ffmpeg roll of source to mar 9 version...

patched-ffmpeg-mt/libavcodec/aac.h

 /*
  * AAC definitions and structures
  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov 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
@@ skipping 14 matching lines @@
  * AAC definitions and structures
  * @author Oded Shimon  ( ods15 ods15 dyndns org )
  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
  */
 
 #ifndef AVCODEC_AAC_H
 #define AVCODEC_AAC_H
 
 #include "avcodec.h"
 #include "dsputil.h"
+#include "fft.h"
 #include "mpeg4audio.h"
+#include "sbr.h"
 
 #include <stdint.h>
 
 #define AAC_INIT_VLC_STATIC(num, size) \
     INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
          ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
         ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
         size);
 
 #define MAX_CHANNELS 64
@@ skipping 161 matching lines @@
  */
 typedef struct {
     IndividualChannelStream ics;
     TemporalNoiseShaping tns;
     Pulse pulse;
     enum BandType band_type[128];             ///< band types
     int band_type_run_end[120];               ///< band type run end points
     float sf[120];                            ///< scalefactors
     int sf_idx[128];                          ///< scalefactor indices (used by encoder)
     uint8_t zeroes[128];                      ///< band is not coded (used by encoder)
-    DECLARE_ALIGNED_16(float, coeffs)[1024];  ///< coefficients for IMDCT
-    DECLARE_ALIGNED_16(float, saved)[1024];   ///< overlap
-    DECLARE_ALIGNED_16(float, ret)[1024];     ///< PCM output
+    DECLARE_ALIGNED(16, float, coeffs)[1024]; ///< coefficients for IMDCT
+    DECLARE_ALIGNED(16, float, saved)[1024];  ///< overlap
+    DECLARE_ALIGNED(16, float, ret)[2048];    ///< PCM output
     PredictorState predictor_state[MAX_PREDICTORS];
 } SingleChannelElement;
 
 /**
  * channel element - generic struct for SCE/CPE/CCE/LFE
  */
 typedef struct {
     // CPE specific
     int common_window;        ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
     int     ms_mode;          ///< Signals mid/side stereo flags coding mode (used by encoder)
     uint8_t ms_mask[128];     ///< Set if mid/side stereo is used for each scalefactor window band
     // shared
     SingleChannelElement ch[2];
     // CCE specific
     ChannelCoupling coup;
+    SpectralBandReplication sbr;
 } ChannelElement;
 
 /**
  * main AAC context
  */
 typedef struct {
     AVCodecContext * avccontext;
 
     MPEG4AudioConfig m4ac;
 
     int is_saved;                 ///< Set if elements have stored overlap from previous frame.
     DynamicRangeControl che_drc;
 
     /**
      * @defgroup elements Channel element related data.
      * @{
      */
     enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
                                                    *   first index as the first 4 raw data block types
                                                    */
     ChannelElement * che[4][MAX_ELEM_ID];
     ChannelElement * tag_che_map[4][MAX_ELEM_ID];
     int tags_mapped;
     /** @} */
 
     /**
      * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
      * @{
      */
-    DECLARE_ALIGNED_16(float, buf_mdct)[1024];
+    DECLARE_ALIGNED(16, float, buf_mdct)[1024];
     /** @} */
 
     /**
      * @defgroup tables   Computed / set up during initialization.
      * @{
      */
     FFTContext mdct;
     FFTContext mdct_small;
     DSPContext dsp;
     int random_state;
     /** @} */
 
     /**
      * @defgroup output   Members used for output interleaving.
      * @{
      */
     float *output_data[MAX_CHANNELS];                 ///< Points to each element's 'ret' buffer (PCM output).
     float add_bias;                                   ///< offset for dsp.float_to_int16
     float sf_scale;                                   ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
     int sf_offset;                                    ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
     /** @} */
 
     DECLARE_ALIGNED(16, float, temp)[128];
 
     enum OCStatus output_configured;
 } AACContext;
 
 #endif /* AVCODEC_AAC_H */