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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 #ifndef WEBRTC_TEST_CALL_TEST_H_ | 10 #ifndef WEBRTC_TEST_CALL_TEST_H_ |
11 #define WEBRTC_TEST_CALL_TEST_H_ | 11 #define WEBRTC_TEST_CALL_TEST_H_ |
12 | 12 |
13 #include <memory> | 13 #include <memory> |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
16 #include "webrtc/call/call.h" | 16 #include "webrtc/call/call.h" |
| 17 #include "webrtc/call/rtp_transport_controller_send.h" |
17 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 18 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
18 #include "webrtc/test/encoder_settings.h" | 19 #include "webrtc/test/encoder_settings.h" |
19 #include "webrtc/test/fake_audio_device.h" | 20 #include "webrtc/test/fake_audio_device.h" |
20 #include "webrtc/test/fake_decoder.h" | 21 #include "webrtc/test/fake_decoder.h" |
21 #include "webrtc/test/fake_encoder.h" | 22 #include "webrtc/test/fake_encoder.h" |
22 #include "webrtc/test/fake_videorenderer.h" | 23 #include "webrtc/test/fake_videorenderer.h" |
23 #include "webrtc/test/frame_generator_capturer.h" | 24 #include "webrtc/test/frame_generator_capturer.h" |
24 #include "webrtc/test/rtp_rtcp_observer.h" | 25 #include "webrtc/test/rtp_rtcp_observer.h" |
25 | 26 |
26 namespace webrtc { | 27 namespace webrtc { |
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60 static const std::map<uint8_t, MediaType> payload_type_map_; | 61 static const std::map<uint8_t, MediaType> payload_type_map_; |
61 | 62 |
62 protected: | 63 protected: |
63 // RunBaseTest overwrites the audio_state and the voice_engine of the send and | 64 // RunBaseTest overwrites the audio_state and the voice_engine of the send and |
64 // receive Call configs to simplify test code and avoid having old VoiceEngine | 65 // receive Call configs to simplify test code and avoid having old VoiceEngine |
65 // APIs in the tests. | 66 // APIs in the tests. |
66 void RunBaseTest(BaseTest* test); | 67 void RunBaseTest(BaseTest* test); |
67 | 68 |
68 void CreateCalls(const Call::Config& sender_config, | 69 void CreateCalls(const Call::Config& sender_config, |
69 const Call::Config& receiver_config); | 70 const Call::Config& receiver_config); |
70 void CreateSenderCall(const Call::Config& config); | 71 void CreateSenderCall(Call::Config config); |
71 void CreateReceiverCall(const Call::Config& config); | 72 void CreateReceiverCall(Call::Config config); |
72 void DestroyCalls(); | 73 void DestroyCalls(); |
73 | 74 |
74 void CreateSendConfig(size_t num_video_streams, | 75 void CreateSendConfig(size_t num_video_streams, |
75 size_t num_audio_streams, | 76 size_t num_audio_streams, |
76 size_t num_flexfec_streams, | 77 size_t num_flexfec_streams, |
77 Transport* send_transport); | 78 Transport* send_transport); |
78 | 79 |
79 void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport); | 80 void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport); |
80 | 81 |
81 void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock, | 82 void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock, |
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92 void CreateAudioStreams(); | 93 void CreateAudioStreams(); |
93 void CreateFlexfecStreams(); | 94 void CreateFlexfecStreams(); |
94 void Start(); | 95 void Start(); |
95 void Stop(); | 96 void Stop(); |
96 void DestroyStreams(); | 97 void DestroyStreams(); |
97 void SetFakeVideoCaptureRotation(VideoRotation rotation); | 98 void SetFakeVideoCaptureRotation(VideoRotation rotation); |
98 | 99 |
99 Clock* const clock_; | 100 Clock* const clock_; |
100 | 101 |
101 std::unique_ptr<webrtc::RtcEventLog> event_log_; | 102 std::unique_ptr<webrtc::RtcEventLog> event_log_; |
| 103 std::unique_ptr<RtpTransportControllerSend> sender_rtp_transport_send_; |
102 std::unique_ptr<Call> sender_call_; | 104 std::unique_ptr<Call> sender_call_; |
103 std::unique_ptr<PacketTransport> send_transport_; | 105 std::unique_ptr<PacketTransport> send_transport_; |
104 VideoSendStream::Config video_send_config_; | 106 VideoSendStream::Config video_send_config_; |
105 VideoEncoderConfig video_encoder_config_; | 107 VideoEncoderConfig video_encoder_config_; |
106 VideoSendStream* video_send_stream_; | 108 VideoSendStream* video_send_stream_; |
107 AudioSendStream::Config audio_send_config_; | 109 AudioSendStream::Config audio_send_config_; |
108 AudioSendStream* audio_send_stream_; | 110 AudioSendStream* audio_send_stream_; |
109 | 111 |
| 112 // Needed at for sending feedback messages. |
| 113 std::unique_ptr<RtpTransportControllerSend> receiver_rtp_transport_send_; |
110 std::unique_ptr<Call> receiver_call_; | 114 std::unique_ptr<Call> receiver_call_; |
111 std::unique_ptr<PacketTransport> receive_transport_; | 115 std::unique_ptr<PacketTransport> receive_transport_; |
112 std::vector<VideoReceiveStream::Config> video_receive_configs_; | 116 std::vector<VideoReceiveStream::Config> video_receive_configs_; |
113 std::vector<VideoReceiveStream*> video_receive_streams_; | 117 std::vector<VideoReceiveStream*> video_receive_streams_; |
114 std::vector<AudioReceiveStream::Config> audio_receive_configs_; | 118 std::vector<AudioReceiveStream::Config> audio_receive_configs_; |
115 std::vector<AudioReceiveStream*> audio_receive_streams_; | 119 std::vector<AudioReceiveStream*> audio_receive_streams_; |
116 std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_; | 120 std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_; |
117 std::vector<FlexfecReceiveStream*> flexfec_receive_streams_; | 121 std::vector<FlexfecReceiveStream*> flexfec_receive_streams_; |
118 | 122 |
119 std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; | 123 std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; |
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220 EndToEndTest(); | 224 EndToEndTest(); |
221 explicit EndToEndTest(unsigned int timeout_ms); | 225 explicit EndToEndTest(unsigned int timeout_ms); |
222 | 226 |
223 bool ShouldCreateReceivers() const override; | 227 bool ShouldCreateReceivers() const override; |
224 }; | 228 }; |
225 | 229 |
226 } // namespace test | 230 } // namespace test |
227 } // namespace webrtc | 231 } // namespace webrtc |
228 | 232 |
229 #endif // WEBRTC_TEST_CALL_TEST_H_ | 233 #endif // WEBRTC_TEST_CALL_TEST_H_ |
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