Hooking up audio network adaptor to VoE.

webrtc/modules/audio_coding/include/audio_coding_module.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 234 matching lines @@
   //
   // Return value:
   //   positive; sampling frequency [Hz] of the current encoder.
   //   -1 if an error has happened.
   //
   virtual int32_t SendFrequency() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // Sets the bitrate to the specified value in bits/sec. If the value is not
   // supported by the codec, it will choose another appropriate value.
+  //
+  // This is only used in audio_coding_module_unittest_oldapi.cc.
+  // TODO(minyue): Remove it when possible.
   virtual void SetBitRate(int bitrate_bps) = 0;
 
   // int32_t RegisterTransportCallback()
   // Register a transport callback which will be called to deliver
   // the encoded buffers whenever Process() is called and a
   // bit-stream is ready.
   //
   // Input:
   //   -transport          : pointer to the callback class
   //                         transport->SendData() is called whenever
@@ skipping 99 matching lines @@
   // loss gnostic encoding to make stream less sensitive to packet losses,
   // through e.g., FEC. No effects on codecs that do not provide such encoding.
   //
   // Input:
   //   -packet_loss_rate   : expected packet loss rate (0 -- 100 inclusive).
   //
   // Return value
   //   -1 if failed to set packet loss rate,
   //   0 if succeeded.
   //
+  // This is only used in audio_coding_module_unittest_oldapi.cc.
+  // TODO(minyue): Remove it when possible.
   virtual int SetPacketLossRate(int packet_loss_rate) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   //   (VAD) Voice Activity Detection
   //
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t SetVAD()
   // If DTX is enabled & the codec does not have internal DTX/VAD
   // WebRtc VAD will be automatically enabled and |enable_vad| is ignored.
@@ skipping 410 matching lines @@
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_