diff --git a/Changelog b/Changelog index 2fb5e3d09b..c56740c856 100644 --- a/Changelog +++ b/Changelog @@ -33,6 +33,7 @@ version : - Microsoft ATC Screen decoder - RTSP listen mode - TechSmith Screen Codec 2 decoder +- AAC encoding via libfdk-aac version 0.8: diff --git a/configure b/configure index 4d83d4bd9f..397be73907 100755 --- a/configure +++ b/configure @@ -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 ${libmp3lame-no}" diff --git a/doc/general.texi b/doc/general.texi index 7e9cfaf550..fcac1143f8 100644 --- a/doc/general.texi +++ b/doc/general.texi @@ -18,8 +18,8 @@ explicitly requested by passing the appropriate flags to @section OpenCORE and VisualOn libraries -Spun off Google Android sources, OpenCore and VisualOn libraries provide -encoders for a number of audio codecs. +Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer +libraries provide encoders for a number of audio codecs. @float NOTE OpenCORE and VisualOn libraries are under the Apache License 2.0 @@ -55,6 +55,14 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for installing the library. Then pass @code{--enable-libvo-amrwbenc} to configure to enable it. +@subsection Fraunhofer AAC library + +Libav can make use of the Fraunhofer AAC library for AAC encoding. + +Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the +instructions for installing the library. +Then pass @code{--enable-libfdk-aac} to configure to enable it. + @section LAME Libav can make use of the LAME library for MP3 encoding. diff --git a/libavcodec/Makefile b/libavcodec/Makefile index ac97d34058..8d38ca24b7 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -595,6 +595,7 @@ OBJS-$(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 diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index 068f191b0b..bd48728ace 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -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); diff --git a/libavcodec/libfdk-aacenc.c b/libavcodec/libfdk-aacenc.c new file mode 100644 index 0000000000..6fda53ca60 --- /dev/null +++ b/libavcodec/libfdk-aacenc.c @@ -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 + +#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, +}; diff --git a/libavcodec/version.h b/libavcodec/version.h index 6f47df96cf..48db12e50a 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -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, \