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avcodec/g723_1dec: reindent after last commit
This commit is contained in:
Paul B Mahol
parent
62dbcb7ddf
commit
7a124138a7
@ -874,137 +874,137 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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return ret;
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for (int ch = 0; ch < avctx->channels; ch++) {
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G723_1_ChannelContext *p = &s->ch[ch];
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int16_t *audio = p->audio;
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G723_1_ChannelContext *p = &s->ch[ch];
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int16_t *audio = p->audio;
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if (unpack_bitstream(p, buf, buf_size) < 0) {
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bad_frame = 1;
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if (p->past_frame_type == ACTIVE_FRAME)
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p->cur_frame_type = ACTIVE_FRAME;
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else
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p->cur_frame_type = UNTRANSMITTED_FRAME;
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}
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if (unpack_bitstream(p, buf, buf_size) < 0) {
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bad_frame = 1;
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if (p->past_frame_type == ACTIVE_FRAME)
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p->cur_frame_type = ACTIVE_FRAME;
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else
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p->cur_frame_type = UNTRANSMITTED_FRAME;
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}
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out = (int16_t *)frame->extended_data[ch];
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out = (int16_t *)frame->extended_data[ch];
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if (p->cur_frame_type == ACTIVE_FRAME) {
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if (!bad_frame)
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p->erased_frames = 0;
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else if (p->erased_frames != 3)
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p->erased_frames++;
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if (p->cur_frame_type == ACTIVE_FRAME) {
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if (!bad_frame)
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p->erased_frames = 0;
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else if (p->erased_frames != 3)
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p->erased_frames++;
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ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
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ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
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ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
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ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
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/* Save the lsp_vector for the next frame */
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memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
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/* Save the lsp_vector for the next frame */
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memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
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/* Generate the excitation for the frame */
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memcpy(p->excitation, p->prev_excitation,
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PITCH_MAX * sizeof(*p->excitation));
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if (!p->erased_frames) {
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int16_t *vector_ptr = p->excitation + PITCH_MAX;
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/* Update interpolation gain memory */
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p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
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p->subframe[3].amp_index) >> 1];
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for (i = 0; i < SUBFRAMES; i++) {
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gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
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p->pitch_lag[i >> 1], i);
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ff_g723_1_gen_acb_excitation(acb_vector,
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&p->excitation[SUBFRAME_LEN * i],
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p->pitch_lag[i >> 1],
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&p->subframe[i], p->cur_rate);
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/* Get the total excitation */
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for (j = 0; j < SUBFRAME_LEN; j++) {
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int v = av_clip_int16(vector_ptr[j] * 2);
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vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
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}
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vector_ptr += SUBFRAME_LEN;
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}
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vector_ptr = p->excitation + PITCH_MAX;
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p->interp_index = comp_interp_index(p, p->pitch_lag[1],
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&p->sid_gain, &p->cur_gain);
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/* Perform pitch postfiltering */
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if (s->postfilter) {
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i = PITCH_MAX;
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for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
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comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
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ppf + j, p->cur_rate);
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for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
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ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
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vector_ptr + i,
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vector_ptr + i + ppf[j].index,
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ppf[j].sc_gain,
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ppf[j].opt_gain,
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1 << 14, 15, SUBFRAME_LEN);
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} else {
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audio = vector_ptr - LPC_ORDER;
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}
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/* Save the excitation for the next frame */
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memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
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/* Generate the excitation for the frame */
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memcpy(p->excitation, p->prev_excitation,
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PITCH_MAX * sizeof(*p->excitation));
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} else {
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p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
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if (p->erased_frames == 3) {
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/* Mute output */
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memset(p->excitation, 0,
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(FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
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memset(p->prev_excitation, 0,
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PITCH_MAX * sizeof(*p->excitation));
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memset(frame->data[0], 0,
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(FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
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} else {
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int16_t *buf = p->audio + LPC_ORDER;
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if (!p->erased_frames) {
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int16_t *vector_ptr = p->excitation + PITCH_MAX;
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/* Regenerate frame */
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residual_interp(p->excitation, buf, p->interp_index,
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p->interp_gain, &p->random_seed);
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/* Update interpolation gain memory */
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p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
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p->subframe[3].amp_index) >> 1];
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for (i = 0; i < SUBFRAMES; i++) {
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gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
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p->pitch_lag[i >> 1], i);
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ff_g723_1_gen_acb_excitation(acb_vector,
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&p->excitation[SUBFRAME_LEN * i],
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p->pitch_lag[i >> 1],
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&p->subframe[i], p->cur_rate);
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/* Get the total excitation */
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for (j = 0; j < SUBFRAME_LEN; j++) {
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int v = av_clip_int16(vector_ptr[j] * 2);
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vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
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}
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vector_ptr += SUBFRAME_LEN;
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}
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vector_ptr = p->excitation + PITCH_MAX;
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p->interp_index = comp_interp_index(p, p->pitch_lag[1],
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&p->sid_gain, &p->cur_gain);
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/* Perform pitch postfiltering */
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if (s->postfilter) {
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i = PITCH_MAX;
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for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
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comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
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ppf + j, p->cur_rate);
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for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
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ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
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vector_ptr + i,
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vector_ptr + i + ppf[j].index,
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ppf[j].sc_gain,
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ppf[j].opt_gain,
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1 << 14, 15, SUBFRAME_LEN);
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} else {
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audio = vector_ptr - LPC_ORDER;
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}
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/* Save the excitation for the next frame */
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memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
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memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
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PITCH_MAX * sizeof(*p->excitation));
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} else {
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p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
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if (p->erased_frames == 3) {
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/* Mute output */
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memset(p->excitation, 0,
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(FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
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memset(p->prev_excitation, 0,
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PITCH_MAX * sizeof(*p->excitation));
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memset(frame->data[0], 0,
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(FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
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} else {
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int16_t *buf = p->audio + LPC_ORDER;
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/* Regenerate frame */
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residual_interp(p->excitation, buf, p->interp_index,
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p->interp_gain, &p->random_seed);
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/* Save the excitation for the next frame */
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memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
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PITCH_MAX * sizeof(*p->excitation));
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}
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}
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}
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p->cng_random_seed = CNG_RANDOM_SEED;
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} else {
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if (p->cur_frame_type == SID_FRAME) {
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p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
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ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
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} else if (p->past_frame_type == ACTIVE_FRAME) {
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p->sid_gain = estimate_sid_gain(p);
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p->cng_random_seed = CNG_RANDOM_SEED;
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} else {
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if (p->cur_frame_type == SID_FRAME) {
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p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
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ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
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} else if (p->past_frame_type == ACTIVE_FRAME) {
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p->sid_gain = estimate_sid_gain(p);
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}
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if (p->past_frame_type == ACTIVE_FRAME)
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p->cur_gain = p->sid_gain;
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else
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p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
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generate_noise(p);
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ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
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/* Save the lsp_vector for the next frame */
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memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
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}
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if (p->past_frame_type == ACTIVE_FRAME)
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p->cur_gain = p->sid_gain;
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else
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p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
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generate_noise(p);
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ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
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/* Save the lsp_vector for the next frame */
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memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
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}
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p->past_frame_type = p->cur_frame_type;
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p->past_frame_type = p->cur_frame_type;
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memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
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for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
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ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
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audio + i, SUBFRAME_LEN, LPC_ORDER,
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0, 1, 1 << 12);
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memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
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memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
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for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
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ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
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audio + i, SUBFRAME_LEN, LPC_ORDER,
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0, 1, 1 << 12);
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memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
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if (s->postfilter) {
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formant_postfilter(p, lpc, p->audio, out);
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} else { // if output is not postfiltered it should be scaled by 2
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for (i = 0; i < FRAME_LEN; i++)
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out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
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}
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if (s->postfilter) {
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formant_postfilter(p, lpc, p->audio, out);
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} else { // if output is not postfiltered it should be scaled by 2
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for (i = 0; i < FRAME_LEN; i++)
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out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
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}
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}
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*got_frame_ptr = 1;
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