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Issue 2978503002: Move RTP keep-alive config from VideoSendStream::Config to Call::Config (Closed)
Patch Set: Typo in comment Created 3 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 #include <algorithm> // max 10 #include <algorithm> // max
(...skipping 3397 matching lines...) Expand 10 before | Expand all | Expand 10 after
3408 RunBaseTest(&test); 3408 RunBaseTest(&test);
3409 } 3409 }
3410 3410
3411 TEST_F(VideoSendStreamTest, SendsKeepAlive) { 3411 TEST_F(VideoSendStreamTest, SendsKeepAlive) {
3412 const int kTimeoutMs = 50; // Really short timeout for testing. 3412 const int kTimeoutMs = 50; // Really short timeout for testing.
3413 3413
3414 class KeepaliveObserver : public test::SendTest { 3414 class KeepaliveObserver : public test::SendTest {
3415 public: 3415 public:
3416 KeepaliveObserver() : SendTest(kDefaultTimeoutMs) {} 3416 KeepaliveObserver() : SendTest(kDefaultTimeoutMs) {}
3417 3417
3418 Call::Config GetSenderCallConfig() override {
3419 Call::Config config = SendTest::GetSenderCallConfig();
3420 config.keepalive_config.timeout_interval_ms = kTimeoutMs;
3421 config.keepalive_config.payload_type =
3422 CallTest::kDefaultKeepalivePayloadType;
3423 return config;
3424 }
3425
3418 private: 3426 private:
3419 Action OnSendRtp(const uint8_t* packet, size_t length) override { 3427 Action OnSendRtp(const uint8_t* packet, size_t length) override {
3420 RTPHeader header; 3428 RTPHeader header;
3421 EXPECT_TRUE(parser_->Parse(packet, length, &header)); 3429 EXPECT_TRUE(parser_->Parse(packet, length, &header));
3422 3430
3423 if (header.payloadType != CallTest::kDefaultKeepalivePayloadType) { 3431 if (header.payloadType != CallTest::kDefaultKeepalivePayloadType) {
3424 // The video stream has started. Stop it now. 3432 // The video stream has started. Stop it now.
3425 if (capturer_) 3433 if (capturer_)
3426 capturer_->Stop(); 3434 capturer_->Stop();
3427 } else { 3435 } else {
3428 observation_complete_.Set(); 3436 observation_complete_.Set();
3429 } 3437 }
3430 3438
3431 return SEND_PACKET; 3439 return SEND_PACKET;
3432 } 3440 }
3433 3441
3434 void ModifyVideoConfigs(
3435 VideoSendStream::Config* send_config,
3436 std::vector<VideoReceiveStream::Config>* receive_configs,
3437 VideoEncoderConfig* encoder_config) override {
3438 send_config->rtp.keep_alive.timeout_interval_ms = kTimeoutMs;
3439 send_config->rtp.keep_alive.payload_type =
3440 CallTest::kDefaultKeepalivePayloadType;
3441 }
3442
3443 void PerformTest() override { 3442 void PerformTest() override {
3444 EXPECT_TRUE(Wait()) << "Timed out while waiting for keep-alive packet."; 3443 EXPECT_TRUE(Wait()) << "Timed out while waiting for keep-alive packet.";
3445 } 3444 }
3446 3445
3447 void OnFrameGeneratorCapturerCreated( 3446 void OnFrameGeneratorCapturerCreated(
3448 test::FrameGeneratorCapturer* frame_generator_capturer) override { 3447 test::FrameGeneratorCapturer* frame_generator_capturer) override {
3449 capturer_ = frame_generator_capturer; 3448 capturer_ = frame_generator_capturer;
3450 } 3449 }
3451 3450
3452 test::FrameGeneratorCapturer* capturer_ = nullptr; 3451 test::FrameGeneratorCapturer* capturer_ = nullptr;
3453 } test; 3452 } test;
3454 3453
3455 RunBaseTest(&test); 3454 RunBaseTest(&test);
3456 } 3455 }
3457 3456
3458 } // namespace webrtc 3457 } // namespace webrtc
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