Fraunhofer AAC codec.

License boilerplate update to follow.

Change-Id: I2810460c11a58b6d148d84673cc031f3685e79b5

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+/****************************************************************************
+
+                     (C) Copyright Fraunhofer IIS (2004)
+                               All Rights Reserved
+
+    Please be advised that this software and/or program delivery is
+    Confidential Information of Fraunhofer and subject to and covered by the
+
+    Fraunhofer IIS Software Evaluation Agreement
+    between Google Inc. and  Fraunhofer
+    effective and in full force since March 1, 2012.
+
+    You may use this software and/or program only under the terms and
+    conditions described in the above mentioned Fraunhofer IIS Software
+    Evaluation Agreement. Any other and/or further use requires a separate agreement.
+
+
+   This software and/or program is protected by copyright law and international
+   treaties. Any reproduction or distribution of this software and/or program,
+   or any portion of it, may result in severe civil and criminal penalties, and
+   will be prosecuted to the maximum extent possible under law.
+
+ $Id$
+
+*******************************************************************************/
+#ifndef __RESAMPLER_H
+#define __RESAMPLER_H
+/*!
+  \file
+  \brief  Fixed Point Resampler Tool Box $Revision: 11752 $
+*/
+
+/*!
+  \mainpage Fixed Point Resampler Library Documentation
+
+  Information in this SDK is subject to change without notice. Companies,
+  names, and data used in examples herein are fictitious unless otherwise
+  noted.
+
+  Product and corporate names may be trademarks or registered trademarks
+  of other companies. They are used for explanation only, with no intent
+  to infringe.
+
+  No part of this publication may be reproduced or utilized in any form or
+  by any means, electronic or mechanical, including photocopying and
+  microfilm, without permission in writing from the publisher.
+*/
+
+#include "common_fix.h"
+
+
+/**************************************************************************/
+/*                         BIQUAD Filter Structure                           */
+/**************************************************************************/
+
+#define MAXNR_SECTIONS  (15)
+
+#ifdef RS_BIQUAD_STATES16
+typedef FIXP_SGL FIXP_BQS;
+#else
+typedef FIXP_DBL FIXP_BQS;
+#endif
+
+typedef struct
+{
+  FIXP_BQS states[MAXNR_SECTIONS+1][2];   /*! state buffer */
+  const FIXP_SGL *coeffa;               /*! pointer to filter coeffs */
+  FIXP_DBL gain;                        /*! overall gain factor */
+  int Wc;                               /*! normalized cutoff freq * 1000 */
+  int noCoeffs;                         /*! number of filter coeffs sets */
+  int ptr;                              /*! index to rinbuffers */
+} LP_FILTER;
+
+
+/**************************************************************************/
+/*                        Downsampler Structure                           */
+/**************************************************************************/
+
+typedef struct
+{
+  LP_FILTER downFilter;           /*! filter instance */
+  int ratio;                      /*! downsampling ration */
+  int delay;                      /*! downsampling delay (source fs)   */
+  int pending;                    /*! number of pending output samples */
+} DOWNSAMPLER;
+
+
+/**
+ * \brief Initialized a given downsampler structure.
+ */
+INT FDKaacEnc_InitDownsampler(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
+                              INT Wc,                   /*!< normalized cutoff freq * 1000 */
+                              INT ratio);               /*!< downsampler ratio */
+
+/**
+ * \brief Downsample a set of audio samples. numInSamples must be at least equal to the
+ *        downsampler ratio.
+ */
+INT FDKaacEnc_Downsample(DOWNSAMPLER *DownSampler,  /*!< pointer to downsampler instance */
+                         INT_PCM *inSamples,        /*!< pointer to input samples */
+                         INT numInSamples,          /*!< number  of input samples  */
+                         INT inStride,              /*!< increment of input samples      */
+                         INT_PCM *outSamples,       /*!< pointer to output samples */
+                         INT *numOutSamples,        /*!< pointer tp number of output samples */
+                         INT outstride);            /*!< increment of output samples */
+
+
+
+#endif /* __RESAMPLER_H */