diff options
| -rw-r--r-- | libavcodec/vorbisenc.c | 16 |
1 files changed, 9 insertions, 7 deletions
diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c index 7c3cd51b53..1777a493dd 100644 --- a/libavcodec/vorbisenc.c +++ b/libavcodec/vorbisenc.c @@ -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, float **audio, int samples) { - int i, channel; + int channel; const float * win = venc->win[0]; int window_len = 1 << (venc->log2_blocksize[0] - 1); float n = (float)(1 << venc->log2_blocksize[0]) / 4.0; - // FIXME use dsp + AVFloatDSPContext *fdsp = venc->fdsp; if (!venc->have_saved && !samples) return 0; @@ -1009,9 +1009,10 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, if (samples) { for (channel = 0; channel < venc->channels; channel++) { - 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]; + float *offset = venc->samples + channel * window_len * 2 + window_len; + + fdsp->vector_fmul_reverse(offset, audio[channel], win, samples); + fdsp->vector_fmul_scalar(offset, offset, 1/n, samples); } } else { for (channel = 0; channel < venc->channels; channel++) @@ -1026,8 +1027,9 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, if (samples) { for (channel = 0; channel < venc->channels; channel++) { 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; } else { |