Merge remote-tracking branch 'qatar/master'

* qatar/master:
  Remove ffmpeg.
  aacenc: Simplify windowing
  aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
  aacenc: Deinterleave input samples before processing.
  aacenc: Store channel count in AACEncContext.
  aacenc: Move Q^3/4 calculation to it's own table
  aacenc: Request normalized float samples instead of converting s16 samples to float.
  aacpsy: Replace an if with FFMAX in LAME windowing.
  aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
  aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
  aacenc: cosmetics: move init() and end() to the bottom of the file.
  aacenc: aac_encode_init() cleanup
  XWD encoder and decoder
  vc1: don't read the interpfrm and bfraction elements for interlaced frames
  mxfdec: fix memleak on mxf_read_close()
  westwood: split the AUD and VQA demuxers into separate files.

Conflicts:
	.gitignore
	Changelog
	Makefile
	configure
	doc/ffmpeg.texi
	ffmpeg.c
	libavcodec/Makefile
	libavcodec/aacenc.c
	libavcodec/allcodecs.c
	libavcodec/avcodec.h
	libavcodec/version.h
	libavformat/Makefile
	libavformat/img2.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 6 insertions, 11 deletions

diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c
index faadb1b870..b3cfb0e75f 100644
--- a/libavcodec/psymodel.c
+++ b/libavcodec/psymodel.c
@@ -112,20 +112,15 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
     return ctx;
 }
 
-void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
-                       const int16_t *audio, int16_t *dest,
-                       int tag, int channels)
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
 {
-    int ch, i;
+    int ch;
+    int frame_size = ctx->avctx->frame_size;
+
     if (ctx->fstate) {
         for (ch = 0; ch < channels; ch++)
-            ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
-                          audio + ch, ctx->avctx->channels,
-                          dest  + ch, ctx->avctx->channels);
-    } else {
-        for (ch = 0; ch < channels; ch++)
-            for (i = 0; i < ctx->avctx->frame_size; i++)
-                dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
+            ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
+                              &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
     }
 }