| Index: webrtc/test/call_test.h
|
| diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
|
| index d8019e55b8c39114a7dc36b0f1345dc3672460ae..f8d52aca9ed1b861d28a2b40d64c892bfcc6474d 100644
|
| --- a/webrtc/test/call_test.h
|
| +++ b/webrtc/test/call_test.h
|
| @@ -14,6 +14,7 @@
|
| #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"
|
| @@ -35,7 +36,9 @@ class CallTest : public ::testing::Test {
|
| 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;
|
| @@ -69,8 +72,12 @@ class CallTest : public ::testing::Test {
|
| 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();
|
| @@ -154,6 +161,9 @@ class BaseTest : public RtpRtcpObserver {
|
| 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);
|
|
|