Remove VoENetwork

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.
  */
 
 #ifndef VOICE_ENGINE_CHANNEL_H_
 #define VOICE_ENGINE_CHANNEL_H_
 
 #include <memory>
 
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/call/audio_sink.h"
+#include "api/call/transport.h"
 #include "api/optional.h"
 #include "common_audio/resampler/include/push_resampler.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_coding/acm2/codec_manager.h"
 #include "modules/audio_coding/acm2/rent_a_codec.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "modules/audio_processing/rms_level.h"
 #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "modules/rtp_rtcp/include/rtp_receiver.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/event.h"
 #include "rtc_base/thread_checker.h"
 #include "voice_engine/audio_level.h"
 #include "voice_engine/include/voe_base.h"
-#include "voice_engine/include/voe_network.h"
 #include "voice_engine/shared_data.h"
 #include "voice_engine/voice_engine_defines.h"
 
 namespace rtc {
 class TimestampWrapAroundHandler;
 }
 
 namespace webrtc {
 
 class AudioDeviceModule;
@@ skipping 150 matching lines @@
   // Codecs
   int32_t GetSendCodec(CodecInst& codec);
   int32_t GetRecCodec(CodecInst& codec);
   int32_t SetSendCodec(const CodecInst& codec);
   void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
   bool EnableAudioNetworkAdaptor(const std::string& config_string);
   void DisableAudioNetworkAdaptor();
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                    int max_frame_length_ms);
 
-  // VoENetwork
+  // Network
   int32_t RegisterExternalTransport(Transport* transport);
   int32_t DeRegisterExternalTransport();
-  int32_t ReceivedRTPPacket(const uint8_t* received_packet,
-                            size_t length,
-                            const PacketTime& packet_time);
   // TODO(nisse, solenberg): Delete when VoENetwork is deleted.

[kwiberg-webrtc, 2017/09/21 10:33:13]

Delete the stuff, or change the text of the TODO?

[the sun, 2017/09/21 10:35:47]

It seems to require a little more work than just deleting, so I'm leaving that
for nisse@, or a separate CL.

   int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
   void OnRtpPacket(const RtpPacketReceived& packet);
 
   // Muting, Volume and Level.
   void SetInputMute(bool enable);
   void SetChannelOutputVolumeScaling(float scaling);
   int GetSpeechOutputLevel() const;
   int GetSpeechOutputLevelFullRange() const;
   // See description of "totalAudioEnergy" in the WebRTC stats spec:
   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
@@ skipping 131 matching lines @@
   std::vector<RtpSource> GetSources() const {
     return rtp_receiver_->GetSources();
   }
 
  private:
   class ProcessAndEncodeAudioTask;
 
   int GetRemoteSSRC(unsigned int& ssrc);
   void OnUplinkPacketLossRate(float packet_loss_rate);
   bool InputMute() const;
-  bool OnRtpPacketWithHeader(const uint8_t* received_packet,
-                             size_t length,
-                             RTPHeader *header);
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length);
 
   bool ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
   bool IsPacketInOrder(const RTPHeader& header) const;
   bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
   int ResendPackets(const uint16_t* sequence_numbers, int length);
   void UpdatePlayoutTimestamp(bool rtcp);
@@ skipping 107 matching lines @@
 
   bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
 
   rtc::TaskQueue* encoder_queue_ = nullptr;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // VOICE_ENGINE_CHANNEL_H_