Add ramp-up tests for transport sequence number with and w/o audio.

webrtc/call/rampup_tests.h

 /*
  *  Copyright (c) 2014 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 22 matching lines @@
  public:
   RampUpTester(size_t num_video_streams,
                size_t num_audio_streams,
                unsigned int start_bitrate_bps,
                const std::string& extension_type,
                bool rtx,
                bool red);
   ~RampUpTester() override;
 
   size_t GetNumVideoStreams() const override;
+  size_t GetNumAudioStreams() const override;
 
   void PerformTest() override;
 
  protected:
   virtual bool PollStats();
 
   void AccumulateStats(const VideoSendStream::StreamStats& stream,
                        size_t* total_packets_sent,
                        size_t* total_sent,
                        size_t* padding_sent,
@@ skipping 19 matching lines @@
 
   Call::Config GetSenderCallConfig() override;
   void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams) override;
   test::PacketTransport* CreateSendTransport(Call* sender_call) override;
   void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override;
+  void ModifyAudioConfigs(
+      AudioSendStream::Config* send_config,
+      std::vector<AudioReceiveStream::Config>* receive_configs) override;
   void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
 
   static bool BitrateStatsPollingThread(void* obj);
 
   const int start_bitrate_bps_;
   bool start_bitrate_verified_;
   int expected_bitrate_bps_;
   int64_t test_start_ms_;
   int64_t ramp_up_finished_ms_;
 
   const std::string extension_type_;
   std::vector<uint32_t> video_ssrcs_;
   std::vector<uint32_t> video_rtx_ssrcs_;
   std::vector<uint32_t> audio_ssrcs_;
   SsrcMap rtx_ssrc_map_;
 
   rtc::PlatformThread poller_thread_;
   Call* sender_call_;
 };
 
 class RampUpDownUpTester : public RampUpTester {
  public:
-  RampUpDownUpTester(size_t num_streams,
+  RampUpDownUpTester(size_t num_video_streams,
+                     size_t num_audio_streams,
                      unsigned int start_bitrate_bps,
                      const std::string& extension_type,
                      bool rtx,
                      bool red);
   ~RampUpDownUpTester() override;
 
  protected:
   bool PollStats() override;
 
  private:
   static const int kHighBandwidthLimitBps = 80000;
   static const int kExpectedHighBitrateBps = 60000;
   static const int kLowBandwidthLimitBps = 20000;
   static const int kExpectedLowBitrateBps = 20000;
   enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
 
   Call::Config GetReceiverCallConfig() override;
 
   std::string GetModifierString() const;
   void EvolveTestState(int bitrate_bps, bool suspended);
 
   TestStates test_state_;
   int64_t state_start_ms_;
   int64_t interval_start_ms_;
   int sent_bytes_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_CALL_RAMPUP_TESTS_H_