OLD | NEW |
---|---|
1 /* | 1 /* |
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2004 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 "webrtc/pc/peerconnectionfactory.h" | 11 #include "webrtc/pc/peerconnectionfactory.h" |
12 | 12 |
13 #include <utility> | 13 #include <utility> |
14 | 14 |
15 #include "webrtc/api/mediaconstraintsinterface.h" | 15 #include "webrtc/api/mediaconstraintsinterface.h" |
16 #include "webrtc/api/mediastreamproxy.h" | 16 #include "webrtc/api/mediastreamproxy.h" |
17 #include "webrtc/api/mediastreamtrackproxy.h" | 17 #include "webrtc/api/mediastreamtrackproxy.h" |
18 #include "webrtc/api/peerconnectionfactoryproxy.h" | 18 #include "webrtc/api/peerconnectionfactoryproxy.h" |
19 #include "webrtc/api/peerconnectionproxy.h" | 19 #include "webrtc/api/peerconnectionproxy.h" |
20 #include "webrtc/api/videosourceproxy.h" | 20 #include "webrtc/api/videosourceproxy.h" |
21 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 21 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
22 #include "webrtc/rtc_base/bind.h" | 22 #include "webrtc/rtc_base/bind.h" |
23 #include "webrtc/rtc_base/checks.h" | 23 #include "webrtc/rtc_base/checks.h" |
24 #include "webrtc/rtc_base/ptr_util.h" | |
25 // Adding 'nogncheck' to disable the gn include headers check to support modular | 24 // Adding 'nogncheck' to disable the gn include headers check to support modular |
26 // WebRTC build targets. | 25 // WebRTC build targets. |
27 // TODO(zhihuang): This wouldn't be necessary if the interface and | 26 // TODO(zhihuang): This wouldn't be necessary if the interface and |
28 // implementation of the media engine were in separate build targets. | 27 // implementation of the media engine were in separate build targets. |
29 #include "webrtc/media/engine/webrtcmediaengine.h" // nogncheck | 28 #include "webrtc/media/engine/webrtcmediaengine.h" // nogncheck |
30 #include "webrtc/media/engine/webrtcvideodecoderfactory.h" // nogncheck | 29 #include "webrtc/media/engine/webrtcvideodecoderfactory.h" // nogncheck |
31 #include "webrtc/media/engine/webrtcvideoencoderfactory.h" // nogncheck | 30 #include "webrtc/media/engine/webrtcvideoencoderfactory.h" // nogncheck |
32 #include "webrtc/modules/audio_device/include/audio_device.h" // nogncheck | 31 #include "webrtc/modules/audio_device/include/audio_device.h" // nogncheck |
33 #include "webrtc/p2p/base/basicpacketsocketfactory.h" | 32 #include "webrtc/p2p/base/basicpacketsocketfactory.h" |
34 #include "webrtc/p2p/client/basicportallocator.h" | 33 #include "webrtc/p2p/client/basicportallocator.h" |
(...skipping 219 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
254 } | 253 } |
255 | 254 |
256 if (!allocator) { | 255 if (!allocator) { |
257 allocator.reset(new cricket::BasicPortAllocator( | 256 allocator.reset(new cricket::BasicPortAllocator( |
258 default_network_manager_.get(), default_socket_factory_.get())); | 257 default_network_manager_.get(), default_socket_factory_.get())); |
259 } | 258 } |
260 network_thread_->Invoke<void>( | 259 network_thread_->Invoke<void>( |
261 RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::SetNetworkIgnoreMask, | 260 RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::SetNetworkIgnoreMask, |
262 allocator.get(), options_.network_ignore_mask)); | 261 allocator.get(), options_.network_ignore_mask)); |
263 | 262 |
264 std::unique_ptr<RtcEventLog> event_log = | 263 std::unique_ptr<RtcEventLog> event_log = |
eladalon
2017/09/05 13:04:32
There's a trade-off between Invoke()-ing twice (bl
| |
265 event_log_factory_ ? event_log_factory_->CreateRtcEventLog() | 264 worker_thread_->Invoke<std::unique_ptr<RtcEventLog>>( |
266 : rtc::MakeUnique<RtcEventLogNullImpl>(); | 265 RTC_FROM_HERE, |
266 rtc::Bind(&PeerConnectionFactory::CreateRtcEventLog_w, this)); | |
267 | 267 |
268 std::unique_ptr<Call> call = worker_thread_->Invoke<std::unique_ptr<Call>>( | 268 std::unique_ptr<Call> call = worker_thread_->Invoke<std::unique_ptr<Call>>( |
269 RTC_FROM_HERE, | 269 RTC_FROM_HERE, |
270 rtc::Bind(&PeerConnectionFactory::CreateCall_w, this, event_log.get())); | 270 rtc::Bind(&PeerConnectionFactory::CreateCall_w, this, event_log.get())); |
271 | 271 |
272 rtc::scoped_refptr<PeerConnection> pc( | 272 rtc::scoped_refptr<PeerConnection> pc( |
273 new rtc::RefCountedObject<PeerConnection>(this, std::move(event_log), | 273 new rtc::RefCountedObject<PeerConnection>(this, std::move(event_log), |
274 std::move(call))); | 274 std::move(call))); |
275 | 275 |
276 if (!pc->Initialize(configuration, std::move(allocator), | 276 if (!pc->Initialize(configuration, std::move(allocator), |
(...skipping 47 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
324 } | 324 } |
325 | 325 |
326 rtc::Thread* PeerConnectionFactory::worker_thread() { | 326 rtc::Thread* PeerConnectionFactory::worker_thread() { |
327 return worker_thread_; | 327 return worker_thread_; |
328 } | 328 } |
329 | 329 |
330 rtc::Thread* PeerConnectionFactory::network_thread() { | 330 rtc::Thread* PeerConnectionFactory::network_thread() { |
331 return network_thread_; | 331 return network_thread_; |
332 } | 332 } |
333 | 333 |
334 std::unique_ptr<RtcEventLog> PeerConnectionFactory::CreateRtcEventLog_w() { | |
335 return event_log_factory_ ? event_log_factory_->CreateRtcEventLog() | |
336 : rtc::MakeUnique<RtcEventLogNullImpl>(); | |
337 } | |
338 | |
334 std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w( | 339 std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w( |
335 RtcEventLog* event_log) { | 340 RtcEventLog* event_log) { |
336 const int kMinBandwidthBps = 30000; | 341 const int kMinBandwidthBps = 30000; |
337 const int kStartBandwidthBps = 300000; | 342 const int kStartBandwidthBps = 300000; |
338 const int kMaxBandwidthBps = 2000000; | 343 const int kMaxBandwidthBps = 2000000; |
339 | 344 |
340 webrtc::Call::Config call_config(event_log); | 345 webrtc::Call::Config call_config(event_log); |
341 if (!channel_manager_->media_engine() || !call_factory_) { | 346 if (!channel_manager_->media_engine() || !call_factory_) { |
342 return nullptr; | 347 return nullptr; |
343 } | 348 } |
344 call_config.audio_state = channel_manager_->media_engine()->GetAudioState(); | 349 call_config.audio_state = channel_manager_->media_engine()->GetAudioState(); |
345 call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps; | 350 call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps; |
346 call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps; | 351 call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps; |
347 call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps; | 352 call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps; |
348 | 353 |
349 return std::unique_ptr<Call>(call_factory_->CreateCall(call_config)); | 354 return std::unique_ptr<Call>(call_factory_->CreateCall(call_config)); |
350 } | 355 } |
351 | 356 |
352 } // namespace webrtc | 357 } // namespace webrtc |
OLD | NEW |