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ffmpeg.wasm-core
Commit 011c9112 authored by Carl Eugen Hoyos's avatar Carl Eugen Hoyos
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lavc/g729dec: Cosmetics, fix indentation after last commit.

parent 641d5215
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Showing with 286 additions and 283 deletions
libavcodec/g729dec.c
View file @ 011c9112
......@@ -272,8 +272,7 @@ static void g729d_get_new_exc(
ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
for(i=0; i<subframe_size; i++)
{
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;
......@@ -289,10 +288,10 @@ static void g729d_get_new_exc(
*/
static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
{
if((past_gain_code[0] >> 1) > past_gain_code[1])
if ((past_gain_code[0] >> 1) > past_gain_code[1])
return 2;
else
return FFMAX(past_onset-1, 0);
return FFMAX(past_onset-1, 0);
}
/**
......@@ -307,24 +306,25 @@ static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const in
{
int i, low_gain_pitch_cnt, voice_decision;
if(past_gain_pitch[0] >= 14745) // 0.9
if (past_gain_pitch[0] >= 14745) { // 0.9
voice_decision = DECISION_VOICE;
else if (past_gain_pitch[0] <= 9830) // 0.6
} else if (past_gain_pitch[0] <= 9830) { // 0.6
voice_decision = DECISION_NOISE;
else
} else {
voice_decision = DECISION_INTERMEDIATE;
}
for(i=0, low_gain_pitch_cnt=0; i<6; i++)
if(past_gain_pitch[i] < 9830)
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)
if (low_gain_pitch_cnt > 2 && !onset)
voice_decision = DECISION_NOISE;
if(!onset && voice_decision > prev_voice_decision + 1)
if (!onset && voice_decision > prev_voice_decision + 1)
voice_decision--;
if(onset && voice_decision < DECISION_VOICE)
if (onset && voice_decision < DECISION_VOICE)
voice_decision++;
return voice_decision;
......@@ -361,30 +361,30 @@ static av_cold int decoder_init(AVCodecContext * avctx)
return AVERROR(ENOMEM);
for (c = 0; c < avctx->channels; c++) {
ctx->gain_coeff = 16384; // 1.0 in (1.14)
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;
}
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->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->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
/* random seed initialization */
ctx->rand_value = 21845;
/* 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)
/* quantized prediction error */
for (i = 0; i < 4; i++)
ctx->quant_energy[i] = -14336; // -14 in (5.10)
ctx++;
ctx++;
}
ff_audiodsp_init(&s->adsp);
......@@ -441,286 +441,289 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
}
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 < 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;
}
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];
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);
for (i = 0; i < buf_size; i++)
frame_erasure |= buf[i];
frame_erasure = !frame_erasure;
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);
}
init_get_bits(&gb, buf, 8*buf_size);
/* 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;
}
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) {
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;
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);
}
}
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.
/* 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;
}
/ 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]);
if (frame_erasure) {
ctx->rand_value = g729_prng(ctx->rand_value);
fc_indexes = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
ctx->past_gain_code[1] = ctx->past_gain_code[0];
ctx->rand_value = g729_prng(ctx->rand_value);
pulses_signs = ctx->rand_value;
}
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];
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;
}
/* 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.
This filter enhances harmonic components of the fixed-codebook vector to
improve the quality of the reconstructed speech.
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.
/ fc_v[i], i < pitch_delay
fc_v[i] = <
\ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
*/
if (packet_type == FORMAT_G729D_6K4) {
gain_corr_factor >>= 1;
ctx->past_gain_code[0] >>= 1;
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
}
#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);
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);
/* 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]);
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);
ctx->gain_coeff = ff_g729_adaptive_gain_control(
gain_before,
gain_after,
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,
ctx->gain_coeff);
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);
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];
/* 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]);
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->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;
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));
/* 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 += packet_type == FORMAT_G729_8K ? G729_8K_BLOCK_SIZE : G729D_6K4_BLOCK_SIZE;
ctx++;
buf += packet_type == FORMAT_G729_8K ? G729_8K_BLOCK_SIZE : G729D_6K4_BLOCK_SIZE;
ctx++;
}
*got_frame_ptr = 1;
......
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