Merge remote-tracking branch 'qatar/master'

* qatar/master:
  build: Fix Ogg demuxer dependencies
  build: Fix FLAC demuxer dependencies
  flac: Move flac functions shared between libraries to flac common code
  build: Fix CAF demuxer dependencies
  build: Fix MP2 muxer dependencies
  build: Add missing build rules for the ISMV muxer
  configure: Drop redundant mxf_d10 test dependency declaration
  Support AAC encoding via the external library fdk-aac
  libavcodec: Add more AAC profiles
  dct/fft-test: use a replacement getopt() if the system has none present.

Conflicts:
	Changelog
	libavcodec/Makefile
	libavcodec/version.h

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

Changelog

@@ -17,6 +17,7 @@ version next:
 - Microsoft ATC Screen decoder
 - RTSP listen mode
 - TechSmith Screen Codec 2 decoder
+- AAC encoding via libfdk-aac
 - showwaves filter
 - LucasArts SMUSH playback support
 - SAMI demuxer and decoder

compat/getopt.c

@@ -0,0 +1,84 @@
+/*
+ * 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
+ */
+
+/*
+ * This file was copied from the following newsgroup posting:
+ *
+ * Newsgroups: mod.std.unix
+ * Subject: public domain AT&T getopt source
+ * Date: 3 Nov 85 19:34:15 GMT
+ *
+ * Here's something you've all been waiting for:  the AT&T public domain
+ * source for getopt(3).  It is the code which was given out at the 1985
+ * UNIFORUM conference in Dallas.  I obtained it by electronic mail
+ * directly from AT&T.  The people there assure me that it is indeed
+ * in the public domain.
+ */
+
+#define EOF (-1)
+
+static int opterr = 1;
+static int optind = 1;
+static int optopt;
+static char *optarg;
+
+#undef fprintf
+
+static int getopt(int argc, char *argv[], char *opts)
+{
+    static int sp = 1;
+    int c;
+    char *cp;
+
+    if (sp == 1)
+        if (optind >= argc ||
+            argv[optind][0] != '-' || argv[optind][1] == '\0')
+            return EOF;
+        else if (!strcmp(argv[optind], "--")) {
+            optind++;
+            return EOF;
+        }
+    optopt = c = argv[optind][sp];
+    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
+        fprintf(stderr, ": illegal option -- %c\n", c);
+        if (argv[optind][++sp] == '\0') {
+            optind++;
+            sp = 1;
+        }
+        return '?';
+    }
+    if (*++cp == ':') {
+        if (argv[optind][sp+1] != '\0')
+            optarg = &argv[optind++][sp+1];
+        else if(++optind >= argc) {
+            fprintf(stderr, ": option requires an argument -- %c\n", c);
+            sp = 1;
+            return '?';
+        } else
+            optarg = argv[optind++];
+        sp = 1;
+    } else {
+        if (argv[optind][++sp] == '\0') {
+            sp = 1;
+            optind++;
+        }
+        optarg = NULL;
+    }
+
+    return c;
+}

configure

@@ -178,6 +178,7 @@ 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-libiec61883     enable iec61883 via libiec61883 [no]
@@ -1053,6 +1054,7 @@ CONFIG_LIST="
     libcelt
     libdc1394
     libfaac
+    libfdk_aac
     libfreetype
     libgsm
     libiec61883
@@ -1210,6 +1212,7 @@ HAVE_LIST="
     fork
     getaddrinfo
     gethrtime
+    getopt
     GetProcessAffinityMask
     GetProcessMemoryInfo
     GetProcessTimes
@@ -1613,6 +1616,7 @@ h264_parser_select="golomb h264dsp h264pred"
 libaacplus_encoder_deps="libaacplus"
 libcelt_decoder_deps="libcelt"
 libfaac_encoder_deps="libfaac"
+libfdk_aac_encoder_deps="libfdk_aac"
 libgsm_decoder_deps="libgsm"
 libgsm_encoder_deps="libgsm"
 libgsm_ms_decoder_deps="libgsm"
@@ -1797,7 +1801,6 @@ test_deps(){
     done
 }
 
-mxf_d10_test_deps="avfilter"
 seek_lavf_mxf_d10_test_deps="mxf_d10_test"
 
 test_deps _muxer _demuxer                                               \
@@ -3178,6 +3181,7 @@ check_func  fcntl
 check_func  fork
 check_func  getaddrinfo $network_extralibs
 check_func  gethrtime
+check_func  getopt
 check_func  getrusage
 check_struct "sys/time.h sys/resource.h" "struct rusage" ru_maxrss
 check_func  gettimeofday
@@ -3308,6 +3312,7 @@ enabled libcelt    && require libcelt celt/celt.h celt_decode -lcelt0 &&
                       { check_lib celt/celt.h celt_decoder_create_custom -lcelt0 ||
                         die "ERROR: libcelt version must be >= 0.11.0."; }
 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
@@ -3684,6 +3689,7 @@ echo "libcdio support           ${libcdio-no}"
 echo "libcelt enabled           ${libcelt-no}"
 echo "libdc1394 support         ${libdc1394-no}"
 echo "libfaac enabled           ${libfaac-no}"
+echo "libfdk-aac enabled        ${libfdk_aac-no}"
 echo "libgsm enabled            ${libgsm-no}"
 echo "libiec61883 support       ${libiec61883-no}"
 echo "libilbc enabled           ${libilbc-no}"

doc/general.texi

@@ -26,8 +26,8 @@ instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp
 
 @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
@@ -63,6 +63,14 @@ 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
 
 FFmpeg can make use of the LAME library for MP3 encoding.

libavcodec/Makefile

@@ -624,22 +624,23 @@ OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o
 # libavformat dependencies
 OBJS-$(CONFIG_ADTS_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_ADX_DEMUXER)             += adx.o
-OBJS-$(CONFIG_CAF_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
+OBJS-$(CONFIG_CAF_DEMUXER)             += mpeg4audio.o mpegaudiodata.o  \
+                                          ac3tab.o
 OBJS-$(CONFIG_DV_DEMUXER)              += dv_profile.o
 OBJS-$(CONFIG_DV_MUXER)                += dv_profile.o timecode.o
-OBJS-$(CONFIG_FLAC_DEMUXER)            += flacdec.o flacdata.o flac.o vorbis_data.o \
+OBJS-$(CONFIG_FLAC_DEMUXER)            += flac.o flacdata.o vorbis_data.o \
                                           vorbis_parser.o xiph.o
-OBJS-$(CONFIG_FLAC_MUXER)              += flacdec.o flacdata.o flac.o vorbis_data.o
+OBJS-$(CONFIG_FLAC_MUXER)              += flac.o flacdata.o vorbis_data.o
 OBJS-$(CONFIG_FLV_DEMUXER)             += mpeg4audio.o
 OBJS-$(CONFIG_GXF_DEMUXER)             += mpeg12data.o
 OBJS-$(CONFIG_IFF_DEMUXER)             += iff.o
+OBJS-$(CONFIG_ISMV_MUXER)              += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_LATM_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER)    += xiph.o mpeg4audio.o vorbis_data.o \
-                                          flacdec.o flacdata.o flac.o
+                                          flac.o flacdata.o
 OBJS-$(CONFIG_MATROSKA_DEMUXER)        += mpeg4audio.o mpegaudiodata.o
-OBJS-$(CONFIG_MATROSKA_MUXER)          += xiph.o mpeg4audio.o \
-                                          flacdec.o flacdata.o flac.o \
-                                          mpegaudiodata.o vorbis_data.o
+OBJS-$(CONFIG_MATROSKA_MUXER)          += mpeg4audio.o mpegaudiodata.o  \
+                                          flac.o flacdata.o vorbis_data.o xiph.o
 OBJS-$(CONFIG_MP2_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
 OBJS-$(CONFIG_MP3_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
 OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o ac3tab.o timecode.o
@@ -648,22 +649,23 @@ OBJS-$(CONFIG_MPEGTS_MUXER)            += mpegvideo.o mpeg4audio.o
 OBJS-$(CONFIG_MPEGTS_DEMUXER)          += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MXF_MUXER)               += timecode.o
 OBJS-$(CONFIG_NUT_MUXER)               += mpegaudiodata.o
-OBJS-$(CONFIG_OGG_DEMUXER)             += flacdec.o flacdata.o flac.o \
-                                          dirac.o mpeg12data.o vorbis_parser.o \
-                                          xiph.o vorbis_data.o
-OBJS-$(CONFIG_OGG_MUXER)               += xiph.o flacdec.o flacdata.o flac.o \
+OBJS-$(CONFIG_OGG_DEMUXER)             += xiph.o flac.o flacdata.o     \
+                                          mpeg12data.o vorbis_parser.o \
+                                          dirac.o vorbis_data.o
+OBJS-$(CONFIG_OGG_MUXER)               += xiph.o flac.o flacdata.o \
                                           vorbis_data.o
 OBJS-$(CONFIG_RTP_MUXER)               += mpeg4audio.o mpegvideo.o xiph.o
 OBJS-$(CONFIG_SPDIF_DEMUXER)           += aacadtsdec.o mpeg4audio.o
-OBJS-$(CONFIG_WEBM_MUXER)              += xiph.o mpeg4audio.o \
-                                          flacdec.o flacdata.o flac.o \
-                                          mpegaudiodata.o vorbis_data.o
+OBJS-$(CONFIG_WEBM_MUXER)              += mpeg4audio.o mpegaudiodata.o  \
+                                          xiph.o flac.o flacdata.o \
+                                          vorbis_data.o
 OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 
 # external codec libraries
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
 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

libavcodec/allcodecs.c

@@ -417,6 +417,7 @@ void avcodec_register_all(void)
     /* external libraries */
     REGISTER_DECODER (LIBCELT, libcelt);
     REGISTER_ENCODER (LIBFAAC, libfaac);
+    REGISTER_ENCODER (LIBFDK_AAC, libfdk_aac);
     REGISTER_ENCDEC  (LIBGSM, libgsm);
     REGISTER_ENCDEC  (LIBGSM_MS, libgsm_ms);
     REGISTER_ENCDEC  (LIBILBC, libilbc);

libavcodec/avcodec.h

@@ -2825,6 +2825,10 @@ typedef struct AVCodecContext {
 #define FF_PROFILE_AAC_LOW  1
 #define FF_PROFILE_AAC_SSR  2
 #define FF_PROFILE_AAC_LTP  3
+#define FF_PROFILE_AAC_HE   4
+#define FF_PROFILE_AAC_HE_V2 28
+#define FF_PROFILE_AAC_LD   22
+#define FF_PROFILE_AAC_ELD  38
 
 #define FF_PROFILE_DTS         20
 #define FF_PROFILE_DTS_ES      30

libavcodec/dct-test.c

@@ -25,10 +25,13 @@
  * Started from sample code by Juan J. Sierralta P.
  */
 
+#include "config.h"
 #include <stdlib.h>
 #include <stdio.h>
 #include <string.h>
+#if HAVE_UNISTD_H
 #include <unistd.h>
+#endif
 #include <math.h>
 
 #include "libavutil/cpu.h"
@@ -519,6 +522,10 @@ static void help(void)
            "-t          speed test\n");
 }
 
+#if !HAVE_GETOPT
+#include "compat/getopt.c"
+#endif
+
 int main(int argc, char **argv)
 {
     int test_idct = 0, test_248_dct = 0;

libavcodec/fft-test.c

@@ -34,7 +34,9 @@
 #include "rdft.h"
 #endif
 #include <math.h>
+#if HAVE_UNISTD_H
 #include <unistd.h>
+#endif
 #include <stdlib.h>
 #include <string.h>
 
@@ -229,6 +231,10 @@ enum tf_transform {
     TRANSFORM_DCT,
 };
 
+#if !HAVE_GETOPT
+#include "compat/getopt.c"
+#endif
+
 int main(int argc, char **argv)
 {
     FFTComplex *tab, *tab1, *tab_ref;

libavcodec/flac.c

@@ -20,6 +20,9 @@
  */
 
 #include "libavutil/crc.h"
+#include "libavutil/log.h"
+#include "bytestream.h"
+#include "get_bits.h"
 #include "flac.h"
 #include "flacdata.h"
 
@@ -150,3 +153,74 @@ int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
 
     return count;
 }
+
+int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,
+                               enum FLACExtradataFormat *format,
+                               uint8_t **streaminfo_start)
+{
+    if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
+        av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
+        return 0;
+    }
+    if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
+        /* extradata contains STREAMINFO only */
+        if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
+            av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
+                   FLAC_STREAMINFO_SIZE-avctx->extradata_size);
+        }
+        *format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
+        *streaminfo_start = avctx->extradata;
+    } else {
+        if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
+            av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
+            return 0;
+        }
+        *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
+        *streaminfo_start = &avctx->extradata[8];
+    }
+    return 1;
+}
+
+void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
+                              const uint8_t *buffer)
+{
+    GetBitContext gb;
+    init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
+
+    skip_bits(&gb, 16); /* skip min blocksize */
+    s->max_blocksize = get_bits(&gb, 16);
+    if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) {
+        av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n",
+               s->max_blocksize);
+        s->max_blocksize = 16;
+    }
+
+    skip_bits(&gb, 24); /* skip min frame size */
+    s->max_framesize = get_bits_long(&gb, 24);
+
+    s->samplerate = get_bits_long(&gb, 20);
+    s->channels = get_bits(&gb, 3) + 1;
+    s->bps = get_bits(&gb, 5) + 1;
+
+    avctx->channels = s->channels;
+    avctx->sample_rate = s->samplerate;
+    avctx->bits_per_raw_sample = s->bps;
+
+    s->samples  = get_bits_long(&gb, 32) << 4;
+    s->samples |= get_bits(&gb, 4);
+
+    skip_bits_long(&gb, 64); /* md5 sum */
+    skip_bits_long(&gb, 64); /* md5 sum */
+}
+
+void avpriv_flac_parse_block_header(const uint8_t *block_header,
+                                int *last, int *type, int *size)
+{
+    int tmp = bytestream_get_byte(&block_header);
+    if (last)
+        *last = tmp & 0x80;
+    if (type)
+        *type = tmp & 0x7F;
+    if (size)
+        *size = bytestream_get_be24(&block_header);
+}

libavcodec/flacdec.c

@@ -75,33 +75,6 @@ static const int64_t flac_channel_layouts[6] = {
 
 static void allocate_buffers(FLACContext *s);
 
-int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,
-                               enum FLACExtradataFormat *format,
-                               uint8_t **streaminfo_start)
-{
-    if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
-        av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
-        return 0;
-    }
-    if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
-        /* extradata contains STREAMINFO only */
-        if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
-            av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
-                   FLAC_STREAMINFO_SIZE-avctx->extradata_size);
-        }
-        *format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
-        *streaminfo_start = avctx->extradata;
-    } else {
-        if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
-            av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
-            return 0;
-        }
-        *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
-        *streaminfo_start = &avctx->extradata[8];
-    }
-    return 1;
-}
-
 static void flac_set_bps(FLACContext *s)
 {
     enum AVSampleFormat req = s->avctx->request_sample_fmt;
@@ -175,52 +148,6 @@ static void allocate_buffers(FLACContext *s)
     }
 }
 
-void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
-                              const uint8_t *buffer)
-{
-    GetBitContext gb;
-    init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
-
-    skip_bits(&gb, 16); /* skip min blocksize */
-    s->max_blocksize = get_bits(&gb, 16);
-    if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) {
-        av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n",
-               s->max_blocksize);
-        s->max_blocksize = 16;
-    }
-
-    skip_bits(&gb, 24); /* skip min frame size */
-    s->max_framesize = get_bits_long(&gb, 24);
-
-    s->samplerate = get_bits_long(&gb, 20);
-    s->channels = get_bits(&gb, 3) + 1;
-    s->bps = get_bits(&gb, 5) + 1;
-
-    avctx->channels = s->channels;
-    avctx->sample_rate = s->samplerate;
-    avctx->bits_per_raw_sample = s->bps;
-
-    s->samples  = get_bits_long(&gb, 32) << 4;
-    s->samples |= get_bits(&gb, 4);
-
-    skip_bits_long(&gb, 64); /* md5 sum */
-    skip_bits_long(&gb, 64); /* md5 sum */
-
-    dump_headers(avctx, s);
-}
-
-void avpriv_flac_parse_block_header(const uint8_t *block_header,
-                                int *last, int *type, int *size)
-{
-    int tmp = bytestream_get_byte(&block_header);
-    if (last)
-        *last = tmp & 0x80;
-    if (type)
-        *type = tmp & 0x7F;
-    if (size)
-        *size = bytestream_get_be24(&block_header);
-}
-
 /**
  * Parse the STREAMINFO from an inline header.
  * @param s the flac decoding context

libavcodec/libfdk-aacenc.c

@@ -0,0 +1,384 @@
+/*
+ * AAC encoder wrapper
+ * 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 <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,
+};

libavcodec/options_table.h

@@ -324,6 +324,10 @@ static const AVOption options[]={
 {"aac_low", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_LOW }, INT_MIN, INT_MAX, A|E, "profile"},
 {"aac_ssr", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_SSR }, INT_MIN, INT_MAX, A|E, "profile"},
 {"aac_ltp", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_LTP }, INT_MIN, INT_MAX, A|E, "profile"},
+{"aac_he", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_HE }, INT_MIN, INT_MAX, A|E, "profile"},
+{"aac_he_v2", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_HE_V2 }, INT_MIN, INT_MAX, A|E, "profile"},
+{"aac_ld", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_LD }, INT_MIN, INT_MAX, A|E, "profile"},
+{"aac_eld", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_AAC_ELD }, INT_MIN, INT_MAX, A|E, "profile"},
 {"dts", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_DTS }, INT_MIN, INT_MAX, A|E, "profile"},
 {"dts_es", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_DTS_ES }, INT_MIN, INT_MAX, A|E, "profile"},
 {"dts_96_24", NULL, 0, AV_OPT_TYPE_CONST, {.dbl = FF_PROFILE_DTS_96_24 }, INT_MIN, INT_MAX, A|E, "profile"},

libavcodec/version.h

@@ -27,7 +27,7 @@
  */
 
 #define LIBAVCODEC_VERSION_MAJOR 54
-#define LIBAVCODEC_VERSION_MINOR  34
+#define LIBAVCODEC_VERSION_MINOR  35
 #define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \

libavformat/Makefile

@@ -132,6 +132,9 @@ OBJS-$(CONFIG_IMAGE2PIPE_DEMUXER)        += img2dec.o img2.o
 OBJS-$(CONFIG_IMAGE2PIPE_MUXER)          += img2enc.o img2.o
 OBJS-$(CONFIG_INGENIENT_DEMUXER)         += ingenientdec.o rawdec.o
 OBJS-$(CONFIG_IPMOVIE_DEMUXER)           += ipmovie.o
+OBJS-$(CONFIG_ISMV_MUXER)                += movenc.o isom.o avc.o       \
+                                            movenchint.o rtpenc_chain.o \
+                                            mov_chan.o
 OBJS-$(CONFIG_ISS_DEMUXER)               += iss.o
 OBJS-$(CONFIG_IV8_DEMUXER)               += iv8.o
 OBJS-$(CONFIG_IVF_DEMUXER)               += ivfdec.o