diff --git a/graphics/ffmpeg/Makefile b/graphics/ffmpeg/Makefile
index f80d60dcf3e..cadf9994fc6 100644
--- a/graphics/ffmpeg/Makefile
+++ b/graphics/ffmpeg/Makefile
@@ -1,11 +1,11 @@
-# $OpenBSD: Makefile,v 1.147 2016/03/10 20:41:22 naddy Exp $
+# $OpenBSD: Makefile,v 1.148 2016/04/07 06:01:19 ajacoutot Exp $
 
 COMMENT=	audio/video converter and streamer
 
 V=		20160113
 DISTNAME=	ffmpeg-git-${V}
 PKGNAME=	ffmpeg-${V}
-REVISION=	6
+REVISION=	7
 CATEGORIES=	graphics multimedia
 MASTER_SITES=	http://comstyle.com/source/
 EXTRACT_SUFX=	.tar.xz
diff --git a/graphics/ffmpeg/patches/patch-libavcodec_aaccoder_twoloop_h b/graphics/ffmpeg/patches/patch-libavcodec_aaccoder_twoloop_h
index e187bcab762..40fde5004d2 100644
--- a/graphics/ffmpeg/patches/patch-libavcodec_aaccoder_twoloop_h
+++ b/graphics/ffmpeg/patches/patch-libavcodec_aaccoder_twoloop_h
@@ -1,4 +1,4 @@
-$OpenBSD: patch-libavcodec_aaccoder_twoloop_h,v 1.3 2015/12/06 08:51:46 ajacoutot Exp $
+$OpenBSD: patch-libavcodec_aaccoder_twoloop_h,v 1.4 2016/04/07 06:01:19 ajacoutot Exp $
 
 AAC encoder: refactor to resynchronize MIPS port
 
@@ -12,9 +12,15 @@ avcodec/aac_tablegen: get rid of hardcoded tables entirely
 
 AAC encoder: improve SF range utilization
 
---- libavcodec/aaccoder_twoloop.h.orig	Sat Dec  5 15:13:02 2015
-+++ libavcodec/aaccoder_twoloop.h	Sat Dec  5 15:01:19 2015
-@@ -0,0 +1,755 @@
+AAC encoder: fix signed integer overflow
+
+AAC encoder: fix undefined behavior
+
+AAC encoder: fix initialization of minsf
+
+--- libavcodec/aaccoder_twoloop.h.orig	Sun Apr  3 18:33:06 2016
++++ libavcodec/aaccoder_twoloop.h	Sun Apr  3 18:33:32 2016
+@@ -0,0 +1,763 @@
 +/*
 + * AAC encoder twoloop coder
 + * Copyright (C) 2008-2009 Konstantin Shishkov
@@ -94,7 +100,7 @@ AAC encoder: improve SF range utilization
 +    int toomanybits, toofewbits;
 +    char nzs[128];
 +    uint8_t nextband[128];
-+    int maxsf[128];
++    int maxsf[128], minsf[128];
 +    float dists[128] = { 0 }, qenergies[128] = { 0 }, uplims[128], euplims[128], energies[128];
 +    float maxvals[128], spread_thr_r[128];
 +    float min_spread_thr_r, max_spread_thr_r;
@@ -311,11 +317,19 @@ AAC encoder: improve SF range utilization
 +    abs_pow34_v(s->scoefs, sce->coeffs, 1024);
 +    ff_quantize_band_cost_cache_init(s);
 +
++    for (i = 0; i < sizeof(minsf) / sizeof(minsf[0]); ++i)
++        minsf[i] = 0;
 +    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
 +        start = w*128;
 +        for (g = 0;  g < sce->ics.num_swb; g++) {
 +            const float *scaled = s->scoefs + start;
++            int minsfidx;
 +            maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled);
++            if (maxvals[w*16+g] > 0) {
++                minsfidx = coef2minsf(maxvals[w*16+g]);
++                for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
++                    minsf[(w+w2)*16+g] = minsfidx;
++            }
 +            start += sce->ics.swb_sizes[g];
 +        }
 +    }
@@ -442,7 +456,7 @@ AAC encoder: improve SF range utilization
 +                recomprd = 1;
 +                for (i = 0; i < 128; i++) {
 +                    if (sce->sf_idx[i] > SCALE_ONE_POS) {
-+                        int new_sf = FFMAX(SCALE_ONE_POS, sce->sf_idx[i] - qstep);
++                        int new_sf = FFMAX3(minsf[i], SCALE_ONE_POS, sce->sf_idx[i] - qstep);
 +                        if (new_sf != sce->sf_idx[i]) {
 +                            sce->sf_idx[i] = new_sf;
 +                            changed = 1;
@@ -612,7 +626,7 @@ AAC encoder: improve SF range utilization
 +                    int cmb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
 +                    int mindeltasf = FFMAX(0, prev - SCALE_MAX_DIFF);
 +                    int maxdeltasf = FFMIN(SCALE_MAX_POS - SCALE_DIV_512, prev + SCALE_MAX_DIFF);
-+                    if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > mindeltasf) {
++                    if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > FFMAX(mindeltasf, minsf[w*16+g])) {
 +                        /* Try to make sure there is some energy in every nonzero band
 +                         * NOTE: This algorithm must be forcibly imbalanced, pushing harder
 +                         *  on holes or more distorted bands at first, otherwise there's
diff --git a/graphics/ffmpeg/patches/patch-libavcodec_aacenc_c b/graphics/ffmpeg/patches/patch-libavcodec_aacenc_c
index 17eb0e58319..d3d7df4d251 100644
--- a/graphics/ffmpeg/patches/patch-libavcodec_aacenc_c
+++ b/graphics/ffmpeg/patches/patch-libavcodec_aacenc_c
@@ -1,4 +1,4 @@
-$OpenBSD: patch-libavcodec_aacenc_c,v 1.14 2016/03/09 17:29:36 ajacoutot Exp $
+$OpenBSD: patch-libavcodec_aacenc_c,v 1.15 2016/04/07 06:01:19 ajacoutot Exp $
 
 aacenc: copy PRNG from the decoder
 
@@ -76,8 +76,10 @@ aacenc: temporarily disable Mid/Side coding with multichannel files
 
 aacenc: use generational cache instead of resetting.
 
+AAC encoder: fix valgrind errors
+
 --- libavcodec/aacenc.c.orig	Wed Jan 13 15:27:48 2016
-+++ libavcodec/aacenc.c	Tue Mar  8 19:23:12 2016
++++ libavcodec/aacenc.c	Wed Apr  6 18:35:10 2016
 @@ -29,6 +29,7 @@
   * add sane pulse detection
   ***********************************/
@@ -169,7 +171,7 @@ aacenc: use generational cache instead of resetting.
      int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
      int chan_el_counter[4];
      FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
-@@ -517,10 +540,12 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+@@ -517,19 +540,22 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
          chans    = tag == TYPE_CPE ? 2 : 1;
          cpe      = &s->cpe[i];
          for (ch = 0; ch < chans; ch++) {
@@ -185,7 +187,18 @@ aacenc: use generational cache instead of resetting.
              samples2 = overlap + 1024;
              la       = samples2 + (448+64);
              if (!frame)
-@@ -537,7 +562,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+                 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
+@@ -537,7 +563,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                   */
                  ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
              } else {
@@ -194,7 +207,32 @@ aacenc: use generational cache instead of resetting.
                                                ics->window_sequence[0]);
              }
              ics->window_sequence[1] = ics->window_sequence[0];
-@@ -571,25 +596,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+@@ -554,10 +580,23 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+             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;
+                     clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
+@@ -571,25 +610,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                  ics->clip_avoidance_factor = 1.0f;
              }
  
@@ -236,7 +274,7 @@ aacenc: use generational cache instead of resetting.
          memset(chan_el_counter, 0, sizeof(chan_el_counter));
          for (i = 0; i < s->chan_map[0]; i++) {
              FFPsyWindowInfo* wi = windows + start_ch;
-@@ -606,15 +640,28 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+@@ -606,15 +654,28 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                  sce = &cpe->ch[ch];
                  coeffs[ch] = sce->coeffs;
                  sce->ics.predictor_present = 0;
@@ -266,7 +304,7 @@ aacenc: use generational cache instead of resetting.
                  s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
              }
              if (chans > 1
-@@ -632,14 +679,14 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+@@ -632,14 +693,14 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
              for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
                  sce = &cpe->ch[ch];
                  s->cur_channel = start_ch + ch;
@@ -283,7 +321,7 @@ aacenc: use generational cache instead of resetting.
              }
              s->cur_channel = start_ch;
              if (s->options.intensity_stereo) { /* Intensity Stereo */
-@@ -656,8 +703,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+@@ -656,8 +717,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                          s->coder->search_for_pred(s, sce);
                      if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
                  }
@@ -294,7 +332,7 @@ aacenc: use generational cache instead of resetting.
                  for (ch = 0; ch < chans; ch++) {
                      sce = &cpe->ch[ch];
                      s->cur_channel = start_ch + ch;
-@@ -666,22 +713,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+@@ -666,22 +727,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                  }
                  s->cur_channel = start_ch;
              }
@@ -333,7 +371,7 @@ aacenc: use generational cache instead of resetting.
                      encode_ms_info(&s->pb, cpe);
                      if (cpe->ms_mode) ms_mode = 1;
                  }
-@@ -693,36 +752,78 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
+@@ -693,36 +766,78 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
              start_ch += chans;
          }
  
@@ -432,7 +470,7 @@ aacenc: use generational cache instead of resetting.
      if (!frame)
          s->last_frame++;
  
-@@ -738,6 +839,8 @@ static av_cold int aac_encode_end(AVCodecContext *avct
+@@ -738,6 +853,8 @@ static av_cold int aac_encode_end(AVCodecContext *avct
  {
      AACEncContext *s = avctx->priv_data;
  
@@ -441,7 +479,7 @@ aacenc: use generational cache instead of resetting.
      ff_mdct_end(&s->mdct1024);
      ff_mdct_end(&s->mdct128);
      ff_psy_end(&s->psy);
-@@ -796,76 +899,123 @@ static av_cold int aac_encode_init(AVCodecContext *avc
+@@ -796,76 +913,123 @@ static av_cold int aac_encode_init(AVCodecContext *avc
      uint8_t grouping[AAC_MAX_CHANNELS];
      int lengths[2];
  
@@ -599,7 +637,7 @@ aacenc: use generational cache instead of resetting.
      ff_af_queue_init(avctx, &s->afq);
  
      return 0;
-@@ -876,27 +1026,16 @@ fail:
+@@ -876,27 +1040,16 @@ fail:
  
  #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
  static const AVOption aacenc_options[] = {
@@ -637,7 +675,7 @@ aacenc: use generational cache instead of resetting.
      {NULL}
  };
  
-@@ -907,6 +1046,11 @@ static const AVClass aacenc_class = {
+@@ -907,6 +1060,11 @@ static const AVClass aacenc_class = {
      LIBAVUTIL_VERSION_INT,
  };
  
@@ -649,7 +687,7 @@ aacenc: use generational cache instead of resetting.
  AVCodec ff_aac_encoder = {
      .name           = "aac",
      .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
-@@ -916,9 +1060,9 @@ AVCodec ff_aac_encoder = {
+@@ -916,9 +1074,9 @@ AVCodec ff_aac_encoder = {
      .init           = aac_encode_init,
      .encode2        = aac_encode_frame,
      .close          = aac_encode_end,
diff --git a/graphics/ffmpeg/patches/patch-libavcodec_aacenc_utils_h b/graphics/ffmpeg/patches/patch-libavcodec_aacenc_utils_h
index 702130b8ae2..505135c8d81 100644
--- a/graphics/ffmpeg/patches/patch-libavcodec_aacenc_utils_h
+++ b/graphics/ffmpeg/patches/patch-libavcodec_aacenc_utils_h
@@ -1,4 +1,4 @@
-$OpenBSD: patch-libavcodec_aacenc_utils_h,v 1.9 2016/03/09 17:29:36 ajacoutot Exp $
+$OpenBSD: patch-libavcodec_aacenc_utils_h,v 1.10 2016/04/07 06:01:19 ajacoutot Exp $
 
 aacenc: copy PRNG from the decoder
 
@@ -28,11 +28,15 @@ aacenc: avoid double in quantize_bands.
 
 aacenc_utils: Use temporary variable.
 
+lavc/aacenc_utils: replace powf(x,y) by expf(logf(x), y)
+
 --- libavcodec/aacenc_utils.h.orig	Wed Jan 13 15:27:48 2016
-+++ libavcodec/aacenc_utils.h	Tue Mar  8 19:24:00 2016
-@@ -29,8 +29,8 @@
++++ libavcodec/aacenc_utils.h	Fri Mar 18 20:59:09 2016
+@@ -28,9 +28,10 @@
+ #ifndef AVCODEC_AACENC_UTILS_H
  #define AVCODEC_AACENC_UTILS_H
  
++#include "libavutil/internal.h"
  #include "aac.h"
 -#include "aac_tablegen_decl.h"
  #include "aacenctab.h"
@@ -40,7 +44,7 @@ aacenc_utils: Use temporary variable.
  
  #define ROUND_STANDARD 0.4054f
  #define ROUND_TO_ZERO 0.1054f
-@@ -45,6 +45,11 @@ static inline void abs_pow34_v(float *out, const float
+@@ -45,6 +46,11 @@ static inline void abs_pow34_v(float *out, const float
      }
  }
  
@@ -52,7 +56,7 @@ aacenc_utils: Use temporary variable.
  /**
   * Quantize one coefficient.
   * @return absolute value of the quantized coefficient
-@@ -61,13 +66,13 @@ static inline void quantize_bands(int *out, const floa
+@@ -61,13 +67,13 @@ static inline void quantize_bands(int *out, const floa
                                    const float rounding)
  {
      int i;
@@ -70,7 +74,7 @@ aacenc_utils: Use temporary variable.
      }
  }
  
-@@ -85,20 +90,65 @@ static inline float find_max_val(int group_len, int sw
+@@ -85,20 +91,68 @@ static inline float find_max_val(int group_len, int sw
  
  static inline int find_min_book(float maxval, int sf)
  {
@@ -115,7 +119,10 @@ aacenc_utils: Use temporary variable.
 +            if (s >= ethresh) {
 +                nzl += 1.0f;
 +            } else {
-+                nzl += powf(s / ethresh, nzslope);
++                if (nzslope == 2.f)
++                    nzl += (s / ethresh) * (s / ethresh);
++                else
++                    nzl += ff_fast_powf(s / ethresh, nzslope);
 +            }
 +        }
 +        if (e2 > thresh) {
@@ -145,7 +152,7 @@ aacenc_utils: Use temporary variable.
  /** Return the minimum scalefactor where the quantized coef does not clip. */
  static inline uint8_t coef2minsf(float coef)
  {
-@@ -128,6 +178,76 @@ static inline int quant_array_idx(const float val, con
+@@ -128,6 +182,76 @@ static inline int quant_array_idx(const float val, con
      return index;
  }
  
@@ -222,7 +229,7 @@ aacenc_utils: Use temporary variable.
  #define ERROR_IF(cond, ...) \
      if (cond) { \
          av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
-@@ -138,6 +258,5 @@ static inline int quant_array_idx(const float val, con
+@@ -138,6 +262,5 @@ static inline int quant_array_idx(const float val, con
      if (cond) { \
          av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
      }
diff --git a/graphics/ffmpeg/patches/patch-libavcodec_aacpsy_c b/graphics/ffmpeg/patches/patch-libavcodec_aacpsy_c
index 686ef57c8ce..f338d3ed164 100644
--- a/graphics/ffmpeg/patches/patch-libavcodec_aacpsy_c
+++ b/graphics/ffmpeg/patches/patch-libavcodec_aacpsy_c
@@ -1,4 +1,4 @@
-$OpenBSD: patch-libavcodec_aacpsy_c,v 1.5 2015/12/06 08:51:46 ajacoutot Exp $
+$OpenBSD: patch-libavcodec_aacpsy_c,v 1.6 2016/04/07 06:01:19 ajacoutot Exp $
 
 AAC encoder: tweak rate-distortion logic
 
@@ -10,9 +10,21 @@ AAC encoder: make pe.min a local minimum
 
 AAC encoder: improve SF range utilization
 
---- libavcodec/aacpsy.c.orig	Sat Dec  5 15:14:00 2015
-+++ libavcodec/aacpsy.c	Sat Dec  5 15:01:19 2015
-@@ -80,6 +80,8 @@
+AAC encoder: fix filling of wi.clipping array
+
+AAC encoder: fix valgrind errors
+
+--- libavcodec/aacpsy.c.orig	Wed Jan 13 15:27:48 2016
++++ libavcodec/aacpsy.c	Wed Apr  6 18:31:01 2016
+@@ -25,6 +25,7 @@
+  */
+ 
+ #include "libavutil/attributes.h"
++#include "libavutil/internal.h"
+ #include "libavutil/libm.h"
+ 
+ #include "avcodec.h"
+@@ -80,6 +81,8 @@
  #define PSY_3GPP_AH_THR_LONG    0.5f
  #define PSY_3GPP_AH_THR_SHORT   0.63f
  
@@ -21,7 +33,7 @@ AAC encoder: improve SF range utilization
  enum {
      PSY_3GPP_AH_NONE,
      PSY_3GPP_AH_INACTIVE,
-@@ -87,6 +89,7 @@ enum {
+@@ -87,6 +90,7 @@ enum {
  };
  
  #define PSY_3GPP_BITS_TO_PE(bits) ((bits) * 1.18f)
@@ -29,7 +41,7 @@ AAC encoder: improve SF range utilization
  
  /* LAME psy model constants */
  #define PSY_LAME_FIR_LEN 21         ///< LAME psy model FIR order
-@@ -157,6 +160,7 @@ typedef struct AacPsyContext{
+@@ -157,6 +161,7 @@ typedef struct AacPsyContext{
      } pe;
      AacPsyCoeffs psy_coef[2][64];
      AacPsyChannel *ch;
@@ -37,7 +49,7 @@ AAC encoder: improve SF range utilization
  }AacPsyContext;
  
  /**
-@@ -299,17 +303,24 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
+@@ -299,17 +304,24 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
      float bark;
      int i, j, g, start;
      float prev, minscale, minath, minsnr, pe_min;
@@ -65,7 +77,22 @@ AAC encoder: improve SF range utilization
      pctx->pe.min       =  8.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
      pctx->pe.max       = 12.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
      ctx->bitres.size   = 6144 - pctx->frame_bits;
-@@ -397,7 +408,7 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsy
+@@ -338,10 +350,10 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
+         for (g = 0; g < ctx->num_bands[j] - 1; g++) {
+             AacPsyCoeffs *coeff = &coeffs[g];
+             float bark_width = coeffs[g+1].barks - coeffs->barks;
+-            coeff->spread_low[0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_LOW);
+-            coeff->spread_hi [0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_HI);
+-            coeff->spread_low[1] = pow(10.0, -bark_width * en_spread_low);
+-            coeff->spread_hi [1] = pow(10.0, -bark_width * en_spread_hi);
++            coeff->spread_low[0] = ff_exp10(-bark_width * PSY_3GPP_THR_SPREAD_LOW);
++            coeff->spread_hi [0] = ff_exp10(-bark_width * PSY_3GPP_THR_SPREAD_HI);
++            coeff->spread_low[1] = ff_exp10(-bark_width * en_spread_low);
++            coeff->spread_hi [1] = ff_exp10(-bark_width * en_spread_hi);
+             pe_min = bark_pe * bark_width;
+             minsnr = exp2(pe_min / band_sizes[g]) - 1.5f;
+             coeff->min_snr = av_clipf(1.0f / minsnr, PSY_SNR_25DB, PSY_SNR_1DB);
+@@ -397,7 +409,7 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsy
                                                   int channel, int prev_type)
  {
      int i, j;
@@ -74,7 +101,7 @@ AAC encoder: improve SF range utilization
      int attack_ratio     = br <= 16000 ? 18 : 10;
      AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
      AacPsyChannel *pch  = &pctx->ch[channel];
-@@ -486,7 +497,7 @@ static int calc_bit_demand(AacPsyContext *ctx, float p
+@@ -486,7 +498,7 @@ static int calc_bit_demand(AacPsyContext *ctx, float p
      const float bitspend_add   = short_window ? PSY_3GPP_SPEND_ADD_S   : PSY_3GPP_SPEND_ADD_L;
      const float clip_low       = short_window ? PSY_3GPP_CLIP_LO_S     : PSY_3GPP_CLIP_LO_L;
      const float clip_high      = short_window ? PSY_3GPP_CLIP_HI_S     : PSY_3GPP_CLIP_HI_L;
@@ -83,7 +110,7 @@ AAC encoder: improve SF range utilization
  
      ctx->fill_level += ctx->frame_bits - bits;
      ctx->fill_level  = av_clip(ctx->fill_level, 0, size);
-@@ -503,11 +514,21 @@ static int calc_bit_demand(AacPsyContext *ctx, float p
+@@ -503,11 +515,21 @@ static int calc_bit_demand(AacPsyContext *ctx, float p
       * Hopefully below is correct.
       */
      bit_factor = 1.0f - bit_save + ((bit_spend - bit_save) / (ctx->pe.max - ctx->pe.min)) * (clipped_pe - ctx->pe.min);
@@ -108,7 +135,7 @@ AAC encoder: improve SF range utilization
  }
  
  static float calc_pe_3gpp(AacPsyBand *band)
-@@ -574,26 +595,30 @@ static float calc_reduced_thr_3gpp(AacPsyBand *band, f
+@@ -574,26 +596,30 @@ static float calc_reduced_thr_3gpp(AacPsyBand *band, f
  
  #ifndef calc_thr_3gpp
  static void calc_thr_3gpp(const FFPsyWindowInfo *wi, const int num_bands, AacPsyChannel *pch,
@@ -144,7 +171,7 @@ AAC encoder: improve SF range utilization
          }
      }
  }
-@@ -634,9 +659,11 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
+@@ -634,9 +660,11 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
      const uint8_t *band_sizes  = ctx->bands[wi->num_windows == 8];
      AacPsyCoeffs  *coeffs      = pctx->psy_coef[wi->num_windows == 8];
      const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
@@ -157,7 +184,16 @@ AAC encoder: improve SF range utilization
  
      //modify thresholds and energies - spread, threshold in quiet, pre-echo control
      for (w = 0; w < wi->num_windows*16; w += 16) {
-@@ -677,16 +704,36 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
+@@ -658,7 +686,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
+ 
+             band->thr_quiet = band->thr = FFMAX(band->thr, coeffs[g].ath);
+             //5.4.2.5 "Pre-echo control"
+-            if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (wi->window_type[1] == LONG_START_SEQUENCE && !w)))
++            if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (!w && wi->window_type[1] == LONG_START_SEQUENCE)))
+                 band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
+                                   PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
+ 
+@@ -677,16 +705,36 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
  
      /* 5.6.1.3.2 "Calculation of the desired perceptual entropy" */
      ctx->ch[channel].entropy = pe;
@@ -203,7 +239,7 @@ AAC encoder: improve SF range utilization
  
      if (desired_pe < pe) {
          /* 5.6.1.3.4 "First Estimation of the reduction value" */
-@@ -788,6 +835,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
+@@ -788,6 +836,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
              psy_band->threshold = band->thr;
              psy_band->energy    = band->energy;
              psy_band->spread    = band->active_lines * 2.0f / band_sizes[g];
@@ -211,3 +247,38 @@ AAC encoder: improve SF range utilization
          }
      }
  
+@@ -927,21 +976,6 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *c
+ 
+     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;
+@@ -970,9 +1004,10 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *c
+         for (i = 0; i < 8; i += wi.grouping[i]) {
+             int w;
+             float clipping = 0.0f;
+-            for (w = 0; w < wi.grouping[i] && !clipping; w++)
++            for (w = 0; w < wi.grouping[i]; w++)
+                 clipping = FFMAX(clipping, clippings[i+w]);
+-            wi.clipping[i] = clipping;
++            for (w = 0; w < wi.grouping[i]; w++)
++                wi.clipping[i+w] = clipping;
+         }
+     }
+ 
diff --git a/graphics/ffmpeg/patches/patch-libavcodec_lpc_c b/graphics/ffmpeg/patches/patch-libavcodec_lpc_c
index 216cdbb3332..dd204c2315f 100644
--- a/graphics/ffmpeg/patches/patch-libavcodec_lpc_c
+++ b/graphics/ffmpeg/patches/patch-libavcodec_lpc_c
@@ -1,10 +1,12 @@
-$OpenBSD: patch-libavcodec_lpc_c,v 1.1 2015/10/13 05:44:18 ajacoutot Exp $
+$OpenBSD: patch-libavcodec_lpc_c,v 1.2 2016/04/07 06:01:19 ajacoutot Exp $
 
 lpc: correctly apply windowing to the samples in the float-only lpc
 
---- libavcodec/lpc.c.orig	Mon Oct 12 23:03:10 2015
-+++ libavcodec/lpc.c	Mon Oct 12 23:03:21 2015
-@@ -173,11 +173,13 @@ double ff_lpc_calc_ref_coefs_f(LPCContext *s, const fl
+lavc/lpc: exploit even symmetry of window function
+
+--- libavcodec/lpc.c.orig	Wed Jan 13 15:27:48 2016
++++ libavcodec/lpc.c	Thu Mar 24 20:27:56 2016
+@@ -173,11 +173,14 @@ double ff_lpc_calc_ref_coefs_f(LPCContext *s, const fl
      int i;
      double signal = 0.0f, avg_err = 0.0f;
      double autoc[MAX_LPC_ORDER+1] = {0}, error[MAX_LPC_ORDER+1] = {0};
@@ -15,9 +17,10 @@ lpc: correctly apply windowing to the samples in the float-only lpc
 -    for (i = 0; i < len; i++)
 -        s->windowed_samples[i] = 1.0f - ((samples[i]-c)/c)*((samples[i]-c)/c);
 +    /* Apply windowing */
-+    for (i = 0; i < len; i++) {
++    for (i = 0; i <= len / 2; i++) {
 +        double weight = a - b*cos((2*M_PI*i)/(len - 1));
 +        s->windowed_samples[i] = weight*samples[i];
++        s->windowed_samples[len-1-i] = weight*samples[len-1-i];
 +    }
  
      s->lpc_compute_autocorr(s->windowed_samples, len, order, autoc);
diff --git a/graphics/ffmpeg/patches/patch-libavutil_internal_h b/graphics/ffmpeg/patches/patch-libavutil_internal_h
new file mode 100644
index 00000000000..fe9412b1f1d
--- /dev/null
+++ b/graphics/ffmpeg/patches/patch-libavutil_internal_h
@@ -0,0 +1,51 @@
+$OpenBSD: patch-libavutil_internal_h,v 1.1 2016/04/07 06:01:19 ajacoutot Exp $
+
+lavu/internal: add ff_exp10
+
+lavc/aacenc_utils: replace powf(x,y) by expf(logf(x), y)
+
+--- libavutil/internal.h.orig	Wed Jan 13 15:27:50 2016
++++ libavutil/internal.h	Wed Apr  6 19:54:16 2016
+@@ -257,6 +257,42 @@ void avpriv_request_sample(void *avc,
+ #endif
+ 
+ /**
++ * Compute 10^x for floating point values. Note: this function is by no means
++ * "correctly rounded", and is meant as a fast, reasonably accurate approximation.
++ * For instance, maximum relative error for the double precision variant is
++ * ~ 1e-13 for very small and very large values.
++ * This is ~2x faster than GNU libm's approach, which is still off by 2ulp on
++ * some inputs.
++ * @param x exponent
++ * @return 10^x
++ */
++static av_always_inline double ff_exp10(double x)
++{
++    return exp2(M_LOG2_10 * x);
++}
++
++static av_always_inline float ff_exp10f(float x)
++{
++    return exp2f(M_LOG2_10 * x);
++}
++
++/**
++ * Compute x^y for floating point x, y. Note: this function is faster than the
++ * libm variant due to mainly 2 reasons:
++ * 1. It does not handle any edge cases. In particular, this is only guaranteed
++ * to work correctly for x > 0.
++ * 2. It is not as accurate as a standard nearly "correctly rounded" libm variant.
++ * @param x base
++ * @param y exponent
++ * @return x^y
++ */
++static av_always_inline float ff_fast_powf(float x, float y)
++{
++    return expf(logf(x) * y);
++}
++
++
++/**
+  * A wrapper for open() setting O_CLOEXEC.
+  */
+ int avpriv_open(const char *filename, int flags, ...);
diff --git a/graphics/ffmpeg/patches/patch-libavutil_libm_h b/graphics/ffmpeg/patches/patch-libavutil_libm_h
new file mode 100644
index 00000000000..34a6740dc19
--- /dev/null
+++ b/graphics/ffmpeg/patches/patch-libavutil_libm_h
@@ -0,0 +1,14 @@
+$OpenBSD: patch-libavutil_libm_h,v 1.1 2016/04/07 06:01:19 ajacoutot Exp $
+
+lavu/libm: add exp10 support
+
+--- libavutil/libm.h.orig	Wed Apr  6 20:45:13 2016
++++ libavutil/libm.h	Wed Apr  6 20:45:27 2016
+@@ -28,6 +28,7 @@
+ #include "config.h"
+ #include "attributes.h"
+ #include "intfloat.h"
++#include "mathematics.h"
+ 
+ #if HAVE_MIPSFPU && HAVE_INLINE_ASM
+ #include "libavutil/mips/libm_mips.h"
diff --git a/graphics/ffmpeg/patches/patch-tests_fate_aac_mak b/graphics/ffmpeg/patches/patch-tests_fate_aac_mak
index e3d1819bb6b..2fb0f9b906a 100644
--- a/graphics/ffmpeg/patches/patch-tests_fate_aac_mak
+++ b/graphics/ffmpeg/patches/patch-tests_fate_aac_mak
@@ -1,4 +1,4 @@
-$OpenBSD: patch-tests_fate_aac_mak,v 1.9 2016/03/07 07:11:35 ajacoutot Exp $
+$OpenBSD: patch-tests_fate_aac_mak,v 1.10 2016/04/07 06:01:19 ajacoutot Exp $
 
 fate: adjust AAC encoder tests values
 
@@ -70,9 +70,11 @@ tests/fate/aac: remove unneeded strict arguments from the encoder tests
 
 fate/aac: Increase fuzz from of fate-aac-pns-encode from 72 to 74 for Loongson
 
+AAC encoder: new regression test
+
 --- tests/fate/aac.mak.orig	Wed Jan 13 15:27:50 2016
-+++ tests/fate/aac.mak	Sun Mar  6 19:41:20 2016
-@@ -142,56 +142,84 @@ FATE_AAC += $(FATE_AAC_CT:%=fate-aac-ct-%)
++++ tests/fate/aac.mak	Sun Apr  3 18:34:00 2016
+@@ -142,56 +142,95 @@ FATE_AAC += $(FATE_AAC_CT:%=fate-aac-ct-%)
  
  FATE_AAC_ENCODE += fate-aac-aref-encode
  fate-aac-aref-encode: ./tests/data/asynth-44100-2.wav
@@ -160,6 +162,17 @@ fate/aac: Increase fuzz from of fate-aac-pns-encode from 72 to 74 for Loongson
 +fate-aac-ltp-encode: CMP_TARGET = 1270
 +fate-aac-ltp-encode: SIZE_TOLERANCE = 3560
 +fate-aac-ltp-encode: FUZZ = 17
++
++#Ticket1784
++FATE_AAC_ENCODE += fate-aac-yoraw-encode
++fate-aac-yoraw-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/yo.raw-short.wav -c:a aac -fflags +bitexact -flags +bitexact
++fate-aac-yoraw-encode: CMP = stddev
++fate-aac-yoraw-encode: REF = $(SAMPLES)/audio-reference/yo.raw-short.wav
++fate-aac-yoraw-encode: CMP_SHIFT = -12288
++fate-aac-yoraw-encode: CMP_TARGET = 259
++fate-aac-yoraw-encode: SIZE_TOLERANCE = 3560
++fate-aac-yoraw-encode: FUZZ = 17
++
 +
  FATE_AAC_ENCODE += fate-aac-pred-encode
 -fate-aac-pred-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -profile:a aac_main -c:a aac -aac_is 0 -aac_pns 0 -b:a 128k