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Diffstat (limited to 'libavcodec/g729dec.c')
| -rw-r--r-- | libavcodec/g729dec.c | 718 |
1 files changed, 718 insertions, 0 deletions
diff --git a/libavcodec/g729dec.c b/libavcodec/g729dec.c new file mode 100644 index 0000000..2e1bf18 --- /dev/null +++ b/libavcodec/g729dec.c @@ -0,0 +1,718 @@ +/* + * 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, +}; |
