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DCA: optimise dca_subsubframe()
8% faster overall on Cortex-A8. Originally committed as revision 22871 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -919,7 +919,8 @@ static int dca_subsubframe(DCAContext * s)
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const float *quant_step_table;
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/* FIXME */
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float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
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LOCAL_ALIGNED_16(float, subband_samples, [DCA_PRIM_CHANNELS_MAX], [DCA_SUBBANDS][8]);
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LOCAL_ALIGNED_16(int, block, [8]);
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/*
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* Audio data
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@ -939,7 +940,6 @@ static int dca_subsubframe(DCAContext * s)
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int abits = s->bitalloc[k][l];
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float quant_step_size = quant_step_table[abits];
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float rscale;
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/*
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* Determine quantization index code book and its type
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@ -953,11 +953,15 @@ static int dca_subsubframe(DCAContext * s)
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*/
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if(!abits){
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memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
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}else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
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} else {
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/* Deal with transients */
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int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
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float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];
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if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
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if(abits <= 7){
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/* Block code */
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int block_code1, block_code2, size, levels;
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int block[8];
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size = abits_sizes[abits-1];
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levels = abits_levels[abits-1];
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@ -967,30 +971,20 @@ static int dca_subsubframe(DCAContext * s)
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decode_blockcode(block_code1, levels, block);
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block_code2 = get_bits(&s->gb, size);
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decode_blockcode(block_code2, levels, &block[4]);
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for (m = 0; m < 8; m++)
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subband_samples[k][l][m] = block[m];
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}else{
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/* no coding */
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for (m = 0; m < 8; m++)
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subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
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block[m] = get_sbits(&s->gb, abits - 3);
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}
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}else{
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/* Huffman coded */
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for (m = 0; m < 8; m++)
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subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
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block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
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}
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/* Deal with transients */
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if (s->transition_mode[k][l] &&
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subsubframe >= s->transition_mode[k][l])
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rscale = quant_step_size * s->scale_factor[k][l][1];
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else
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rscale = quant_step_size * s->scale_factor[k][l][0];
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rscale *= s->scalefactor_adj[k][sel];
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for (m = 0; m < 8; m++)
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subband_samples[k][l][m] *= rscale;
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s->dsp.int32_to_float_fmul_scalar(subband_samples[k][l],
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block, rscale, 8);
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}
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/*
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* Inverse ADPCM if in prediction mode
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