commit e74eef19bd3f101208dc72b98038e42fc523a351
author stefan <stefan@webrtc.org> Fri Jan 08 14:47:13 2016
committer Commit bot <commit-bot@chromium.org> Fri Jan 08 14:47:21 2016
tree 04e853053e199d2344f4b32b688e99377ef220e0
parent 37ebcf0ce5ad1685bcf659ea75960beb96019647

Add CreateSend/ReceiveTransport() methods to CallTest.

This allows the test to create its own transports if it, for instance, needs to do demuxing.

BUG=webrtc:5416

Review URL: https://codereview.webrtc.org/1573453002

Cr-Commit-Position: refs/heads/master@{#11187}

webrtc/call/call_perf_tests.cc

diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 79f1ff6..86cd9ae 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -384,8 +384,12 @@
   class CaptureNtpTimeObserver : public test::EndToEndTest,
                                  public VideoRenderer {
    public:
-    CaptureNtpTimeObserver(int threshold_ms, int start_time_ms, int run_time_ms)
+    CaptureNtpTimeObserver(const FakeNetworkPipe::Config& net_config,
+                           int threshold_ms,
+                           int start_time_ms,
+                           int run_time_ms)
         : EndToEndTest(kLongTimeoutMs),
+          net_config_(net_config),
           clock_(Clock::GetRealTimeClock()),
           threshold_ms_(threshold_ms),
           start_time_ms_(start_time_ms),
@@ -396,6 +400,11 @@
           rtp_start_timestamp_(0) {}
 
    private:
+    test::PacketTransport* CreateSendTransport(Call* sender_call) override {
+      return new test::PacketTransport(
+          sender_call, this, test::PacketTransport::kSender, net_config_);
+    }
+
     void RenderFrame(const VideoFrame& video_frame,
                      int time_to_render_ms) override {
       rtc::CritScope lock(&crit_);
@@ -480,6 +489,7 @@
     }
 
     rtc::CriticalSection crit_;
+    const FakeNetworkPipe::Config net_config_;
     Clock* const clock_;
     int threshold_ms_;
     int start_time_ms_;
@@ -490,9 +500,9 @@
     uint32_t rtp_start_timestamp_;
     typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
     FrameCaptureTimeList capture_time_list_ GUARDED_BY(&crit_);
-  } test(threshold_ms, start_time_ms, run_time_ms);
+  } test(net_config, threshold_ms, start_time_ms, run_time_ms);
 
-  RunBaseTest(&test, net_config);
+  RunBaseTest(&test);
 }
 
 TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
@@ -548,7 +558,7 @@
     test::DelayedEncoder encoder_;
   } test(tested_load, encode_delay_ms);
 
-  RunBaseTest(&test, FakeNetworkPipe::Config());
+  RunBaseTest(&test);
 }
 
 TEST_F(CallPerfTest, ReceivesCpuUnderuse) {
@@ -642,7 +652,7 @@
   } test(pad_to_min_bitrate);
 
   fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);
-  RunBaseTest(&test, FakeNetworkPipe::Config());
+  RunBaseTest(&test);
 }
 
 TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); }
@@ -738,7 +748,7 @@
     VideoEncoderConfig encoder_config_;
   } test;
 
-  RunBaseTest(&test, FakeNetworkPipe::Config());
+  RunBaseTest(&test);
 }
 
 }  // namespace webrtc