[FFmpeg-cvslog] AAC encoder: fix valgrind errors

[Claudio Freire]

ffmpeg | branch: master | Claudio Freire <klaussfreire at gmail.com> | Tue Apr  5 23:13:44 2016 -0300| [8005b6de4f88c9e3739a3a4ceda4288804788df9] | committer: Claudio Freire

AAC encoder: fix valgrind errors

Move wi.clipping computation outside of psy_lame_window, LFE
channels don't even call that, and make the LFE path also
initialize window_type[1] which is needed by analyze_channel

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8005b6de4f88c9e3739a3a4ceda4288804788df9
---

 libavcodec/aacenc.c |   18 ++++++++++++++++--
 libavcodec/aacpsy.c |   15 ---------------
 2 files changed, 16 insertions(+), 17 deletions(-)

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 023260a..2653cef 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -554,10 +554,11 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             if (!frame)
                 la = NULL;
             if (tag == TYPE_LFE) {
-                wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
+                wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
                 wi[ch].window_shape   = 0;
                 wi[ch].num_windows    = 1;
                 wi[ch].grouping[0]    = 1;
+                wi[ch].clipping[0]    = 0;
 
                 /* Only the lowest 12 coefficients are used in a LFE channel.
                  * The expression below results in only the bottom 8 coefficients
@@ -582,9 +583,22 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             ics->tns_max_bands      = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
                                         ff_tns_max_bands_128 [s->samplerate_index]:
                                         ff_tns_max_bands_1024[s->samplerate_index];
-            clip_avoidance_factor = 0.0f;
+
             for (w = 0; w < ics->num_windows; w++)
                 ics->group_len[w] = wi[ch].grouping[w];
+
+            /* Calculate input sample maximums and evaluate clipping risk */
+            clip_avoidance_factor = 0.0f;
+            for (w = 0; w < ics->num_windows; w++) {
+                const float *wbuf = overlap + w * 128;
+                const int wlen = 2048 / ics->num_windows;
+                float max = 0;
+                int j;
+                /* mdct input is 2 * output */
+                for (j = 0; j < wlen; j++)
+                    max = FFMAX(max, fabsf(wbuf[j]));
+                wi[ch].clipping[w] = max;
+            }
             for (w = 0; w < ics->num_windows; w++) {
                 if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
                     ics->window_clipping[w] = 1;
diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
index 7453f2d..6dbec3c 100644
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -975,21 +975,6 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
 
     lame_apply_block_type(pch, &wi, uselongblock);
 
-    /* Calculate input sample maximums and evaluate clipping risk */
-    if (audio) {
-        for (i = 0; i < AAC_NUM_BLOCKS_SHORT; i++) {
-            const float *wbuf = audio + i * AAC_BLOCK_SIZE_SHORT;
-            float max = 0;
-            int j;
-            for (j = 0; j < AAC_BLOCK_SIZE_SHORT; j++)
-                max = FFMAX(max, fabsf(wbuf[j]));
-            clippings[i] = max;
-        }
-    } else {
-        for (i = 0; i < 8; i++)
-            clippings[i] = 0;
-    }
-
     wi.window_type[1] = prev_type;
     if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
         float clipping = 0.0f;