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Side by Side Diff: webrtc/video/end_to_end_tests.cc

Issue 1452883002: Require negotiation to send transport cc feedback over RTCP. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 5 years, 1 month 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> 10 #include <algorithm>
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1578 test::FrameGeneratorCapturer** frame_generator) override { 1578 test::FrameGeneratorCapturer** frame_generator) override {
1579 send_config->rtp.extensions.push_back( 1579 send_config->rtp.extensions.push_back(
1580 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); 1580 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
1581 } 1581 }
1582 1582
1583 void UpdateReceiveConfig( 1583 void UpdateReceiveConfig(
1584 size_t stream_index, 1584 size_t stream_index,
1585 VideoReceiveStream::Config* receive_config) override { 1585 VideoReceiveStream::Config* receive_config) override {
1586 receive_config->rtp.extensions.push_back( 1586 receive_config->rtp.extensions.push_back(
1587 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId)); 1587 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
1588 receive_config->rtp.transport_cc_feedback = true;
1588 } 1589 }
1589 1590
1590 test::DirectTransport* CreateReceiveTransport( 1591 test::DirectTransport* CreateReceiveTransport(
1591 Call* receiver_call) override { 1592 Call* receiver_call) override {
1592 return new TransportFeedbackObserver(receiver_call, &done_); 1593 return new TransportFeedbackObserver(receiver_call, &done_);
1593 } 1594 }
1594 1595
1595 private: 1596 private:
1596 rtc::Event done_; 1597 rtc::Event done_;
1597 } tester; 1598 } tester;
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3180 EXPECT_TRUE(default_receive_config.rtp.rtx.empty()) 3181 EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
3181 << "Enabling RTX requires rtpmap: rtx negotiation."; 3182 << "Enabling RTX requires rtpmap: rtx negotiation.";
3182 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) 3183 EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
3183 << "Enabling RTP extensions require negotiation."; 3184 << "Enabling RTP extensions require negotiation.";
3184 3185
3185 VerifyEmptyNackConfig(default_receive_config.rtp.nack); 3186 VerifyEmptyNackConfig(default_receive_config.rtp.nack);
3186 VerifyEmptyFecConfig(default_receive_config.rtp.fec); 3187 VerifyEmptyFecConfig(default_receive_config.rtp.fec);
3187 } 3188 }
3188 3189
3189 } // namespace webrtc 3190 } // namespace webrtc
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