iSAC: Move global trig tables into the codec instance

webrtc/modules/audio_coding/codecs/isac/main/source/structs.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 410 matching lines @@
   double       loFiltGain[SUBFRAMES];
   double       hiFiltGain[SUBFRAMES];
   /* Upper boundary of interval W */
   uint32_t W_upper;
   uint32_t streamval;
   /* Index to the current position in bytestream */
   uint32_t stream_index;
   uint8_t  stream[3];
 } transcode_obj;
 
+typedef struct {
+  // TODO(kwiberg): The size of these tables could be reduced by storing floats
+  // instead of doubles, and by making use of the identity cos(x) =
+  // sin(x+pi/2). They could also be made global constants that we fill in at
+  // compile time.
+  double costab1[FRAMESAMPLES_HALF];
+  double sintab1[FRAMESAMPLES_HALF];
+  double costab2[FRAMESAMPLES_QUARTER];
+  double sintab2[FRAMESAMPLES_QUARTER];
+} TransformTables;
 
 typedef struct {
   // lower-band codec instance
   ISACLBStruct              instLB;
   // upper-band codec instance
   ISACUBStruct              instUB;
 
   // Bandwidth Estimator and model for the rate.
   BwEstimatorstr            bwestimator_obj;
   RateModel                 rate_data_obj;
@@ skipping 29 matching lines @@
   // Maximum allowed rate, measured in Bytes per 30 ms.
   int16_t               maxRateBytesPer30Ms;
   // Maximum allowed payload-size, measured in Bytes.
   int16_t               maxPayloadSizeBytes;
   /* The expected sampling rate of the input signal. Valid values are 16000,
    * 32000 and 48000. This is not the operation sampling rate of the codec.
    * Input signals at 48 kHz are resampled to 32 kHz, then encoded. */
   uint16_t in_sample_rate_hz;
   /* State for the input-resampler. It is only used for 48 kHz input signals. */
   int16_t state_in_resampler[SIZE_RESAMPLER_STATE];
+
+  // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
+  TransformTables transform_tables;
 } ISACMainStruct;
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */