[FFmpeg-cvslog] r23646 - in trunk/libavcodec: mpegaudio.h mpegaudiodec.c
vitor
subversion
Sat Jun 19 11:56:06 CEST 2010
Author: vitor
Date: Sat Jun 19 11:56:05 2010
New Revision: 23646
Log:
Factorize the mpegaudio windowing code in a function and call it by a
function pointer. Should allow for ASM optimizations.
Modified:
trunk/libavcodec/mpegaudio.h
trunk/libavcodec/mpegaudiodec.c
Modified: trunk/libavcodec/mpegaudio.h
==============================================================================
--- trunk/libavcodec/mpegaudio.h Sat Jun 19 01:17:20 2010 (r23645)
+++ trunk/libavcodec/mpegaudio.h Sat Jun 19 11:56:05 2010 (r23646)
@@ -156,6 +156,8 @@ typedef struct MPADecodeContext {
int dither_state;
int error_recognition;
AVCodecContext* avctx;
+ void (*apply_window_mp3)(MPA_INT *synth_buf, MPA_INT *window,
+ int *dither_state, OUT_INT *samples, int incr);
} MPADecodeContext;
/* layer 3 huffman tables */
@@ -175,7 +177,8 @@ void ff_mpa_synth_filter(MPA_INT *synth_
INTFLOAT sb_samples[SBLIMIT]);
void ff_mpa_synth_init_float(MPA_INT *window);
-void ff_mpa_synth_filter_float(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
+void ff_mpa_synth_filter_float(MPADecodeContext *s,
+ MPA_INT *synth_buf_ptr, int *synth_buf_offset,
MPA_INT *window, int *dither_state,
OUT_INT *samples, int incr,
INTFLOAT sb_samples[SBLIMIT]);
Modified: trunk/libavcodec/mpegaudiodec.c
==============================================================================
--- trunk/libavcodec/mpegaudiodec.c Sat Jun 19 01:17:20 2010 (r23645)
+++ trunk/libavcodec/mpegaudiodec.c Sat Jun 19 11:56:05 2010 (r23646)
@@ -69,6 +69,8 @@
static void compute_antialias_integer(MPADecodeContext *s, GranuleDef *g);
static void compute_antialias_float(MPADecodeContext *s, GranuleDef *g);
+static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
+ int *dither_state, OUT_INT *samples, int incr);
/* vlc structure for decoding layer 3 huffman tables */
static VLC huff_vlc[16];
@@ -305,6 +307,7 @@ static av_cold int decode_init(AVCodecCo
int i, j, k;
s->avctx = avctx;
+ s->apply_window_mp3 = apply_window_mp3_c;
avctx->sample_fmt= OUT_FMT;
s->error_recognition= avctx->error_recognition;
@@ -836,41 +839,20 @@ void av_cold RENAME(ff_mpa_synth_init)(M
}
}
-/* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
- 32 samples. */
-/* XXX: optimize by avoiding ring buffer usage */
-void RENAME(ff_mpa_synth_filter)(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
- MPA_INT *window, int *dither_state,
- OUT_INT *samples, int incr,
- INTFLOAT sb_samples[SBLIMIT])
+static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
+ int *dither_state, OUT_INT *samples, int incr)
{
- register MPA_INT *synth_buf;
register const MPA_INT *w, *w2, *p;
- int j, offset;
+ int j;
OUT_INT *samples2;
#if CONFIG_FLOAT
float sum, sum2;
#elif FRAC_BITS <= 15
- int32_t tmp[32];
int sum, sum2;
#else
int64_t sum, sum2;
#endif
- offset = *synth_buf_offset;
- synth_buf = synth_buf_ptr + offset;
-
-#if FRAC_BITS <= 15 && !CONFIG_FLOAT
- dct32(tmp, sb_samples);
- for(j=0;j<32;j++) {
- /* NOTE: can cause a loss in precision if very high amplitude
- sound */
- synth_buf[j] = av_clip_int16(tmp[j]);
- }
-#else
- dct32(synth_buf, sb_samples);
-#endif
-
/* copy to avoid wrap */
memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
@@ -909,10 +891,63 @@ void RENAME(ff_mpa_synth_filter)(MPA_INT
SUM8(MLSS, sum, w + 32, p);
*samples = round_sample(&sum);
*dither_state= sum;
+}
+
+
+/* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
+ 32 samples. */
+/* XXX: optimize by avoiding ring buffer usage */
+#if CONFIG_FLOAT
+void ff_mpa_synth_filter_float(MPADecodeContext *s, float *synth_buf_ptr,
+ int *synth_buf_offset,
+ float *window, int *dither_state,
+ float *samples, int incr,
+ float sb_samples[SBLIMIT])
+{
+ float *synth_buf;
+ int offset;
+
+ offset = *synth_buf_offset;
+ synth_buf = synth_buf_ptr + offset;
+
+ dct32(synth_buf, sb_samples);
+ s->apply_window_mp3(synth_buf, window, dither_state, samples, incr);
+
+ offset = (offset - 32) & 511;
+ *synth_buf_offset = offset;
+}
+#else
+void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
+ MPA_INT *window, int *dither_state,
+ OUT_INT *samples, int incr,
+ INTFLOAT sb_samples[SBLIMIT])
+{
+ register MPA_INT *synth_buf;
+ int offset;
+#if FRAC_BITS <= 15
+ int32_t tmp[32];
+#endif
+
+ offset = *synth_buf_offset;
+ synth_buf = synth_buf_ptr + offset;
+
+#if FRAC_BITS <= 15 && !CONFIG_FLOAT
+ dct32(tmp, sb_samples);
+ for(j=0;j<32;j++) {
+ /* NOTE: can cause a loss in precision if very high amplitude
+ sound */
+ synth_buf[j] = av_clip_int16(tmp[j]);
+ }
+#else
+ dct32(synth_buf, sb_samples);
+#endif
+
+ apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
offset = (offset - 32) & 511;
*synth_buf_offset = offset;
}
+#endif
#define C3 FIXHR(0.86602540378443864676/2)
@@ -2227,7 +2262,11 @@ static int mp_decode_frame(MPADecodeCont
for(ch=0;ch<s->nb_channels;ch++) {
samples_ptr = samples + ch;
for(i=0;i<nb_frames;i++) {
- RENAME(ff_mpa_synth_filter)(s->synth_buf[ch], &(s->synth_buf_offset[ch]),
+ RENAME(ff_mpa_synth_filter)(
+#if CONFIG_FLOAT
+ s,
+#endif
+ s->synth_buf[ch], &(s->synth_buf_offset[ch]),
RENAME(ff_mpa_synth_window), &s->dither_state,
samples_ptr, s->nb_channels,
s->sb_samples[ch][i]);
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