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Factorize the mpegaudio windowing code in a function and call it by a

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function pointer. Should allow for ASM optimizations.

Originally committed as revision 23646 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Vitor Sessak 2010-06-19 09:56:05 +00:00
parent 57eb217ac4
commit 4d49a5a785
2 changed files with 68 additions and 26 deletions
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5
libavcodec/mpegaudio.h
View File

@ -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_buf_ptr, int *synth_buf_offset,
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]);

89
libavcodec/mpegaudiodec.c
View File

@ -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(AVCodecContext * avctx)
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)(MPA_INT *window)
}
}
/* 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 *synth_buf_ptr, int *synth_buf_offset,
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(MPADecodeContext *s,
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]);