/* * Interface to libaacplus for aac+ (sbr+ps) encoding * Copyright (c) 2010 tipok <piratfm@gmail.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 libaacplus for aac+ (sbr+ps) encoding. */ #include <aacplus.h> #include "avcodec.h" #include "internal.h" typedef struct aacPlusAudioContext { aacplusEncHandle aacplus_handle; unsigned long max_output_bytes; unsigned long samples_input; } aacPlusAudioContext; static av_cold int aacPlus_encode_init(AVCodecContext *avctx) { aacPlusAudioContext *s = avctx->priv_data; aacplusEncConfiguration *aacplus_cfg; /* number of channels */ if (avctx->channels < 1 || avctx->channels > 2) { av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels); return -1; } s->aacplus_handle = aacplusEncOpen(avctx->sample_rate, avctx->channels, &s->samples_input, &s->max_output_bytes); if(!s->aacplus_handle) { av_log(avctx, AV_LOG_ERROR, "can't open encoder\n"); return -1; } /* check aacplus version */ aacplus_cfg = aacplusEncGetCurrentConfiguration(s->aacplus_handle); /* put the options in the configuration struct */ if(avctx->profile != FF_PROFILE_AAC_LOW && avctx->profile != FF_PROFILE_UNKNOWN) { av_log(avctx, AV_LOG_ERROR, "invalid AAC profile: %d, only LC supported\n", avctx->profile); aacplusEncClose(s->aacplus_handle); return -1; } aacplus_cfg->bitRate = avctx->bit_rate; aacplus_cfg->bandWidth = avctx->cutoff; aacplus_cfg->outputFormat = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER); aacplus_cfg->inputFormat = AACPLUS_INPUT_16BIT; if (!aacplusEncSetConfiguration(s->aacplus_handle, aacplus_cfg)) { av_log(avctx, AV_LOG_ERROR, "libaacplus doesn't support this output format!\n"); return -1; } avctx->frame_size = s->samples_input / avctx->channels; #if FF_API_OLD_ENCODE_AUDIO avctx->coded_frame= avcodec_alloc_frame(); avctx->coded_frame->key_frame= 1; #endif /* Set decoder specific info */ avctx->extradata_size = 0; if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) { unsigned char *buffer = NULL; unsigned long decoder_specific_info_size; if (aacplusEncGetDecoderSpecificInfo(s->aacplus_handle, &buffer, &decoder_specific_info_size) == 1) { avctx->extradata = av_malloc(decoder_specific_info_size + FF_INPUT_BUFFER_PADDING_SIZE); avctx->extradata_size = decoder_specific_info_size; memcpy(avctx->extradata, buffer, avctx->extradata_size); } #undef free free(buffer); #define free please_use_av_free } return 0; } static int aacPlus_encode_frame(AVCodecContext *avctx, AVPacket *pkt, const AVFrame *frame, int *got_packet) { aacPlusAudioContext *s = avctx->priv_data; int32_t *input_buffer = (int32_t *)frame->data[0]; int ret; if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes))) return ret; pkt->size = aacplusEncEncode(s->aacplus_handle, input_buffer, s->samples_input, pkt->data, pkt->size); *got_packet = 1; pkt->pts = frame->pts; return 0; } static av_cold int aacPlus_encode_close(AVCodecContext *avctx) { aacPlusAudioContext *s = avctx->priv_data; #if FF_API_OLD_ENCODE_AUDIO av_freep(&avctx->coded_frame); #endif av_freep(&avctx->extradata); aacplusEncClose(s->aacplus_handle); return 0; } AVCodec ff_libaacplus_encoder = { .name = "libaacplus", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_AAC, .priv_data_size = sizeof(aacPlusAudioContext), .init = aacPlus_encode_init, .encode2 = aacPlus_encode_frame, .close = aacPlus_encode_close, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("libaacplus AAC+ (Advanced Audio Codec with SBR+PS)"), };