xsubdec: Convert to the new bitstream reader

[ffmpeg] / libavcodec / aacenc.h

diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
index 86c68d3e167cb267a6713f08c059adc09d574180..dec445ce346dca9298d6b8333493c61cf59aa723 100644 (file)
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -2,34 +2,38 @@
  * 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 {
@@ -48,14 +52,17 @@ extern AACCoefficientsEncoder ff_aac_coders[];
  * 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;
-    DECLARE_ALIGNED(16, FFTSample, output)[2048]; ///< temporary buffer for MDCT input coefficients
-    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;
@@ -64,8 +71,15 @@ typedef struct AACEncContext {
     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 */