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Issue 417203002: Add browser test to connect chormium sink to remote tracks, verify that no crash. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Created 6 years, 4 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "base/command_line.h" 5 #include "base/command_line.h"
6 #include "base/file_util.h" 6 #include "base/file_util.h"
7 #include "base/process/process_handle.h" 7 #include "base/process/process_handle.h"
8 #include "base/strings/string_number_conversions.h" 8 #include "base/strings/string_number_conversions.h"
9 #include "base/strings/stringprintf.h" 9 #include "base/strings/stringprintf.h"
10 #include "base/threading/platform_thread.h" 10 #include "base/threading/platform_thread.h"
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201 // TODO(henrika): see http://crbug.com/362437 and http://crbug.com/359389 201 // TODO(henrika): see http://crbug.com/362437 and http://crbug.com/359389
202 // for details. 202 // for details.
203 CommandLine::ForCurrentProcess()->AppendSwitch( 203 CommandLine::ForCurrentProcess()->AppendSwitch(
204 switches::kDisableWebRtcHWDecoding); 204 switches::kDisableWebRtcHWDecoding);
205 #endif 205 #endif
206 const std::string javascript = GenerateGetUserMediaCall( 206 const std::string javascript = GenerateGetUserMediaCall(
207 "callAndForwardRemoteStream", 1280, 1280, 720, 720, 10, 30); 207 "callAndForwardRemoteStream", 1280, 1280, 720, 720, 10, 30);
208 MakeTypicalPeerConnectionCall(javascript); 208 MakeTypicalPeerConnectionCall(javascript);
209 } 209 }
210 210
211 IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
212 NoCrashWhenConnectChromiumSinkToRemoteTrack) {
213 MakeTypicalPeerConnectionCall("ConnectChromiumSinkToRemoteAudioTrack();");
214 }
215
211 // This test will make a complete PeerConnection-based call but remove the 216 // This test will make a complete PeerConnection-based call but remove the
212 // MSID and bundle attribute from the initial offer to verify that 217 // MSID and bundle attribute from the initial offer to verify that
213 // video is playing for the call even if the initiating client don't support 218 // video is playing for the call even if the initiating client don't support
214 // MSID. http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02 219 // MSID. http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02
215 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) 220 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
216 // Timing out on ARM linux, see http://crbug.com/240373 221 // Timing out on ARM linux, see http://crbug.com/240373
217 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\ 222 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\
218 DISABLED_CanSetupAudioAndVideoCallWithoutMsidAndBundle 223 DISABLED_CanSetupAudioAndVideoCallWithoutMsidAndBundle
219 #else 224 #else
220 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\ 225 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\
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529 EXPECT_TRUE(base::PathExists(unique_dump_file)); 534 EXPECT_TRUE(base::PathExists(unique_dump_file));
530 int64 file_size = 0; 535 int64 file_size = 0;
531 EXPECT_TRUE(base::GetFileSize(unique_dump_file, &file_size)); 536 EXPECT_TRUE(base::GetFileSize(unique_dump_file, &file_size));
532 EXPECT_GT(file_size, 0); 537 EXPECT_GT(file_size, 0);
533 538
534 base::DeleteFile(unique_dump_file, false); 539 base::DeleteFile(unique_dump_file, false);
535 } 540 }
536 } 541 }
537 542
538 } // namespace content 543 } // namespace content
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