lavc: remove libfaac wrapper

There is really no need for two aac wrappers, we already have
libfdk-aac which is better. Not to mention that faac doesn't
even support HEv1, or HEv2. It's also under a license which is
unusable for distribution, so it would only be useful to people
who will compile their own ffmpeg, only use it themselves (which
at that point should just use fdk-aac).

Signed-off-by: Josh de Kock <josh@itanimul.li>

11 files changed, 6 insertions, 368 deletions

diff --git a/Changelog b/Changelog
index d7f5dc9fed..2c2f32dba6 100644
--- a/Changelog
+++ b/Changelog
@@ -36,6 +36,7 @@ version <next>:
 - sdl2 support for ffplay
 - sdl1 output device and sdl1 support removed
 - extended mov edit list support
+- libfaac encoder removed
 
 
 version 3.1:
diff --git a/LICENSE.md b/LICENSE.md
index d08c747ae5..a384fa0bef 100644
--- a/LICENSE.md
+++ b/LICENSE.md
@@ -115,8 +115,6 @@ The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
 incompatible with the GPLv2 and v3. To the best of our knowledge, they are
 compatible with the LGPL.
 
-The FAAC library is incompatible with all versions of GPL and LGPL.
-
 The NVENC library, while its header file is licensed under the compatible MIT
 license, requires a proprietary binary blob at run time, and is deemed to be
 incompatible with the GPL. We are not certain if it is compatible with the
diff --git a/configure b/configure
index 52162603b0..df6ffa2d18 100755
--- a/configure
+++ b/configure
@@ -225,7 +225,6 @@ External library support:
                            and libraw1394 [no]
   --enable-libebur128      enable libebur128 for EBU R128 measurement,
                            needed for loudnorm filter [no]
-  --enable-libfaac         enable AAC encoding via libfaac [no]
   --enable-libfdk-aac      enable AAC de/encoding via libfdk-aac [no]
   --enable-libflite        enable flite (voice synthesis) support via libflite [no]
   --enable-libfontconfig   enable libfontconfig, useful for drawtext filter [no]
@@ -1488,7 +1487,6 @@ EXTERNAL_LIBRARY_LIST="
     libcelt
     libdc1394
     libebur128
-    libfaac
     libfdk_aac
     libflite
     libfontconfig
@@ -2773,8 +2771,6 @@ pcm_mulaw_at_encoder_select="audio_frame_queue"
 chromaprint_muxer_deps="chromaprint"
 h264_videotoolbox_encoder_deps="videotoolbox_encoder pthreads"
 libcelt_decoder_deps="libcelt"
-libfaac_encoder_deps="libfaac"
-libfaac_encoder_select="audio_frame_queue"
 libfdk_aac_decoder_deps="libfdk_aac"
 libfdk_aac_encoder_deps="libfdk_aac"
 libfdk_aac_encoder_select="audio_frame_queue"
@@ -5072,7 +5068,6 @@ die_license_disabled gpl x11grab
 
 die_license_disabled nonfree cuda
 die_license_disabled nonfree cuvid
-die_license_disabled nonfree libfaac
 die_license_disabled nonfree libnpp
 enabled gpl && die_license_disabled_gpl nonfree libfdk_aac
 enabled gpl && die_license_disabled_gpl nonfree openssl
@@ -5682,7 +5677,6 @@ enabled libcelt           && require libcelt celt/celt.h celt_decode -lcelt0 &&
                                die "ERROR: libcelt must be installed and version must be >= 0.11.0."; }
 enabled libcaca           && require_pkg_config caca caca.h caca_create_canvas
 enabled libebur128        && require ebur128 ebur128.h ebur128_relative_threshold -lebur128
-enabled libfaac           && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac
 enabled libfdk_aac        && { use_pkg_config fdk-aac "fdk-aac/aacenc_lib.h" aacEncOpen ||
                                { require libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac &&
                                  warn "using libfdk without pkg-config"; } }
diff --git a/doc/encoders.texi b/doc/encoders.texi
index 73ebd9cd0b..1f4044ef0f 100644
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -612,111 +612,6 @@ and slightly improves compression.
 
 @end table
 
-@anchor{libfaac}
-@section libfaac
-
-libfaac AAC (Advanced Audio Coding) encoder wrapper.
-
-This encoder is of much lower quality and is more unstable than any other AAC
-encoders, so it's highly recommended to instead use other encoders, like
-@ref{aacenc,,the native FFmpeg AAC encoder}.
-
-This encoder also requires the presence of the libfaac headers and library
-during configuration. You need to explicitly configure the build with
-@code{--enable-libfaac --enable-nonfree}.
-
-@subsection Options
-
-The following shared FFmpeg codec options are recognized.
-
-The following options are supported by the libfaac wrapper. The
-@command{faac}-equivalent of the options are listed in parentheses.
-
-@table @option
-@item b (@emph{-b})
-Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
-is not explicitly specified, it is automatically set to a suitable
-value depending on the selected profile. @command{faac} bitrate is
-expressed in kilobits/s.
-
-Note that libfaac does not support CBR (Constant Bit Rate) but only
-ABR (Average Bit Rate).
-
-If VBR mode is enabled this option is ignored.
-
-@item ar (@emph{-R})
-Set audio sampling rate (in Hz).
-
-@item ac (@emph{-c})
-Set the number of audio channels.
-
-@item cutoff (@emph{-C})
-Set cutoff frequency. If not specified (or explicitly set to 0) it
-will use a value automatically computed by the library. Default value
-is 0.
-
-@item profile
-Set audio profile.
-
-The following profiles are recognized:
-@table @samp
-@item aac_main
-Main AAC (Main)
-
-@item aac_low
-Low Complexity AAC (LC)
-
-@item aac_ssr
-Scalable Sample Rate (SSR)
-
-@item aac_ltp
-Long Term Prediction (LTP)
-@end table
-
-If not specified it is set to @samp{aac_low}.
-
-@item flags +qscale
-Set constant quality VBR (Variable Bit Rate) mode.
-
-@item global_quality
-Set quality in VBR mode as an integer number of lambda units.
-
-Only relevant when VBR mode is enabled with @code{flags +qscale}.  The
-value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
-and used to set the quality value used by libfaac. A reasonable range
-for the option value in QP units is [10-500], the higher the value the
-higher the quality.
-
-@item q (@emph{-q})
-Enable VBR mode when set to a non-negative value, and set constant
-quality value as a double floating point value in QP units.
-
-The value sets the quality value used by libfaac. A reasonable range
-for the option value is [10-500], the higher the value the higher the
-quality.
-
-This option is valid only using the @command{ffmpeg} command-line
-tool. For library interface users, use @option{global_quality}.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
-container:
-@example
-ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
-@end example
-
-@item
-Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
-LTP AAC profile:
-@example
-ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
-@end example
-@end itemize
-
 @anchor{libfdk-aac-enc}
 @section libfdk_aac
 
diff --git a/doc/ffserver.conf b/doc/ffserver.conf
index 7a30fb6c3b..e3f99bbfb3 100644
--- a/doc/ffserver.conf
+++ b/doc/ffserver.conf
@@ -317,7 +317,7 @@ StartSendOnKey
 #AVPresetVideo baseline
 #AVOptionVideo flags +global_header
 #
-#AudioCodec libfaac
+#AudioCodec aac
 #AudioBitRate 32
 #AudioChannels 2
 #AudioSampleRate 22050
diff --git a/doc/general.texi b/doc/general.texi
index a0f1122d67..f08c3cd66c 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -887,7 +887,7 @@ following image formats are supported:
 @item 8SVX exponential       @tab     @tab  X
 @item 8SVX fibonacci         @tab     @tab  X
 @item AAC                    @tab EX  @tab  X
-    @tab encoding supported through internal encoder and external libraries libfaac and libfdk-aac
+    @tab encoding supported through internal encoder and external library libfdk-aac
 @item AAC+                   @tab  E  @tab  IX
     @tab encoding supported through external library libfdk-aac
 @item AC-3                   @tab IX  @tab  IX
diff --git a/doc/muxers.texi b/doc/muxers.texi
index 2c937c76aa..9ec2e31a5c 100644
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -1437,9 +1437,9 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_fr
 
 @item
 Convert the @file{in.mkv} to TS segments using the @code{libx264}
-and @code{libfaac} encoders:
+and @code{aac} encoders:
 @example
-ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts
+ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts
 @end example
 
 @item
diff --git a/doc/platform.texi b/doc/platform.texi
index 21b135f55d..4729d41797 100644
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -314,7 +314,7 @@ These library packages are only available from
 @uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
 
 @example
-yasm, libSDL-devel, libfaac-devel, libgsm-devel, libmp3lame-devel,
+yasm, libSDL-devel, libgsm-devel, libmp3lame-devel,
 libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
 @end example
 
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index a0413ff3e0..a1560ba4d8 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -862,7 +862,6 @@ OBJS-$(CONFIG_ILBC_AT_ENCODER)            += audiotoolboxenc.o
 OBJS-$(CONFIG_PCM_ALAW_AT_ENCODER)        += audiotoolboxenc.o
 OBJS-$(CONFIG_PCM_MULAW_AT_ENCODER)       += audiotoolboxenc.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
-OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
 OBJS-$(CONFIG_LIBFDK_AAC_DECODER)         += libfdk-aacdec.o
 OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsmdec.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index b403bf2acd..fed740ab15 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -594,7 +594,6 @@ void avcodec_register_all(void)
     REGISTER_DECODER(QDMC_AT,           qdmc_at);
     REGISTER_DECODER(QDM2_AT,           qdm2_at);
     REGISTER_DECODER(LIBCELT,           libcelt);
-    REGISTER_ENCODER(LIBFAAC,           libfaac);
     REGISTER_ENCDEC (LIBFDK_AAC,        libfdk_aac);
     REGISTER_ENCDEC (LIBGSM,            libgsm);
     REGISTER_ENCDEC (LIBGSM_MS,         libgsm_ms);
diff --git a/libavcodec/libfaac.c b/libavcodec/libfaac.c
deleted file mode 100644
index 98b3ba8183..0000000000
--- a/libavcodec/libfaac.c
+++ /dev/null
@@ -1,248 +0,0 @@
-/*
- * Interface to libfaac for aac encoding
- * Copyright (c) 2002 Gildas Bazin <gbazin@netcourrier.com>
- *
- * 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
- */
-
-/**
- * @file
- * Interface to libfaac for aac encoding.
- */
-
-#include <faac.h>
-
-#include "libavutil/channel_layout.h"
-#include "libavutil/common.h"
-#include "avcodec.h"
-#include "audio_frame_queue.h"
-#include "internal.h"
-
-
-/* libfaac has an encoder delay of 1024 samples */
-#define FAAC_DELAY_SAMPLES 1024
-
-typedef struct FaacAudioContext {
-    faacEncHandle faac_handle;
-    AudioFrameQueue afq;
-} FaacAudioContext;
-
-static av_cold int Faac_encode_close(AVCodecContext *avctx)
-{
-    FaacAudioContext *s = avctx->priv_data;
-
-    av_freep(&avctx->extradata);
-    ff_af_queue_close(&s->afq);
-
-    if (s->faac_handle)
-        faacEncClose(s->faac_handle);
-
-    return 0;
-}
-
-static const int channel_maps[][6] = {
-    { 2, 0, 1 },          //< C L R
-    { 2, 0, 1, 3 },       //< C L R Cs
-    { 2, 0, 1, 3, 4 },    //< C L R Ls Rs
-    { 2, 0, 1, 4, 5, 3 }, //< C L R Ls Rs LFE
-};
-
-static av_cold int Faac_encode_init(AVCodecContext *avctx)
-{
-    FaacAudioContext *s = avctx->priv_data;
-    faacEncConfigurationPtr faac_cfg;
-    unsigned long samples_input, max_bytes_output;
-    int ret;
-
-    /* number of channels */
-    if (avctx->channels < 1 || avctx->channels > 6) {
-        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels);
-        ret = AVERROR(EINVAL);
-        goto error;
-    }
-
-    s->faac_handle = faacEncOpen(avctx->sample_rate,
-                                 avctx->channels,
-                                 &samples_input, &max_bytes_output);
-    if (!s->faac_handle) {
-        av_log(avctx, AV_LOG_ERROR, "error in faacEncOpen()\n");
-        ret = AVERROR_UNKNOWN;
-        goto error;
-    }
-
-    /* check faac version */
-    faac_cfg = faacEncGetCurrentConfiguration(s->faac_handle);
-    if (faac_cfg->version != FAAC_CFG_VERSION) {
-        av_log(avctx, AV_LOG_ERROR, "wrong libfaac version (compiled for: %d, using %d)\n", FAAC_CFG_VERSION, faac_cfg->version);
-        ret = AVERROR(EINVAL);
-        goto error;
-    }
-
-    /* put the options in the configuration struct */
-    switch(avctx->profile) {
-        case FF_PROFILE_AAC_MAIN:
-            faac_cfg->aacObjectType = MAIN;
-            break;
-        case FF_PROFILE_UNKNOWN:
-        case FF_PROFILE_AAC_LOW:
-            faac_cfg->aacObjectType = LOW;
-            break;
-        case FF_PROFILE_AAC_SSR:
-            faac_cfg->aacObjectType = SSR;
-            break;
-        case FF_PROFILE_AAC_LTP:
-            faac_cfg->aacObjectType = LTP;
-            break;
-        default:
-            av_log(avctx, AV_LOG_ERROR, "invalid AAC profile\n");
-            ret = AVERROR(EINVAL);
-            goto error;
-    }
-    faac_cfg->mpegVersion = MPEG4;
-    faac_cfg->useTns = 0;
-    faac_cfg->allowMidside = 1;
-    faac_cfg->bitRate = avctx->bit_rate / avctx->channels;
-    faac_cfg->bandWidth = avctx->cutoff;
-    if(avctx->flags & AV_CODEC_FLAG_QSCALE) {
-        faac_cfg->bitRate = 0;
-        faac_cfg->quantqual = avctx->global_quality / FF_QP2LAMBDA;
-    }
-    faac_cfg->outputFormat = 1;
-    faac_cfg->inputFormat = FAAC_INPUT_16BIT;
-    if (avctx->channels > 2)
-        memcpy(faac_cfg->channel_map, channel_maps[avctx->channels-3],
-               avctx->channels * sizeof(int));
-
-    avctx->frame_size = samples_input / avctx->channels;
-
-    /* Set decoder specific info */
-    avctx->extradata_size = 0;
-    if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) {
-
-        unsigned char *buffer = NULL;
-        unsigned long decoder_specific_info_size;
-
-        if (!faacEncGetDecoderSpecificInfo(s->faac_handle, &buffer,
-                                           &decoder_specific_info_size)) {
-            avctx->extradata = av_malloc(decoder_specific_info_size + AV_INPUT_BUFFER_PADDING_SIZE);
-            if (!avctx->extradata) {
-                ret = AVERROR(ENOMEM);
-                goto error;
-            }
-            avctx->extradata_size = decoder_specific_info_size;
-            memcpy(avctx->extradata, buffer, avctx->extradata_size);
-            faac_cfg->outputFormat = 0;
-        }
-        free(buffer);
-    }
-
-    if (!faacEncSetConfiguration(s->faac_handle, faac_cfg)) {
-        int i;
-        for (i = avctx->bit_rate/1000; i ; i--) {
-            faac_cfg->bitRate = 1000*i / avctx->channels;
-            if (faacEncSetConfiguration(s->faac_handle, faac_cfg))
-                break;
-        }
-        if (!i) {
-            av_log(avctx, AV_LOG_ERROR, "libfaac doesn't support this output format!\n");
-            ret = AVERROR(EINVAL);
-            goto error;
-        } else {
-            avctx->bit_rate = 1000*i;
-            av_log(avctx, AV_LOG_WARNING, "libfaac doesn't support the specified bitrate, using %dkbit/s instead\n", i);
-        }
-    }
-
-    avctx->initial_padding = FAAC_DELAY_SAMPLES;
-    ff_af_queue_init(avctx, &s->afq);
-
-    return 0;
-error:
-    Faac_encode_close(avctx);
-    return ret;
-}
-
-static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                             const AVFrame *frame, int *got_packet_ptr)
-{
-    FaacAudioContext *s = avctx->priv_data;
-    int bytes_written, ret;
-    int num_samples  = frame ? frame->nb_samples : 0;
-    void *samples    = frame ? frame->data[0]    : NULL;
-
-    if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels, 0)) < 0)
-        return ret;
-
-    bytes_written = faacEncEncode(s->faac_handle, samples,
-                                  num_samples * avctx->channels,
-                                  avpkt->data, avpkt->size);
-    if (bytes_written < 0) {
-        av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n");
-        return bytes_written;
-    }
-
-    /* add current frame to the queue */
-    if (frame) {
-        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
-            return ret;
-    }
-
-    if (!bytes_written)
-        return 0;
-
-    /* Get the next frame pts/duration */
-    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
-                       &avpkt->duration);
-
-    avpkt->size = bytes_written;
-    *got_packet_ptr = 1;
-    return 0;
-}
-
-static const AVProfile profiles[] = {
-    { FF_PROFILE_AAC_MAIN, "Main" },
-    { FF_PROFILE_AAC_LOW,  "LC"   },
-    { FF_PROFILE_AAC_SSR,  "SSR"  },
-    { FF_PROFILE_AAC_LTP,  "LTP"  },
-    { FF_PROFILE_UNKNOWN },
-};
-
-static const uint64_t faac_channel_layouts[] = {
-    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_libfaac_encoder = {
-    .name           = "libfaac",
-    .long_name      = NULL_IF_CONFIG_SMALL("libfaac AAC (Advanced Audio Coding)"),
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_AAC,
-    .priv_data_size = sizeof(FaacAudioContext),
-    .init           = Faac_encode_init,
-    .encode2        = Faac_encode_frame,
-    .close          = Faac_encode_close,
-    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
-                                                     AV_SAMPLE_FMT_NONE },
-    .profiles       = NULL_IF_CONFIG_SMALL(profiles),
-    .channel_layouts = faac_channel_layouts,
-};