@@ -10,6 +10,7 @@ version next:

 - atempo filter

 - ffprobe -show_data option

 - RTMPT protocol support

+- iLBC encoding/decoding via libilbc

 version 0.11:

@@ -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}"

@@ -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

@@ -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

@@ -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 \

@@ -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);

@@ -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;

@@ -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]

@@ -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);

@@ -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);

@@ -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;

@@ -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;

@@ -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

new file mode 100644

@@ -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,

+};

@@ -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

@@ -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;

+            }

         }

     }

@@ -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, \

@@ -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());

@@ -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 \

@@ -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);

@@ -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",

 };

@@ -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)) {

new file mode 100644

@@ -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;