Refactoring full stack and loopback tests

webrtc/video/video_quality_test.h

Index: webrtc/video/video_quality_test.h
diff --git a/webrtc/video/video_quality_test.h b/webrtc/video/video_quality_test.h
new file mode 100644
index 0000000000000000000000000000000000000000..ccac58fa46e7ee0aa63999f2706dc050b9bad9ac
--- /dev/null
+++ b/webrtc/video/video_quality_test.h
@@ -0,0 +1,88 @@
+/*
+ *  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.
+ */
+#ifndef WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_
+#define WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_
+
+#include <string>
+
+#include "webrtc/test/call_test.h"
+#include "webrtc/test/frame_generator.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
+
+namespace webrtc {
+
+class VideoQualityTest : public test::CallTest {
+ public:
+  // Parameters are grouped into smaller structs to make it easier to fill out

[sprang_webrtc, 2015/09/11 08:01:34]

s/fill out/set

[ivica, 2015/09/14 17:12:55]

Done.

+  // the desired elements and skip unused, using aggregate initialization.
+  // Unfortunately, C++11 (as opposed to C11) doesn't support unnamed structs,
+  // which makes the implementation of VideoQualityTest a bit uglier.
+  // Also, C++11 doesn't support both aggregate initialization and default
+  // values simultaneously, so default values are manually set in
+  // FillDefaultParams(Params *).
+  struct Params {
+    struct {
+      size_t width;
+      size_t height;
+      int32_t fps;
+      size_t min_bitrate_bps;
+      size_t target_bitrate_bps;
+      size_t max_bitrate_bps;
+
+      std::string codec;                // Default: "VP8"
+      size_t num_temporal_layers;       // Default: 2
+      size_t start_bitrate_bps;         // Default: 400 * 1000
+      size_t min_transmit_bps;          // Default: 0
+      size_t tl_discard_threshold;
+    } general;
+    struct {  // Video-specific settings.
+      std::string clip_name;
+    } video;
+    struct {  // Screenshare-specific settings.
+      bool enabled;
+      int32_t slide_change_interval;    // Default: 10
+      int32_t scroll_duration;
+    } screenshare;
+    struct {  // Analyzer settings.
+      std::string test_label;
+      double avg_psnr_threshold;
+      double avg_ssim_threshold;
+      int test_durations_secs;
+      std::string graph_data_output_filename;
+    } analyzer;
+    FakeNetworkPipe::Config pipe;
+    bool logs;
+  };
+
+  VideoQualityTest();
+  void RunWithAnalyzer(const Params &params);
+  void RunWithVideoRenderer(const Params &params);
+
+ protected:
+  void TestBody() override;
+
+  void CreateCapturer(const Params &params, VideoCaptureInput *input);
+  void FillDefaultParams(Params *params);
+  void ValidateParams(const Params &params);
+  void SetupFullStack(const Params &params,
+                      newapi::Transport *send_transport,
+                      newapi::Transport *recv_transport);
+  void SetupScreenshare(const Params &params);
+
+  rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr_;
+  rtc::scoped_ptr<test::FrameGenerator> frame_generator_;
+  rtc::scoped_ptr<VideoEncoder> encoder_;
+  VideoCodecUnion codec_settings_;
+  Clock* const clock_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_