| /* |
| * 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 "audiodsp.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 { |
| AudioDSPContext adsp; |
| |
| /// 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; |
| } |
| |
| /** |
| * 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_audiodsp_init(&ctx->adsp); |
| ctx->adsp.scalarproduct_int16 = scalarproduct_int16_c; |
| |
| 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 |
| AVFrame *frame = data; |
| |
| frame->nb_samples = SUBFRAME_SIZE<<1; |
| if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| return ret; |
| out_frame = (int16_t*) frame->data[0]; |
| |
| if (buf_size % 10 == 0) { |
| 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 = av_parity(ac_index >> 2) == 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 = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits); |
| |
| 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->adsp, 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->adsp, |
| &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; |
| return packet_type == FORMAT_G729_8K ? 10 : 8; |
| } |
| |
| AVCodec ff_g729_decoder = { |
| .name = "g729", |
| .long_name = NULL_IF_CONFIG_SMALL("G.729"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_G729, |
| .priv_data_size = sizeof(G729Context), |
| .init = decoder_init, |
| .decode = decode_frame, |
| .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1, |
| }; |
