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avcodec/vorbisenc: Use fdsp for applying windows 

Using fdsp improves readability and allows using architecture-specific
optimizations.

Signed-off-by: Tyler Jones <tdjones879@gmail.com>
This commit is contained in:
Tyler Jones 2017-05-30 09:14:36 -06:00 committed by Rostislav Pehlivanov
parent 610864dc36
commit 79941602a3
1 changed files with 9 additions and 7 deletions
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16
libavcodec/vorbisenc.c
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@ -988,11 +988,11 @@ static int residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
static int apply_window_and_mdct(vorbis_enc_context *venc, static int apply_window_and_mdct(vorbis_enc_context *venc,
float **audio, int samples) float **audio, int samples)
{ {
int i, channel; int channel;
const float * win = venc->win[0]; const float * win = venc->win[0];
int window_len = 1 << (venc->log2_blocksize[0] - 1); int window_len = 1 << (venc->log2_blocksize[0] - 1);
float n = (float)(1 << venc->log2_blocksize[0]) / 4.0; float n = (float)(1 << venc->log2_blocksize[0]) / 4.0;
// FIXME use dsp AVFloatDSPContext *fdsp = venc->fdsp;
if (!venc->have_saved && !samples) if (!venc->have_saved && !samples)
return 0; return 0;
@ -1009,9 +1009,10 @@ static int apply_window_and_mdct(vorbis_enc_context *venc,
if (samples) { if (samples) {
for (channel = 0; channel < venc->channels; channel++) { for (channel = 0; channel < venc->channels; channel++) {
float * offset = venc->samples + channel*window_len*2 + window_len; float *offset = venc->samples + channel * window_len * 2 + window_len;
for (i = 0; i < samples; i++)
offset[i] = audio[channel][i] / n * win[window_len - i - 1]; fdsp->vector_fmul_reverse(offset, audio[channel], win, samples);
fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
} }
} else { } else {
for (channel = 0; channel < venc->channels; channel++) for (channel = 0; channel < venc->channels; channel++)
@ -1026,8 +1027,9 @@ static int apply_window_and_mdct(vorbis_enc_context *venc,
if (samples) { if (samples) {
for (channel = 0; channel < venc->channels; channel++) { for (channel = 0; channel < venc->channels; channel++) {
float *offset = venc->saved + channel * window_len; float *offset = venc->saved + channel * window_len;
for (i = 0; i < samples; i++)
offset[i] = audio[channel][i] / n * win[i]; fdsp->vector_fmul(offset, audio[channel], win, samples);
fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
} }
venc->have_saved = 1; venc->have_saved = 1;
} else { } else {