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ffmpeg-mt
Commit a5405890 authored by jbr's avatar jbr
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downmix before imdct unless different size transforms are used. about 20% 

faster 5.1-to-stereo downmixing.

git-svn-id: file:///var/local/repositories/ffmpeg/trunk@12397 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b
parent c47e6afb
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Showing with 76 additions and 21 deletions
libavcodec/ac3dec.c
View file @ a5405890
......@@ -171,6 +171,7 @@ typedef struct {
int fixed_coeffs[AC3_MAX_CHANNELS][256]; ///> fixed-point transform coefficients
DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients
int downmixed; ///< indicates if coeffs are currently downmixed
/* For IMDCT. */
MDCTContext imdct_512; ///< for 512 sample IMDCT
......@@ -179,9 +180,9 @@ typedef struct {
float add_bias; ///< offset for float_to_int16 conversion
float mul_bias; ///< scaling for float_to_int16 conversion
DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]); ///< output after imdct transform and windowing
DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][256]); ///< output after imdct transform and windowing
DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output
DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]); ///< delay - added to the next block
DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS][256]); ///< delay - added to the next block
DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform
DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing
DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients
......@@ -287,6 +288,7 @@ static int ac3_decode_init(AVCodecContext *avctx)
avctx->request_channels <= 2) {
avctx->channels = avctx->request_channels;
}
s->downmixed = 1;
return 0;
}
......@@ -708,15 +710,9 @@ static void do_imdct_256(AC3DecodeContext *s, int chindex)
* Convert frequency domain coefficients to time-domain audio samples.
* reference: Section 7.9.4 Transformation Equations
*/
static inline void do_imdct(AC3DecodeContext *s)
static inline void do_imdct(AC3DecodeContext *s, int channels)
{
int ch;
int channels;
/* Don't perform the IMDCT on the LFE channel unless it's used in the output */
channels = s->fbw_channels;
if(s->output_mode & AC3_OUTPUT_LFEON)
channels++;
for (ch=1; ch<=channels; ch++) {
if (s->block_switch[ch]) {
......@@ -739,7 +735,8 @@ static inline void do_imdct(AC3DecodeContext *s)
/**
* Downmix the output to mono or stereo.
*/
static void ac3_downmix(AC3DecodeContext *s)
static void ac3_downmix(AC3DecodeContext *s,
float samples[AC3_MAX_CHANNELS][256], int ch_offset)
{
int i, j;
float v0, v1;
......@@ -747,20 +744,48 @@ static void ac3_downmix(AC3DecodeContext *s)
for(i=0; i<256; i++) {
v0 = v1 = 0.0f;
for(j=0; j<s->fbw_channels; j++) {
v0 += s->output[j][i] * s->downmix_coeffs[j][0];
v1 += s->output[j][i] * s->downmix_coeffs[j][1];
v0 += samples[j+ch_offset][i] * s->downmix_coeffs[j][0];
v1 += samples[j+ch_offset][i] * s->downmix_coeffs[j][1];
}
v0 *= s->downmix_coeff_adjust[0];
v1 *= s->downmix_coeff_adjust[1];
if(s->output_mode == AC3_CHMODE_MONO) {
s->output[0][i] = (v0 + v1) * LEVEL_MINUS_3DB;
samples[ch_offset][i] = (v0 + v1) * LEVEL_MINUS_3DB;
} else if(s->output_mode == AC3_CHMODE_STEREO) {
s->output[0][i] = v0;
s->output[1][i] = v1;
samples[ ch_offset][i] = v0;
samples[1+ch_offset][i] = v1;
}
}
}
/**
* Upmix delay samples from stereo to original channel layout.
*/
static void ac3_upmix_delay(AC3DecodeContext *s)
{
int channel_data_size = sizeof(s->delay[0]);
switch(s->channel_mode) {
case AC3_CHMODE_DUALMONO:
case AC3_CHMODE_STEREO:
/* upmix mono to stereo */
memcpy(s->delay[1], s->delay[0], channel_data_size);
break;
case AC3_CHMODE_2F2R:
memset(s->delay[3], 0, channel_data_size);
case AC3_CHMODE_2F1R:
memset(s->delay[2], 0, channel_data_size);
break;
case AC3_CHMODE_3F2R:
memset(s->delay[4], 0, channel_data_size);
case AC3_CHMODE_3F1R:
memset(s->delay[3], 0, channel_data_size);
case AC3_CHMODE_3F:
memcpy(s->delay[2], s->delay[1], channel_data_size);
memset(s->delay[1], 0, channel_data_size);
break;
}
}
/**
* Parse an audio block from AC-3 bitstream.
*/
......@@ -769,14 +794,20 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
int fbw_channels = s->fbw_channels;
int channel_mode = s->channel_mode;
int i, bnd, seg, ch;
int different_transforms;
int downmix_output;
GetBitContext *gbc = &s->gbc;
uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
/* block switch flags */
for (ch = 1; ch <= fbw_channels; ch++)
different_transforms = 0;
for (ch = 1; ch <= fbw_channels; ch++) {
s->block_switch[ch] = get_bits1(gbc);
if(ch > 1 && s->block_switch[ch] != s->block_switch[1])
different_transforms = 1;
}
/* dithering flags */
s->dither_all = 1;
......@@ -1048,12 +1079,36 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
}
}
do_imdct(s);
/* downmix and MDCT. order depends on whether block switching is used for
any channel in this block. this is because coefficients for the long
and short transforms cannot be mixed. */
downmix_output = s->channels != s->out_channels &&
!((s->output_mode & AC3_OUTPUT_LFEON) &&
s->fbw_channels == s->out_channels);
if(different_transforms) {
/* the delay samples have already been downmixed, so we upmix the delay
samples in order to reconstruct all channels before downmixing. */
if(s->downmixed) {
s->downmixed = 0;
ac3_upmix_delay(s);
}
/* downmix output if needed */
if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
s->fbw_channels == s->out_channels)) {
ac3_downmix(s);
do_imdct(s, s->channels);
if(downmix_output) {
ac3_downmix(s, s->output, 0);
}
} else {
if(downmix_output) {
ac3_downmix(s, s->transform_coeffs, 1);
}
if(!s->downmixed) {
s->downmixed = 1;
ac3_downmix(s, s->delay, 0);
}
do_imdct(s, s->out_channels);
}
/* convert float to 16-bit integer */
......
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