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@ -26,6 +26,7 @@ |
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#include "lpc.h" |
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#include "celp_math.h" |
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#include "celp_filters.h" |
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#include "dsputil.h" |
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#define MAX_BACKWARD_FILTER_ORDER 36 |
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#define MAX_BACKWARD_FILTER_LEN 40 |
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@ -35,8 +36,9 @@ |
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#define RA288_BLOCKS_PER_FRAME 32 |
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typedef struct { |
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float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
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float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
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DSPContext dsp; |
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DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A)
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DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB)
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/** speech data history (spec: SB).
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* Its first 70 coefficients are updated only at backward filtering. |
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@ -57,16 +59,12 @@ typedef struct { |
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static av_cold int ra288_decode_init(AVCodecContext *avctx) |
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{ |
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RA288Context *ractx = avctx->priv_data; |
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
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dsputil_init(&ractx->dsp, avctx); |
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return 0; |
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} |
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static void apply_window(float *tgt, const float *m1, const float *m2, int n) |
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{ |
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while (n--) |
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*tgt++ = *m1++ * *m2++; |
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} |
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static void convolve(float *tgt, const float *src, int len, int n) |
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{ |
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for (; n >= 0; n--) |
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@ -123,15 +121,18 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) |
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* @param out2 pointer to the recursive part of the output |
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* @param window pointer to the windowing function table |
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*/ |
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static void do_hybrid_window(int order, int n, int non_rec, float *out, |
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static void do_hybrid_window(RA288Context *ractx, |
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int order, int n, int non_rec, float *out, |
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float *hist, float *out2, const float *window) |
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{ |
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int i; |
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float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; |
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float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; |
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float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC]; |
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LOCAL_ALIGNED_16(float, work)[FFALIGN(MAX_BACKWARD_FILTER_ORDER + |
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MAX_BACKWARD_FILTER_LEN + |
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MAX_BACKWARD_FILTER_NONREC, 8)]; |
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apply_window(work, window, hist, order + n + non_rec); |
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ractx->dsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 8)); |
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convolve(buffer1, work + order , n , order); |
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convolve(buffer2, work + order + n, non_rec, order); |
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@ -148,16 +149,17 @@ static void do_hybrid_window(int order, int n, int non_rec, float *out, |
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/**
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* Backward synthesis filter, find the LPC coefficients from past speech data. |
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*/ |
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static void backward_filter(float *hist, float *rec, const float *window, |
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static void backward_filter(RA288Context *ractx, |
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float *hist, float *rec, const float *window, |
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float *lpc, const float *tab, |
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int order, int n, int non_rec, int move_size) |
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{ |
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float temp[MAX_BACKWARD_FILTER_ORDER+1]; |
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do_hybrid_window(order, n, non_rec, temp, hist, rec, window); |
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do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); |
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if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) |
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apply_window(lpc, lpc, tab, order); |
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ractx->dsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 8)); |
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memmove(hist, hist + n, move_size*sizeof(*hist)); |
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} |
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@ -198,10 +200,10 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data, |
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out += RA288_BLOCK_SIZE; |
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if ((i & 7) == 3) { |
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backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, |
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backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window, |
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ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); |
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backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, |
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backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window, |
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ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); |
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} |
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} |
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