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authorJustin Ruggles <justin.ruggles@gmail.com>2007-12-16 04:25:50 +0000
committerJustin Ruggles <justin.ruggles@gmail.com>2007-12-16 04:25:50 +0000
commitb40211ff676da07be3a01c7ae810052da15a0913 (patch)
tree6d39a9e4bf079cc7d8a226959e7b34d32421b7b5
parent15c57ced2fd5999ca89521d8b291582b3ae022c1 (diff)
downloadffmpeg-b40211ff676da07be3a01c7ae810052da15a0913.tar.gz
downmix before the IMDCT if no block switching is used
Originally committed as revision 11228 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavcodec/ac3dec.c47
1 files changed, 34 insertions, 13 deletions
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 33f87ab7ff..1a7cfc1241 100644
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -750,27 +750,28 @@ static inline void do_imdct(AC3DecodeContext *ctx)
/**
* Downmix the output to mono or stereo.
*/
-static void ac3_downmix(float samples[AC3_MAX_CHANNELS][256], int fbw_channels,
- int output_mode, float coef[AC3_MAX_CHANNELS][2])
+static void ac3_downmix(float samples[][256], int fbw_channels,
+ int output_mode, float coef[AC3_MAX_CHANNELS][2],
+ int ch_offset)
{
int i, j;
float v0, v1, s0, s1;
for(i=0; i<256; i++) {
v0 = v1 = s0 = s1 = 0.0f;
- for(j=0; j<fbw_channels; j++) {
- v0 += samples[j][i] * coef[j][0];
- v1 += samples[j][i] * coef[j][1];
- s0 += coef[j][0];
- s1 += coef[j][1];
+ for(j=ch_offset; j<fbw_channels+ch_offset; j++) {
+ v0 += samples[j][i] * coef[j-ch_offset][0];
+ v1 += samples[j][i] * coef[j-ch_offset][1];
+ s0 += coef[j-ch_offset][0];
+ s1 += coef[j-ch_offset][1];
}
v0 /= s0;
v1 /= s1;
if(output_mode == AC3_CHMODE_MONO) {
- samples[0][i] = (v0 + v1) * LEVEL_MINUS_3DB;
+ samples[ch_offset][i] = (v0 + v1) * LEVEL_MINUS_3DB;
} else if(output_mode == AC3_CHMODE_STEREO) {
- samples[0][i] = v0;
- samples[1][i] = v1;
+ samples[ch_offset][i] = v0;
+ samples[ch_offset+1][i] = v1;
}
}
}
@@ -785,12 +786,16 @@ static int ac3_parse_audio_block(AC3DecodeContext *ctx, int blk)
int i, bnd, seg, ch;
GetBitContext *gb = &ctx->gb;
uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
+ int any_block_switching = 0;
+ int num_channels_bak, fbw_channels_bak;
memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
/* block switch flags */
- for (ch = 1; ch <= fbw_channels; ch++)
+ for (ch = 1; ch <= fbw_channels; ch++) {
ctx->block_switch[ch] = get_bits1(gb);
+ any_block_switching |= ctx->block_switch[ch];
+ }
/* dithering flags */
ctx->dither_all = 1;
@@ -1062,13 +1067,26 @@ static int ac3_parse_audio_block(AC3DecodeContext *ctx, int blk)
}
}
+ /* if no block switching is used, downmixing can be done before IMDCT */
+ num_channels_bak = ctx->channels;
+ fbw_channels_bak = ctx->fbw_channels;
+ if(!any_block_switching) {
+ if(ctx->channels != ctx->out_channels && !((ctx->output_mode & AC3_OUTPUT_LFEON) &&
+ ctx->fbw_channels == ctx->out_channels)) {
+ ac3_downmix(ctx->transform_coeffs, ctx->fbw_channels,
+ ctx->output_mode, ctx->downmix_coeffs, 1);
+ ctx->channels = ctx->out_channels;
+ ctx->fbw_channels = ctx->channels - (ctx->output_mode & AC3_OUTPUT_LFEON);
+ }
+ }
+
do_imdct(ctx);
- /* downmix output if needed */
+ /* downmix output now if it wasn't done before IMDCT */
if(ctx->channels != ctx->out_channels && !((ctx->output_mode & AC3_OUTPUT_LFEON) &&
ctx->fbw_channels == ctx->out_channels)) {
ac3_downmix(ctx->output, ctx->fbw_channels, ctx->output_mode,
- ctx->downmix_coeffs);
+ ctx->downmix_coeffs, 0);
}
/* convert float to 16-bit integer */
@@ -1079,6 +1097,9 @@ static int ac3_parse_audio_block(AC3DecodeContext *ctx, int blk)
ctx->dsp.float_to_int16(ctx->int_output[ch], ctx->output[ch], 256);
}
+ ctx->channels = num_channels_bak;
+ ctx->fbw_channels = fbw_channels_bak;
+
return 0;
}