dca: use defines for subband related constants
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
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8563f98871
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4 changed files with 20 additions and 18 deletions
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@ -140,8 +140,6 @@ static const uint64_t dca_core_channel_layout[] = {
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#define DCA_NSYNCAUX 0x9A1105A0
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#define SAMPLES_PER_SUBBAND 8 // number of samples per subband per subsubframe
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/** Bit allocation */
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typedef struct BitAlloc {
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int offset; ///< code values offset
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@ -544,7 +542,7 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
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}
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static void qmf_32_subbands(DCAContext *s, int chans,
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float samples_in[32][SAMPLES_PER_SUBBAND], float *samples_out,
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float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], float *samples_out,
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float scale)
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{
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const float *prCoeff;
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@ -592,7 +590,8 @@ static QMF64_table *qmf64_precompute(void)
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/* FIXME: Totally unoptimized. Based on the reference code and
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* http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
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* for doubling the size. */
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static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPLES_PER_SUBBAND],
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static void qmf_64_subbands(DCAContext *s, int chans,
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float samples_in[DCA_SUBBANDS_X96K][SAMPLES_PER_SUBBAND],
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float *samples_out, float scale)
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{
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float raXin[64];
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@ -601,7 +600,7 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPL
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float *raZ = s->dca_chan[chans].subband_fir_noidea;
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unsigned i, j, k, subindex;
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for (i = s->audio_header.subband_activity[chans]; i < 64; i++)
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for (i = s->audio_header.subband_activity[chans]; i < DCA_SUBBANDS_X96K; i++)
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raXin[i] = 0.0;
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for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
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for (i = 0; i < s->audio_header.subband_activity[chans]; i++)
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@ -622,14 +621,14 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPL
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raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
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}
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for (i = 0; i < 64; i++) {
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for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
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float out = raZ[i];
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for (j = 0; j < 1024; j += 128)
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out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
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*samples_out++ = out * scale;
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}
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for (i = 0; i < 64; i++) {
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for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
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float hist = 0.0;
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for (j = 0; j < 1024; j += 128)
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hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
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@ -938,7 +937,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
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int k;
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if (upsample) {
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LOCAL_ALIGNED(32, float, samples, [64], [SAMPLES_PER_SUBBAND]);
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LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS_X96K], [SAMPLES_PER_SUBBAND]);
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if (!s->qmf64_table) {
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s->qmf64_table = qmf64_precompute();
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@ -952,7 +951,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
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s->dca_chan[k].subband_samples[block_index];
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s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
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64 * SAMPLES_PER_SUBBAND);
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DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND);
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if (s->channel_order_tab[k] >= 0)
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qmf_64_subbands(s, k, samples,
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@ -962,14 +961,14 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
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}
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} else {
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/* 32 subbands QMF */
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LOCAL_ALIGNED(32, float, samples, [32], [SAMPLES_PER_SUBBAND]);
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LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS], [SAMPLES_PER_SUBBAND]);
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for (k = 0; k < s->audio_header.prim_channels; k++) {
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int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
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s->dca_chan[k].subband_samples[block_index];
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s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
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32 * SAMPLES_PER_SUBBAND);
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DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
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if (s->channel_order_tab[k] >= 0)
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qmf_32_subbands(s, k, samples,
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