ffmpeg roll of source to mar 9 version...

patched-ffmpeg-mt/libavcodec/atrac1.c

 /*
  * Atrac 1 compatible decoder
  * Copyright (c) 2009 Maxim Poliakovski
  * Copyright (c) 2009 Benjamin Larsson
  *
  * 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 17 matching lines @@
 
 /* Many thanks to Tim Craig for all the help! */
 
 #include <math.h>
 #include <stddef.h>
 #include <stdio.h>
 
 #include "avcodec.h"
 #include "get_bits.h"
 #include "dsputil.h"
+#include "fft.h"
 
 #include "atrac.h"
 #include "atrac1data.h"
 
 #define AT1_MAX_BFU      52                 ///< max number of block floating units in a sound unit
 #define AT1_SU_SIZE      212                ///< number of bytes in a sound unit
 #define AT1_SU_SAMPLES   512                ///< number of samples in a sound unit
 #define AT1_FRAME_SIZE   AT1_SU_SIZE * 2
 #define AT1_SU_MAX_BITS  AT1_SU_SIZE * 8
 #define AT1_MAX_CHANNELS 2
 
 #define AT1_QMF_BANDS    3
 #define IDX_LOW_BAND     0
 #define IDX_MID_BAND     1
 #define IDX_HIGH_BAND    2
 
 /**
  * Sound unit struct, one unit is used per channel
  */
 typedef struct {
     int                 log2_block_count[AT1_QMF_BANDS];    ///< log2 number of blocks in a band
     int                 num_bfus;                           ///< number of Block Floating Units
     float*              spectrum[2];
-    DECLARE_ALIGNED_16(float, spec1)[AT1_SU_SAMPLES];     ///< mdct buffer
-    DECLARE_ALIGNED_16(float, spec2)[AT1_SU_SAMPLES];     ///< mdct buffer
-    DECLARE_ALIGNED_16(float, fst_qmf_delay)[46];         ///< delay line for the 1st stacked QMF filter
-    DECLARE_ALIGNED_16(float, snd_qmf_delay)[46];         ///< delay line for the 2nd stacked QMF filter
-    DECLARE_ALIGNED_16(float, last_qmf_delay)[256+23];    ///< delay line for the last stacked QMF filter
+    DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES];     ///< mdct buffer
+    DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES];     ///< mdct buffer
+    DECLARE_ALIGNED(16, float, fst_qmf_delay)[46];         ///< delay line for the 1st stacked QMF filter
+    DECLARE_ALIGNED(16, float, snd_qmf_delay)[46];         ///< delay line for the 2nd stacked QMF filter
+    DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23];    ///< delay line for the last stacked QMF filter
 } AT1SUCtx;
 
 /**
  * The atrac1 context, holds all needed parameters for decoding
  */
 typedef struct {
     AT1SUCtx            SUs[AT1_MAX_CHANNELS];              ///< channel sound unit
-    DECLARE_ALIGNED_16(float, spec)[AT1_SU_SAMPLES];      ///< the mdct spectrum buffer
+    DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES];      ///< the mdct spectrum buffer
 
-    DECLARE_ALIGNED_16(float,  low)[256];
-    DECLARE_ALIGNED_16(float,  mid)[256];
-    DECLARE_ALIGNED_16(float, high)[512];
+    DECLARE_ALIGNED(16, float,  low)[256];
+    DECLARE_ALIGNED(16, float,  mid)[256];
+    DECLARE_ALIGNED(16, float, high)[512];
     float*              bands[3];
-    DECLARE_ALIGNED_16(float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
+    DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
     FFTContext          mdct_ctx[3];
     int                 channels;
     DSPContext          dsp;
 } AT1Ctx;
 
 /** size of the transform in samples in the long mode for each QMF band */
 static const uint16_t samples_per_band[3] = {128, 128, 256};
 static const uint8_t   mdct_long_nbits[3] = {7, 7, 8};
 
 
@@ skipping 154 matching lines @@
             } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
                 memset(&spec[pos], 0, num_specs * sizeof(float));
             }
         }
     }
 
     return 0;
 }
 
 
-void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
+static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
 {
     float temp[256];
     float iqmf_temp[512 + 46];
 
     /* combine low and middle bands */
     atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
 
     /* delay the signal of the high band by 23 samples */
     memcpy( su->last_qmf_delay,    &su->last_qmf_delay[256], sizeof(float) *  23);
     memcpy(&su->last_qmf_delay[23], q->bands[2],             sizeof(float) * 256);
@@ skipping 107 matching lines @@
 AVCodec atrac1_decoder = {
     .name = "atrac1",
     .type = CODEC_TYPE_AUDIO,
     .id = CODEC_ID_ATRAC1,
     .priv_data_size = sizeof(AT1Ctx),
     .init = atrac1_decode_init,
     .close = atrac1_decode_end,
     .decode = atrac1_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
 };