Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(217)

Side by Side Diff: webrtc/sdk/objc/Framework/Classes/RTCPeerConnection.mm

Issue 2110113003: Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop f… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 4 years, 5 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
OLDNEW
1 /* 1 /*
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2015 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 189 matching lines...) Expand 10 before | Expand all | Expand 10 after
200 didRemoveIceCandidates:ice_candidates]; 200 didRemoveIceCandidates:ice_candidates];
201 } 201 }
202 202
203 } // namespace webrtc 203 } // namespace webrtc
204 204
205 205
206 @implementation RTCPeerConnection { 206 @implementation RTCPeerConnection {
207 NSMutableArray *_localStreams; 207 NSMutableArray *_localStreams;
208 std::unique_ptr<webrtc::PeerConnectionDelegateAdapter> _observer; 208 std::unique_ptr<webrtc::PeerConnectionDelegateAdapter> _observer;
209 rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection; 209 rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection;
210 BOOL _hasStartedRtcEventLog;
210 } 211 }
211 212
212 @synthesize delegate = _delegate; 213 @synthesize delegate = _delegate;
213 214
214 - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory 215 - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
215 configuration:(RTCConfiguration *)configuration 216 configuration:(RTCConfiguration *)configuration
216 constraints:(RTCMediaConstraints *)constraints 217 constraints:(RTCMediaConstraints *)constraints
217 delegate:(id<RTCPeerConnectionDelegate>)delegate { 218 delegate:(id<RTCPeerConnectionDelegate>)delegate {
218 NSParameterAssert(factory); 219 NSParameterAssert(factory);
219 std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config( 220 std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
(...skipping 129 matching lines...) Expand 10 before | Expand all | Expand 10 after
349 } 350 }
350 351
351 - (void)setRemoteDescription:(RTCSessionDescription *)sdp 352 - (void)setRemoteDescription:(RTCSessionDescription *)sdp
352 completionHandler:(void (^)(NSError *error))completionHandler { 353 completionHandler:(void (^)(NSError *error))completionHandler {
353 rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer( 354 rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
354 new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>( 355 new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
355 completionHandler)); 356 completionHandler));
356 _peerConnection->SetRemoteDescription(observer, sdp.nativeDescription); 357 _peerConnection->SetRemoteDescription(observer, sdp.nativeDescription);
357 } 358 }
358 359
360 - (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
361 maxSizeInBytes:(int64_t)maxSizeInBytes {
362 RTC_DCHECK(filePath.length);
363 RTC_DCHECK_GT(maxSizeInBytes, 0);
364 RTC_DCHECK(!_hasStartedRtcEventLog);
365 if (_hasStartedRtcEventLog) {
366 RTCLogError(@"Event logging already started.");
367 return NO;
368 }
369 int fd = open(filePath.UTF8String, O_WRONLY | O_CREAT | O_TRUNC,
370 S_IRUSR | S_IWUSR);
371 if (fd < 0) {
372 RTCLogError(@"Error opening file: %@. Error: %d", filePath, errno);
373 return NO;
374 }
375 _hasStartedRtcEventLog =
376 _peerConnection->StartRtcEventLog(fd, maxSizeInBytes);
377 return _hasStartedRtcEventLog;
378 }
379
380 - (void)stopRtcEventLog {
381 _peerConnection->StopRtcEventLog();
382 _hasStartedRtcEventLog = NO;
383 }
384
359 - (RTCRtpSender *)senderWithKind:(NSString *)kind 385 - (RTCRtpSender *)senderWithKind:(NSString *)kind
360 streamId:(NSString *)streamId { 386 streamId:(NSString *)streamId {
361 std::string nativeKind = [NSString stdStringForString:kind]; 387 std::string nativeKind = [NSString stdStringForString:kind];
362 std::string nativeStreamId = [NSString stdStringForString:streamId]; 388 std::string nativeStreamId = [NSString stdStringForString:streamId];
363 rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender( 389 rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender(
364 _peerConnection->CreateSender(nativeKind, nativeStreamId)); 390 _peerConnection->CreateSender(nativeKind, nativeStreamId));
365 return nativeSender ? 391 return nativeSender ?
366 [[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender] 392 [[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender]
367 : nil; 393 : nil;
368 } 394 }
(...skipping 185 matching lines...) Expand 10 before | Expand all | Expand 10 after
554 case RTCStatsOutputLevelDebug: 580 case RTCStatsOutputLevelDebug:
555 return webrtc::PeerConnectionInterface::kStatsOutputLevelDebug; 581 return webrtc::PeerConnectionInterface::kStatsOutputLevelDebug;
556 } 582 }
557 } 583 }
558 584
559 - (rtc::scoped_refptr<webrtc::PeerConnectionInterface>)nativePeerConnection { 585 - (rtc::scoped_refptr<webrtc::PeerConnectionInterface>)nativePeerConnection {
560 return _peerConnection; 586 return _peerConnection;
561 } 587 }
562 588
563 @end 589 @end
OLDNEW
« no previous file with comments | « webrtc/pc/channelmanager.cc ('k') | webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory.mm » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698