Chiebot-Mirror
/
FFmpeg
mirror of https://github.com/FFmpeg/FFmpeg.git
8
0
Fork 0
Code Issues Projects Releases Wiki Activity

downmix before imdct unless different size transforms are used. about 20%

Browse Source 
faster 5.1-to-stereo downmixing.

Originally committed as revision 12397 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Justin Ruggles 17 years ago
parent 95049fecb2
commit 38dae9c3a6
1 changed files with 76 additions and 21 deletions
Whitespace
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. 97
      libavcodec/ac3dec.c

97
libavcodec/ac3dec.c
Unescape View File

@ -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;
@ -747,17 +744,45 @@ 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;
}
}
@ -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 */

Write Preview
Loading…
Cancel
Save