static int g723_1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { G723_1_Context *p = avctx->priv_data; AVFrame *frame = data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; int dec_mode = buf[0] & 3; PPFParam ppf[SUBFRAMES]; int16_t cur_lsp[LPC_ORDER]; int16_t lpc[SUBFRAMES * LPC_ORDER]; int16_t acb_vector[SUBFRAME_LEN]; int16_t *out; int bad_frame = 0, i, j, ret; int16_t *audio = p->audio; if (buf_size < frame_size[dec_mode]) { if (buf_size) av_log(avctx, AV_LOG_WARNING, "Expected %d bytes, got %d - skipping packet\n", frame_size[dec_mode], buf_size); *got_frame_ptr = 0; return buf_size; } 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; } frame->nb_samples = FRAME_LEN; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; out = (int16_t *)frame->data[0]; 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] << 1); vector_ptr[j] = av_clip_int16(v + acb_vector[j]); } vector_ptr += SUBFRAME_LEN; } 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 (p->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, 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); } 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; 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 (p->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; return frame_size[dec_mode]; }