iSAC: Functions for importing and exporting bandwidth est. info

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.
  */
 
 /*
  * structs.h
  *
  * This header file contains all the structs used in the ISAC codec
  *
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
 
+#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/settings.h"
 #include "webrtc/typedefs.h"
 
 typedef struct Bitstreamstruct {
 
   uint8_t   stream[STREAM_SIZE_MAX];
   uint32_t  W_upper;
   uint32_t  streamval;
   uint32_t  stream_index;
@@ skipping 185 matching lines @@
   int in_wait_period;
 
   int change_to_WB;
 
   uint32_t                 senderTimestamp;
   uint32_t                 receiverTimestamp;
   //enum IsacSamplingRate incomingStreamSampFreq;
   uint16_t                 numConsecLatePkts;
   float                        consecLatency;
   int16_t                  inWaitLatePkts;
+
+  IsacBandwidthInfo external_bw_info;
 } BwEstimatorstr;
 
 
 typedef struct {
 
   /* boolean, flags if previous packet exceeded B.N. */
   int    PrevExceed;
   /* ms */
   int    ExceedAgo;
   /* packets left to send in current burst */
@@ skipping 250 matching lines @@
    * 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_ */