1 files changed, 21 insertions, 13 deletions

diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c
index a2af611..22d2497 100644
--- a/libavcodec/psymodel.c
+++ b/libavcodec/psymodel.c
@@ -2,20 +2,20 @@
  * audio encoder psychoacoustic model
  * Copyright (C) 2008 Konstantin Shishkov
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -35,10 +35,10 @@ av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
     int i, j, k = 0;
 
     ctx->avctx = avctx;
-    ctx->ch        = av_mallocz(sizeof(ctx->ch[0]) * avctx->channels * 2);
-    ctx->group     = av_mallocz(sizeof(ctx->group[0]) * num_groups);
-    ctx->bands     = av_malloc (sizeof(ctx->bands[0])     * num_lens);
-    ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
+    ctx->ch        = av_mallocz_array(sizeof(ctx->ch[0]), avctx->channels * 2);
+    ctx->group     = av_mallocz_array(sizeof(ctx->group[0]), num_groups);
+    ctx->bands     = av_malloc_array (sizeof(ctx->bands[0]),      num_lens);
+    ctx->num_bands = av_malloc_array (sizeof(ctx->num_bands[0]),  num_lens);
     memcpy(ctx->bands,     bands,     sizeof(ctx->bands[0])     *  num_lens);
     memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) *  num_lens);
 
@@ -75,7 +75,7 @@ FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
 
 av_cold void ff_psy_end(FFPsyContext *ctx)
 {
-    if (ctx->model->end)
+    if (ctx->model && ctx->model->end)
         ctx->model->end(ctx);
     av_freep(&ctx->bands);
     av_freep(&ctx->num_bands);
@@ -88,6 +88,7 @@ typedef struct FFPsyPreprocessContext{
     float stereo_att;
     struct FFIIRFilterCoeffs *fcoeffs;
     struct FFIIRFilterState **fstate;
+    struct FFIIRFilterContext fiir;
 }FFPsyPreprocessContext;
 
 #define FILT_ORDER 4
@@ -103,15 +104,21 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
     if (avctx->cutoff > 0)
         cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
 
-    if (cutoff_coeff)
+    if (!cutoff_coeff && avctx->codec_id == AV_CODEC_ID_AAC)
+        cutoff_coeff = 2.0 * AAC_CUTOFF(avctx) / avctx->sample_rate;
+
+    if (cutoff_coeff && cutoff_coeff < 0.98)
     ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
                                              FF_FILTER_MODE_LOWPASS, FILT_ORDER,
                                              cutoff_coeff, 0.0, 0.0);
     if (ctx->fcoeffs) {
-        ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
+        ctx->fstate = av_mallocz_array(sizeof(ctx->fstate[0]), avctx->channels);
         for (i = 0; i < avctx->channels; i++)
             ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
     }
+
+    ff_iir_filter_init(&ctx->fiir);
+
     return ctx;
 }
 
@@ -119,11 +126,12 @@ void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int ch
 {
     int ch;
     int frame_size = ctx->avctx->frame_size;
+    FFIIRFilterContext *iir = &ctx->fiir;
 
     if (ctx->fstate) {
         for (ch = 0; ch < channels; ch++)
-            ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
-                              &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
+            iir->filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
+                            &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
     }
 }