Merge remote-tracking branch 'qatar/master'

* qatar/master: (24 commits)
  flvdec: remove incomplete, disabled seeking code
  mem: add support for _aligned_malloc() as found on Windows
  lavc: Extend the documentation for avcodec_init_packet
  flvdec: remove incomplete, disabled seeking code
  http: replace atoll() with strtoll()
  mpegts: remove unused/incomplete/broken seeking code
  af_amix: allow float planar sample format as input
  af_amix: use AVFloatDSPContext.vector_fmac_scalar()
  float_dsp: add x86-optimized functions for vector_fmac_scalar()
  float_dsp: Move vector_fmac_scalar() from libavcodec to libavutil
  lavr: Add x86-optimized function for flt to s32 conversion
  lavr: Add x86-optimized function for flt to s16 conversion
  lavr: Add x86-optimized functions for s32 to flt conversion
  lavr: Add x86-optimized functions for s32 to s16 conversion
  lavr: Add x86-optimized functions for s16 to flt conversion
  lavr: Add x86-optimized function for s16 to s32 conversion
  rtpenc: Support packetizing iLBC
  rtpdec: Add a depacketizer for iLBC
  Implement the iLBC storage file format
  mov: Support muxing/demuxing iLBC
  ...

Conflicts:
	Changelog
	configure
	libavcodec/avcodec.h
	libavcodec/dsputil.c
	libavcodec/version.h
	libavformat/movenc.c
	libavformat/mpegts.c
	libavformat/version.h
	libavutil/mem.c

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

44 files changed, 952 insertions, 231 deletions

diff --git a/Changelog b/Changelog
index 2e92165f01..0f4168a02c 100644
--- a/Changelog
+++ b/Changelog
@@ -10,6 +10,7 @@ version next:
 - atempo filter
 - ffprobe -show_data option
 - RTMPT protocol support
+- iLBC encoding/decoding via libilbc
 
 
 version 0.11:
diff --git a/configure b/configure
index 7de8149edb..bf6a1f94a6 100755
--- a/configure
+++ b/configure
@@ -180,6 +180,7 @@ External library support:
   --enable-libfaac         enable FAAC support via libfaac [no]
   --enable-libfreetype     enable libfreetype [no]
   --enable-libgsm          enable GSM support via libgsm [no]
+  --enable-libilbc         enable iLBC de/encoding via libilbc [no]
   --enable-libmodplug      enable ModPlug via libmodplug [no]
   --enable-libmp3lame      enable MP3 encoding via libmp3lame [no]
   --enable-libnut          enable NUT (de)muxing via libnut,
@@ -1051,6 +1052,7 @@ CONFIG_LIST="
     libfaac
     libfreetype
     libgsm
+    libilbc
     libmodplug
     libmp3lame
     libnut
@@ -1168,6 +1170,7 @@ HAVE_LIST="
     $ARCH_EXT_LIST
     $HAVE_LIST_PUB
     $THREADS_LIST
+    aligned_malloc
     aligned_stack
     alsa_asoundlib_h
     altivec_h
@@ -1588,6 +1591,8 @@ libgsm_decoder_deps="libgsm"
 libgsm_encoder_deps="libgsm"
 libgsm_ms_decoder_deps="libgsm"
 libgsm_ms_encoder_deps="libgsm"
+libilbc_decoder_deps="libilbc"
+libilbc_encoder_deps="libilbc"
 libmodplug_demuxer_deps="libmodplug"
 libmp3lame_encoder_deps="libmp3lame"
 libopencore_amrnb_decoder_deps="libopencore_amrnb"
@@ -3144,6 +3149,7 @@ check_func  ${malloc_prefix}memalign            && enable memalign
 check_func  mkstemp
 check_func  mmap
 check_func  ${malloc_prefix}posix_memalign      && enable posix_memalign
+check_func_headers malloc.h _aligned_malloc     && enable aligned_malloc
 check_func  setrlimit
 check_func  strerror_r
 check_func  strptime
@@ -3254,6 +3260,7 @@ enabled libcelt    && require libcelt celt/celt.h celt_decode -lcelt0 &&
 enabled libfaac    && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac
 enabled libfreetype && require_pkg_config freetype2 "ft2build.h freetype/freetype.h" FT_Init_FreeType
 enabled libgsm     && require  libgsm gsm/gsm.h gsm_create -lgsm
+enabled libilbc    && require  libilbc ilbc.h WebRtcIlbcfix_InitDecode -lilbc
 enabled libmodplug && require  libmodplug libmodplug/modplug.h ModPlug_Load -lmodplug
 enabled libmp3lame && require  "libmp3lame >= 3.98.3" lame/lame.h lame_set_VBR_quality -lmp3lame
 enabled libnut     && require  libnut libnut.h nut_demuxer_init -lnut
@@ -3522,7 +3529,7 @@ if test $target_os = "haiku"; then
     disable posix_memalign
 fi
 
-! enabled_any memalign posix_memalign &&
+! enabled_any memalign posix_memalign aligned_malloc &&
     enabled_any $need_memalign && enable memalign_hack
 
 # add_dep lib dep
@@ -3628,6 +3635,7 @@ echo "libcelt enabled           ${libcelt-no}"
 echo "libdc1394 support         ${libdc1394-no}"
 echo "libfaac enabled           ${libfaac-no}"
 echo "libgsm enabled            ${libgsm-no}"
+echo "libilbc enabled           ${libilbc-no}"
 echo "libmodplug enabled        ${libmodplug-no}"
 echo "libmp3lame enabled        ${libmp3lame-no}"
 echo "libnut enabled            ${libnut-no}"
diff --git a/doc/general.texi b/doc/general.texi
index 12d7c68962..6981399843 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -93,6 +93,17 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
+@section libilbc
+
+iLBC is a narrowband speech codec that has been made freely available
+by Google as part of the WebRTC project. libilbc is a packaging friendly
+copy of the iLBC codec. Libav can make use of the libilbc library for
+iLBC encoding and decoding.
+
+Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
+installing the library. Then pass @code{--enable-libilbc} to configure to
+enable it.
+
 
 
 @chapter Supported File Formats, Codecs or Features
@@ -191,6 +202,7 @@ library:
 @item IEC61937 encapsulation @tab X @tab X
 @item IFF                       @tab   @tab X
     @tab Interchange File Format
+@item iLBC                      @tab X @tab X
 @item Interplay MVE             @tab   @tab X
     @tab Format used in various Interplay computer games.
 @item IV8                       @tab   @tab X
@@ -749,6 +761,8 @@ following image formats are supported:
 @item GSM Microsoft variant  @tab  E  @tab  X
     @tab encoding supported through external library libgsm
 @item IAC (Indeo Audio Coder)  @tab     @tab  X
+@item iLBC (Internet Low Bitrate Codec) @tab  E  @tab  E
+    @tab encoding and decoding supported through external library libilbc
 @item IMC (Intel Music Coder)  @tab     @tab  X
 @item MACE (Macintosh Audio Compression/Expansion) 3:1  @tab     @tab  X
 @item MACE (Macintosh Audio Compression/Expansion) 6:1  @tab     @tab  X
diff --git a/doc/protocols.texi b/doc/protocols.texi
index a157d344d5..a335fa7d21 100644
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -247,6 +247,10 @@ times to construct arbitrary AMF sequences.
 Version of the Flash plugin used to run the SWF player. The default
 is LNX 9,0,124,2.
 
+@item rtmp_flush_interval
+Number of packets flushed in the same request (RTMPT only). The default
+is 10.
+
 @item rtmp_live
 Specify that the media is a live stream. No resuming or seeking in
 live streams is possible. The default value is @code{any}, which means the
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 851fe1db74..1eeba3a8cd 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -659,6 +659,8 @@ OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_ENCODER)          += libgsm.o
+OBJS-$(CONFIG_LIBILBC_DECODER)            += libilbc.o
+OBJS-$(CONFIG_LIBILBC_ENCODER)            += libilbc.o
 OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o mpegaudiodecheader.o \
                                              audio_frame_queue.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o \
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 4067537403..e4af0d8d79 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -410,6 +410,7 @@ void avcodec_register_all(void)
     REGISTER_ENCODER (LIBFAAC, libfaac);
     REGISTER_ENCDEC  (LIBGSM, libgsm);
     REGISTER_ENCDEC  (LIBGSM_MS, libgsm_ms);
+    REGISTER_ENCDEC  (LIBILBC, libilbc);
     REGISTER_ENCODER (LIBMP3LAME, libmp3lame);
     REGISTER_ENCDEC  (LIBOPENCORE_AMRNB, libopencore_amrnb);
     REGISTER_DECODER (LIBOPENCORE_AMRWB, libopencore_amrwb);
diff --git a/libavcodec/arm/dsputil_init_neon.c b/libavcodec/arm/dsputil_init_neon.c
index 46b22a412c..ef5a8df85f 100644
--- a/libavcodec/arm/dsputil_init_neon.c
+++ b/libavcodec/arm/dsputil_init_neon.c
@@ -154,8 +154,6 @@ void ff_vector_fmul_window_neon(float *dst, const float *src0,
                                 const float *src1, const float *win, int len);
 void ff_vector_fmul_scalar_neon(float *dst, const float *src, float mul,
                                 int len);
-void ff_vector_fmac_scalar_neon(float *dst, const float *src, float mul,
-                                int len);
 void ff_butterflies_float_neon(float *v1, float *v2, int len);
 float ff_scalarproduct_float_neon(const float *v1, const float *v2, int len);
 void ff_vector_fmul_reverse_neon(float *dst, const float *src0,
@@ -329,7 +327,6 @@ void ff_dsputil_init_neon(DSPContext *c, AVCodecContext *avctx)
 
     c->vector_fmul_window         = ff_vector_fmul_window_neon;
     c->vector_fmul_scalar         = ff_vector_fmul_scalar_neon;
-    c->vector_fmac_scalar         = ff_vector_fmac_scalar_neon;
     c->butterflies_float          = ff_butterflies_float_neon;
     c->scalarproduct_float        = ff_scalarproduct_float_neon;
     c->vector_fmul_reverse        = ff_vector_fmul_reverse_neon;
diff --git a/libavcodec/arm/dsputil_neon.S b/libavcodec/arm/dsputil_neon.S
index 7ddd76a153..66b3f17d3d 100644
--- a/libavcodec/arm/dsputil_neon.S
+++ b/libavcodec/arm/dsputil_neon.S
@@ -682,54 +682,6 @@ NOVFP   vdup.32         q8,  r2
         .unreq          len
 endfunc
 
-function ff_vector_fmac_scalar_neon, export=1
-VFP     len .req r2
-VFP     acc .req r3
-NOVFP   len .req r3
-NOVFP   acc .req r2
-VFP     vdup.32         q15, d0[0]
-NOVFP   vdup.32         q15, r2
-        bics            r12, len, #15
-        mov             acc, r0
-        beq             3f
-        vld1.32         {q0},     [r1,:128]!
-        vld1.32         {q8},     [acc,:128]!
-        vld1.32         {q1},     [r1,:128]!
-        vld1.32         {q9},     [acc,:128]!
-1:      vmla.f32        q8,  q0,  q15
-        vld1.32         {q2},     [r1,:128]!
-        vld1.32         {q10},    [acc,:128]!
-        vmla.f32        q9,  q1,  q15
-        vld1.32         {q3},     [r1,:128]!
-        vld1.32         {q11},    [acc,:128]!
-        vmla.f32        q10, q2,  q15
-        vst1.32         {q8},     [r0,:128]!
-        vmla.f32        q11, q3,  q15
-        vst1.32         {q9},     [r0,:128]!
-        subs            r12, r12, #16
-        beq             2f
-        vld1.32         {q0},     [r1,:128]!
-        vld1.32         {q8},     [acc,:128]!
-        vst1.32         {q10},    [r0,:128]!
-        vld1.32         {q1},     [r1,:128]!
-        vld1.32         {q9},     [acc,:128]!
-        vst1.32         {q11},    [r0,:128]!
-        b               1b
-2:      vst1.32         {q10},    [r0,:128]!
-        vst1.32         {q11},    [r0,:128]!
-        ands            len, len, #15
-        it              eq
-        bxeq            lr
-3:      vld1.32         {q0},     [r1,:128]!
-        vld1.32         {q8},     [acc,:128]!
-        vmla.f32        q8,  q0,  q15
-        vst1.32         {q8},     [r0,:128]!
-        subs            len, len, #4
-        bgt             3b
-        bx              lr
-        .unreq          len
-endfunc
-
 function ff_butterflies_float_neon, export=1
 1:      vld1.32         {q0},[r0,:128]
         vld1.32         {q1},[r1,:128]
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 43931da831..b3c2991069 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -407,6 +407,7 @@ enum CodecID {
     CODEC_ID_BMV_AUDIO,
     CODEC_ID_RALF,
     CODEC_ID_IAC,
+    CODEC_ID_ILBC,
     CODEC_ID_FFWAVESYNTH = MKBETAG('F','F','W','S'),
     CODEC_ID_8SVX_RAW    = MKBETAG('8','S','V','X'),
     CODEC_ID_SONIC       = MKBETAG('S','O','N','C'),
@@ -3470,6 +3471,9 @@ void av_destruct_packet(AVPacket *pkt);
 /**
  * Initialize optional fields of a packet with default values.
  *
+ * Note, this does not touch the data and size members, which have to be
+ * initialized separately.
+ *
  * @param pkt packet
  */
 void av_init_packet(AVPacket *pkt);
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index f40440c1ca..76b258bf05 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -27,6 +27,7 @@
 #include <stdio.h>
 
 #include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
 #include "libavutil/intmath.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/mathematics.h"
@@ -384,7 +385,7 @@ typedef struct {
     int profile;
 
     int debug_flag;             ///< used for suppressing repeated error messages output
-    DSPContext dsp;
+    AVFloatDSPContext fdsp;
     FFTContext imdct;
     SynthFilterContext synth;
     DCADSPContext dcadsp;
@@ -2042,8 +2043,8 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
             float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel]     * 256;
             float *lt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
             float *rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
-            s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
-            s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
+            s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
+            s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
         }
 
         if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
@@ -2085,7 +2086,7 @@ static av_cold int dca_decode_init(AVCodecContext *avctx)
     s->avctx = avctx;
     dca_init_vlcs();
 
-    ff_dsputil_init(&s->dsp, avctx);
+    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
     ff_mdct_init(&s->imdct, 6, 1, 1.0);
     ff_synth_filter_init(&s->synth);
     ff_dcadsp_init(&s->dcadsp);
diff --git a/libavcodec/dirac.c b/libavcodec/dirac.c
index 3ff33f6554..57074bcc8c 100644
--- a/libavcodec/dirac.c
+++ b/libavcodec/dirac.c
@@ -127,7 +127,7 @@ static const enum PixelFormat dirac_pix_fmt[2][3] = {
 static int parse_source_parameters(AVCodecContext *avctx, GetBitContext *gb,
                                    dirac_source_params *source)
 {
-    AVRational frame_rate = (AVRational){0,0};
+    AVRational frame_rate = {0,0};
     unsigned luma_depth = 8, luma_offset = 16;
     int idx;
 
diff --git a/libavcodec/dsputil.c b/libavcodec/dsputil.c
index 81521ea376..442b9005f8 100644
--- a/libavcodec/dsputil.c
+++ b/libavcodec/dsputil.c
@@ -2509,14 +2509,6 @@ static void vector_fmul_scalar_c(float *dst, const float *src, float mul,
         dst[i] = src[i] * mul;
 }
 
-static void vector_fmac_scalar_c(float *dst, const float *src, float mul,
-                                 int len)
-{
-    int i;
-    for (i = 0; i < len; i++)
-        dst[i] += src[i] * mul;
-}
-
 static void butterflies_float_c(float *av_restrict v1, float *av_restrict v2,
                                 int len)
 {
@@ -3060,7 +3052,6 @@ av_cold void ff_dsputil_init(DSPContext* c, AVCodecContext *avctx)
     c->butterflies_float = butterflies_float_c;
     c->butterflies_float_interleave = butterflies_float_interleave_c;
     c->vector_fmul_scalar = vector_fmul_scalar_c;
-    c->vector_fmac_scalar = vector_fmac_scalar_c;
 
     c->shrink[0]= av_image_copy_plane;
     c->shrink[1]= ff_shrink22;
diff --git a/libavcodec/dsputil.h b/libavcodec/dsputil.h
index 67dd269fd4..e1aefe1eb6 100644
--- a/libavcodec/dsputil.h
+++ b/libavcodec/dsputil.h
@@ -421,17 +421,6 @@ typedef struct DSPContext {
     void (*vector_fmul_scalar)(float *dst, const float *src, float mul,
                                int len);
     /**
-     * Multiply a vector of floats by a scalar float and add to
-     * destination vector.  Source and destination vectors must
-     * overlap exactly or not at all.
-     * @param dst result vector, 16-byte aligned
-     * @param src input vector, 16-byte aligned
-     * @param mul scalar value
-     * @param len length of vector, multiple of 4
-     */
-    void (*vector_fmac_scalar)(float *dst, const float *src, float mul,
-                               int len);
-    /**
      * Calculate the scalar product of two vectors of floats.
      * @param v1  first vector, 16-byte aligned
      * @param v2  second vector, 16-byte aligned
diff --git a/libavcodec/libilbc.c b/libavcodec/libilbc.c
new file mode 100644
index 0000000000..0893c6c794
--- /dev/null
+++ b/libavcodec/libilbc.c
@@ -0,0 +1,209 @@
+/*
+ * iLBC decoder/encoder stub
+ * Copyright (c) 2012 Martin Storsjo
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <ilbc.h>
+
+#include "avcodec.h"
+#include "libavutil/opt.h"
+#include "internal.h"
+
+static int get_mode(AVCodecContext *avctx)
+{
+    if (avctx->block_align == 38)
+        return 20;
+    else if (avctx->block_align == 50)
+        return 30;
+    else if (avctx->bit_rate > 0)
+        return avctx->bit_rate <= 14000 ? 30 : 20;
+    else
+        return -1;
+}
+
+typedef struct ILBCDecContext {
+    const AVClass *class;
+    AVFrame frame;
+    iLBC_Dec_Inst_t decoder;
+    int enhance;
+} ILBCDecContext;
+
+static const AVOption ilbc_dec_options[] = {
+    { "enhance", "Enhance the decoded audio (adds delay)", offsetof(ILBCDecContext, enhance), AV_OPT_TYPE_INT, { 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM },
+    { NULL }
+};
+
+static const AVClass ilbc_dec_class = {
+    "libilbc", av_default_item_name, ilbc_dec_options, LIBAVUTIL_VERSION_INT
+};
+
+static av_cold int ilbc_decode_init(AVCodecContext *avctx)
+{
+    ILBCDecContext *s  = avctx->priv_data;
+    int mode;
+
+    if ((mode = get_mode(avctx)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "iLBC frame mode not indicated\n");
+        return AVERROR(EINVAL);
+    }
+
+    WebRtcIlbcfix_InitDecode(&s->decoder, mode, s->enhance);
+    avcodec_get_frame_defaults(&s->frame);
+    avctx->coded_frame = &s->frame;
+
+    avctx->channels = 1;
+    avctx->sample_rate = 8000;
+    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+    return 0;
+}
+
+static int ilbc_decode_frame(AVCodecContext *avctx, void *data,
+                             int *got_frame_ptr, AVPacket *avpkt)
+{
+    const uint8_t *buf = avpkt->data;
+    int buf_size       = avpkt->size;
+    ILBCDecContext *s  = avctx->priv_data;
+    int ret;
+
+    if (s->decoder.no_of_bytes > buf_size) {
+        av_log(avctx, AV_LOG_ERROR, "iLBC frame too short (%u, should be %u)\n",
+               buf_size, s->decoder.no_of_bytes);
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->frame.nb_samples = s->decoder.blockl;
+    if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
+    }
+
+    WebRtcIlbcfix_DecodeImpl((WebRtc_Word16*) s->frame.data[0],
+                             (const WebRtc_UWord16*) buf, &s->decoder, 1);
+
+    *got_frame_ptr   = 1;
+    *(AVFrame *)data = s->frame;
+
+    return s->decoder.no_of_bytes;
+}
+
+AVCodec ff_libilbc_decoder = {
+    .name           = "libilbc",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_ILBC,
+    .priv_data_size = sizeof(ILBCDecContext),
+    .init           = ilbc_decode_init,
+    .decode         = ilbc_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("Internet Low Bitrate Codec (iLBC)"),
+    .priv_class     = &ilbc_dec_class,
+};
+
+typedef struct ILBCEncContext {
+    const AVClass *class;
+    iLBC_Enc_Inst_t encoder;
+    int mode;
+} ILBCEncContext;
+
+static const AVOption ilbc_enc_options[] = {
+    { "mode", "iLBC mode (20 or 30 ms frames)", offsetof(ILBCEncContext, mode), AV_OPT_TYPE_INT, { 20 }, 20, 30, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
+    { NULL }
+};
+
+static const AVClass ilbc_enc_class = {
+    "libilbc", av_default_item_name, ilbc_enc_options, LIBAVUTIL_VERSION_INT
+};
+
+static av_cold int ilbc_encode_init(AVCodecContext *avctx)
+{
+    ILBCEncContext *s = avctx->priv_data;
+    int mode;
+
+    if (avctx->sample_rate != 8000) {
+        av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
+        return AVERROR(EINVAL);
+    }
+
+    if (avctx->channels != 1) {
+        av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
+        return AVERROR(EINVAL);
+    }
+
+    if ((mode = get_mode(avctx)) > 0)
+        s->mode = mode;
+    else
+        s->mode = s->mode != 30 ? 20 : 30;
+    WebRtcIlbcfix_InitEncode(&s->encoder, s->mode);
+
+    avctx->block_align = s->encoder.no_of_bytes;
+    avctx->frame_size  = s->encoder.blockl;
+#if FF_API_OLD_ENCODE_AUDIO
+    avctx->coded_frame = avcodec_alloc_frame();
+    if (!avctx->coded_frame)
+        return AVERROR(ENOMEM);
+#endif
+
+    return 0;
+}
+
+static av_cold int ilbc_encode_close(AVCodecContext *avctx)
+{
+#if FF_API_OLD_ENCODE_AUDIO
+    av_freep(&avctx->coded_frame);
+#endif
+    return 0;
+}
+
+static int ilbc_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                             const AVFrame *frame, int *got_packet_ptr)
+{
+    ILBCEncContext *s = avctx->priv_data;
+    int ret;
+
+    if ((ret = ff_alloc_packet(avpkt, 50))) {
+        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+        return ret;
+    }
+
+    WebRtcIlbcfix_EncodeImpl((WebRtc_UWord16*) avpkt->data, (const WebRtc_Word16*) frame->data[0], &s->encoder);
+
+    avpkt->size     = s->encoder.no_of_bytes;
+    *got_packet_ptr = 1;
+    return 0;
+}
+
+static const AVCodecDefault ilbc_encode_defaults[] = {
+    { "b", "0" },
+    { NULL }
+};
+
+AVCodec ff_libilbc_encoder = {
+    .name           = "libilbc",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_ILBC,
+    .priv_data_size = sizeof(ILBCEncContext),
+    .init           = ilbc_encode_init,
+    .encode2        = ilbc_encode_frame,
+    .close          = ilbc_encode_close,
+    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+                                                     AV_SAMPLE_FMT_NONE },
+    .long_name      = NULL_IF_CONFIG_SMALL("Internet Low Bitrate Codec (iLBC)"),
+    .defaults       = ilbc_encode_defaults,
+    .priv_class     = &ilbc_enc_class,
+};
diff --git a/libavcodec/ratecontrol.c b/libavcodec/ratecontrol.c
index 9223dce2da..c4624f9e3c 100644
--- a/libavcodec/ratecontrol.c
+++ b/libavcodec/ratecontrol.c
@@ -514,14 +514,6 @@ static double predict_size(Predictor *p, double q, double var)
      return p->coeff*var / (q*p->count);
 }
 
-/*
-static double predict_qp(Predictor *p, double size, double var)
-{
-//printf("coeff:%f, count:%f, var:%f, size:%f//\n", p->coeff, p->count, var, size);
-     return p->coeff*var / (size*p->count);
-}
-*/
-
 static void update_predictor(Predictor *p, double q, double var, double size)
 {
     double new_coeff= size*q / (var + 1);
@@ -561,10 +553,6 @@ static void adaptive_quantization(MpegEncContext *s, double q){
         int mb_y = mb_xy / s->mb_stride;
         int mb_distance;
         float mb_factor = 0.0;
-#if 0
-        if(spat_cplx < q/3) spat_cplx= q/3; //FIXME finetune
-        if(temp_cplx < q/3) temp_cplx= q/3; //FIXME finetune
-#endif
         if(spat_cplx < 4) spat_cplx= 4; //FIXME finetune
         if(temp_cplx < 4) temp_cplx= 4; //FIXME finetune
 
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 315dc57da4..e3a4e93883 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -2138,6 +2138,11 @@ int av_get_audio_frame_duration(AVCodecContext *avctx, int frame_bytes)
             case 29: return 288;
             case 37: return 480;
             }
+        } else if (id == CODEC_ID_ILBC) {
+            switch (ba) {
+            case 38: return 160;
+            case 50: return 240;
+            }
         }
     }
 
diff --git a/libavcodec/version.h b/libavcodec/version.h
index b3644ba24c..2c3a941067 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -27,7 +27,7 @@
  */
 
 #define LIBAVCODEC_VERSION_MAJOR 54
-#define LIBAVCODEC_VERSION_MINOR  25
+#define LIBAVCODEC_VERSION_MINOR  26
 #define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c
index 81586981be..dcfa5e4f94 100644
--- a/libavfilter/af_amix.c
+++ b/libavfilter/af_amix.c
@@ -32,6 +32,7 @@
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
+#include "libavutil/float_dsp.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
@@ -152,6 +153,7 @@ static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t p
 
 typedef struct MixContext {
     const AVClass *class;       /**< class for AVOptions */
+    AVFloatDSPContext fdsp;
 
     int nb_inputs;              /**< number of inputs */
     int active_inputs;          /**< number of input currently active */
@@ -160,6 +162,7 @@ typedef struct MixContext {
 
     int nb_channels;            /**< number of channels */
     int sample_rate;            /**< sample rate */
+    int planar;
     AVAudioFifo **fifos;        /**< audio fifo for each input */
     uint8_t *input_state;       /**< current state of each input */
     float *input_scale;         /**< mixing scale factor for each input */
@@ -224,6 +227,7 @@ static int config_output(AVFilterLink *outlink)
     int i;
     char buf[64];
 
+    s->planar          = av_sample_fmt_is_planar(outlink->format);
     s->sample_rate     = outlink->sample_rate;
     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
     s->next_pts        = AV_NOPTS_VALUE;
@@ -264,14 +268,6 @@ static int config_output(AVFilterLink *outlink)
     return 0;
 }
 
-/* TODO: move optimized version from DSPContext to libavutil */
-static void vector_fmac_scalar(float *dst, const float *src, float mul, int len)
-{
-    int i;
-    for (i = 0; i < len; i++)
-        dst[i] += src[i] * mul;
-}
-
 /**
  * Read samples from the input FIFOs, mix, and write to the output link.
  */
@@ -294,11 +290,20 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
 
     for (i = 0; i < s->nb_inputs; i++) {
         if (s->input_state[i] == INPUT_ON) {
+            int planes, plane_size, p;
+
             av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
                                nb_samples);
-            vector_fmac_scalar((float *)out_buf->extended_data[0],
-                               (float *) in_buf->extended_data[0],
-                               s->input_scale[i], nb_samples * s->nb_channels);
+
+            planes     = s->planar ? s->nb_channels : 1;
+            plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
+            plane_size = FFALIGN(plane_size, 16);
+
+            for (p = 0; p < planes; p++) {
+                s->fdsp.vector_fmac_scalar((float *)out_buf->extended_data[p],
+                                           (float *) in_buf->extended_data[p],
+                                           s->input_scale[i], plane_size);
+            }
         }
     }
     avfilter_unref_buffer(in_buf);
@@ -501,6 +506,8 @@ static int init(AVFilterContext *ctx, const char *args, void *opaque)
         ff_insert_inpad(ctx, i, &pad);
     }
 
+    avpriv_float_dsp_init(&s->fdsp, 0);
+
     return 0;
 }
 
@@ -527,6 +534,7 @@ static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats = NULL;
     ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
+    ff_add_format(&formats, AV_SAMPLE_FMT_FLTP);
     ff_set_common_formats(ctx, formats);
     ff_set_common_channel_layouts(ctx, ff_all_channel_layouts());
     ff_set_common_samplerates(ctx, ff_all_samplerates());
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 3dca060994..f38b8a575c 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -123,6 +123,8 @@ OBJS-$(CONFIG_ICO_DEMUXER)               += icodec.o
 OBJS-$(CONFIG_IDCIN_DEMUXER)             += idcin.o
 OBJS-$(CONFIG_IDF_DEMUXER)               += bintext.o
 OBJS-$(CONFIG_IFF_DEMUXER)               += iff.o
+OBJS-$(CONFIG_ILBC_DEMUXER)              += ilbc.o
+OBJS-$(CONFIG_ILBC_MUXER)                += ilbc.o
 OBJS-$(CONFIG_IMAGE2_DEMUXER)            += img2dec.o img2.o
 OBJS-$(CONFIG_IMAGE2_MUXER)              += img2enc.o img2.o
 OBJS-$(CONFIG_IMAGE2PIPE_DEMUXER)        += img2dec.o img2.o
@@ -281,6 +283,7 @@ OBJS-$(CONFIG_RTPDEC)                    += rdt.o         \
                                             rtpdec_h263.o \
                                             rtpdec_h263_rfc2190.o \
                                             rtpdec_h264.o \
+                                            rtpdec_ilbc.o \
                                             rtpdec_latm.o \
                                             rtpdec_mpeg4.o \
                                             rtpdec_qcelp.o \
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index ed9227e5b6..b5738e4d97 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -116,6 +116,7 @@ void av_register_all(void)
     REGISTER_DEMUXER  (IDCIN, idcin);
     REGISTER_DEMUXER  (IDF, idf);
     REGISTER_DEMUXER  (IFF, iff);
+    REGISTER_MUXDEMUX (ILBC, ilbc);
     REGISTER_MUXDEMUX (IMAGE2, image2);
     REGISTER_MUXDEMUX (IMAGE2PIPE, image2pipe);
     REGISTER_DEMUXER  (INGENIENT, ingenient);
diff --git a/libavformat/flvdec.c b/libavformat/flvdec.c
index aa016ee4de..31f306872f 100644
--- a/libavformat/flvdec.c
+++ b/libavformat/flvdec.c
@@ -759,33 +759,6 @@ static int flv_read_seek(AVFormatContext *s, int stream_index,
     return avio_seek_time(s->pb, stream_index, ts, flags);
 }
 
-#if 0 /* don't know enough to implement this */
-static int flv_read_seek2(AVFormatContext *s, int stream_index,
-    int64_t min_ts, int64_t ts, int64_t max_ts, int flags)
-{
-    int ret = AVERROR(ENOSYS);
-
-    if (ts - min_ts > (uint64_t)(max_ts - ts)) flags |= AVSEEK_FLAG_BACKWARD;
-
-    if (!s->pb->seekable) {
-        if (stream_index < 0) {
-            stream_index = av_find_default_stream_index(s);
-            if (stream_index < 0)
-                return -1;
-
-            /* timestamp for default must be expressed in AV_TIME_BASE units */
-            ts = av_rescale_rnd(ts, 1000, AV_TIME_BASE,
-                flags & AVSEEK_FLAG_BACKWARD ? AV_ROUND_DOWN : AV_ROUND_UP);
-        }
-        ret = avio_seek_time(s->pb, stream_index, ts, flags);
-    }
-
-    if (ret == AVERROR(ENOSYS))
-        ret = av_seek_frame(s, stream_index, ts, flags);
-    return ret;
-}
-#endif
-
 AVInputFormat ff_flv_demuxer = {
     .name           = "flv",
     .long_name      = NULL_IF_CONFIG_SMALL("FLV format"),
@@ -794,9 +767,6 @@ AVInputFormat ff_flv_demuxer = {
     .read_header    = flv_read_header,
     .read_packet    = flv_read_packet,
     .read_seek      = flv_read_seek,
-#if 0
-    .read_seek2     = flv_read_seek2,
-#endif
     .read_close     = flv_read_close,
     .extensions     = "flv",
 };
diff --git a/libavformat/http.c b/libavformat/http.c
index fb4a83ad0e..96a7eeda0a 100644
--- a/libavformat/http.c
+++ b/libavformat/http.c
@@ -308,15 +308,15 @@ static int process_line(URLContext *h, char *line, int line_count,
             strcpy(s->location, p);
             *new_location = 1;
         } else if (!av_strcasecmp (tag, "Content-Length") && s->filesize == -1) {
-            s->filesize = atoll(p);
+            s->filesize = strtoll(p, NULL, 10);
         } else if (!av_strcasecmp (tag, "Content-Range")) {
             /* "bytes $from-$to/$document_size" */
             const char *slash;
             if (!strncmp (p, "bytes ", 6)) {
                 p += 6;
-                s->off = atoll(p);
+                s->off = strtoll(p, NULL, 10);
                 if ((slash = strchr(p, '/')) && strlen(slash) > 0)
-                    s->filesize = atoll(slash+1);
+                    s->filesize = strtoll(slash+1, NULL, 10);
             }
             h->is_streamed = 0; /* we _can_ in fact seek */
         } else if (!av_strcasecmp(tag, "Accept-Ranges") && !strncmp(p, "bytes", 5)) {
diff --git a/libavformat/ilbc.c b/libavformat/ilbc.c
new file mode 100644
index 0000000000..d4f4583545
--- /dev/null
+++ b/libavformat/ilbc.c
@@ -0,0 +1,141 @@
+/*
+ * iLBC storage file format
+ * Copyright (c) 2012 Martin Storsjo
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avformat.h"
+#include "internal.h"
+
+static const char mode20_header[] = "#!iLBC20\n";
+static const char mode30_header[] = "#!iLBC30\n";
+
+static int ilbc_write_header(AVFormatContext *s)
+{
+    AVIOContext *pb = s->pb;
+    AVCodecContext *enc;
+
+    if (s->nb_streams != 1) {
+        av_log(s, AV_LOG_ERROR, "Unsupported number of streams\n");
+        return AVERROR(EINVAL);
+    }
+    enc = s->streams[0]->codec;
+
+    if (enc->codec_id != CODEC_ID_ILBC) {
+        av_log(s, AV_LOG_ERROR, "Unsupported codec\n");
+        return AVERROR(EINVAL);
+    }
+
+    if (enc->block_align == 50) {
+        avio_write(pb, mode30_header, sizeof(mode30_header) - 1);
+    } else if (enc->block_align == 38) {
+        avio_write(pb, mode20_header, sizeof(mode20_header) - 1);
+    } else {
+        av_log(s, AV_LOG_ERROR, "Unsupported mode\n");
+        return AVERROR(EINVAL);
+    }
+    avio_flush(pb);
+    return 0;
+}
+
+static int ilbc_write_packet(AVFormatContext *s, AVPacket *pkt)
+{
+    avio_write(s->pb, pkt->data, pkt->size);
+    avio_flush(s->pb);
+    return 0;
+}
+
+static int ilbc_probe(AVProbeData *p)
+{
+    // Only check for "#!iLBC" which matches both formats
+    if (!memcmp(p->buf, mode20_header, 6))
+        return AVPROBE_SCORE_MAX;
+    else
+        return 0;
+}
+
+static int ilbc_read_header(AVFormatContext *s)
+{
+    AVIOContext *pb = s->pb;
+    AVStream *st;
+    uint8_t header[9];
+
+    avio_read(pb, header, 9);
+
+    st = avformat_new_stream(s, NULL);
+    if (!st)
+        return AVERROR(ENOMEM);
+    st->codec->codec_id = CODEC_ID_ILBC;
+    st->codec->sample_rate = 8000;
+    st->codec->channels = 1;
+    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+    st->start_time = 0;
+    avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+    if (!memcmp(header, mode20_header, sizeof(mode20_header) - 1)) {
+        st->codec->block_align = 38;
+        st->codec->bit_rate = 15200;
+    } else if (!memcmp(header, mode30_header, sizeof(mode30_header) - 1)) {
+        st->codec->block_align = 50;
+        st->codec->bit_rate = 13333;
+    } else {
+        av_log(s, AV_LOG_ERROR, "Unrecognized iLBC file header\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    return 0;
+}
+
+static int ilbc_read_packet(AVFormatContext *s,
+                          AVPacket *pkt)
+{
+    AVCodecContext *enc = s->streams[0]->codec;
+    int ret;
+
+    if ((ret = av_new_packet(pkt, enc->block_align)) < 0)
+        return ret;
+
+    pkt->stream_index = 0;
+    pkt->pos = avio_tell(s->pb);
+    pkt->duration = enc->block_align == 38 ? 160 : 240;
+    if ((ret = avio_read(s->pb, pkt->data, enc->block_align)) != enc->block_align) {
+        av_free_packet(pkt);
+        return ret < 0 ? ret : AVERROR(EIO);
+    }
+
+    return 0;
+}
+
+AVInputFormat ff_ilbc_demuxer = {
+    .name         = "ilbc",
+    .long_name    = NULL_IF_CONFIG_SMALL("iLBC storage file format"),
+    .read_probe   = ilbc_probe,
+    .read_header  = ilbc_read_header,
+    .read_packet  = ilbc_read_packet,
+    .flags        = AVFMT_GENERIC_INDEX,
+};
+
+AVOutputFormat ff_ilbc_muxer = {
+    .name         = "ilbc",
+    .long_name    = NULL_IF_CONFIG_SMALL("iLBC storage file format"),
+    .mime_type    = "audio/iLBC",
+    .extensions   = "lbc",
+    .audio_codec  = CODEC_ID_ILBC,
+    .write_header = ilbc_write_header,
+    .write_packet = ilbc_write_packet,
+    .flags        = AVFMT_NOTIMESTAMPS,
+};
diff --git a/libavformat/isom.c b/libavformat/isom.c
index e4575033e2..0ddbb813ec 100644
--- a/libavformat/isom.c
+++ b/libavformat/isom.c
@@ -259,6 +259,7 @@ const AVCodecTag ff_codec_movaudio_tags[] = {
     { CODEC_ID_DVAUDIO,         MKTAG('v', 'd', 'v', 'a') },
     { CODEC_ID_DVAUDIO,         MKTAG('d', 'v', 'c', 'a') },
     { CODEC_ID_GSM,             MKTAG('a', 'g', 's', 'm') },
+    { CODEC_ID_ILBC,            MKTAG('i', 'l', 'b', 'c') },
     { CODEC_ID_MACE3,           MKTAG('M', 'A', 'C', '3') },
     { CODEC_ID_MACE6,           MKTAG('M', 'A', 'C', '6') },
     { CODEC_ID_MP1,             MKTAG('.', 'm', 'p', '1') },
diff --git a/libavformat/mov.c b/libavformat/mov.c
index 9548f215f3..3aab48ffff 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -1540,6 +1540,7 @@ int ff_mov_read_stsd_entries(MOVContext *c, AVIOContext *pb, int entries)
     case CODEC_ID_GSM:
     case CODEC_ID_ADPCM_MS:
     case CODEC_ID_ADPCM_IMA_WAV:
+    case CODEC_ID_ILBC:
         st->codec->block_align = sc->bytes_per_frame;
         break;
     case CODEC_ID_ALAC:
diff --git a/libavformat/movenc.c b/libavformat/movenc.c
index d5eb25bcc2..89072d5ded 100644
--- a/libavformat/movenc.c
+++ b/libavformat/movenc.c
@@ -3371,7 +3371,8 @@ static int mov_write_header(AVFormatContext *s)
                 av_log(s, AV_LOG_WARNING, "track %d: codec frame size is not set\n", i);
                 track->audio_vbr = 1;
             }else if(st->codec->codec_id == CODEC_ID_ADPCM_MS ||
-                     st->codec->codec_id == CODEC_ID_ADPCM_IMA_WAV){
+                     st->codec->codec_id == CODEC_ID_ADPCM_IMA_WAV ||
+                     st->codec->codec_id == CODEC_ID_ILBC){
                 if (!st->codec->block_align) {
                     av_log(s, AV_LOG_ERROR, "track %d: codec block align is not set for adpcm\n", i);
                     goto error;
diff --git a/libavformat/mpegts.c b/libavformat/mpegts.c
index 3acca0230c..3a5ed696c8 100644
--- a/libavformat/mpegts.c
+++ b/libavformat/mpegts.c
@@ -19,8 +19,6 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-//#define USE_SYNCPOINT_SEARCH
-
 #include "libavutil/crc.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/log.h"
@@ -2161,92 +2159,6 @@ static int64_t mpegts_get_dts(AVFormatContext *s, int stream_index,
     return AV_NOPTS_VALUE;
 }
 
-#ifdef USE_SYNCPOINT_SEARCH
-
-static int read_seek2(AVFormatContext *s,
-                      int stream_index,
-                      int64_t min_ts,
-                      int64_t target_ts,
-                      int64_t max_ts,
-                      int flags)
-{
-    int64_t pos;
-
-    int64_t ts_ret, ts_adj;
-    int stream_index_gen_search;
-    AVStream *st;
-    AVParserState *backup;
-
-    backup = ff_store_parser_state(s);
-
-    // detect direction of seeking for search purposes
-    flags |= (target_ts - min_ts > (uint64_t)(max_ts - target_ts)) ?
-             AVSEEK_FLAG_BACKWARD : 0;
-
-    if (flags & AVSEEK_FLAG_BYTE) {
-        // use position directly, we will search starting from it
-        pos = target_ts;
-    } else {
-        // search for some position with good timestamp match
-        if (stream_index < 0) {
-            stream_index_gen_search = av_find_default_stream_index(s);
-            if (stream_index_gen_search < 0) {
-                ff_restore_parser_state(s, backup);
-                return -1;
-            }
-
-            st = s->streams[stream_index_gen_search];
-            // timestamp for default must be expressed in AV_TIME_BASE units
-            ts_adj = av_rescale(target_ts,
-                                st->time_base.den,
-                                AV_TIME_BASE * (int64_t)st->time_base.num);
-        } else {
-            ts_adj = target_ts;
-            stream_index_gen_search = stream_index;
-        }
-        pos = ff_gen_search(s, stream_index_gen_search, ts_adj,
-                            0, INT64_MAX, -1,
-                            AV_NOPTS_VALUE,
-                            AV_NOPTS_VALUE,
-                            flags, &ts_ret, mpegts_get_pcr);
-        if (pos < 0) {
-            ff_restore_parser_state(s, backup);
-            return -1;
-        }
-    }
-
-    // search for actual matching keyframe/starting position for all streams
-    if (ff_gen_syncpoint_search(s, stream_index, pos,
-                                min_ts, target_ts, max_ts,
-                                flags) < 0) {
-        ff_restore_parser_state(s, backup);
-        return -1;
-    }
-
-    ff_free_parser_state(s, backup);
-    return 0;
-}
-
-static int read_seek(AVFormatContext *s, int stream_index, int64_t target_ts, int flags)
-{
-    int ret;
-    if (flags & AVSEEK_FLAG_BACKWARD) {
-        flags &= ~AVSEEK_FLAG_BACKWARD;
-        ret = read_seek2(s, stream_index, INT64_MIN, target_ts, target_ts, flags);
-        if (ret < 0)
-            // for compatibility reasons, seek to the best-fitting timestamp
-            ret = read_seek2(s, stream_index, INT64_MIN, target_ts, INT64_MAX, flags);
-    } else {
-        ret = read_seek2(s, stream_index, target_ts, target_ts, INT64_MAX, flags);
-        if (ret < 0)
-            // for compatibility reasons, seek to the best-fitting timestamp
-            ret = read_seek2(s, stream_index, INT64_MIN, target_ts, INT64_MAX, flags);
-    }
-    return ret;
-}
-
-#endif
-
 /**************************************************************/
 /* parsing functions - called from other demuxers such as RTP */
 
@@ -2313,9 +2225,6 @@ AVInputFormat ff_mpegts_demuxer = {
     .read_close     = mpegts_read_close,
     .read_timestamp = mpegts_get_dts,
     .flags          = AVFMT_SHOW_IDS | AVFMT_TS_DISCONT,
-#ifdef USE_SYNCPOINT_SEARCH
-    .read_seek2     = read_seek2,
-#endif
 };
 
 AVInputFormat ff_mpegtsraw_demuxer = {
@@ -2327,8 +2236,5 @@ AVInputFormat ff_mpegtsraw_demuxer = {
     .read_close     = mpegts_read_close,
     .read_timestamp = mpegts_get_dts,
     .flags          = AVFMT_SHOW_IDS | AVFMT_TS_DISCONT,
-#ifdef USE_SYNCPOINT_SEARCH
-    .read_seek2     = read_seek2,
-#endif
     .priv_class     = &mpegtsraw_class,
 };
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index 07ec62cc43..4bdda4d097 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -76,6 +76,7 @@ typedef struct RTMPContext {
     uint8_t*      flv_data;                   ///< buffer with data for demuxer
     int           flv_size;                   ///< current buffer size
     int           flv_off;                    ///< number of bytes read from current buffer
+    int           flv_nb_packets;             ///< number of flv packets published
     RTMPPacket    out_pkt;                    ///< rtmp packet, created from flv a/v or metadata (for output)
     uint32_t      client_report_size;         ///< number of bytes after which client should report to server
     uint32_t      bytes_read;                 ///< number of bytes read from server
@@ -90,6 +91,7 @@ typedef struct RTMPContext {
     char*         swfurl;                     ///< url of the swf player
     int           server_bw;                  ///< server bandwidth
     int           client_buffer_time;         ///< client buffer time in ms
+    int           flush_interval;             ///< number of packets flushed in the same request (RTMPT only)
 } RTMPContext;
 
 #define PLAYER_KEY_OPEN_PART_LEN 30   ///< length of partial key used for first client digest signing
@@ -1361,9 +1363,14 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
             rt->flv_size = 0;
             rt->flv_off = 0;
             rt->flv_header_bytes = 0;
+            rt->flv_nb_packets++;
         }
     } while (buf_temp - buf < size);
 
+    if (rt->flv_nb_packets < rt->flush_interval)
+        return size;
+    rt->flv_nb_packets = 0;
+
     /* set stream into nonblocking mode */
     rt->stream->flags |= AVIO_FLAG_NONBLOCK;
 
@@ -1404,6 +1411,7 @@ static const AVOption rtmp_options[] = {
     {"rtmp_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_INT, {3000}, 0, INT_MAX, DEC|ENC},
     {"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
     {"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+    {"rtmp_flush_interval", "Number of packets flushed in the same request (RTMPT only).", OFFSET(flush_interval), AV_OPT_TYPE_INT, {10}, 0, INT_MAX, ENC},
     {"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
     {"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
     {"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index b7240a2d51..3a272e33c7 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -68,6 +68,7 @@ void av_register_rtp_dynamic_payload_handlers(void)
     ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
+    ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
diff --git a/libavformat/rtpdec_formats.h b/libavformat/rtpdec_formats.h
index 73ffe1f49e..a0c61a84b9 100644
--- a/libavformat/rtpdec_formats.h
+++ b/libavformat/rtpdec_formats.h
@@ -45,6 +45,7 @@ extern RTPDynamicProtocolHandler ff_h263_1998_dynamic_handler;
 extern RTPDynamicProtocolHandler ff_h263_2000_dynamic_handler;
 extern RTPDynamicProtocolHandler ff_h263_rfc2190_dynamic_handler;
 extern RTPDynamicProtocolHandler ff_h264_dynamic_handler;
+extern RTPDynamicProtocolHandler ff_ilbc_dynamic_handler;
 extern RTPDynamicProtocolHandler ff_mp4a_latm_dynamic_handler;
 extern RTPDynamicProtocolHandler ff_mp4v_es_dynamic_handler;
 extern RTPDynamicProtocolHandler ff_mpeg4_generic_dynamic_handler;
diff --git a/libavformat/rtpdec_ilbc.c b/libavformat/rtpdec_ilbc.c
new file mode 100644
index 0000000000..247c779bfc
--- /dev/null
+++ b/libavformat/rtpdec_ilbc.c
@@ -0,0 +1,73 @@
+/*
+ * RTP iLBC Depacketizer, RFC 3952
+ * Copyright (c) 2012 Martin Storsjo
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avformat.h"
+#include "rtpdec_formats.h"
+#include "libavutil/avstring.h"
+
+static int ilbc_parse_fmtp(AVStream *stream, PayloadContext *data,
+                           char *attr, char *value)
+{
+    if (!strcmp(attr, "mode")) {
+        int mode = atoi(value);
+        switch (mode) {
+        case 20:
+            stream->codec->block_align = 38;
+            break;
+        case 30:
+            stream->codec->block_align = 50;
+            break;
+        default:
+            av_log(NULL, AV_LOG_ERROR, "Unsupported iLBC mode %d\n", mode);
+            return AVERROR(EINVAL);
+        }
+    }
+    return 0;
+}
+
+static int ilbc_parse_sdp_line(AVFormatContext *s, int st_index,
+                               PayloadContext *data, const char *line)
+{
+    const char *p;
+    AVStream *st;
+
+    if (st_index < 0)
+        return 0;
+    st = s->streams[st_index];
+
+    if (av_strstart(line, "fmtp:", &p)) {
+        int ret = ff_parse_fmtp(st, data, p, ilbc_parse_fmtp);
+        if (ret < 0)
+            return ret;
+        if (!st->codec->block_align) {
+            av_log(s, AV_LOG_ERROR, "No iLBC mode set\n");
+            return AVERROR(EINVAL);
+        }
+    }
+    return 0;
+}
+
+RTPDynamicProtocolHandler ff_ilbc_dynamic_handler = {
+    .enc_name         = "iLBC",
+    .codec_type       = AVMEDIA_TYPE_AUDIO,
+    .codec_id         = CODEC_ID_ILBC,
+    .parse_sdp_a_line = ilbc_parse_sdp_line,
+};
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index e16e610820..f7e2cf0630 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -74,6 +74,7 @@ static int is_supported(enum CodecID id)
     case CODEC_ID_VP8:
     case CODEC_ID_ADPCM_G722:
     case CODEC_ID_ADPCM_G726:
+    case CODEC_ID_ILBC:
         return 1;
     default:
         return 0;
@@ -187,6 +188,16 @@ static int rtp_write_header(AVFormatContext *s1)
          * 8000, even if the sample rate is 16000. See RFC 3551. */
         avpriv_set_pts_info(st, 32, 1, 8000);
         break;
+    case CODEC_ID_ILBC:
+        if (st->codec->block_align != 38 && st->codec->block_align != 50) {
+            av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
+            goto fail;
+        }
+        if (!s->max_frames_per_packet)
+            s->max_frames_per_packet = 1;
+        s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
+                                         s->max_payload_size / st->codec->block_align);
+        goto defaultcase;
     case CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:
         if (!s->max_frames_per_packet)
@@ -395,6 +406,36 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1,
     }
 }
 
+static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
+{
+    RTPMuxContext *s = s1->priv_data;
+    AVStream *st = s1->streams[0];
+    int frame_duration = av_get_audio_frame_duration(st->codec, 0);
+    int frame_size = st->codec->block_align;
+    int frames = size / frame_size;
+
+    while (frames > 0) {
+        int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
+
+        if (!s->num_frames) {
+            s->buf_ptr = s->buf;
+            s->timestamp = s->cur_timestamp;
+        }
+        memcpy(s->buf_ptr, buf, n * frame_size);
+        frames           -= n;
+        s->num_frames    += n;
+        s->buf_ptr       += n * frame_size;
+        buf              += n * frame_size;
+        s->cur_timestamp += n * frame_duration;
+
+        if (s->num_frames == s->max_frames_per_packet) {
+            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
+            s->num_frames = 0;
+        }
+    }
+    return 0;
+}
+
 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     RTPMuxContext *s = s1->priv_data;
@@ -483,6 +524,9 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
     case CODEC_ID_VP8:
         ff_rtp_send_vp8(s1, pkt->data, size);
         break;
+    case CODEC_ID_ILBC:
+        rtp_send_ilbc(s1, pkt->data, size);
+        break;
     default:
         /* better than nothing : send the codec raw data */
         rtp_send_raw(s1, pkt->data, size);
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 891afc92a4..68ca247b00 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -1279,7 +1279,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
                      "%s/UDP;multicast", trans_pref);
         }
         if (s->oformat) {
-            av_strlcat(transport, ";mode=receive", sizeof(transport));
+            av_strlcat(transport, ";mode=record", sizeof(transport));
         } else if (rt->server_type == RTSP_SERVER_REAL ||
                    rt->server_type == RTSP_SERVER_WMS)
             av_strlcat(transport, ";mode=play", sizeof(transport));
diff --git a/libavformat/sdp.c b/libavformat/sdp.c
index 5d5e9515f7..352dea0335 100644
--- a/libavformat/sdp.c
+++ b/libavformat/sdp.c
@@ -549,6 +549,12 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
                                          c->sample_rate);
             break;
         }
+        case CODEC_ID_ILBC:
+            av_strlcatf(buff, size, "a=rtpmap:%d iLBC/%d\r\n"
+                                    "a=fmtp:%d mode=%d\r\n",
+                                     payload_type, c->sample_rate,
+                                     payload_type, c->block_align == 38 ? 20 : 30);
+            break;
         default:
             /* Nothing special to do here... */
             break;
diff --git a/libavformat/version.h b/libavformat/version.h
index 0b1bff2764..3b61ea34fc 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFORMAT_VERSION_MAJOR 54
-#define LIBAVFORMAT_VERSION_MINOR  8
+#define LIBAVFORMAT_VERSION_MINOR  9
 #define LIBAVFORMAT_VERSION_MICRO 100
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
diff --git a/libavresample/x86/audio_convert.asm b/libavresample/x86/audio_convert.asm
index ba59f3314f..7b3cc223c7 100644
--- a/libavresample/x86/audio_convert.asm
+++ b/libavresample/x86/audio_convert.asm
@@ -1,6 +1,7 @@
 ;******************************************************************************
 ;* x86 optimized Format Conversion Utils
 ;* Copyright (c) 2008 Loren Merritt
+;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
 ;*
 ;* This file is part of Libav.
 ;*
@@ -21,9 +22,217 @@
 
 %include "x86inc.asm"
 %include "x86util.asm"
+%include "util.asm"
+
+SECTION_RODATA 32
+
+pf_s32_inv_scale: times 8 dd 0x30000000
+pf_s32_scale:     times 8 dd 0x4f000000
+pf_s16_inv_scale: times 4 dd 0x38000000
+pf_s16_scale:     times 4 dd 0x47000000
 
 SECTION_TEXT
 
+;------------------------------------------------------------------------------
+; void ff_conv_s16_to_s32(int32_t *dst, const int16_t *src, int len);
+;------------------------------------------------------------------------------
+
+INIT_XMM sse2
+cglobal conv_s16_to_s32, 3,3,3, dst, src, len
+    lea      lenq, [2*lend]
+    lea      dstq, [dstq+2*lenq]
+    add      srcq, lenq
+    neg      lenq
+.loop:
+    mova       m2, [srcq+lenq]
+    pxor       m0, m0
+    pxor       m1, m1
+    punpcklwd  m0, m2
+    punpckhwd  m1, m2
+    mova  [dstq+2*lenq       ], m0
+    mova  [dstq+2*lenq+mmsize], m1
+    add      lenq, mmsize
+    jl .loop
+    REP_RET
+
+;------------------------------------------------------------------------------
+; void ff_conv_s16_to_flt(float *dst, const int16_t *src, int len);
+;------------------------------------------------------------------------------
+
+%macro CONV_S16_TO_FLT 0
+cglobal conv_s16_to_flt, 3,3,3, dst, src, len
+    lea      lenq, [2*lend]
+    add      srcq, lenq
+    lea      dstq, [dstq + 2*lenq]
+    neg      lenq
+    mova       m2, [pf_s16_inv_scale]
+    ALIGN 16
+.loop:
+    mova       m0, [srcq+lenq]
+    S16_TO_S32_SX 0, 1
+    cvtdq2ps   m0, m0
+    cvtdq2ps   m1, m1
+    mulps      m0, m2
+    mulps      m1, m2
+    mova  [dstq+2*lenq       ], m0
+    mova  [dstq+2*lenq+mmsize], m1
+    add      lenq, mmsize
+    jl .loop
+    REP_RET
+%endmacro
+
+INIT_XMM sse2
+CONV_S16_TO_FLT
+INIT_XMM sse4
+CONV_S16_TO_FLT
+
+;------------------------------------------------------------------------------
+; void ff_conv_s32_to_s16(int16_t *dst, const int32_t *src, int len);
+;------------------------------------------------------------------------------
+
+%macro CONV_S32_TO_S16 0
+cglobal conv_s32_to_s16, 3,3,4, dst, src, len
+    lea     lenq, [2*lend]
+    lea     srcq, [srcq+2*lenq]
+    add     dstq, lenq
+    neg     lenq
+.loop:
+    mova      m0, [srcq+2*lenq         ]
+    mova      m1, [srcq+2*lenq+  mmsize]
+    mova      m2, [srcq+2*lenq+2*mmsize]
+    mova      m3, [srcq+2*lenq+3*mmsize]
+    psrad     m0, 16
+    psrad     m1, 16
+    psrad     m2, 16
+    psrad     m3, 16
+    packssdw  m0, m1
+    packssdw  m2, m3
+    mova  [dstq+lenq       ], m0
+    mova  [dstq+lenq+mmsize], m2
+    add     lenq, mmsize*2
+    jl .loop
+%if mmsize == 8
+    emms
+    RET
+%else
+    REP_RET
+%endif
+%endmacro
+
+INIT_MMX mmx
+CONV_S32_TO_S16
+INIT_XMM sse2
+CONV_S32_TO_S16
+
+;------------------------------------------------------------------------------
+; void ff_conv_s32_to_flt(float *dst, const int32_t *src, int len);
+;------------------------------------------------------------------------------
+
+%macro CONV_S32_TO_FLT 0
+cglobal conv_s32_to_flt, 3,3,3, dst, src, len
+    lea     lenq, [4*lend]
+    add     srcq, lenq
+    add     dstq, lenq
+    neg     lenq
+    mova      m0, [pf_s32_inv_scale]
+    ALIGN 16
+.loop:
+    cvtdq2ps  m1, [srcq+lenq       ]
+    cvtdq2ps  m2, [srcq+lenq+mmsize]
+    mulps     m1, m1, m0
+    mulps     m2, m2, m0
+    mova  [dstq+lenq       ], m1
+    mova  [dstq+lenq+mmsize], m2
+    add     lenq, mmsize*2
+    jl .loop
+%if mmsize == 32
+    vzeroupper
+    RET
+%else
+    REP_RET
+%endif
+%endmacro
+
+INIT_XMM sse2
+CONV_S32_TO_FLT
+%if HAVE_AVX
+INIT_YMM avx
+CONV_S32_TO_FLT
+%endif
+
+;------------------------------------------------------------------------------
+; void ff_conv_flt_to_s16(int16_t *dst, const float *src, int len);
+;------------------------------------------------------------------------------
+
+INIT_XMM sse2
+cglobal conv_flt_to_s16, 3,3,5, dst, src, len
+    lea     lenq, [2*lend]
+    lea     srcq, [srcq+2*lenq]
+    add     dstq, lenq
+    neg     lenq
+    mova      m4, [pf_s16_scale]
+.loop:
+    mova      m0, [srcq+2*lenq         ]
+    mova      m1, [srcq+2*lenq+1*mmsize]
+    mova      m2, [srcq+2*lenq+2*mmsize]
+    mova      m3, [srcq+2*lenq+3*mmsize]
+    mulps     m0, m4
+    mulps     m1, m4
+    mulps     m2, m4
+    mulps     m3, m4
+    cvtps2dq  m0, m0
+    cvtps2dq  m1, m1
+    cvtps2dq  m2, m2
+    cvtps2dq  m3, m3
+    packssdw  m0, m1
+    packssdw  m2, m3
+    mova  [dstq+lenq       ], m0
+    mova  [dstq+lenq+mmsize], m2
+    add     lenq, mmsize*2
+    jl .loop
+    REP_RET
+
+;------------------------------------------------------------------------------
+; void ff_conv_flt_to_s32(int32_t *dst, const float *src, int len);
+;------------------------------------------------------------------------------
+
+%macro CONV_FLT_TO_S32 0
+cglobal conv_flt_to_s32, 3,3,5, dst, src, len
+    lea     lenq, [lend*4]
+    add     srcq, lenq
+    add     dstq, lenq
+    neg     lenq
+    mova      m4, [pf_s32_scale]
+.loop:
+    mulps     m0, m4, [srcq+lenq         ]
+    mulps     m1, m4, [srcq+lenq+1*mmsize]
+    mulps     m2, m4, [srcq+lenq+2*mmsize]
+    mulps     m3, m4, [srcq+lenq+3*mmsize]
+    cvtps2dq  m0, m0
+    cvtps2dq  m1, m1
+    cvtps2dq  m2, m2
+    cvtps2dq  m3, m3
+    mova  [dstq+lenq         ], m0
+    mova  [dstq+lenq+1*mmsize], m1
+    mova  [dstq+lenq+2*mmsize], m2
+    mova  [dstq+lenq+3*mmsize], m3
+    add     lenq, mmsize*4
+    jl .loop
+%if mmsize == 32
+    vzeroupper
+    RET
+%else
+    REP_RET
+%endif
+%endmacro
+
+INIT_XMM sse2
+CONV_FLT_TO_S32
+%if HAVE_AVX
+INIT_YMM avx
+CONV_FLT_TO_S32
+%endif
+
 ;-----------------------------------------------------------------------------
 ; void ff_conv_fltp_to_flt_6ch(float *dst, float *const *src, int len,
 ;                              int channels);
diff --git a/libavresample/x86/audio_convert_init.c b/libavresample/x86/audio_convert_init.c
index 206aede751..f41d974445 100644
--- a/libavresample/x86/audio_convert_init.c
+++ b/libavresample/x86/audio_convert_init.c
@@ -22,6 +22,22 @@
 #include "libavutil/cpu.h"
 #include "libavresample/audio_convert.h"
 
+extern void ff_conv_s16_to_s32_sse2(int16_t *dst, const int32_t *src, int len);
+
+extern void ff_conv_s16_to_flt_sse2(float *dst, const int16_t *src, int len);
+extern void ff_conv_s16_to_flt_sse4(float *dst, const int16_t *src, int len);
+
+extern void ff_conv_s32_to_s16_mmx (int16_t *dst, const int32_t *src, int len);
+extern void ff_conv_s32_to_s16_sse2(int16_t *dst, const int32_t *src, int len);
+
+extern void ff_conv_s32_to_flt_sse2(float *dst, const int32_t *src, int len);
+extern void ff_conv_s32_to_flt_avx (float *dst, const int32_t *src, int len);
+
+extern void ff_conv_flt_to_s16_sse2(int16_t *dst, const float *src, int len);
+
+extern void ff_conv_flt_to_s32_sse2(int32_t *dst, const float *src, int len);
+extern void ff_conv_flt_to_s32_avx (int32_t *dst, const float *src, int len);
+
 extern void ff_conv_fltp_to_flt_6ch_mmx (float *dst, float *const *src, int len);
 extern void ff_conv_fltp_to_flt_6ch_sse4(float *dst, float *const *src, int len);
 extern void ff_conv_fltp_to_flt_6ch_avx (float *dst, float *const *src, int len);
@@ -32,6 +48,8 @@ av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
     int mm_flags = av_get_cpu_flags();
 
     if (mm_flags & AV_CPU_FLAG_MMX && HAVE_MMX) {
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
+                                  0, 1, 8, "MMX", ff_conv_s32_to_s16_mmx);
         ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
                                   6, 1, 4, "MMX", ff_conv_fltp_to_flt_6ch_mmx);
     }
@@ -43,5 +61,31 @@ av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
         ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
                                   6, 16, 4, "AVX", ff_conv_fltp_to_flt_6ch_avx);
     }
+    if (mm_flags & AV_CPU_FLAG_SSE2 && HAVE_SSE) {
+        if (!(mm_flags & AV_CPU_FLAG_SSE2SLOW)) {
+            ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
+                                      0, 16, 16, "SSE2", ff_conv_s32_to_s16_sse2);
+        }
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16,
+                                  0, 16, 8, "SSE2", ff_conv_s16_to_s32_sse2);
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16,
+                                  0, 16, 8, "SSE2", ff_conv_s16_to_flt_sse2);
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32,
+                                  0, 16, 8, "SSE2", ff_conv_s32_to_flt_sse2);
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT,
+                                  0, 16, 16, "SSE2", ff_conv_flt_to_s16_sse2);
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT,
+                                  0, 16, 16, "SSE2", ff_conv_flt_to_s32_sse2);
+    }
+    if (mm_flags & AV_CPU_FLAG_SSE4 && HAVE_SSE) {
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16,
+                                  0, 16, 8, "SSE4", ff_conv_s16_to_flt_sse4);
+    }
+    if (mm_flags & AV_CPU_FLAG_AVX && HAVE_AVX) {
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32,
+                                  0, 32, 16, "AVX", ff_conv_s32_to_flt_avx);
+        ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT,
+                                  0, 32, 32, "AVX", ff_conv_flt_to_s32_avx);
+    }
 #endif
 }
diff --git a/libavutil/arm/float_dsp_init_neon.c b/libavutil/arm/float_dsp_init_neon.c
index fa6d0d7d15..3ca0288b31 100644
--- a/libavutil/arm/float_dsp_init_neon.c
+++ b/libavutil/arm/float_dsp_init_neon.c
@@ -26,7 +26,11 @@
 
 void ff_vector_fmul_neon(float *dst, const float *src0, const float *src1, int len);
 
+void ff_vector_fmac_scalar_neon(float *dst, const float *src, float mul,
+                                int len);
+
 void ff_float_dsp_init_neon(AVFloatDSPContext *fdsp)
 {
     fdsp->vector_fmul = ff_vector_fmul_neon;
+    fdsp->vector_fmac_scalar = ff_vector_fmac_scalar_neon;
 }
diff --git a/libavutil/arm/float_dsp_neon.S b/libavutil/arm/float_dsp_neon.S
index d66fa09424..03b164388f 100644
--- a/libavutil/arm/float_dsp_neon.S
+++ b/libavutil/arm/float_dsp_neon.S
@@ -62,3 +62,51 @@ function ff_vector_fmul_neon, export=1
 3:      vst1.32         {d16-d19},[r0,:128]!
         bx              lr
 endfunc
+
+function ff_vector_fmac_scalar_neon, export=1
+VFP     len .req r2
+VFP     acc .req r3
+NOVFP   len .req r3
+NOVFP   acc .req r2
+VFP     vdup.32         q15, d0[0]
+NOVFP   vdup.32         q15, r2
+        bics            r12, len, #15
+        mov             acc, r0
+        beq             3f
+        vld1.32         {q0},     [r1,:128]!
+        vld1.32         {q8},     [acc,:128]!
+        vld1.32         {q1},     [r1,:128]!
+        vld1.32         {q9},     [acc,:128]!
+1:      vmla.f32        q8,  q0,  q15
+        vld1.32         {q2},     [r1,:128]!
+        vld1.32         {q10},    [acc,:128]!
+        vmla.f32        q9,  q1,  q15
+        vld1.32         {q3},     [r1,:128]!
+        vld1.32         {q11},    [acc,:128]!
+        vmla.f32        q10, q2,  q15
+        vst1.32         {q8},     [r0,:128]!
+        vmla.f32        q11, q3,  q15
+        vst1.32         {q9},     [r0,:128]!
+        subs            r12, r12, #16
+        beq             2f
+        vld1.32         {q0},     [r1,:128]!
+        vld1.32         {q8},     [acc,:128]!
+        vst1.32         {q10},    [r0,:128]!
+        vld1.32         {q1},     [r1,:128]!
+        vld1.32         {q9},     [acc,:128]!
+        vst1.32         {q11},    [r0,:128]!
+        b               1b
+2:      vst1.32         {q10},    [r0,:128]!
+        vst1.32         {q11},    [r0,:128]!
+        ands            len, len, #15
+        it              eq
+        bxeq            lr
+3:      vld1.32         {q0},     [r1,:128]!
+        vld1.32         {q8},     [acc,:128]!
+        vmla.f32        q8,  q0,  q15
+        vst1.32         {q8},     [r0,:128]!
+        subs            len, len, #4
+        bgt             3b
+        bx              lr
+        .unreq          len
+endfunc
diff --git a/libavutil/float_dsp.c b/libavutil/float_dsp.c
index 87cfd88268..f5a8360c86 100644
--- a/libavutil/float_dsp.c
+++ b/libavutil/float_dsp.c
@@ -31,9 +31,18 @@ static void vector_fmul_c(float *dst, const float *src0, const float *src1,
         dst[i] = src0[i] * src1[i];
 }
 
+static void vector_fmac_scalar_c(float *dst, const float *src, float mul,
+                                 int len)
+{
+    int i;
+    for (i = 0; i < len; i++)
+        dst[i] += src[i] * mul;
+}
+
 void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
 {
     fdsp->vector_fmul = vector_fmul_c;
+    fdsp->vector_fmac_scalar = vector_fmac_scalar_c;
 
 #if ARCH_ARM
     ff_float_dsp_init_arm(fdsp);
diff --git a/libavutil/float_dsp.h b/libavutil/float_dsp.h
index 02c4ab7bde..735eb34c36 100644
--- a/libavutil/float_dsp.h
+++ b/libavutil/float_dsp.h
@@ -35,6 +35,22 @@ typedef struct AVFloatDSPContext {
      */
     void (*vector_fmul)(float *dst, const float *src0, const float *src1,
                         int len);
+
+    /**
+     * Multiply a vector of floats by a scalar float and add to
+     * destination vector.  Source and destination vectors must
+     * overlap exactly or not at all.
+     *
+     * @param dst result vector
+     *            constraints: 32-byte aligned
+     * @param src input vector
+     *            constraints: 32-byte aligned
+     * @param mul scalar value
+     * @param len length of vector
+     *            constraints: multiple of 16
+     */
+    void (*vector_fmac_scalar)(float *dst, const float *src, float mul,
+                               int len);
 } AVFloatDSPContext;
 
 /**
diff --git a/libavutil/mem.c b/libavutil/mem.c
index de22ad8db8..385ace0702 100644
--- a/libavutil/mem.c
+++ b/libavutil/mem.c
@@ -94,6 +94,8 @@ void *av_malloc(size_t size)
     if (size) //OS X on SDK 10.6 has a broken posix_memalign implementation
     if (posix_memalign(&ptr,ALIGN,size))
         ptr = NULL;
+#elif HAVE_ALIGNED_MALLOC
+    ptr = _aligned_malloc(size, ALIGN);
 #elif HAVE_MEMALIGN
     ptr = memalign(ALIGN,size);
     /* Why 64?
@@ -145,6 +147,8 @@ void *av_realloc(void *ptr, size_t size)
     ptr= realloc((char*)ptr - diff, size + diff);
     if(ptr) ptr = (char*)ptr + diff;
     return ptr;
+#elif HAVE_ALIGNED_MALLOC
+    return _aligned_realloc(ptr, size + !size, ALIGN);
 #else
     return realloc(ptr, size + !size);
 #endif
@@ -170,6 +174,8 @@ void av_free(void *ptr)
 #if CONFIG_MEMALIGN_HACK
     if (ptr)
         free((char*)ptr - ((char*)ptr)[-1]);
+#elif HAVE_ALIGNED_MALLOC
+    _aligned_free(ptr);
 #else
     free(ptr);
 #endif
diff --git a/libavutil/x86/float_dsp.asm b/libavutil/x86/float_dsp.asm
index 6ed716c026..f68e0bfe2d 100644
--- a/libavutil/x86/float_dsp.asm
+++ b/libavutil/x86/float_dsp.asm
@@ -21,6 +21,7 @@
 ;******************************************************************************
 
 %include "x86inc.asm"
+%include "x86util.asm"
 
 SECTION .text
 
@@ -55,3 +56,49 @@ VECTOR_FMUL
 INIT_YMM avx
 VECTOR_FMUL
 %endif
+
+;------------------------------------------------------------------------------
+; void ff_vector_fmac_scalar(float *dst, const float *src, float mul, int len)
+;------------------------------------------------------------------------------
+
+%macro VECTOR_FMAC_SCALAR 0
+%if UNIX64
+cglobal vector_fmac_scalar, 3,3,3, dst, src, len
+%else
+cglobal vector_fmac_scalar, 4,4,3, dst, src, mul, len
+%endif
+%if WIN64
+    SWAP 0, 2
+%endif
+%if ARCH_X86_32
+    VBROADCASTSS m0, mulm
+%else
+    shufps     xmm0, xmm0, 0
+%if cpuflag(avx)
+    vinsertf128  m0, m0, xmm0, 1
+%endif
+%endif
+    lea    lenq, [lend*4-2*mmsize]
+.loop
+    mulps    m1, m0, [srcq+lenq       ]
+    mulps    m2, m0, [srcq+lenq+mmsize]
+    addps    m1, m1, [dstq+lenq       ]
+    addps    m2, m2, [dstq+lenq+mmsize]
+    mova  [dstq+lenq       ], m1
+    mova  [dstq+lenq+mmsize], m2
+    sub    lenq, 2*mmsize
+    jge .loop
+%if mmsize == 32
+    vzeroupper
+    RET
+%else
+    REP_RET
+%endif
+%endmacro
+
+INIT_XMM sse
+VECTOR_FMAC_SCALAR
+%if HAVE_AVX
+INIT_YMM avx
+VECTOR_FMAC_SCALAR
+%endif
diff --git a/libavutil/x86/float_dsp_init.c b/libavutil/x86/float_dsp_init.c
index 8f6980cbc2..3e05b9d4ca 100644
--- a/libavutil/x86/float_dsp_init.c
+++ b/libavutil/x86/float_dsp_init.c
@@ -26,6 +26,11 @@ extern void ff_vector_fmul_sse(float *dst, const float *src0, const float *src1,
 extern void ff_vector_fmul_avx(float *dst, const float *src0, const float *src1,
                                int len);
 
+extern void ff_vector_fmac_scalar_sse(float *dst, const float *src, float mul,
+                                      int len);
+extern void ff_vector_fmac_scalar_avx(float *dst, const float *src, float mul,
+                                      int len);
+
 void ff_float_dsp_init_x86(AVFloatDSPContext *fdsp)
 {
 #if HAVE_YASM
@@ -33,9 +38,11 @@ void ff_float_dsp_init_x86(AVFloatDSPContext *fdsp)
 
     if (mm_flags & AV_CPU_FLAG_SSE && HAVE_SSE) {
         fdsp->vector_fmul = ff_vector_fmul_sse;
+        fdsp->vector_fmac_scalar = ff_vector_fmac_scalar_sse;
     }
     if (mm_flags & AV_CPU_FLAG_AVX && HAVE_AVX) {
         fdsp->vector_fmul = ff_vector_fmul_avx;
+        fdsp->vector_fmac_scalar = ff_vector_fmac_scalar_avx;
     }
 #endif
 }