[FFmpeg-cvslog] r14705 - in trunk/libavcodec: ac3dec.c ac3dec.h
lorenm
subversion
Tue Aug 12 03:30:24 CEST 2008
Author: lorenm
Date: Tue Aug 12 03:30:24 2008
New Revision: 14705
Log:
use imdct_half in ac3
Modified:
trunk/libavcodec/ac3dec.c
trunk/libavcodec/ac3dec.h
Modified: trunk/libavcodec/ac3dec.c
==============================================================================
--- trunk/libavcodec/ac3dec.c (original)
+++ trunk/libavcodec/ac3dec.c Tue Aug 12 03:30:24 2008
@@ -589,47 +589,6 @@ static void do_rematrixing(AC3DecodeCont
}
/**
- * Perform the 256-point IMDCT
- */
-static void do_imdct_256(AC3DecodeContext *s, int chindex)
-{
- int i, k;
- DECLARE_ALIGNED_16(float, x[128]);
- FFTComplex z[2][64];
- float *o_ptr = s->tmp_output;
-
- for(i=0; i<2; i++) {
- /* de-interleave coefficients */
- for(k=0; k<128; k++) {
- x[k] = s->transform_coeffs[chindex][2*k+i];
- }
-
- /* run standard IMDCT */
- ff_imdct_calc(&s->imdct_256, o_ptr, x);
-
- /* reverse the post-rotation & reordering from standard IMDCT */
- for(k=0; k<32; k++) {
- z[i][32+k].re = -o_ptr[128+2*k];
- z[i][32+k].im = -o_ptr[2*k];
- z[i][31-k].re = o_ptr[2*k+1];
- z[i][31-k].im = o_ptr[128+2*k+1];
- }
- }
-
- /* apply AC-3 post-rotation & reordering */
- for(k=0; k<64; k++) {
- o_ptr[ 2*k ] = -z[0][ k].im;
- o_ptr[ 2*k+1] = z[0][63-k].re;
- o_ptr[128+2*k ] = -z[0][ k].re;
- o_ptr[128+2*k+1] = z[0][63-k].im;
- o_ptr[256+2*k ] = -z[1][ k].re;
- o_ptr[256+2*k+1] = z[1][63-k].im;
- o_ptr[384+2*k ] = z[1][ k].im;
- o_ptr[384+2*k+1] = -z[1][63-k].re;
- }
-}
-
-/**
* Inverse MDCT Transform.
* Convert frequency domain coefficients to time-domain audio samples.
* reference: Section 7.9.4 Transformation Equations
@@ -640,18 +599,20 @@ static inline void do_imdct(AC3DecodeCon
for (ch=1; ch<=channels; ch++) {
if (s->block_switch[ch]) {
- do_imdct_256(s, ch);
+ int i;
+ float *x = s->tmp_output+128;
+ for(i=0; i<128; i++)
+ x[i] = s->transform_coeffs[ch][2*i];
+ ff_imdct_half(&s->imdct_256, s->tmp_output, x);
+ s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 0, 128);
+ for(i=0; i<128; i++)
+ x[i] = s->transform_coeffs[ch][2*i+1];
+ ff_imdct_half(&s->imdct_256, s->delay[ch-1], x);
} else {
- ff_imdct_calc(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
+ ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
+ s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 0, 128);
+ memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float));
}
- /* For the first half of the block, apply the window, add the delay
- from the previous block, and send to output */
- s->dsp.vector_fmul_add_add(s->output[ch-1], s->tmp_output,
- s->window, s->delay[ch-1], 0, 256, 1);
- /* For the second half of the block, apply the window and store the
- samples to delay, to be combined with the next block */
- s->dsp.vector_fmul_reverse(s->delay[ch-1], s->tmp_output+256,
- s->window, 256);
}
}
@@ -686,7 +647,7 @@ static void ac3_downmix(AC3DecodeContext
*/
static void ac3_upmix_delay(AC3DecodeContext *s)
{
- int channel_data_size = sizeof(s->delay[0]);
+ int channel_data_size = 128*sizeof(float);
switch(s->channel_mode) {
case AC3_CHMODE_DUALMONO:
case AC3_CHMODE_STEREO:
@@ -1050,6 +1011,7 @@ static int decode_audio_block(AC3DecodeC
if(!s->downmixed) {
s->downmixed = 1;
+ // FIXME delay[] is half the size of the other downmixes
ac3_downmix(s, s->delay, 0);
}
Modified: trunk/libavcodec/ac3dec.h
==============================================================================
--- trunk/libavcodec/ac3dec.h (original)
+++ trunk/libavcodec/ac3dec.h Tue Aug 12 03:30:24 2008
@@ -165,7 +165,7 @@ typedef struct {
DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][AC3_MAX_COEFS]); ///< transform coefficients
DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]); ///< delay - added to the next block
DECLARE_ALIGNED_16(float, window[AC3_BLOCK_SIZE]); ///< window coefficients
- DECLARE_ALIGNED_16(float, tmp_output[AC3_BLOCK_SIZE*2]); ///< temporary storage for output before windowing
+ DECLARE_ALIGNED_16(float, tmp_output[AC3_BLOCK_SIZE]); ///< temporary storage for output before windowing
DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]); ///< output after imdct transform and windowing
DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][AC3_BLOCK_SIZE]); ///< final 16-bit integer output
///@}
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