From 0131e70af51ccaeb7faadef001a1aa1fea0271e2 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Sat, 29 Oct 2011 00:42:48 -0400 Subject: ra288: utilize DSPContext.vector_fmul() --- libavcodec/ra288.c | 34 ++++++++++++++++++---------------- 1 file changed, 18 insertions(+), 16 deletions(-) (limited to 'libavcodec/ra288.c') diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c index d82e52df2f..c58bc31f62 100644 --- a/libavcodec/ra288.c +++ b/libavcodec/ra288.c @@ -26,6 +26,7 @@ #include "lpc.h" #include "celp_math.h" #include "celp_filters.h" +#include "dsputil.h" #define MAX_BACKWARD_FILTER_ORDER 36 #define MAX_BACKWARD_FILTER_LEN 40 @@ -35,8 +36,9 @@ #define RA288_BLOCKS_PER_FRAME 32 typedef struct { - float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) - float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) + DSPContext dsp; + DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A) + DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB) /** speech data history (spec: SB). * Its first 70 coefficients are updated only at backward filtering. @@ -57,16 +59,12 @@ typedef struct { static av_cold int ra288_decode_init(AVCodecContext *avctx) { + RA288Context *ractx = avctx->priv_data; avctx->sample_fmt = AV_SAMPLE_FMT_FLT; + dsputil_init(&ractx->dsp, avctx); return 0; } -static void apply_window(float *tgt, const float *m1, const float *m2, int n) -{ - while (n--) - *tgt++ = *m1++ * *m2++; -} - static void convolve(float *tgt, const float *src, int len, int n) { for (; n >= 0; n--) @@ -123,15 +121,18 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) * @param out2 pointer to the recursive part of the output * @param window pointer to the windowing function table */ -static void do_hybrid_window(int order, int n, int non_rec, float *out, +static void do_hybrid_window(RA288Context *ractx, + int order, int n, int non_rec, float *out, float *hist, float *out2, const float *window) { int i; float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; - float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC]; + LOCAL_ALIGNED_16(float, work)[FFALIGN(MAX_BACKWARD_FILTER_ORDER + + MAX_BACKWARD_FILTER_LEN + + MAX_BACKWARD_FILTER_NONREC, 8)]; - apply_window(work, window, hist, order + n + non_rec); + ractx->dsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 8)); convolve(buffer1, work + order , n , order); convolve(buffer2, work + order + n, non_rec, order); @@ -148,16 +149,17 @@ static void do_hybrid_window(int order, int n, int non_rec, float *out, /** * Backward synthesis filter, find the LPC coefficients from past speech data. */ -static void backward_filter(float *hist, float *rec, const float *window, +static void backward_filter(RA288Context *ractx, + float *hist, float *rec, const float *window, float *lpc, const float *tab, int order, int n, int non_rec, int move_size) { float temp[MAX_BACKWARD_FILTER_ORDER+1]; - do_hybrid_window(order, n, non_rec, temp, hist, rec, window); + do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) - apply_window(lpc, lpc, tab, order); + ractx->dsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 8)); memmove(hist, hist + n, move_size*sizeof(*hist)); } @@ -198,10 +200,10 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data, out += RA288_BLOCK_SIZE; if ((i & 7) == 3) { - backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, + backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window, ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); - backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, + backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window, ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); } } -- cgit v1.2.3