- Remove use of VoERTP_RTCP::SetLocalSSRC() for receive streams; recreate them instead.

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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 103 matching lines @@
     return experimental_ns_enabled_;
   }
 
  private:
   bool experimental_ns_enabled_;
 };
 
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
-      public webrtc::VoEHardware, public webrtc::VoERTP_RTCP,
+      public webrtc::VoEHardware,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
     Channel() {
       memset(&send_codec, 0, sizeof(send_codec));
     }
     bool playout = false;
     float volume_scale = 1.0f;
     bool vad = false;
     bool codec_fec = false;
     int max_encoding_bandwidth = 0;
     bool opus_dtx = false;
     int cn8_type = 13;
     int cn16_type = 105;
-    uint32_t send_ssrc = 0;
     int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
     int neteq_capacity = -1;
     bool neteq_fast_accelerate = false;
   };
 
   FakeWebRtcVoiceEngine() {
     memset(&agc_config_, 0, sizeof(agc_config_));
   }
   ~FakeWebRtcVoiceEngine() override {
     RTC_CHECK(channels_.empty());
   }
 
   bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
 
   bool IsInited() const { return inited_; }
   int GetLastChannel() const { return last_channel_; }
   int GetNumChannels() const { return static_cast<int>(channels_.size()); }
-  uint32_t GetLocalSSRC(int channel) {
-    return channels_[channel]->send_ssrc;
-  }
   bool GetPlayout(int channel) {
     return channels_[channel]->playout;
   }
   bool GetVAD(int channel) {
     return channels_[channel]->vad;
   }
   bool GetOpusDtx(int channel) {
     return channels_[channel]->opus_dtx;
   }
   bool GetCodecFEC(int channel) {
@@ skipping 255 matching lines @@
   WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
   WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
   bool BuiltInAECIsAvailable() const override { return false; }
   WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
   bool BuiltInAGCIsAvailable() const override { return false; }
   WEBRTC_STUB(EnableBuiltInNS, (bool enable));
   bool BuiltInNSIsAvailable() const override { return false; }
 
-  // webrtc::VoERTP_RTCP
-  WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->send_ssrc = ssrc;
-    return 0;
-  }
-  WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc));
-  WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
-  WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
-      unsigned char id));
-  WEBRTC_STUB(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
-      unsigned char id));
-  WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
-      unsigned char id));
-  WEBRTC_STUB(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable,
-      unsigned char id));
-  WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable));
-  WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled));
-  WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256]));
-  WEBRTC_STUB(GetRTCP_CNAME, (int channel, char cname[256]));
-  WEBRTC_STUB(GetRemoteRTCP_CNAME, (int channel, char* cname));
-  WEBRTC_STUB(GetRemoteRTCPData, (int channel, unsigned int& NTPHigh,
-                                  unsigned int& NTPLow,
-                                  unsigned int& timestamp,
-                                  unsigned int& playoutTimestamp,
-                                  unsigned int* jitter,
-                                  unsigned short* fractionLost));
-  WEBRTC_STUB(GetRemoteRTCPReportBlocks,
-              (int channel, std::vector<webrtc::ReportBlock>* receive_blocks));
-  WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs,
-                                 unsigned int& maxJitterMs,
-                                 unsigned int& discardedPackets));
-  WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats));
-  WEBRTC_STUB(SetREDStatus, (int channel, bool enable, int redPayloadtype));
-  WEBRTC_STUB(GetREDStatus, (int channel, bool& enable, int& redPayloadtype));
-  WEBRTC_STUB(SetNACKStatus, (int channel, bool enable, int maxNoPackets));
-
   // webrtc::VoEVolumeControl
   WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
   WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
   WEBRTC_STUB(SetMicVolume, (unsigned int));
   WEBRTC_STUB(GetMicVolume, (unsigned int&));
   WEBRTC_STUB(SetInputMute, (int, bool));
   WEBRTC_STUB(GetInputMute, (int, bool&));
   WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&));
   WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&));
   WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&));
@@ skipping 150 matching lines @@
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   int playout_fail_channel_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_