@@ -33,6 +33,7 @@ version <next>:

 - Microsoft ATC Screen decoder

 - RTSP listen mode

 - TechSmith Screen Codec 2 decoder

+- AAC encoding via libfdk-aac

 version 0.8:

@@ -170,6 +170,7 @@ External library support:

   --enable-libdc1394       enable IIDC-1394 grabbing using libdc1394

                            and libraw1394 [no]

   --enable-libfaac         enable FAAC support via libfaac [no]

+  --enable-libfdk-aac      enable AAC support via libfdk-aac [no]

   --enable-libfreetype     enable libfreetype [no]

   --enable-libgsm          enable GSM support via libgsm [no]

   --enable-libilbc         enable iLBC de/encoding via libilbc [no]

@@ -943,6 +944,7 @@ CONFIG_LIST="

     libcdio

     libdc1394

     libfaac

+    libfdk_aac

     libfreetype

     libgsm

     libilbc

@@ -1448,6 +1450,7 @@ h264_parser_select="golomb h264dsp h264pred"

 # external libraries

 libfaac_encoder_deps="libfaac"

+libfdk_aac_encoder_deps="libfdk_aac"

 libgsm_decoder_deps="libgsm"

 libgsm_encoder_deps="libgsm"

 libgsm_ms_decoder_deps="libgsm"

@@ -2968,6 +2971,7 @@ enabled avisynth   && require2 vfw32 "windows.h vfw.h" AVIFileInit -lavifil32

 enabled frei0r     && { check_header frei0r.h || die "ERROR: frei0r.h header not found"; }

 enabled gnutls     && require_pkg_config gnutls gnutls/gnutls.h gnutls_global_init

 enabled libfaac    && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac

+enabled libfdk_aac && require  libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac

 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

@@ -3259,6 +3263,7 @@ echo "gnutls enabled            ${gnutls-no}"

 echo "libcdio support           ${libcdio-no}"

 echo "libdc1394 support         ${libdc1394-no}"

 echo "libfaac enabled           ${libfaac-no}"

+echo "libfdk-aac enabled        ${libfdk_aac-no}"

 echo "libgsm enabled            ${libgsm-no}"

 echo "libilbc enabled           ${libilbc-no}"

 echo "libmp3lame enabled        ${libmp3lame-no}"

@@ -18,8 +18,8 @@ explicitly requested by passing the appropriate flags to

 @section OpenCORE and VisualOn libraries

-Spun off Google Android sources, OpenCore and VisualOn libraries provide

-encoders for a number of audio codecs.

+Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer

+libraries provide encoders for a number of audio codecs.

 @float NOTE

 OpenCORE and VisualOn libraries are under the Apache License 2.0

@@ -55,6 +55,14 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the

 instructions for installing the library.

 Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.

+@subsection Fraunhofer AAC library

+

+Libav can make use of the Fraunhofer AAC library for AAC encoding.

+

+Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the

+instructions for installing the library.

+Then pass @code{--enable-libfdk-aac} to configure to enable it.

+

 @section LAME

 Libav can make use of the LAME library for MP3 encoding.

@@ -595,6 +595,7 @@ OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o

 # external codec libraries

 OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o audio_frame_queue.o

+OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o audio_frame_queue.o

 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o

 OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o

 OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsm.o

@@ -380,6 +380,7 @@ void avcodec_register_all(void)

     /* external libraries */

     REGISTER_ENCODER (LIBFAAC, libfaac);

+    REGISTER_ENCODER (LIBFDK_AAC, libfdk_aac);

     REGISTER_ENCDEC  (LIBGSM, libgsm);

     REGISTER_ENCDEC  (LIBGSM_MS, libgsm_ms);

     REGISTER_ENCDEC  (LIBILBC, libilbc);

new file mode 100644

@@ -0,0 +1,384 @@

+/*

+ * AAC encoder wrapper

+ * Copyright (c) 2012 Martin Storsjo

+ *

+ * This file is part of Libav.

+ *

+ * Libav is free software; you can redistribute it and/or

+ * modify it under the terms of the GNU Lesser General Public

+ * License as published by the Free Software Foundation; either

+ * version 2.1 of the License, or (at your option) any later version.

+ *

+ * Libav is distributed in the hope that it will be useful,

+ * but WITHOUT ANY WARRANTY; without even the implied warranty of

+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

+ * Lesser General Public License for more details.

+ *

+ * You should have received a copy of the GNU Lesser General Public

+ * License along with Libav; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

+ */

+

+#include <fdk-aac/aacenc_lib.h>

+

+#include "avcodec.h"

+#include "audio_frame_queue.h"

+#include "internal.h"

+#include "libavutil/audioconvert.h"

+#include "libavutil/opt.h"

+

+typedef struct AACContext {

+    const AVClass *class;

+    HANDLE_AACENCODER handle;

+    int afterburner;

+    int eld_sbr;

+    int signaling;

+

+    AudioFrameQueue afq;

+} AACContext;

+

+static const AVOption aac_enc_options[] = {

+    { "afterburner", "Afterburner (improved quality)", offsetof(AACContext, afterburner), AV_OPT_TYPE_INT, { 1 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },

+    { "eld_sbr", "Enable SBR for ELD (for SBR in other configurations, use the -profile parameter)", offsetof(AACContext, eld_sbr), AV_OPT_TYPE_INT, { 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },

+    { "signaling", "SBR/PS signaling style", offsetof(AACContext, signaling), AV_OPT_TYPE_INT, { -1 }, -1, 2, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },

+    { "default", "Choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)", 0, AV_OPT_TYPE_CONST, { -1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },

+    { "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { 0 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },

+    { "explicit_sbr", "Explicit SBR, implicit PS signaling", 0, AV_OPT_TYPE_CONST, { 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },

+    { "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },

+    { NULL }

+};

+

+static const AVClass aac_enc_class = {

+    "libfdk_aac", av_default_item_name, aac_enc_options, LIBAVUTIL_VERSION_INT

+};

+

+static const char *aac_get_error(AACENC_ERROR err)

+{

+    switch (err) {

+    case AACENC_OK:

+        return "No error";

+    case AACENC_INVALID_HANDLE:

+        return "Invalid handle";

+    case AACENC_MEMORY_ERROR:

+        return "Memory allocation error";

+    case AACENC_UNSUPPORTED_PARAMETER:

+        return "Unsupported parameter";

+    case AACENC_INVALID_CONFIG:

+        return "Invalid config";

+    case AACENC_INIT_ERROR:

+        return "Initialization error";

+    case AACENC_INIT_AAC_ERROR:

+        return "AAC library initialization error";

+    case AACENC_INIT_SBR_ERROR:

+        return "SBR library initialization error";

+    case AACENC_INIT_TP_ERROR:

+        return "Transport library initialization error";

+    case AACENC_INIT_META_ERROR:

+        return "Metadata library initialization error";

+    case AACENC_ENCODE_ERROR:

+        return "Encoding error";

+    case AACENC_ENCODE_EOF:

+        return "End of file";

+    default:

+        return "Unknown error";

+    }

+}

+

+static int aac_encode_close(AVCodecContext *avctx)

+{

+    AACContext *s = avctx->priv_data;

+

+    if (s->handle)

+        aacEncClose(&s->handle);

+#if FF_API_OLD_ENCODE_AUDIO

+    av_freep(&avctx->coded_frame);

+#endif

+    av_freep(&avctx->extradata);

+    ff_af_queue_close(&s->afq);

+

+    return 0;

+}

+

+static av_cold int aac_encode_init(AVCodecContext *avctx)

+{

+    AACContext *s = avctx->priv_data;

+    int ret = AVERROR(EINVAL);

+    AACENC_InfoStruct info = { 0 };

+    CHANNEL_MODE mode;

+    AACENC_ERROR err;

+    int aot = FF_PROFILE_AAC_LOW + 1;

+    int sce = 0, cpe = 0;

+

+    if ((err = aacEncOpen(&s->handle, 0, avctx->channels)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR, "Unable to open the encoder: %s\n",

+               aac_get_error(err));

+        goto error;

+    }

+

+    if (avctx->profile != FF_PROFILE_UNKNOWN)

+        aot = avctx->profile + 1;

+

+    if ((err = aacEncoder_SetParam(s->handle, AACENC_AOT, aot)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR, "Unable to set the AOT %d: %s\n",

+               aot, aac_get_error(err));

+        goto error;

+    }

+

+    if (aot == FF_PROFILE_AAC_ELD + 1 && s->eld_sbr) {

+        if ((err = aacEncoder_SetParam(s->handle, AACENC_SBR_MODE,

+                                       1)) != AACENC_OK) {

+            av_log(avctx, AV_LOG_ERROR, "Unable to enable SBR for ELD: %s\n",

+                   aac_get_error(err));

+            goto error;

+        }

+    }

+

+    if ((err = aacEncoder_SetParam(s->handle, AACENC_SAMPLERATE,

+                                   avctx->sample_rate)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR, "Unable to set the sample rate %d: %s\n",

+               avctx->sample_rate, aac_get_error(err));

+        goto error;

+    }

+

+    switch (avctx->channels) {

+    case 1: mode = MODE_1;       sce = 1; cpe = 0; break;

+    case 2: mode = MODE_2;       sce = 0; cpe = 1; break;

+    case 3: mode = MODE_1_2;     sce = 1; cpe = 1; break;

+    case 4: mode = MODE_1_2_1;   sce = 2; cpe = 1; break;

+    case 5: mode = MODE_1_2_2;   sce = 1; cpe = 2; break;

+    case 6: mode = MODE_1_2_2_1; sce = 2; cpe = 2; break;

+    default:

+        av_log(avctx, AV_LOG_ERROR,

+               "Unsupported number of channels %d\n", avctx->channels);

+        goto error;

+    }

+

+    if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELMODE,

+                                   mode)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR,

+               "Unable to set channel mode %d: %s\n", mode, aac_get_error(err));

+        goto error;

+    }

+

+    if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELORDER,

+                                   1)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR,

+               "Unable to set wav channel order %d: %s\n",

+               mode, aac_get_error(err));

+        goto error;

+    }

+

+    if (avctx->flags & CODEC_FLAG_QSCALE) {

+        int mode = avctx->global_quality;

+        if (mode <  1 || mode > 5) {

+            av_log(avctx, AV_LOG_WARNING,

+                   "VBR quality %d out of range, should be 1-5\n", mode);

+            mode = av_clip(mode, 1, 5);

+        }

+        if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATEMODE,

+                                       mode)) != AACENC_OK) {

+            av_log(avctx, AV_LOG_ERROR, "Unable to set the VBR bitrate mode %d: %s\n",

+                   mode, aac_get_error(err));

+            goto error;

+        }

+    } else {

+        if (avctx->bit_rate <= 0) {

+            if (avctx->profile == FF_PROFILE_AAC_HE_V2) {

+                sce = 1;

+                cpe = 0;

+            }

+            avctx->bit_rate = (96*sce + 128*cpe) * avctx->sample_rate / 44;

+            if (avctx->profile == FF_PROFILE_AAC_HE ||

+                avctx->profile == FF_PROFILE_AAC_HE_V2 ||

+                s->eld_sbr)

+                avctx->bit_rate /= 2;

+        }

+        if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATE,

+                                       avctx->bit_rate)) != AACENC_OK) {

+            av_log(avctx, AV_LOG_ERROR, "Unable to set the bitrate %d: %s\n",

+                   avctx->bit_rate, aac_get_error(err));

+            goto error;

+        }

+    }

+

+    /* Choose bitstream format - if global header is requested, use

+     * raw access units, otherwise use ADTS. */

+    if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX,

+                                   avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : 2)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",

+               aac_get_error(err));

+        goto error;

+    }

+

+    /* If no signaling mode is chosen, use explicit hierarchical signaling

+     * if using mp4 mode (raw access units, with global header) and

+     * implicit signaling if using ADTS. */

+    if (s->signaling < 0)

+        s->signaling = avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 2 : 0;

+

+    if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE,

+                                   s->signaling)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR, "Unable to set signaling mode %d: %s\n",

+               s->signaling, aac_get_error(err));

+        goto error;

+    }

+

+    if ((err = aacEncoder_SetParam(s->handle, AACENC_AFTERBURNER,

+                                   s->afterburner)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR, "Unable to set afterburner to %d: %s\n",

+               s->afterburner, aac_get_error(err));

+        goto error;

+    }

+

+    if ((err = aacEncEncode(s->handle, NULL, NULL, NULL, NULL)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR, "Unable to initialize the encoder: %s\n",

+               aac_get_error(err));

+        return AVERROR(EINVAL);

+    }

+

+    if ((err = aacEncInfo(s->handle, &info)) != AACENC_OK) {

+        av_log(avctx, AV_LOG_ERROR, "Unable to get encoder info: %s\n",

+               aac_get_error(err));

+        goto error;

+    }

+

+#if FF_API_OLD_ENCODE_AUDIO

+    avctx->coded_frame = avcodec_alloc_frame();

+    if (!avctx->coded_frame) {

+        ret = AVERROR(ENOMEM);

+        goto error;

+    }

+#endif

+    avctx->frame_size = info.frameLength;

+    avctx->delay      = info.encoderDelay;

+    ff_af_queue_init(avctx, &s->afq);

+

+    if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {

+        avctx->extradata_size = info.confSize;

+        avctx->extradata      = av_mallocz(avctx->extradata_size +

+                                           FF_INPUT_BUFFER_PADDING_SIZE);

+        if (!avctx->extradata) {

+            ret = AVERROR(ENOMEM);

+            goto error;

+        }

+

+        memcpy(avctx->extradata, info.confBuf, info.confSize);

+    }

+    return 0;

+error:

+    aac_encode_close(avctx);

+    return ret;

+}

+

+static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

+                            const AVFrame *frame, int *got_packet_ptr)

+{

+    AACContext    *s        = avctx->priv_data;

+    AACENC_BufDesc in_buf   = { 0 }, out_buf = { 0 };

+    AACENC_InArgs  in_args  = { 0 };

+    AACENC_OutArgs out_args = { 0 };

+    int in_buffer_identifier = IN_AUDIO_DATA;

+    int in_buffer_size, in_buffer_element_size;

+    int out_buffer_identifier = OUT_BITSTREAM_DATA;

+    int out_buffer_size, out_buffer_element_size;

+    void *in_ptr, *out_ptr;

+    int ret;

+    AACENC_ERROR err;

+

+    /* handle end-of-stream small frame and flushing */

+    if (!frame) {

+        in_args.numInSamples = -1;

+    } else {

+        in_ptr                   = frame->data[0];

+        in_buffer_size           = 2 * avctx->channels * frame->nb_samples;

+        in_buffer_element_size   = 2;

+

+        in_args.numInSamples     = avctx->channels * frame->nb_samples;

+        in_buf.numBufs           = 1;

+        in_buf.bufs              = &in_ptr;

+        in_buf.bufferIdentifiers = &in_buffer_identifier;

+        in_buf.bufSizes          = &in_buffer_size;

+        in_buf.bufElSizes        = &in_buffer_element_size;

+

+        /* add current frame to the queue */

+        if ((ret = ff_af_queue_add(&s->afq, frame) < 0))

+            return ret;

+    }

+

+    /* The maximum packet size is 6144 bits aka 768 bytes per channel. */

+    if ((ret = ff_alloc_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) {

+        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");

+        return ret;

+    }

+

+    out_ptr                   = avpkt->data;

+    out_buffer_size           = avpkt->size;

+    out_buffer_element_size   = 1;

+    out_buf.numBufs           = 1;

+    out_buf.bufs              = &out_ptr;

+    out_buf.bufferIdentifiers = &out_buffer_identifier;

+    out_buf.bufSizes          = &out_buffer_size;

+    out_buf.bufElSizes        = &out_buffer_element_size;

+

+    if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args,

+                            &out_args)) != AACENC_OK) {

+        if (!frame && err == AACENC_ENCODE_EOF)

+            return 0;

+        av_log(avctx, AV_LOG_ERROR, "Unable to encode frame: %s\n",

+               aac_get_error(err));

+        return AVERROR(EINVAL);

+    }

+

+    if (!out_args.numOutBytes)

+        return 0;

+

+    /* Get the next frame pts & duration */

+    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,

+                       &avpkt->duration);

+

+    avpkt->size     = out_args.numOutBytes;

+    *got_packet_ptr = 1;

+    return 0;

+}

+

+static const AVProfile profiles[] = {

+    { FF_PROFILE_AAC_LOW,   "LC"       },

+    { FF_PROFILE_AAC_HE,    "HE-AAC"   },

+    { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },

+    { FF_PROFILE_AAC_LD,    "LD"       },

+    { FF_PROFILE_AAC_ELD,   "ELD"      },

+    { FF_PROFILE_UNKNOWN },

+};

+

+static const AVCodecDefault aac_encode_defaults[] = {

+    { "b", "0" },

+    { NULL }

+};

+

+static const uint64_t aac_channel_layout[] = {

+    AV_CH_LAYOUT_MONO,

+    AV_CH_LAYOUT_STEREO,

+    AV_CH_LAYOUT_SURROUND,

+    AV_CH_LAYOUT_4POINT0,

+    AV_CH_LAYOUT_5POINT0_BACK,

+    AV_CH_LAYOUT_5POINT1_BACK,

+    0,

+};

+

+AVCodec ff_libfdk_aac_encoder = {

+    .name            = "libfdk_aac",

+    .type            = AVMEDIA_TYPE_AUDIO,

+    .id              = CODEC_ID_AAC,

+    .priv_data_size  = sizeof(AACContext),

+    .init            = aac_encode_init,

+    .encode2         = aac_encode_frame,

+    .close           = aac_encode_close,

+    .capabilities    = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,

+    .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,

+                                                      AV_SAMPLE_FMT_NONE },

+    .long_name       = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"),

+    .priv_class      = &aac_enc_class,

+    .defaults        = aac_encode_defaults,

+    .profiles        = profiles,

+    .channel_layouts = aac_channel_layout,

+};

@@ -27,7 +27,7 @@

  */

 #define LIBAVCODEC_VERSION_MAJOR 54

-#define LIBAVCODEC_VERSION_MINOR 18

+#define LIBAVCODEC_VERSION_MINOR 19

 #define LIBAVCODEC_VERSION_MICRO  0

 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \