* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AACENC_H
#define AVCODEC_AACENC_H
+#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "aac.h"
-
+#include "audio_frame_queue.h"
#include "psymodel.h"
+typedef struct AACEncOptions {
+ int stereo_mode;
+} AACEncOptions;
+
struct AACEncContext;
typedef struct AACCoefficientsEncoder {
* AAC encoder context
*/
typedef struct AACEncContext {
+ AVClass *av_class;
+ AACEncOptions options; ///< encoding options
PutBitContext pb;
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp;
- int16_t *samples; ///< saved preprocessed input
+ AVFloatDSPContext fdsp;
+ float *planar_samples[6]; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
+ int channels; ///< channel count
+ const uint8_t *chan_map; ///< channel configuration map
ChannelElement *cpe; ///< channel elements
FFPsyContext psy;
int cur_channel;
int last_frame;
float lambda;
+ AudioFrameQueue afq;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
- DECLARE_ALIGNED(16, float, scoefs)[1024]; ///< scaled coefficients
+ DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
+
+ struct {
+ float *samples;
+ } buffer;
} AACEncContext;
+extern float ff_aac_pow34sf_tab[428];
+
#endif /* AVCODEC_AACENC_H */