2011-09-26 03:28:21 +03:00
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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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2012-03-24 20:41:55 +03:00
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#include "avcodec.h"
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#include "internal.h"
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2011-09-26 03:28:21 +03:00
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typedef struct aacPlusAudioContext {
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aacplusEncHandle aacplus_handle;
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2012-03-24 20:41:55 +03:00
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unsigned long max_output_bytes;
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unsigned long samples_input;
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2011-09-26 03:28:21 +03:00
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} aacPlusAudioContext;
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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 -1;
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}
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2012-03-24 20:41:55 +03:00
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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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2011-09-26 03:28:21 +03:00
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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 -1;
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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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/* put the options in the configuration struct */
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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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aacplusEncClose(s->aacplus_handle);
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return -1;
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}
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aacplus_cfg->bitRate = avctx->bit_rate;
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aacplus_cfg->bandWidth = avctx->cutoff;
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2011-11-24 15:53:19 +03:00
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aacplus_cfg->outputFormat = !(avctx->flags & CODEC_FLAG_GLOBAL_HEADER);
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2011-09-26 03:28:21 +03:00
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aacplus_cfg->inputFormat = 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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return -1;
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}
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2012-03-24 20:41:55 +03:00
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avctx->frame_size = s->samples_input / avctx->channels;
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2011-09-26 03:28:21 +03:00
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2012-03-24 20:41:55 +03:00
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#if FF_API_OLD_ENCODE_AUDIO
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2011-09-26 03:28:21 +03:00
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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2012-03-24 20:41:55 +03:00
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#endif
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2011-09-26 03:28:21 +03:00
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/* Set decoder specific info */
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avctx->extradata_size = 0;
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if (avctx->flags & 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 + FF_INPUT_BUFFER_PADDING_SIZE);
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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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#undef free
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free(buffer);
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#define free please_use_av_free
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}
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return 0;
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}
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2012-03-24 20:41:55 +03:00
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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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2011-09-26 03:28:21 +03:00
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{
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aacPlusAudioContext *s = avctx->priv_data;
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2012-03-24 20:41:55 +03:00
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int32_t *input_buffer = (int32_t *)frame->data[0];
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int ret;
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2011-09-26 03:28:21 +03:00
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2012-03-24 20:41:55 +03:00
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if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes)))
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return ret;
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2011-09-26 03:28:21 +03:00
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2012-03-24 20:41:55 +03:00
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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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2011-09-26 03:28:21 +03:00
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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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2012-03-24 20:41:55 +03:00
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#if FF_API_OLD_ENCODE_AUDIO
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2011-09-26 03:28:21 +03:00
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av_freep(&avctx->coded_frame);
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2012-03-24 20:41:55 +03:00
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#endif
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2011-09-26 03:28:21 +03:00
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av_freep(&avctx->extradata);
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aacplusEncClose(s->aacplus_handle);
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return 0;
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}
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AVCodec ff_libaacplus_encoder = {
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2011-12-30 22:00:53 +03:00
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.name = "libaacplus",
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.type = AVMEDIA_TYPE_AUDIO,
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2012-08-07 23:45:46 +03:00
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.id = AV_CODEC_ID_AAC,
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2011-12-30 22:00:53 +03:00
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.priv_data_size = sizeof(aacPlusAudioContext),
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.init = aacPlus_encode_init,
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2012-03-24 20:41:55 +03:00
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.encode2 = aacPlus_encode_frame,
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2011-12-30 22:00:53 +03:00
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.close = aacPlus_encode_close,
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2012-06-12 23:41:18 +03:00
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.long_name = NULL_IF_CONFIG_SMALL("libaacplus AAC+ (Advanced Audio Codec with SBR+PS)"),
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2011-09-26 03:28:21 +03:00
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};
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