Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1059)

Side by Side Diff: webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc

Issue 3002173002: Delete unneeded includes of atomic32.h. (Closed)
Patch Set: Created 3 years, 3 months 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/video/payload_router.h ('k') | no next file » | 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) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 10
11 #include <memory> 11 #include <memory>
12 12
13 #include "webrtc/rtc_base/criticalsection.h" 13 #include "webrtc/rtc_base/criticalsection.h"
14 #include "webrtc/system_wrappers/include/atomic32.h"
15 #include "webrtc/system_wrappers/include/event_wrapper.h" 14 #include "webrtc/system_wrappers/include/event_wrapper.h"
16 #include "webrtc/test/testsupport/fileutils.h" 15 #include "webrtc/test/testsupport/fileutils.h"
17 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" 16 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
18 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h" 17 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
19 18
20 class TestRtpObserver : public webrtc::VoERTPObserver { 19 class TestRtpObserver : public webrtc::VoERTPObserver {
21 public: 20 public:
22 TestRtpObserver() : changed_ssrc_event_(webrtc::EventWrapper::Create()) {} 21 TestRtpObserver() : changed_ssrc_event_(webrtc::EventWrapper::Create()) {}
23 virtual ~TestRtpObserver() {} 22 virtual ~TestRtpObserver() {}
24 virtual void OnIncomingCSRCChanged(int channel, 23 virtual void OnIncomingCSRCChanged(int channel,
(...skipping 84 matching lines...) Expand 10 before | Expand all | Expand 10 after
109 108
110 Sleep(1000); 109 Sleep(1000);
111 110
112 unsigned int ssrc; 111 unsigned int ssrc;
113 EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc)); 112 EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc));
114 EXPECT_EQ(local_ssrc, ssrc); 113 EXPECT_EQ(local_ssrc, ssrc);
115 114
116 EXPECT_EQ(0, voe_rtp_rtcp_->GetRemoteSSRC(channel_, ssrc)); 115 EXPECT_EQ(0, voe_rtp_rtcp_->GetRemoteSSRC(channel_, ssrc));
117 EXPECT_EQ(local_ssrc, ssrc); 116 EXPECT_EQ(local_ssrc, ssrc);
118 } 117 }
OLDNEW
« no previous file with comments | « webrtc/video/payload_router.h ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698