Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(387)

Side by Side Diff: webrtc/api/peerconnectionendtoend_unittest.cc

Issue 2593313002: Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. (Closed)
Patch Set: Use a different variable (rtc_enable_sctp). Created 3 years, 12 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/api/peerconnection_unittest.cc ('k') | webrtc/api/rtcstats_integrationtest.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 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright 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 10
(...skipping 176 matching lines...) Expand 10 before | Expand all | Expand 10 after
187 FakeConstraints pc_constraints; 187 FakeConstraints pc_constraints;
188 pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, 188 pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
189 false); 189 false);
190 CreatePcs(&pc_constraints); 190 CreatePcs(&pc_constraints);
191 GetAndAddUserMedia(); 191 GetAndAddUserMedia();
192 Negotiate(); 192 Negotiate();
193 WaitForCallEstablished(); 193 WaitForCallEstablished();
194 } 194 }
195 #endif // !defined(ADDRESS_SANITIZER) 195 #endif // !defined(ADDRESS_SANITIZER)
196 196
197 #ifdef HAVE_SCTP
197 // Verifies that a DataChannel created before the negotiation can transition to 198 // Verifies that a DataChannel created before the negotiation can transition to
198 // "OPEN" and transfer data. 199 // "OPEN" and transfer data.
199 TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { 200 TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
200 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); 201 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
201 202
202 CreatePcs(); 203 CreatePcs();
203 204
204 webrtc::DataChannelInit init; 205 webrtc::DataChannelInit init;
205 rtc::scoped_refptr<DataChannelInterface> caller_dc( 206 rtc::scoped_refptr<DataChannelInterface> caller_dc(
206 caller_->CreateDataChannel("data", init)); 207 caller_->CreateDataChannel("data", init));
(...skipping 105 matching lines...) Expand 10 before | Expand all | Expand 10 after
312 313
313 caller_dc_1->Send(webrtc::DataBuffer(message_1)); 314 caller_dc_1->Send(webrtc::DataBuffer(message_1));
314 EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait); 315 EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
315 316
316 caller_dc_2->Send(webrtc::DataBuffer(message_2)); 317 caller_dc_2->Send(webrtc::DataBuffer(message_2));
317 EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait); 318 EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
318 319
319 EXPECT_EQ(1U, dc_1_observer->received_message_count()); 320 EXPECT_EQ(1U, dc_1_observer->received_message_count());
320 EXPECT_EQ(1U, dc_2_observer->received_message_count()); 321 EXPECT_EQ(1U, dc_2_observer->received_message_count());
321 } 322 }
323 #endif // HAVE_SCTP
322 324
323 #ifdef HAVE_QUIC 325 #ifdef HAVE_QUIC
324 // Test that QUIC data channels can be used and that messages go to the correct 326 // Test that QUIC data channels can be used and that messages go to the correct
325 // remote data channel when both peers want to use QUIC. It is assumed that the 327 // remote data channel when both peers want to use QUIC. It is assumed that the
326 // application has externally negotiated the data channel parameters. 328 // application has externally negotiated the data channel parameters.
327 TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) { 329 TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) {
328 config_.enable_quic = true; 330 config_.enable_quic = true;
329 CreatePcs(); 331 CreatePcs();
330 332
331 webrtc::DataChannelInit init_1; 333 webrtc::DataChannelInit init_1;
(...skipping 57 matching lines...) Expand 10 before | Expand all | Expand 10 after
389 EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait); 391 EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
390 392
391 callee_dc_2->Send(webrtc::DataBuffer(message_2)); 393 callee_dc_2->Send(webrtc::DataBuffer(message_2));
392 EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait); 394 EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
393 395
394 EXPECT_EQ(1U, dc_1_observer->received_message_count()); 396 EXPECT_EQ(1U, dc_1_observer->received_message_count());
395 EXPECT_EQ(1U, dc_2_observer->received_message_count()); 397 EXPECT_EQ(1U, dc_2_observer->received_message_count());
396 } 398 }
397 #endif // HAVE_QUIC 399 #endif // HAVE_QUIC
398 400
401 #ifdef HAVE_SCTP
399 // Verifies that a DataChannel added from an OPEN message functions after 402 // Verifies that a DataChannel added from an OPEN message functions after
400 // a channel has been previously closed (webrtc issue 3778). 403 // a channel has been previously closed (webrtc issue 3778).
401 // This previously failed because the new channel re-uses the ID of the closed 404 // This previously failed because the new channel re-uses the ID of the closed
402 // channel, and the closed channel was incorrectly still assigned to the id. 405 // channel, and the closed channel was incorrectly still assigned to the id.
403 // TODO(deadbeef): This is disabled because there's currently a race condition 406 // TODO(deadbeef): This is disabled because there's currently a race condition
404 // caused by the fact that a data channel signals that it's closed before it 407 // caused by the fact that a data channel signals that it's closed before it
405 // really is. Re-enable this test once that's fixed. 408 // really is. Re-enable this test once that's fixed.
406 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453 409 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
407 TEST_F(PeerConnectionEndToEndTest, 410 TEST_F(PeerConnectionEndToEndTest,
408 DISABLED_DataChannelFromOpenWorksAfterClose) { 411 DISABLED_DataChannelFromOpenWorksAfterClose) {
(...skipping 39 matching lines...) Expand 10 before | Expand all | Expand 10 after
448 WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0); 451 WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
449 // This removes the reference to the remote data channel that we hold. 452 // This removes the reference to the remote data channel that we hold.
450 callee_signaled_data_channels_.clear(); 453 callee_signaled_data_channels_.clear();
451 caller_dc->Close(); 454 caller_dc->Close();
452 EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait); 455 EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
453 456
454 // Wait for a bit longer so the remote data channel will receive the 457 // Wait for a bit longer so the remote data channel will receive the
455 // close message and be destroyed. 458 // close message and be destroyed.
456 rtc::Thread::Current()->ProcessMessages(100); 459 rtc::Thread::Current()->ProcessMessages(100);
457 } 460 }
461 #endif // HAVE_SCTP
OLDNEW
« no previous file with comments | « webrtc/api/peerconnection_unittest.cc ('k') | webrtc/api/rtcstats_integrationtest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698