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use imdct_half in ac3

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Originally committed as revision 14705 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Loren Merritt 2008-08-12 01:30:24 +00:00
parent 862b98d42c
commit 916d5d6c32
2 changed files with 15 additions and 53 deletions
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66
libavcodec/ac3dec.c
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@ -588,47 +588,6 @@ static void do_rematrixing(AC3DecodeContext *s)
}
}
/**
* 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.
@ -640,18 +599,20 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)
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 *s,
*/
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(AC3DecodeContext *s, int blk)
if(!s->downmixed) {
s->downmixed = 1;
// FIXME delay[] is half the size of the other downmixes
ac3_downmix(s, s->delay, 0);
}

2
libavcodec/ac3dec.h
View File

@ -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
///@}