Be less pessimistic about turning "default" receive streams into signaled streams.

webrtc/media/engine/fakewebrtccall.h

 /*
  *  Copyright (c) 2015 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 25 matching lines @@
 namespace cricket {
 class FakeAudioSendStream final : public webrtc::AudioSendStream {
  public:
   struct TelephoneEvent {
     int payload_type = -1;
     int payload_frequency = -1;
     int event_code = 0;
     int duration_ms = 0;
   };
 
-  explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
+  explicit FakeAudioSendStream(
+      int id, const webrtc::AudioSendStream::Config& config);
 
+  int id() const { return id_; }
   const webrtc::AudioSendStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioSendStream::Stats& stats);
   TelephoneEvent GetLatestTelephoneEvent() const;
   bool IsSending() const { return sending_; }
   bool muted() const { return muted_; }
 
  private:
   // webrtc::AudioSendStream implementation.
   void Start() override { sending_ = true; }
   void Stop() override { sending_ = false; }
 
   bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
                           int duration_ms) override;
   void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
+  int id_ = -1;
   TelephoneEvent latest_telephone_event_;
   webrtc::AudioSendStream::Config config_;
   webrtc::AudioSendStream::Stats stats_;
   bool sending_ = false;
   bool muted_ = false;
 };
 
 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
  public:
   explicit FakeAudioReceiveStream(
-      const webrtc::AudioReceiveStream::Config& config);
+      int id, const webrtc::AudioReceiveStream::Config& config);
 
+  int id() const { return id_; }
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
   int received_packets() const { return received_packets_; }
   bool VerifyLastPacket(const uint8_t* data, size_t length) const;
   const webrtc::AudioSinkInterface* sink() const { return sink_.get(); }
   float gain() const { return gain_; }
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
                   const webrtc::PacketTime& packet_time);
   bool started() const { return started_; }
 
  private:
   // webrtc::AudioReceiveStream implementation.
   void Start() override { started_ = true; }
   void Stop() override { started_ = false; }
 
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
 
+  int id_ = -1;
   webrtc::AudioReceiveStream::Config config_;
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_ = 0;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
   float gain_ = 1.0f;
   rtc::Buffer last_packet_;
   bool started_ = false;
 };
 
 class FakeVideoSendStream final
@@ skipping 180 matching lines @@
                                  webrtc::NetworkState state) override;
   void OnTransportOverheadChanged(webrtc::MediaType media,
                                   int transport_overhead_per_packet) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   webrtc::Call::Config config_;
   webrtc::NetworkState audio_network_state_;
   webrtc::NetworkState video_network_state_;
   rtc::SentPacket last_sent_packet_;
   int last_sent_nonnegative_packet_id_ = -1;
+  int next_stream_id_ = 665;
   webrtc::Call::Stats stats_;
   std::vector<FakeVideoSendStream*> video_send_streams_;
   std::vector<FakeAudioSendStream*> audio_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
   std::vector<FakeFlexfecReceiveStream*> flexfec_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 
   int audio_transport_overhead_;
   int video_transport_overhead_;
 };
 
 }  // namespace cricket
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_