- 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
+- RealText demuxer and decoder
+- Heart Of Darkness PAF playback support
+- iec61883 device
+- asettb filter
+
+
+version 0.11:
+
+- Fixes: CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777,
+ CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785,
+ CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790,
+ CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795,
+ CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800,
+ CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804,
+- v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder
+- setfield filter
+- CDXL demuxer and decoder
+- Apple ProRes encoder
+- ffprobe -count_packets and -count_frames options
+- Sun Rasterfile Encoder
+- ID3v2 attached pictures reading and writing
+- WMA Lossless decoder
+- bluray protocol
+- blackdetect filter
+- libutvideo encoder wrapper (--enable-libutvideo)
+- swapuv filter
+- bbox filter
+- XBM encoder and decoder
+- RealAudio Lossless decoder
+- ZeroCodec decoder
+- tile video filter
+- Metal Gear Solid: The Twin Snakes demuxer
+- OpenEXR image decoder
+- removelogo filter
+- drop support for ffmpeg without libavfilter
+- drawtext video filter: fontconfig support
+- ffmpeg -benchmark_all option
+- super2xsai filter ported from libmpcodecs
+- add libavresample audio conversion library for compatibility
+- MicroDVD decoder
+- Avid Meridien (AVUI) encoder and decoder
+- accept + prefix to -pix_fmt option to disable automatic conversions.
+- complete audio filtering in libavfilter and ffmpeg
+- add fps filter
+- vorbis parser
+- png parser
+- audio mix filter
-version 0.8:
+version 0.10:
+- Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936,
+ CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944,
+ CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949,
+ CVE-2011-3950, CVE-2011-3951, CVE-2011-3952
+- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
+- SBaGen (SBG) binaural beats script demuxer
+- OpenMG Audio muxer
+- Timecode extraction in DV and MOV
+- thumbnail video filter
+- XML output in ffprobe
+- asplit audio filter
+- tinterlace video filter
+- astreamsync audio filter
+- amerge audio filter
+- ISMV (Smooth Streaming) muxer
- GSM audio parser
- SMJPEG muxer
-
-
-version 0.8_beta2:
-
+- XWD encoder and decoder
- Automatic thread count based on detection number of (available) CPU cores
-- Deprecate libpostproc. If desired, the switch --enable-postproc will
- enable it but it may be removed in a later Libav release.
+- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
+- ffprobe -show_error option
+- Avid 1:1 10-bit RGB Packer codec
+- v308 Quicktime Uncompressed 4:4:4 encoder and decoder
+- yuv4 libquicktime packed 4:2:0 encoder and decoder
+- ffprobe -show_frames option
+- silencedetect audio filter
+- ffprobe -show_program_version, -show_library_versions, -show_versions options
- rv34: frame-level multi-threading
- optimized iMDCT transform on x86 using SSE for for mpegaudiodec
+- Improved PGS subtitle decoder
+- dumpgraph option to lavfi device
+- r210 and r10k encoders
+- ffwavesynth decoder
+- aviocat tool
+- ffeval tool
-version 0.8_beta1:
+version 0.9:
+- openal input device added
+- boxblur filter added
- BWF muxer
- Flash Screen Video 2 decoder
-- ffplay/ffprobe/ffserver renamed to avplay/avprobe/avserver
-- ffmpeg deprecated, added avconv, which is almost the same for now, except
+- lavfi input device added
+- added avconv, which is almost the same for now, except
for a few incompatible changes in the options, which will hopefully make them
easier to use. The changes are:
* The options placement is now strictly enforced! While in theory the
--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]
--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,
+ native (de)muxer exists [no]
--enable-libopencore-amrnb enable AMR-NB de/encoding via libopencore-amrnb [no]
--enable-libopencore-amrwb enable AMR-WB decoding via libopencore-amrwb [no]
--enable-libopencv enable video filtering via libopencv [no]
gpl
gray
hardcoded_tables
+ libaacplus
+ libass
+ libbluray
libcdio
+ libcelt
libdc1394
libfaac
+ libfdk_aac
libfreetype
libgsm
+ libiec61883
libilbc
+ libmodplug
libmp3lame
+ libnut
libopencore_amrnb
libopencore_amrwb
libopencv
h264_parser_select="golomb h264dsp h264pred"
# external libraries
+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"
# these are off by default, so fail if requested and not available
enabled avisynth && require2 vfw32 "windows.h vfw.h" AVIFileInit -lavifil32
+enabled fontconfig && require_pkg_config fontconfig "fontconfig/fontconfig.h" FcInit
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 libiec61883 && require libiec61883 libiec61883/iec61883.h iec61883_cmp_connect -lraw1394 -lavc1394 -lrom1394 -liec61883
+enabled libaacplus && require "libaacplus >= 2.0.0" aacplus.h aacplusEncOpen -laacplus
+enabled libass && require_pkg_config libass ass/ass.h ass_library_init
+enabled libbluray && require libbluray libbluray/bluray.h bd_open -lbluray
+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
echo "AVISynth enabled ${avisynth-no}"
echo "frei0r enabled ${frei0r-no}"
echo "gnutls enabled ${gnutls-no}"
+echo "libaacplus enabled ${libaacplus-no}"
+echo "libass enabled ${libass-no}"
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}"
+echo "libmodplug enabled ${libmodplug-no}"
echo "libmp3lame enabled ${libmp3lame-no}"
+echo "libnut enabled ${libnut-no}"
echo "libopencore-amrnb support ${libopencore_amrnb-no}"
echo "libopencore-amrwb support ${libopencore_amrwb-no}"
echo "libopencv support ${libopencv-no}"
# libavformat dependencies
OBJS-$(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-$(CONFIG_DV_MUXER) += dv_profile.o
-OBJS-$(CONFIG_FLAC_DEMUXER) += flac.o flacdata.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-$(CONFIG_FLAC_MUXER) += flac.o flacdata.o
++OBJS-$(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 \
+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 xiph.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-$(CONFIG_MOV_DEMUXER) += mpeg4audio.o mpegaudiodata.o ac3tab.o
+OBJS-$(CONFIG_MOV_DEMUXER) += mpeg4audio.o mpegaudiodata.o ac3tab.o timecode.o
OBJS-$(CONFIG_MOV_MUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(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
-OBJS-$(CONFIG_OGG_MUXER) += xiph.o flac.o flacdata.o
++ dirac.o vorbis_data.o
++OBJS-$(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
++ 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
--- /dev/null
- * This file is part of Libav.
+ /*
+ * AAC encoder wrapper
+ * Copyright (c) 2012 Martin Storsjo
+ *
- * Libav is free software; you can redistribute it and/or
++ * This file is part of FFmpeg.
+ *
- * Libav is distributed in the hope that it will be useful,
++ * 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.
+ *
- * License along with Libav; if not, write to the Free Software
++ * 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,
+ };
projects / ffmpeg.git / commitdiff