diff options
Diffstat (limited to 'libavcodec/dcadec.c')
| -rw-r--r-- | libavcodec/dcadec.c | 296 |
1 files changed, 153 insertions, 143 deletions
diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c index bf28f3b..b7f6a34 100644 --- a/libavcodec/dcadec.c +++ b/libavcodec/dcadec.c @@ -238,8 +238,8 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, return AVERROR_INVALIDDATA; } - s->total_channels = nchans + base_channel; - s->prim_channels = s->total_channels; + s->audio_header.total_channels = nchans + base_channel; + s->audio_header.prim_channels = s->audio_header.total_channels; /* obtain speaker layout mask & downmix coefficients for XXCH */ if (xxch) { @@ -266,11 +266,11 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, s->xxch_dmix_sf[s->xxch_chset] = scale_factor; - for (i = base_channel; i < s->prim_channels; i++) { + for (i = base_channel; i < s->audio_header.prim_channels; i++) { mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask); } - for (j = base_channel; j < s->prim_channels; j++) { + for (j = base_channel; j < s->audio_header.prim_channels; j++) { memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0])); s->xxch_dmix_embedded |= (embedded_downmix << j); for (i = 0; i < s->xxch_nbits_spk_mask; i++) { @@ -294,40 +294,44 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, } } - if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) - s->prim_channels = DCA_PRIM_CHANNELS_MAX; + if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX) + s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX; - for (i = base_channel; i < s->prim_channels; i++) { - s->subband_activity[i] = get_bits(&s->gb, 5) + 2; - if (s->subband_activity[i] > DCA_SUBBANDS) - s->subband_activity[i] = DCA_SUBBANDS; + for (i = base_channel; i < s->audio_header.prim_channels; i++) { + s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2; + if (s->audio_header.subband_activity[i] > DCA_SUBBANDS) + s->audio_header.subband_activity[i] = DCA_SUBBANDS; } - for (i = base_channel; i < s->prim_channels; i++) { - s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1; - if (s->vq_start_subband[i] > DCA_SUBBANDS) - s->vq_start_subband[i] = DCA_SUBBANDS; + for (i = base_channel; i < s->audio_header.prim_channels; i++) { + s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1; + if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS) + s->audio_header.vq_start_subband[i] = DCA_SUBBANDS; } - get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3); - get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2); - get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3); - get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3); + get_array(&s->gb, s->audio_header.joint_intensity + base_channel, + s->audio_header.prim_channels - base_channel, 3); + get_array(&s->gb, s->audio_header.transient_huffman + base_channel, + s->audio_header.prim_channels - base_channel, 2); + get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel, + s->audio_header.prim_channels - base_channel, 3); + get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel, + s->audio_header.prim_channels - base_channel, 3); /* Get codebooks quantization indexes */ if (!base_channel) - memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman)); + memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.quant_index_huffman)); for (j = 1; j < 11; j++) - for (i = base_channel; i < s->prim_channels; i++) - s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); + for (i = base_channel; i < s->audio_header.prim_channels; i++) + s->audio_header.quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); /* Get scale factor adjustment */ for (j = 0; j < 11; j++) - for (i = base_channel; i < s->prim_channels; i++) - s->scalefactor_adj[i][j] = 1; + for (i = base_channel; i < s->audio_header.prim_channels; i++) + s->audio_header.scalefactor_adj[i][j] = 1; for (j = 1; j < 11; j++) - for (i = base_channel; i < s->prim_channels; i++) - if (s->quant_index_huffman[i][j] < thr[j]) - s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; + for (i = base_channel; i < s->audio_header.prim_channels; i++) + if (s->audio_header.quant_index_huffman[i][j] < thr[j]) + s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; if (!xxch) { if (s->crc_present) { @@ -406,7 +410,7 @@ static int dca_parse_frame_header(DCAContext *s) s->output |= DCA_LFE; /* Primary audio coding header */ - s->subframes = get_bits(&s->gb, 4) + 1; + s->audio_header.subframes = get_bits(&s->gb, 4) + 1; return dca_parse_audio_coding_header(s, 0, 0); } @@ -445,53 +449,53 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3); } - for (j = base_channel; j < s->prim_channels; j++) { - for (k = 0; k < s->subband_activity[j]; k++) - s->prediction_mode[j][k] = get_bits(&s->gb, 1); + for (j = base_channel; j < s->audio_header.prim_channels; j++) { + for (k = 0; k < s->audio_header.subband_activity[j]; k++) + s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1); } /* Get prediction codebook */ - for (j = base_channel; j < s->prim_channels; j++) { - for (k = 0; k < s->subband_activity[j]; k++) { - if (s->prediction_mode[j][k] > 0) { + for (j = base_channel; j < s->audio_header.prim_channels; j++) { + for (k = 0; k < s->audio_header.subband_activity[j]; k++) { + if (s->dca_chan[j].prediction_mode[k] > 0) { /* (Prediction coefficient VQ address) */ - s->prediction_vq[j][k] = get_bits(&s->gb, 12); + s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12); } } } /* Bit allocation index */ - for (j = base_channel; j < s->prim_channels; j++) { - for (k = 0; k < s->vq_start_subband[j]; k++) { - if (s->bitalloc_huffman[j] == 6) - s->bitalloc[j][k] = get_bits(&s->gb, 5); - else if (s->bitalloc_huffman[j] == 5) - s->bitalloc[j][k] = get_bits(&s->gb, 4); - else if (s->bitalloc_huffman[j] == 7) { + for (j = base_channel; j < s->audio_header.prim_channels; j++) { + for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) { + if (s->audio_header.bitalloc_huffman[j] == 6) + s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5); + else if (s->audio_header.bitalloc_huffman[j] == 5) + s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4); + else if (s->audio_header.bitalloc_huffman[j] == 7) { av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation index\n"); return AVERROR_INVALIDDATA; } else { - s->bitalloc[j][k] = - get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]); + s->dca_chan[j].bitalloc[k] = + get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]); } - if (s->bitalloc[j][k] > 26) { + if (s->dca_chan[j].bitalloc[k] > 26) { ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n", - j, k, s->bitalloc[j][k]); + j, k, s->dca_chan[j].bitalloc[k]); return AVERROR_INVALIDDATA; } } } /* Transition mode */ - for (j = base_channel; j < s->prim_channels; j++) { - for (k = 0; k < s->subband_activity[j]; k++) { - s->transition_mode[j][k] = 0; + for (j = base_channel; j < s->audio_header.prim_channels; j++) { + for (k = 0; k < s->audio_header.subband_activity[j]; k++) { + s->dca_chan[j].transition_mode[k] = 0; if (s->subsubframes[s->current_subframe] > 1 && - k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) { - s->transition_mode[j][k] = - get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]); + k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) { + s->dca_chan[j].transition_mode[k] = + get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]); } } } @@ -499,14 +503,14 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) if (get_bits_left(&s->gb) < 0) return AVERROR_INVALIDDATA; - for (j = base_channel; j < s->prim_channels; j++) { + for (j = base_channel; j < s->audio_header.prim_channels; j++) { const uint32_t *scale_table; int scale_sum, log_size; - memset(s->scale_factor[j], 0, - s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2); + memset(s->dca_chan[j].scale_factor, 0, + s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2); - if (s->scalefactor_huffman[j] == 6) { + if (s->audio_header.scalefactor_huffman[j] == 6) { scale_table = ff_dca_scale_factor_quant7; log_size = 7; } else { @@ -517,45 +521,46 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) /* When huffman coded, only the difference is encoded */ scale_sum = 0; - for (k = 0; k < s->subband_activity[j]; k++) { - if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) { - scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size); - s->scale_factor[j][k][0] = scale_table[scale_sum]; + for (k = 0; k < s->audio_header.subband_activity[j]; k++) { + if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) { + scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size); + s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum]; } - if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) { + if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) { /* Get second scale factor */ - scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size); - s->scale_factor[j][k][1] = scale_table[scale_sum]; + scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size); + s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum]; } } } /* Joint subband scale factor codebook select */ - for (j = base_channel; j < s->prim_channels; j++) { + for (j = base_channel; j < s->audio_header.prim_channels; j++) { /* Transmitted only if joint subband coding enabled */ - if (s->joint_intensity[j] > 0) - s->joint_huff[j] = get_bits(&s->gb, 3); + if (s->audio_header.joint_intensity[j] > 0) + s->dca_chan[j].joint_huff = get_bits(&s->gb, 3); } if (get_bits_left(&s->gb) < 0) return AVERROR_INVALIDDATA; /* Scale factors for joint subband coding */ - for (j = base_channel; j < s->prim_channels; j++) { + for (j = base_channel; j < s->audio_header.prim_channels; j++) { int source_channel; /* Transmitted only if joint subband coding enabled */ - if (s->joint_intensity[j] > 0) { + if (s->audio_header.joint_intensity[j] > 0) { int scale = 0; - source_channel = s->joint_intensity[j] - 1; + source_channel = s->audio_header.joint_intensity[j] - 1; /* When huffman coded, only the difference is encoded * (is this valid as well for joint scales ???) */ - for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) { - scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7); - s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ + for (k = s->audio_header.subband_activity[j]; + k < s->audio_header.subband_activity[source_channel]; k++) { + scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7); + s->dca_chan[j].joint_scale_factor[k] = scale; /*joint_scale_table[scale]; */ } if (!(s->debug_flag & 0x02)) { @@ -580,10 +585,10 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) */ /* VQ encoded high frequency subbands */ - for (j = base_channel; j < s->prim_channels; j++) - for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) + for (j = base_channel; j < s->audio_header.prim_channels; j++) + for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++) /* 1 vector -> 32 samples */ - s->high_freq_vq[j][k] = get_bits(&s->gb, 10); + s->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10); /* Low frequency effect data */ if (!base_channel && s->lfe) { @@ -622,7 +627,7 @@ static void qmf_32_subbands(DCAContext *s, int chans, { const float *prCoeff; - int sb_act = s->subband_activity[chans]; + int sb_act = s->audio_header.subband_activity[chans]; scale *= sqrt(1 / 8.0); @@ -633,9 +638,9 @@ static void qmf_32_subbands(DCAContext *s, int chans, prCoeff = ff_dca_fir_32bands_perfect; s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct, - s->subband_fir_hist[chans], - &s->hist_index[chans], - s->subband_fir_noidea[chans], prCoeff, + s->dca_chan[chans].subband_fir_hist, + &s->dca_chan[chans].hist_index, + s->dca_chan[chans].subband_fir_noidea, prCoeff, samples_out, s->raXin, scale); } @@ -670,14 +675,14 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPL { float raXin[64]; float A[32], B[32]; - float *raX = s->subband_fir_hist[chans]; - float *raZ = s->subband_fir_noidea[chans]; + float *raX = s->dca_chan[chans].subband_fir_hist; + float *raZ = s->dca_chan[chans].subband_fir_noidea; unsigned i, j, k, subindex; - for (i = s->subband_activity[chans]; i < 64; i++) + for (i = s->audio_header.subband_activity[chans]; i < 64; i++) raXin[i] = 0.0; for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) { - for (i = 0; i < s->subband_activity[chans]; i++) + for (i = 0; i < s->audio_header.subband_activity[chans]; i++) raXin[i] = samples_in[i][subindex]; for (k = 0; k < 32; k++) { @@ -866,8 +871,6 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) const float *quant_step_table; - /* FIXME */ - float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index]; LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]); /* @@ -880,17 +883,18 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) else quant_step_table = ff_dca_lossy_quant_d; - for (k = base_channel; k < s->prim_channels; k++) { + for (k = base_channel; k < s->audio_header.prim_channels; k++) { + float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index]; float rscale[DCA_SUBBANDS]; if (get_bits_left(&s->gb) < 0) return AVERROR_INVALIDDATA; - for (l = 0; l < s->vq_start_subband[k]; l++) { + for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) { int m; /* Select the mid-tread linear quantizer */ - int abits = s->bitalloc[k][l]; + int abits = s->dca_chan[k].bitalloc[l]; float quant_step_size = quant_step_table[abits]; @@ -899,7 +903,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) */ /* Select quantization index code book */ - int sel = s->quant_index_huffman[k][abits]; + int sel = s->audio_header.quant_index_huffman[k][abits]; /* * Extract bits from the bit stream @@ -909,9 +913,10 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0])); } else { /* Deal with transients */ - int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l]; - rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] * - s->scalefactor_adj[k][sel]; + int sfi = s->dca_chan[k].transition_mode[l] && + subsubframe >= s->dca_chan[k].transition_mode[l]; + rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] * + s->audio_header.scalefactor_adj[k][sel]; if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) { if (abits <= 7) { @@ -944,54 +949,61 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) } } - s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0], - block, rscale, SAMPLES_PER_SUBBAND * s->vq_start_subband[k]); + s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0], + block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]); - for (l = 0; l < s->vq_start_subband[k]; l++) { + for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) { int m; /* * Inverse ADPCM if in prediction mode */ - if (s->prediction_mode[k][l]) { + if (s->dca_chan[k].prediction_mode[l]) { int n; if (s->predictor_history) - subband_samples[k][l][0] += (ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] * - s->subband_samples_hist[k][l][3] + - ff_dca_adpcm_vb[s->prediction_vq[k][l]][1] * - s->subband_samples_hist[k][l][2] + - ff_dca_adpcm_vb[s->prediction_vq[k][l]][2] * - s->subband_samples_hist[k][l][1] + - ff_dca_adpcm_vb[s->prediction_vq[k][l]][3] * - s->subband_samples_hist[k][l][0]) * + subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * + s->dca_chan[k].subband_samples_hist[l][3] + + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] * + s->dca_chan[k].subband_samples_hist[l][2] + + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] * + s->dca_chan[k].subband_samples_hist[l][1] + + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] * + s->dca_chan[k].subband_samples_hist[l][0]) * (1.0f / 8192); for (m = 1; m < SAMPLES_PER_SUBBAND; m++) { - float sum = ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] * - subband_samples[k][l][m - 1]; + float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * + subband_samples[l][m - 1]; for (n = 2; n <= 4; n++) if (m >= n) - sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] * - subband_samples[k][l][m - n]; + sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * + subband_samples[l][m - n]; else if (s->predictor_history) - sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] * - s->subband_samples_hist[k][l][m - n + 4]; - subband_samples[k][l][m] += sum * (1.0f / 8192); + sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * + s->dca_chan[k].subband_samples_hist[l][m - n + 4]; + subband_samples[l][m] += sum * (1.0f / 8192); } } + } + /* Backup predictor history for adpcm */ + for (l = 0; l < DCA_SUBBANDS; l++) + AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]); + /* * Decode VQ encoded high frequencies */ - if (s->subband_activity[k] > s->vq_start_subband[k]) { + if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) { if (!(s->debug_flag & 0x01)) { av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); s->debug_flag |= 0x01; } - s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k], + + s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq, ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND, - s->scale_factor[k], s->vq_start_subband[k], - s->subband_activity[k]); + s->dca_chan[k].scale_factor, + s->audio_header.vq_start_subband[k], + s->audio_header.subband_activity[k]); } } @@ -1003,17 +1015,11 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) } } - /* Backup predictor history for adpcm */ - for (k = base_channel; k < s->prim_channels; k++) - for (l = 0; l < s->vq_start_subband[k]; l++) - AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]); - return 0; } static int dca_filter_channels(DCAContext *s, int block_index, int upsample) { - float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index]; int k; if (upsample) { @@ -1024,18 +1030,22 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample) } /* 64 subbands QMF */ - for (k = 0; k < s->prim_channels; k++) { + for (k = 0; k < s->audio_header.prim_channels; k++) { + float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index]; + if (s->channel_order_tab[k] >= 0) - qmf_64_subbands(s, k, subband_samples[k], + qmf_64_subbands(s, k, subband_samples, s->samples_chanptr[s->channel_order_tab[k]], /* Upsampling needs a factor 2 here. */ M_SQRT2 / 32768.0); } } else { /* 32 subbands QMF */ - for (k = 0; k < s->prim_channels; k++) { + for (k = 0; k < s->audio_header.prim_channels; k++) { + float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index]; + if (s->channel_order_tab[k] >= 0) - qmf_32_subbands(s, k, subband_samples[k], + qmf_32_subbands(s, k, subband_samples, s->samples_chanptr[s->channel_order_tab[k]], M_SQRT1_2 / 32768.0); } @@ -1062,7 +1072,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample) /* FIXME: This downmixing is probably broken with upsample. * Probably totally broken also with XLL in general. */ /* Downmixing to Stereo */ - if (s->prim_channels + !!s->lfe > 2 && + if (s->audio_header.prim_channels + !!s->lfe > 2 && s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef, s->channel_order_tab); @@ -1139,7 +1149,7 @@ static int dca_subframe_footer(DCAContext *s, int base_channel) return AVERROR_INVALIDDATA; } for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) { - for (in = 0; in < s->prim_channels + !!s->lfe; in++) { + for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) { uint16_t tmp = get_bits(&s->gb, 9); if ((tmp & 0xFF) > 241) { av_log(s->avctx, AV_LOG_ERROR, @@ -1185,9 +1195,9 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index) int ret; /* Sanity check */ - if (s->current_subframe >= s->subframes) { + if (s->current_subframe >= s->audio_header.subframes) { av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", - s->current_subframe, s->subframes); + s->current_subframe, s->audio_header.subframes); return AVERROR_INVALIDDATA; } @@ -1207,7 +1217,7 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index) s->current_subsubframe = 0; s->current_subframe++; } - if (s->current_subframe >= s->subframes) { + if (s->current_subframe >= s->audio_header.subframes) { /* Read subframe footer */ if ((ret = dca_subframe_footer(s, base_channel))) return ret; @@ -1260,7 +1270,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s) /* loop over the channel data sets */ /* only decode as many channels as we've decoded base data for */ for(chset = 0, chan_base = 0; - chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->prim_channels; + chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->audio_header.prim_channels; chan_base += n_xbr_ch[chset++]) { int start_posn = get_bits_count(&s->gb); int subsubframe = 0; @@ -1293,7 +1303,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s) int nbits; int scale_table_size; - if (s->scalefactor_huffman[chan_base+i] == 6) { + if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) { scale_table = ff_dca_scale_factor_quant7; scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7); } else { @@ -1312,7 +1322,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s) } scale_table_high[i][j][0] = scale_table[index]; - if(xbr_tmode && s->transition_mode[i][j]) { + if(xbr_tmode && s->dca_chan[i].transition_mode[j]) { int index = get_bits(&s->gb, nbits); if (index >= scale_table_size) { av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index); @@ -1330,9 +1340,9 @@ int ff_dca_xbr_parse_frame(DCAContext *s) for(j = 0; j < active_bands[chset][i]; j++) { const int xbr_abits = abits_high[i][j]; const float quant_step_size = ff_dca_lossless_quant_d[xbr_abits]; - const int sfi = xbr_tmode && s->transition_mode[i][j] && subsubframe >= s->transition_mode[i][j]; + const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j]; const float rscale = quant_step_size * scale_table_high[i][j][sfi]; - float *subband_samples = s->subband_samples[k][chan_base+i][j]; + float *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j]; int block[8]; if(xbr_abits <= 0) @@ -1421,7 +1431,7 @@ int ff_dca_xxch_decode_frame(DCAContext *s) for (chset = 0; chset < num_chsets; chset++) { chstart = get_bits_count(&s->gb); - base_channel = s->prim_channels; + base_channel = s->audio_header.prim_channels; s->xxch_chset = chset; /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs. @@ -1479,7 +1489,7 @@ static int scan_for_extensions(AVCodecContext *avctx) case DCA_SYNCWORD_XCH: { int ext_amode, xch_fsize; - s->xch_base_channel = s->prim_channels; + s->xch_base_channel = s->audio_header.prim_channels; /* validate sync word using XCHFSIZE field */ xch_fsize = show_bits(&s->gb, 10); @@ -1576,7 +1586,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core if (s->amode < 16) { avctx->channel_layout = ff_dca_core_channel_layout[s->amode]; - if (s->prim_channels + !!s->lfe > 2 && + if (s->audio_header.prim_channels + !!s->lfe > 2 && avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { /* * Neither the core's auxiliary data nor our default tables contain @@ -1622,7 +1632,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core if (num_core_channels + !!s->lfe > 2 && avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { *channels = 2; - s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO; + s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO; avctx->channel_layout = AV_CH_LAYOUT_STEREO; } else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) { @@ -1642,7 +1652,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core channel_mask = s->xxch_core_spkmask; { - *channels = s->prim_channels + !!s->lfe; + *channels = s->audio_header.prim_channels + !!s->lfe; for (i = 0; i < s->xxch_chset; i++) { channel_mask |= s->xxch_spk_masks[i]; } @@ -1751,9 +1761,9 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, } /* record number of core channels incase less than max channels are requested */ - num_core_channels = s->prim_channels; + num_core_channels = s->audio_header.prim_channels; - if (s->prim_channels + !!s->lfe > 2 && + if (s->audio_header.prim_channels + !!s->lfe > 2 && avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { /* Stereo downmix coefficients * @@ -1778,7 +1788,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, if (num_core_channels + !!s->lfe > FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) { avpriv_request_sample(s->avctx, "Downmixing %d channels", - s->prim_channels + !!s->lfe); + s->audio_header.prim_channels + !!s->lfe); return AVERROR_PATCHWELCOME; } for (i = 0; i < num_core_channels + !!s->lfe; i++) { @@ -1805,7 +1815,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, avctx->profile = s->profile; - full_channels = channels = s->prim_channels + !!s->lfe; + full_channels = channels = s->audio_header.prim_channels + !!s->lfe; ret = set_channel_layout(avctx, &channels, num_core_channels); if (ret < 0) |
