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authorPaul B Mahol <onemda@gmail.com>2018-12-15 13:22:52 +0100
committerPaul B Mahol <onemda@gmail.com>2018-12-18 18:52:39 +0100
commit7a124138a719d3597aa0391711fd4e82eaaff917 (patch)
tree7baa721031e8df160fd73345e1be3b7f4e3c31c5 /libavcodec
parent62dbcb7ddf106ce43018d3087f8fc49286a1fbc5 (diff)
avcodec/g723_1dec: reindent after last commit
Diffstat (limited to 'libavcodec')
-rw-r--r--libavcodec/g723_1dec.c234
1 files changed, 117 insertions, 117 deletions
diff --git a/libavcodec/g723_1dec.c b/libavcodec/g723_1dec.c
index 798d464c71..bc8629fbec 100644
--- a/libavcodec/g723_1dec.c
+++ b/libavcodec/g723_1dec.c
@@ -874,137 +874,137 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
return ret;
for (int ch = 0; ch < avctx->channels; ch++) {
- G723_1_ChannelContext *p = &s->ch[ch];
- int16_t *audio = p->audio;
+ G723_1_ChannelContext *p = &s->ch[ch];
+ int16_t *audio = p->audio;
+
+ if (unpack_bitstream(p, buf, buf_size) < 0) {
+ bad_frame = 1;
+ if (p->past_frame_type == ACTIVE_FRAME)
+ p->cur_frame_type = ACTIVE_FRAME;
+ else
+ p->cur_frame_type = UNTRANSMITTED_FRAME;
+ }
- if (unpack_bitstream(p, buf, buf_size) < 0) {
- bad_frame = 1;
- if (p->past_frame_type == ACTIVE_FRAME)
- p->cur_frame_type = ACTIVE_FRAME;
- else
- p->cur_frame_type = UNTRANSMITTED_FRAME;
- }
+ out = (int16_t *)frame->extended_data[ch];
- out = (int16_t *)frame->extended_data[ch];
-
- if (p->cur_frame_type == ACTIVE_FRAME) {
- if (!bad_frame)
- p->erased_frames = 0;
- else if (p->erased_frames != 3)
- p->erased_frames++;
-
- ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
- ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
-
- /* Save the lsp_vector for the next frame */
- memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
-
- /* Generate the excitation for the frame */
- memcpy(p->excitation, p->prev_excitation,
- PITCH_MAX * sizeof(*p->excitation));
- if (!p->erased_frames) {
- int16_t *vector_ptr = p->excitation + PITCH_MAX;
-
- /* Update interpolation gain memory */
- p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
- p->subframe[3].amp_index) >> 1];
- for (i = 0; i < SUBFRAMES; i++) {
- gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
- p->pitch_lag[i >> 1], i);
- ff_g723_1_gen_acb_excitation(acb_vector,
- &p->excitation[SUBFRAME_LEN * i],
- p->pitch_lag[i >> 1],
- &p->subframe[i], p->cur_rate);
- /* Get the total excitation */
- for (j = 0; j < SUBFRAME_LEN; j++) {
- int v = av_clip_int16(vector_ptr[j] * 2);
- vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
- }
- vector_ptr += SUBFRAME_LEN;
- }
+ if (p->cur_frame_type == ACTIVE_FRAME) {
+ if (!bad_frame)
+ p->erased_frames = 0;
+ else if (p->erased_frames != 3)
+ p->erased_frames++;
- vector_ptr = p->excitation + PITCH_MAX;
-
- p->interp_index = comp_interp_index(p, p->pitch_lag[1],
- &p->sid_gain, &p->cur_gain);
-
- /* Perform pitch postfiltering */
- if (s->postfilter) {
- i = PITCH_MAX;
- for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
- ppf + j, p->cur_rate);
-
- for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
- vector_ptr + i,
- vector_ptr + i + ppf[j].index,
- ppf[j].sc_gain,
- ppf[j].opt_gain,
- 1 << 14, 15, SUBFRAME_LEN);
- } else {
- audio = vector_ptr - LPC_ORDER;
- }
+ ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
+ ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
- /* Save the excitation for the next frame */
- memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
+ /* Save the lsp_vector for the next frame */
+ memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
+
+ /* Generate the excitation for the frame */
+ memcpy(p->excitation, p->prev_excitation,
PITCH_MAX * sizeof(*p->excitation));
- } else {
- p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
- if (p->erased_frames == 3) {
- /* Mute output */
- memset(p->excitation, 0,
- (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
- memset(p->prev_excitation, 0,
- PITCH_MAX * sizeof(*p->excitation));
- memset(frame->data[0], 0,
- (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
- } else {
- int16_t *buf = p->audio + LPC_ORDER;
+ if (!p->erased_frames) {
+ int16_t *vector_ptr = p->excitation + PITCH_MAX;
+
+ /* Update interpolation gain memory */
+ p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
+ p->subframe[3].amp_index) >> 1];
+ for (i = 0; i < SUBFRAMES; i++) {
+ gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
+ p->pitch_lag[i >> 1], i);
+ ff_g723_1_gen_acb_excitation(acb_vector,
+ &p->excitation[SUBFRAME_LEN * i],
+ p->pitch_lag[i >> 1],
+ &p->subframe[i], p->cur_rate);
+ /* Get the total excitation */
+ for (j = 0; j < SUBFRAME_LEN; j++) {
+ int v = av_clip_int16(vector_ptr[j] * 2);
+ vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
+ }
+ vector_ptr += SUBFRAME_LEN;
+ }
- /* Regenerate frame */
- residual_interp(p->excitation, buf, p->interp_index,
- p->interp_gain, &p->random_seed);
+ vector_ptr = p->excitation + PITCH_MAX;
+
+ p->interp_index = comp_interp_index(p, p->pitch_lag[1],
+ &p->sid_gain, &p->cur_gain);
+
+ /* Perform pitch postfiltering */
+ if (s->postfilter) {
+ i = PITCH_MAX;
+ for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
+ ppf + j, p->cur_rate);
+
+ for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
+ vector_ptr + i,
+ vector_ptr + i + ppf[j].index,
+ ppf[j].sc_gain,
+ ppf[j].opt_gain,
+ 1 << 14, 15, SUBFRAME_LEN);
+ } else {
+ audio = vector_ptr - LPC_ORDER;
+ }
/* Save the excitation for the next frame */
- memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
+ memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
PITCH_MAX * sizeof(*p->excitation));
+ } else {
+ p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
+ if (p->erased_frames == 3) {
+ /* Mute output */
+ memset(p->excitation, 0,
+ (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
+ memset(p->prev_excitation, 0,
+ PITCH_MAX * sizeof(*p->excitation));
+ memset(frame->data[0], 0,
+ (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
+ } else {
+ int16_t *buf = p->audio + LPC_ORDER;
+
+ /* Regenerate frame */
+ residual_interp(p->excitation, buf, p->interp_index,
+ p->interp_gain, &p->random_seed);
+
+ /* Save the excitation for the next frame */
+ memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
+ PITCH_MAX * sizeof(*p->excitation));
+ }
+ }
+ p->cng_random_seed = CNG_RANDOM_SEED;
+ } else {
+ if (p->cur_frame_type == SID_FRAME) {
+ p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
+ ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
+ } else if (p->past_frame_type == ACTIVE_FRAME) {
+ p->sid_gain = estimate_sid_gain(p);
}
- }
- p->cng_random_seed = CNG_RANDOM_SEED;
- } else {
- if (p->cur_frame_type == SID_FRAME) {
- p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
- ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
- } else if (p->past_frame_type == ACTIVE_FRAME) {
- p->sid_gain = estimate_sid_gain(p);
- }
- if (p->past_frame_type == ACTIVE_FRAME)
- p->cur_gain = p->sid_gain;
- else
- p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
- generate_noise(p);
- ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
- /* Save the lsp_vector for the next frame */
- memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
- }
+ if (p->past_frame_type == ACTIVE_FRAME)
+ p->cur_gain = p->sid_gain;
+ else
+ p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
+ generate_noise(p);
+ ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
+ /* Save the lsp_vector for the next frame */
+ memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
+ }
- p->past_frame_type = p->cur_frame_type;
+ p->past_frame_type = p->cur_frame_type;
- memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
- for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
- audio + i, SUBFRAME_LEN, LPC_ORDER,
- 0, 1, 1 << 12);
- memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
+ memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
+ for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
+ audio + i, SUBFRAME_LEN, LPC_ORDER,
+ 0, 1, 1 << 12);
+ memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
- if (s->postfilter) {
- formant_postfilter(p, lpc, p->audio, out);
- } else { // if output is not postfiltered it should be scaled by 2
- for (i = 0; i < FRAME_LEN; i++)
- out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
- }
+ if (s->postfilter) {
+ formant_postfilter(p, lpc, p->audio, out);
+ } else { // if output is not postfiltered it should be scaled by 2
+ for (i = 0; i < FRAME_LEN; i++)
+ out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
+ }
}
*got_frame_ptr = 1;