Hooking up audio network adaptor to VoE.

webrtc/voice_engine/channel.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 14 matching lines @@
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/level_indicator.h"
-#include "webrtc/voice_engine/network_predictor.h"
 #include "webrtc/voice_engine/shared_data.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace rtc {
 class TimestampWrapAroundHandler;
 }
 
 namespace webrtc {
 
 class AudioDeviceModule;
@@ skipping 155 matching lines @@
   int32_t SetSendCodec(const CodecInst& codec);
   void SetBitRate(int bitrate_bps);
   int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX);
   int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX);
   int32_t SetRecPayloadType(const CodecInst& codec);
   int32_t GetRecPayloadType(CodecInst& codec);
   int32_t SetSendCNPayloadType(int type, PayloadFrequencies frequency);
   int SetOpusMaxPlaybackRate(int frequency_hz);
   int SetOpusDtx(bool enable_dtx);
   int GetOpusDtx(bool* enabled);
+  bool EnableAudioNetworkAdaptor(const std::string& config_string);
+  void DisableAudioNetworkAdaptor();
+  void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                   int max_frame_length_ms);
 
   // VoENetwork
   int32_t RegisterExternalTransport(Transport* transport);
   int32_t DeRegisterExternalTransport();
   int32_t ReceivedRTPPacket(const uint8_t* received_packet,
                             size_t length,
                             const PacketTime& packet_time);
   int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
 
   // VoEFile
@@ skipping 305 matching lines @@
   int8_t _lastPayloadType;
   bool _includeAudioLevelIndication;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
   // VoEVideoSync
   rtc::CriticalSection video_sync_lock_;
   // VoEAudioProcessing
   bool restored_packet_in_use_;
   // RtcpBandwidthObserver
   std::unique_ptr<VoERtcpObserver> rtcp_observer_;
-  std::unique_ptr<NetworkPredictor> network_predictor_;
   // An associated send channel.
   rtc::CriticalSection assoc_send_channel_lock_;
   ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 
   bool pacing_enabled_;
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_