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/*
* 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)"),
};
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