Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(89)

Side by Side Diff: webrtc/test/direct_transport.cc

Issue 2458863002: Start probes only after network is connected. (Closed)
Patch Set: sync Created 4 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/test/call_test.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 "webrtc/test/direct_transport.h" 10 #include "webrtc/test/direct_transport.h"
(...skipping 10 matching lines...) Expand all
21 21
22 DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config, 22 DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config,
23 Call* send_call) 23 Call* send_call)
24 : send_call_(send_call), 24 : send_call_(send_call),
25 packet_event_(false, false), 25 packet_event_(false, false),
26 thread_(NetworkProcess, this, "NetworkProcess"), 26 thread_(NetworkProcess, this, "NetworkProcess"),
27 clock_(Clock::GetRealTimeClock()), 27 clock_(Clock::GetRealTimeClock()),
28 shutting_down_(false), 28 shutting_down_(false),
29 fake_network_(clock_, config) { 29 fake_network_(clock_, config) {
30 thread_.Start(); 30 thread_.Start();
31 if (send_call_) {
32 send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
33 send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
34 }
31 } 35 }
32 36
33 DirectTransport::~DirectTransport() { StopSending(); } 37 DirectTransport::~DirectTransport() { StopSending(); }
34 38
35 void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) { 39 void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
36 fake_network_.SetConfig(config); 40 fake_network_.SetConfig(config);
37 } 41 }
38 42
39 void DirectTransport::StopSending() { 43 void DirectTransport::StopSending() {
40 { 44 {
(...skipping 40 matching lines...) Expand 10 before | Expand all | Expand 10 after
81 fake_network_.Process(); 85 fake_network_.Process();
82 int64_t wait_time_ms = fake_network_.TimeUntilNextProcess(); 86 int64_t wait_time_ms = fake_network_.TimeUntilNextProcess();
83 if (wait_time_ms > 0) { 87 if (wait_time_ms > 0) {
84 packet_event_.Wait(static_cast<int>(wait_time_ms)); 88 packet_event_.Wait(static_cast<int>(wait_time_ms));
85 } 89 }
86 rtc::CritScope crit(&lock_); 90 rtc::CritScope crit(&lock_);
87 return shutting_down_ ? false : true; 91 return shutting_down_ ? false : true;
88 } 92 }
89 } // namespace test 93 } // namespace test
90 } // namespace webrtc 94 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/test/call_test.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698