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authorJustin Ruggles <justin.ruggles@gmail.com>2011-10-29 00:42:48 -0400
committerJustin Ruggles <justin.ruggles@gmail.com>2011-11-08 12:36:48 -0500
commit0131e70af51ccaeb7faadef001a1aa1fea0271e2 (patch)
tree1c3bca7f08296bc20e5d572ed2567c6e6adc4c84 /libavcodec/ra288.c
parent03e5d6118c239edc0f420d8912454d8b37f789f3 (diff)
ra288: utilize DSPContext.vector_fmul()
Diffstat (limited to 'libavcodec/ra288.c')
-rw-r--r--libavcodec/ra288.c34
1 files changed, 18 insertions, 16 deletions
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);
}
}