/* * 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 #include #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 uint8_t block_size; } G729FormatDescription; typedef struct { /// 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]; } G729ChannelContext; typedef struct { AudioDSPContext adsp; G729ChannelContext *channel_context; } 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, .block_size = G729_8K_BLOCK_SIZE, }; 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, .block_size = G729D_6K4_BLOCK_SIZE, }; /** * @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; 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) { int64_t res = 0; while (order--) res += *v1++ * *v2++; if (res > INT32_MAX) return INT32_MAX; else if (res < INT32_MIN) return INT32_MIN; return res; } static av_cold int decoder_init(AVCodecContext * avctx) { G729Context *s = avctx->priv_data; G729ChannelContext *ctx; int c,i,k; if (avctx->channels < 1 || avctx->channels > 2) { av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels); return AVERROR(EINVAL); } avctx->sample_fmt = AV_SAMPLE_FMT_S16P; /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */ avctx->frame_size = SUBFRAME_SIZE << 1; ctx = s->channel_context = av_mallocz(sizeof(G729ChannelContext) * avctx->channels); if (!ctx) return AVERROR(ENOMEM); for (c = 0; c < avctx->channels; c++) { 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) ctx++; } ff_audiodsp_init(&s->adsp); s->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 c, i; int16_t *tmp; G729Formats packet_type; G729Context *s = avctx->priv_data; G729ChannelContext *ctx = s->channel_context; 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; AVFrame *frame = data; frame->nb_samples = SUBFRAME_SIZE<<1; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; if (buf_size && buf_size % (G729_8K_BLOCK_SIZE * avctx->channels) == 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 == G729D_6K4_BLOCK_SIZE * avctx->channels) { 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 (c = 0; c < avctx->channels; c++) { int frame_erasure = 0; ///< frame erasure detected during decoding int bad_pitch = 0; ///< parity check failed int is_periodic = 0; ///< whether one of the subframes is declared as periodic or not out_frame = (int16_t*)frame->data[c]; for (i = 0; i < format->block_size; i++) frame_erasure |= buf[i]; frame_erasure = !frame_erasure; init_get_bits(&gb, buf, 8*format->block_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 */ if (SUBFRAME_SIZE > pitch_delay_int[i]) 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(&s->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( &s->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)); buf += format->block_size; ctx++; } *got_frame_ptr = 1; return format->block_size * avctx->channels; } static av_cold int decode_close(AVCodecContext *avctx) { G729Context *s = avctx->priv_data; av_freep(&s->channel_context); return 0; } 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, .close = decode_close, .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1, };