mirror of https://github.com/FFmpeg/FFmpeg.git
TODO: bump minor It's inferior in quality to fdk-aac and has an arguably more problematic license. As early as 2012, a HydrogenAudio user reported: > It has however one huge advantage: much better quality at low bitrates than > faac and libaacplus. (https://hydrogenaud.io/index.php?PHPSESSID=ckiq394pdglka0kj2fin6ij8t7&topic=95989.msg804633#msg804633) I myself have made a few spectrograms for a comparison of the two encoders as well. The FDK output is consistently better than the libaacplus one, in all bitrates I tested. libaacplus license is 3GPP + LGPLv2. 3GPP copyright notice is completely proprietory, as follows: > No part may be reproduced except as authorized by written permission. > > The copyright and the foregoing restriction extend to reproduction in > all media. > > © 2008, 3GPP Organizational Partners (ARIB, ATIS, CCSA, ETSI, TTA, TTC). > > All rights reserved. (The latest 26410-d00 zip from 3GPP has the same notice, but the copyright year is changed to 2015) The copyright part of the FDK AAC license (section 2) is a copyleft license that permits redistribution under certain conditions (and therefore the LGPL + libfdk-aac combination is not prohibited by configure): > Redistribution and use in source and binary forms, with or without > modification, are permitted without payment of copyright license fees > provided that you satisfy the following conditions: > > You must retain the complete text of this software license in > redistributions of the FDK AAC Codec or your modifications thereto in > source code form. > > You must retain the complete text of this software license in the > documentation and/or other materials provided with redistributions of > the FDK AAC Codec or your modifications thereto in binary form. > > You must make available free of charge copies of the complete source > code of the FDK AAC Codec and your modifications thereto to recipients > of copies in binary form. > > The name of Fraunhofer may not be used to endorse or promote products > derived from this library without prior written permission. > > You may not charge copyright license fees for anyone to use, copy or > distribute the FDK AAC Codec software or your modifications thereto. > > Your modified versions of the FDK AAC Codec must carry prominent > notices stating that you changed the software and the date of any > change. For modified versions of the FDK AAC Codec, the term > "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the > term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec > Library for Android."pull/172/merge
Timothy Gu
9 years ago
parent
5893e87537
commit
9ba54c1b82
8 changed files with 6 additions and 162 deletions
@ -1,150 +0,0 @@ |
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/*
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* Interface to libaacplus for aac+ (sbr+ps) encoding |
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* Copyright (c) 2010 tipok <piratfm@gmail.com> |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/**
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* @file |
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* Interface to libaacplus for aac+ (sbr+ps) encoding. |
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*/ |
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#include <aacplus.h> |
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#include "avcodec.h" |
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#include "internal.h" |
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typedef struct aacPlusAudioContext { |
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aacplusEncHandle aacplus_handle; |
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unsigned long max_output_bytes; |
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unsigned long samples_input; |
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} aacPlusAudioContext; |
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static int aacPlus_encode_close(AVCodecContext *avctx); |
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static av_cold int aacPlus_encode_init(AVCodecContext *avctx) |
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{ |
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aacPlusAudioContext *s = avctx->priv_data; |
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aacplusEncConfiguration *aacplus_cfg; |
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/* number of channels */ |
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if (avctx->channels < 1 || avctx->channels > 2) { |
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av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels); |
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return AVERROR(EINVAL); |
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} |
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if (avctx->profile != FF_PROFILE_AAC_LOW && avctx->profile != FF_PROFILE_UNKNOWN) { |
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av_log(avctx, AV_LOG_ERROR, "invalid AAC profile: %d, only LC supported\n", avctx->profile); |
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return AVERROR(EINVAL); |
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} |
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s->aacplus_handle = aacplusEncOpen(avctx->sample_rate, avctx->channels, |
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&s->samples_input, &s->max_output_bytes); |
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if (!s->aacplus_handle) { |
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av_log(avctx, AV_LOG_ERROR, "can't open encoder\n"); |
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return AVERROR(EINVAL); |
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} |
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/* check aacplus version */ |
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aacplus_cfg = aacplusEncGetCurrentConfiguration(s->aacplus_handle); |
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aacplus_cfg->bitRate = avctx->bit_rate; |
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aacplus_cfg->bandWidth = avctx->cutoff; |
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aacplus_cfg->outputFormat = !(avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER); |
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aacplus_cfg->inputFormat = avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? AACPLUS_INPUT_FLOAT : AACPLUS_INPUT_16BIT; |
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if (!aacplusEncSetConfiguration(s->aacplus_handle, aacplus_cfg)) { |
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av_log(avctx, AV_LOG_ERROR, "libaacplus doesn't support this output format!\n"); |
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aacPlus_encode_close(avctx); |
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return AVERROR(EINVAL); |
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} |
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avctx->frame_size = s->samples_input / avctx->channels; |
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/* Set decoder specific info */ |
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avctx->extradata_size = 0; |
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if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) { |
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unsigned char *buffer = NULL; |
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unsigned long decoder_specific_info_size; |
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if (aacplusEncGetDecoderSpecificInfo(s->aacplus_handle, &buffer, |
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&decoder_specific_info_size) == 1) { |
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avctx->extradata = av_malloc(decoder_specific_info_size + AV_INPUT_BUFFER_PADDING_SIZE); |
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if (!avctx->extradata) { |
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free(buffer); |
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aacPlus_encode_close(avctx); |
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return AVERROR(ENOMEM); |
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} |
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avctx->extradata_size = decoder_specific_info_size; |
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memcpy(avctx->extradata, buffer, avctx->extradata_size); |
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} |
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free(buffer); |
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} |
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return 0; |
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} |
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static int aacPlus_encode_frame(AVCodecContext *avctx, AVPacket *pkt, |
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const AVFrame *frame, int *got_packet) |
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{ |
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aacPlusAudioContext *s = avctx->priv_data; |
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int32_t *input_buffer = (int32_t *)frame->data[0]; |
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int ret; |
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if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes, 0)) < 0) |
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return ret; |
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pkt->size = aacplusEncEncode(s->aacplus_handle, input_buffer, |
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s->samples_input, pkt->data, pkt->size); |
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*got_packet = 1; |
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pkt->pts = frame->pts; |
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return 0; |
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} |
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static av_cold int aacPlus_encode_close(AVCodecContext *avctx) |
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{ |
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aacPlusAudioContext *s = avctx->priv_data; |
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av_freep(&avctx->extradata); |
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aacplusEncClose(s->aacplus_handle); |
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s->aacplus_handle = NULL; |
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return 0; |
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} |
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static const AVProfile profiles[] = { |
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{ FF_PROFILE_AAC_LOW, "LC" }, |
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{ FF_PROFILE_UNKNOWN }, |
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}; |
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AVCodec ff_libaacplus_encoder = { |
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.name = "libaacplus", |
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.long_name = NULL_IF_CONFIG_SMALL("libaacplus AAC+ (Advanced Audio Codec with SBR+PS)"), |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_AAC, |
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.priv_data_size = sizeof(aacPlusAudioContext), |
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.init = aacPlus_encode_init, |
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.encode2 = aacPlus_encode_frame, |
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.close = aacPlus_encode_close, |
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, |
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AV_SAMPLE_FMT_FLT, |
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AV_SAMPLE_FMT_NONE }, |
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.profiles = profiles, |
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.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, |
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AV_CH_LAYOUT_STEREO, |
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0 }, |
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}; |
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