diff options
author | Paul B Mahol <onemda@gmail.com> | 2021-10-03 11:37:50 +0200 |
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committer | Paul B Mahol <onemda@gmail.com> | 2021-10-03 11:42:15 +0200 |
commit | d2cf2cb0bd3ebfeabab6663b59a82f2c69a75e87 (patch) | |
tree | f7fbbdcc2fffe1f5208fdd22daefd1d93ad85510 /libavcodec/amrnbdec.c | |
parent | f282c34c009e3653ec160c3880e64fc1a9300d0e (diff) |
avcodec/amr*dec: reindent
Diffstat (limited to 'libavcodec/amrnbdec.c')
-rw-r--r-- | libavcodec/amrnbdec.c | 228 |
1 files changed, 114 insertions, 114 deletions
diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c index 472fa85f87..5efaa7d1fc 100644 --- a/libavcodec/amrnbdec.c +++ b/libavcodec/amrnbdec.c @@ -181,21 +181,21 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx) for (int ch = 0; ch < avctx->channels; ch++) { AMRContext *p = &s->ch[ch]; - // p->excitation always points to the same position in p->excitation_buf - p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1]; + // p->excitation always points to the same position in p->excitation_buf + p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1]; - for (i = 0; i < LP_FILTER_ORDER; i++) { - p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15); - p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15); - } + for (i = 0; i < LP_FILTER_ORDER; i++) { + p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15); + p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15); + } - for (i = 0; i < 4; i++) - p->prediction_error[i] = MIN_ENERGY; + for (i = 0; i < 4; i++) + p->prediction_error[i] = MIN_ENERGY; - ff_acelp_filter_init(&p->acelpf_ctx); - ff_acelp_vectors_init(&p->acelpv_ctx); - ff_celp_filter_init(&p->celpf_ctx); - ff_celp_math_init(&p->celpm_ctx); + ff_acelp_filter_init(&p->acelpf_ctx); + ff_acelp_vectors_init(&p->acelpv_ctx); + ff_celp_filter_init(&p->celpf_ctx); + ff_celp_math_init(&p->celpm_ctx); } return 0; @@ -980,112 +980,112 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int channel_size; int i, subframe; - p->cur_frame_mode = unpack_bitstream(p, buf, buf_size); + p->cur_frame_mode = unpack_bitstream(p, buf, buf_size); channel_size = frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC - if (p->cur_frame_mode == NO_DATA) { - av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n"); - return AVERROR_INVALIDDATA; - } - if (p->cur_frame_mode == MODE_DTX) { - avpriv_report_missing_feature(avctx, "dtx mode"); - av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n"); - return AVERROR_PATCHWELCOME; - } - - if (p->cur_frame_mode == MODE_12k2) { - lsf2lsp_5(p); - } else - lsf2lsp_3(p); - - for (i = 0; i < 4; i++) - ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5); - - for (subframe = 0; subframe < 4; subframe++) { - const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe]; - - decode_pitch_vector(p, amr_subframe, subframe); - - decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses, - p->cur_frame_mode, subframe); - - // The fixed gain (section 6.1.3) depends on the fixed vector - // (section 6.1.2), but the fixed vector calculation uses - // pitch sharpening based on the on the pitch gain (section 6.1.3). - // So the correct order is: pitch gain, pitch sharpening, fixed gain. - decode_gains(p, amr_subframe, p->cur_frame_mode, subframe, - &fixed_gain_factor); - - pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse); - - if (fixed_sparse.pitch_lag == 0) { - av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n"); + if (p->cur_frame_mode == NO_DATA) { + av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n"); return AVERROR_INVALIDDATA; } - ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0, - AMR_SUBFRAME_SIZE); - - p->fixed_gain[4] = - ff_amr_set_fixed_gain(fixed_gain_factor, - p->celpm_ctx.dot_productf(p->fixed_vector, - p->fixed_vector, - AMR_SUBFRAME_SIZE) / - AMR_SUBFRAME_SIZE, - p->prediction_error, - energy_mean[p->cur_frame_mode], energy_pred_fac); - - // The excitation feedback is calculated without any processing such - // as fixed gain smoothing. This isn't mentioned in the specification. - for (i = 0; i < AMR_SUBFRAME_SIZE; i++) - p->excitation[i] *= p->pitch_gain[4]; - ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4], - AMR_SUBFRAME_SIZE); - - // In the ref decoder, excitation is stored with no fractional bits. - // This step prevents buzz in silent periods. The ref encoder can - // emit long sequences with pitch factor greater than one. This - // creates unwanted feedback if the excitation vector is nonzero. - // (e.g. test sequence T19_795.COD in 3GPP TS 26.074) - for (i = 0; i < AMR_SUBFRAME_SIZE; i++) - p->excitation[i] = truncf(p->excitation[i]); - - // Smooth fixed gain. - // The specification is ambiguous, but in the reference source, the - // smoothed value is NOT fed back into later fixed gain smoothing. - synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe], - p->lsf_avg, p->cur_frame_mode); - - synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector, - synth_fixed_gain, spare_vector); - - if (synthesis(p, p->lpc[subframe], synth_fixed_gain, - synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0)) - // overflow detected -> rerun synthesis scaling pitch vector down - // by a factor of 4, skipping pitch vector contribution emphasis - // and adaptive gain control - synthesis(p, p->lpc[subframe], synth_fixed_gain, - synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1); - - postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); - - // update buffers and history - ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE); - update_state(p); - } + if (p->cur_frame_mode == MODE_DTX) { + avpriv_report_missing_feature(avctx, "dtx mode"); + av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n"); + return AVERROR_PATCHWELCOME; + } + + if (p->cur_frame_mode == MODE_12k2) { + lsf2lsp_5(p); + } else + lsf2lsp_3(p); + + for (i = 0; i < 4; i++) + ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5); + + for (subframe = 0; subframe < 4; subframe++) { + const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe]; + + decode_pitch_vector(p, amr_subframe, subframe); + + decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses, + p->cur_frame_mode, subframe); + + // The fixed gain (section 6.1.3) depends on the fixed vector + // (section 6.1.2), but the fixed vector calculation uses + // pitch sharpening based on the on the pitch gain (section 6.1.3). + // So the correct order is: pitch gain, pitch sharpening, fixed gain. + decode_gains(p, amr_subframe, p->cur_frame_mode, subframe, + &fixed_gain_factor); + + pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse); + + if (fixed_sparse.pitch_lag == 0) { + av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n"); + return AVERROR_INVALIDDATA; + } + ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0, + AMR_SUBFRAME_SIZE); + + p->fixed_gain[4] = + ff_amr_set_fixed_gain(fixed_gain_factor, + p->celpm_ctx.dot_productf(p->fixed_vector, + p->fixed_vector, + AMR_SUBFRAME_SIZE) / + AMR_SUBFRAME_SIZE, + p->prediction_error, + energy_mean[p->cur_frame_mode], energy_pred_fac); + + // The excitation feedback is calculated without any processing such + // as fixed gain smoothing. This isn't mentioned in the specification. + for (i = 0; i < AMR_SUBFRAME_SIZE; i++) + p->excitation[i] *= p->pitch_gain[4]; + ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4], + AMR_SUBFRAME_SIZE); + + // In the ref decoder, excitation is stored with no fractional bits. + // This step prevents buzz in silent periods. The ref encoder can + // emit long sequences with pitch factor greater than one. This + // creates unwanted feedback if the excitation vector is nonzero. + // (e.g. test sequence T19_795.COD in 3GPP TS 26.074) + for (i = 0; i < AMR_SUBFRAME_SIZE; i++) + p->excitation[i] = truncf(p->excitation[i]); + + // Smooth fixed gain. + // The specification is ambiguous, but in the reference source, the + // smoothed value is NOT fed back into later fixed gain smoothing. + synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe], + p->lsf_avg, p->cur_frame_mode); + + synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector, + synth_fixed_gain, spare_vector); + + if (synthesis(p, p->lpc[subframe], synth_fixed_gain, + synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0)) + // overflow detected -> rerun synthesis scaling pitch vector down + // by a factor of 4, skipping pitch vector contribution emphasis + // and adaptive gain control + synthesis(p, p->lpc[subframe], synth_fixed_gain, + synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1); + + postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); + + // update buffers and history + ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE); + update_state(p); + } - p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out, - buf_out, highpass_zeros, - highpass_poles, - highpass_gain * AMR_SAMPLE_SCALE, - p->high_pass_mem, AMR_BLOCK_SIZE); - - /* Update averaged lsf vector (used for fixed gain smoothing). - * - * Note that lsf_avg should not incorporate the current frame's LSFs - * for fixed_gain_smooth. - * The specification has an incorrect formula: the reference decoder uses - * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */ - p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], - 0.84, 0.16, LP_FILTER_ORDER); + p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out, + buf_out, highpass_zeros, + highpass_poles, + highpass_gain * AMR_SAMPLE_SCALE, + p->high_pass_mem, AMR_BLOCK_SIZE); + + /* Update averaged lsf vector (used for fixed gain smoothing). + * + * Note that lsf_avg should not incorporate the current frame's LSFs + * for fixed_gain_smooth. + * The specification has an incorrect formula: the reference decoder uses + * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */ + p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], + 0.84, 0.16, LP_FILTER_ORDER); buf += channel_size; buf_size -= channel_size; } |