Let ViEEncoder handle resolution changes.

webrtc/test/call_test.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.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
 #define WEBRTC_TEST_CALL_TEST_H_
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/call.h"
+#include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/fake_videorenderer.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 
 namespace webrtc {
 
 class VoEBase;
 class VoECodec;
 
 namespace test {
 
 class BaseTest;
 
 class CallTest : public ::testing::Test {
  public:
   CallTest();
   virtual ~CallTest();
 
   static const size_t kNumSsrcs = 3;
-
+  static const int kDefaultWidth = 320;
+  static const int kDefaultHeight = 180;
+  static const int kDefaultFramerate = 30;
   static const int kDefaultTimeoutMs;
   static const int kLongTimeoutMs;
   static const uint8_t kVideoSendPayloadType;
   static const uint8_t kSendRtxPayloadType;
   static const uint8_t kFakeVideoSendPayloadType;
   static const uint8_t kRedPayloadType;
   static const uint8_t kRtxRedPayloadType;
   static const uint8_t kUlpfecPayloadType;
   static const uint8_t kAudioSendPayloadType;
   static const uint32_t kSendRtxSsrcs[kNumSsrcs];
@@ skipping 13 matching lines @@
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
   void DestroyCalls();
 
   void CreateSendConfig(size_t num_video_streams,
                         size_t num_audio_streams,
                         Transport* send_transport);
   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
 
-  void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock, float speed);
-  void CreateFrameGeneratorCapturer();
+  void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
+                                             float speed,
+                                             int framerate,
+                                             int width,
+                                             int height);
+  void CreateFrameGeneratorCapturer(int framerate, int width, int height);
   void CreateFakeAudioDevices();
 
   void CreateVideoStreams();
   void CreateAudioStreams();
   void Start();
   void Stop();
   void DestroyStreams();
   void SetFakeVideoCaptureRotation(VideoRotation rotation);
 
   Clock* const clock_;
@@ skipping 64 matching lines @@
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
   virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
   virtual test::PacketTransport* CreateReceiveTransport();
 
   virtual void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config);
+  virtual void ModifyVideoCaptureStartResolution(int* width,
+                                                 int* heigt,
+                                                 int* frame_rate);
   virtual void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams);
 
   virtual void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs);
   virtual void OnAudioStreamsCreated(
       AudioSendStream* send_stream,
       const std::vector<AudioReceiveStream*>& receive_streams);
@@ skipping 13 matching lines @@
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_