ffmpeg/libavcodec/libaacplus.c

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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 "avcodec.h"
#include <aacplus.h>
typedef struct aacPlusAudioContext {
aacplusEncHandle aacplus_handle;
} aacPlusAudioContext;
static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
{
aacPlusAudioContext *s = avctx->priv_data;
aacplusEncConfiguration *aacplus_cfg;
unsigned long samples_input, max_bytes_output;
/* 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,
&samples_input, &max_bytes_output);
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 = samples_input / avctx->channels;
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
/* 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,
unsigned char *frame, int buf_size, void *data)
{
aacPlusAudioContext *s = avctx->priv_data;
int bytes_written;
bytes_written = aacplusEncEncode(s->aacplus_handle,
data,
avctx->frame_size * avctx->channels,
frame,
buf_size);
return bytes_written;
}
static av_cold int aacPlus_encode_close(AVCodecContext *avctx)
{
aacPlusAudioContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame);
av_freep(&avctx->extradata);
aacplusEncClose(s->aacplus_handle);
return 0;
}
AVCodec ff_libaacplus_encoder = {
.name = "libaacplus",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(aacPlusAudioContext),
.init = aacPlus_encode_init,
.encode = aacPlus_encode_frame,
.close = aacPlus_encode_close,
Merge remote-tracking branch 'qatar/master' * qatar/master: (71 commits) movenc: Allow writing to a non-seekable output if using empty moov movenc: Support adding isml (smooth streaming live) metadata libavcodec: Don't crash in avcodec_encode_audio if time_base isn't set sunrast: Document the different Sun Raster file format types. sunrast: Add a check for experimental type. libspeexenc: use AVSampleFormat instead of deprecated/removed SampleFormat lavf: remove disabled FF_API_SET_PTS_INFO cruft lavf: remove disabled FF_API_OLD_INTERRUPT_CB cruft lavf: remove disabled FF_API_REORDER_PRIVATE cruft lavf: remove disabled FF_API_SEEK_PUBLIC cruft lavf: remove disabled FF_API_STREAM_COPY cruft lavf: remove disabled FF_API_PRELOAD cruft lavf: remove disabled FF_API_NEW_STREAM cruft lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft lavf: remove disabled FF_API_MUXRATE cruft lavf: remove disabled FF_API_FILESIZE cruft lavf: remove disabled FF_API_TIMESTAMP cruft lavf: remove disabled FF_API_LOOP_OUTPUT cruft lavf: remove disabled FF_API_LOOP_INPUT cruft lavf: remove disabled FF_API_AVSTREAM_QUALITY cruft ... Conflicts: doc/APIchanges libavcodec/8bps.c libavcodec/avcodec.h libavcodec/libx264.c libavcodec/mjpegbdec.c libavcodec/options.c libavcodec/sunrast.c libavcodec/utils.c libavcodec/version.h libavcodec/x86/h264_deblock.asm libavdevice/libdc1394.c libavdevice/v4l2.c libavformat/avformat.h libavformat/avio.c libavformat/avio.h libavformat/aviobuf.c libavformat/dv.c libavformat/mov.c libavformat/utils.c libavformat/version.h libavformat/wtv.c libavutil/Makefile libavutil/file.c libswscale/x86/input.asm libswscale/x86/swscale_mmx.c libswscale/x86/swscale_template.c tests/ref/lavf/ffm Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-28 03:23:26 +00:00
.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)"),
};