[FFmpeg-cvslog] ra288: utilize DSPContext.vector_fmul()
Justin Ruggles
git at videolan.org
Wed Nov 9 03:31:05 CET 2011
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sat Oct 29 00:42:48 2011 -0400| [0131e70af51ccaeb7faadef001a1aa1fea0271e2] | committer: Justin Ruggles
ra288: utilize DSPContext.vector_fmul()
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=0131e70af51ccaeb7faadef001a1aa1fea0271e2
---
libavcodec/ra288.c | 34 ++++++++++++++++++----------------
libavcodec/ra288.h | 9 +++++----
2 files changed, 23 insertions(+), 20 deletions(-)
diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c
index d82e52d..c58bc31 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);
}
}
diff --git a/libavcodec/ra288.h b/libavcodec/ra288.h
index 8857f40..1c98c16 100644
--- a/libavcodec/ra288.h
+++ b/libavcodec/ra288.h
@@ -23,6 +23,7 @@
#define AVCODEC_RA288_H
#include <stdint.h>
+#include "dsputil.h"
static const float amptable[8]={
0.515625, 0.90234375, 1.57910156, 2.76342773,
@@ -96,7 +97,7 @@ static const int16_t codetable[128][5]={
{ 3746, -606, 53, -269, -3301}, { 606, 2018, -1316, 4064, 398}
};
-static const float syn_window[111]={
+DECLARE_ALIGNED(16, static const float, syn_window)[FFALIGN(111, 8)]={
0.576690972, 0.580838025, 0.585013986, 0.589219987, 0.59345597, 0.597723007,
0.602020264, 0.606384277, 0.610748291, 0.615142822, 0.619598389, 0.624084473,
0.628570557, 0.633117676, 0.637695313, 0.642272949, 0.646911621, 0.651580811,
@@ -118,7 +119,7 @@ static const float syn_window[111]={
0.142852783, 0.0954284668,0.0477600098
};
-static const float gain_window[38]={
+DECLARE_ALIGNED(16, static const float, gain_window)[FFALIGN(38, 8)]={
0.505699992, 0.524200022, 0.54339999, 0.563300014, 0.583953857, 0.60534668,
0.627502441, 0.650482178, 0.674316406, 0.699005127, 0.724578857, 0.75112915,
0.778625488, 0.807128906, 0.836669922, 0.86730957, 0.899078369, 0.932006836,
@@ -129,7 +130,7 @@ static const float gain_window[38]={
};
/** synthesis bandwidth broadening table */
-static const float syn_bw_tab[36]={
+DECLARE_ALIGNED(16, static const float, syn_bw_tab)[FFALIGN(36, 8)] = {
0.98828125, 0.976699829, 0.965254128, 0.953942537, 0.942763507, 0.931715488,
0.920796931, 0.910006344, 0.899342179, 0.888803005, 0.878387332, 0.868093729,
0.857920766, 0.847867012, 0.837931097, 0.828111589, 0.818407178, 0.808816493,
@@ -139,7 +140,7 @@ static const float syn_bw_tab[36]={
};
/** gain bandwidth broadening table */
-static const float gain_bw_tab[10]={
+DECLARE_ALIGNED(16, static const float, gain_bw_tab)[FFALIGN(10, 8)] = {
0.90625, 0.821289063, 0.74432373, 0.674499512, 0.61126709,
0.553955078, 0.50201416, 0.454956055, 0.41229248, 0.373657227
};
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