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| author | Tyler Jones <tdjones879@gmail.com> | 2017-05-30 09:14:36 -0600 |
|---|---|---|
| committer | Rostislav Pehlivanov <atomnuker@gmail.com> | 2017-06-05 16:35:03 +0100 |
| commit | 79941602a317b45dcf3d1c4d700fe4678e02a0fe (patch) | |
| tree | aa63494865a091c00523b482226c46dc5e6adba1 | |
| parent | 610864dc36e2c485b4508b774b0199e67e9e8999 (diff) | |
| download | ffmpeg-streaming-79941602a317b45dcf3d1c4d700fe4678e02a0fe.zip ffmpeg-streaming-79941602a317b45dcf3d1c4d700fe4678e02a0fe.tar.gz | |
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>
| -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 7c3cd51..1777a49 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 { |
