain_corr_factor = 0; - } else { - if (packet_type == FORMAT_G729D_6K4) { - ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] + - cb_gain_2nd_6k4[gc_2nd_index][0]; - gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] + - cb_gain_2nd_6k4[gc_2nd_index][1]; - - /* Without check below overflow can occur in ff_acelp_update_past_gain. - It is not issue for G.729, because gain_corr_factor in it's case is always - greater than 1024, while in G.729D it can be even zero. */ - gain_corr_factor = FFMAX(gain_corr_factor, 1024); -#ifndef G729_BITEXACT - gain_corr_factor >>= 1; -#endif - } else { - ctx->past_gain_pitch[0] = 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]; - } - - /* Decode the fixed-codebook gain. */ - ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor, - fc, MR_ENERGY, - ctx->quant_energy, - ma_prediction_coeff, - SUBFRAME_SIZE, 4); -#ifdef G729_BITEXACT - /* - This correction required to get bit-exact result with - reference code, because gain_corr_factor in G.729D is - two times larger than in original G.729. - - If bit-exact result is not issue then gain_corr_factor - can be simpler divided by 2 before call to g729_get_gain_code - instead of using correction below. - */ - if (packet_type == FORMAT_G729D_6K4) { - gain_corr_factor >>= 1; - ctx->past_gain_code[0] >>= 1; - } -#endif - } - ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure); - - /* Routine requires rounding to lowest. */ - ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE, - ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3, - ff_acelp_interp_filter, 6, - (pitch_delay_3x % 3) << 1, - 10, SUBFRAME_SIZE); - - ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE, - ctx->exc + i * SUBFRAME_SIZE, fc, - (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0], - ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0], - 1 << 13, 14, SUBFRAME_SIZE); - - memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t)); - - if (ff_celp_lp_synthesis_filter( - synth+10, - &lp[i][1], - ctx->exc + i * SUBFRAME_SIZE, - SUBFRAME_SIZE, - 10, - 1, - 0, - 0x800)) - /* Overflow occurred, downscale excitation signal... */ - for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++) - ctx->exc_base[j] >>= 2; - - /* ... and make synthesis again. */ - if (packet_type == FORMAT_G729D_6K4) { - int16_t exc_new[SUBFRAME_SIZE]; - - ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code); - ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch); - - g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE); - - ff_celp_lp_synthesis_filter( - synth+10, - &lp[i][1], - exc_new, - SUBFRAME_SIZE, - 10, - 0, - 0, - 0x800); - } else { - ff_celp_lp_synthesis_filter( - synth+10, - &lp[i][1], - ctx->exc + i * SUBFRAME_SIZE, - SUBFRAME_SIZE, - 10, - 0, - 0, - 0x800); - } - /* Save data (without postfilter) for use in next subframe. */ - memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t)); - - /* Calculate gain of unfiltered signal for use in AGC. */ - gain_before = 0; - for (j = 0; j < SUBFRAME_SIZE; j++) - gain_before += FFABS(synth[j+10]); - - /* Call postfilter and also update voicing decision for use in next frame. */ - ff_g729_postfilter( - &ctx->dsp, - &ctx->ht_prev_data, - &is_periodic, - &lp[i][0], - pitch_delay_int[0], - ctx->residual, - ctx->res_filter_data, - ctx->pos_filter_data, - synth+10, - SUBFRAME_SIZE); - - /* Calculate gain of filtered signal for use in AGC. */ - gain_after = 0; - for(j=0; j<SUBFRAME_SIZE; j++) - gain_after += FFABS(synth[j+10]); - - ctx->gain_coeff = ff_g729_adaptive_gain_control( - gain_before, - gain_after, - synth+10, - SUBFRAME_SIZE, - ctx->gain_coeff); - - if (frame_erasure) - ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX); - else - ctx->pitch_delay_int_prev = pitch_delay_int[i]; - - memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t)); - ff_acelp_high_pass_filter( - out_frame + i*SUBFRAME_SIZE, - ctx->hpf_f, - synth+10, - SUBFRAME_SIZE); - memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t)); - } - - ctx->was_periodic = is_periodic; - - /* Save signal for use in next frame. */ - memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t)); - - *got_frame_ptr = 1; - *(AVFrame*)data = ctx->frame; - return buf_size; -} - -AVCodec ff_g729_decoder = { - .name = "g729", - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_G729, - .priv_data_size = sizeof(G729Context), - .init = decoder_init, - .decode = decode_frame, - .capabilities = CODEC_CAP_DR1, - .long_name = NULL_IF_CONFIG_SMALL("G.729"), -}; |