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| 1 /* | 1 /* |
| 2 * Copyright 2011 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2011 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 #ifndef WEBRTC_PC_PEERCONNECTIONFACTORY_H_ | 11 #ifndef WEBRTC_PC_PEERCONNECTIONFACTORY_H_ |
| 12 #define WEBRTC_PC_PEERCONNECTIONFACTORY_H_ | 12 #define WEBRTC_PC_PEERCONNECTIONFACTORY_H_ |
| 13 | 13 |
| 14 #include <memory> | 14 #include <memory> |
| 15 #include <string> | 15 #include <string> |
| 16 | 16 |
| 17 #include "webrtc/api/mediastreaminterface.h" | 17 #include "webrtc/api/mediastreaminterface.h" |
| 18 #include "webrtc/api/peerconnectioninterface.h" | 18 #include "webrtc/api/peerconnectioninterface.h" |
| 19 #include "webrtc/base/scoped_ref_ptr.h" | 19 #include "webrtc/base/scoped_ref_ptr.h" |
| 20 #include "webrtc/base/thread.h" | 20 #include "webrtc/base/thread.h" |
| 21 #include "webrtc/base/rtccertificategenerator.h" | 21 #include "webrtc/base/rtccertificategenerator.h" |
| 22 #include "webrtc/pc/channelmanager.h" | 22 #include "webrtc/pc/channelmanager.h" |
| 23 #include "webrtc/pc/mediacontroller.h" | |
| 24 | 23 |
| 25 namespace rtc { | 24 namespace rtc { |
| 26 class BasicNetworkManager; | 25 class BasicNetworkManager; |
| 27 class BasicPacketSocketFactory; | 26 class BasicPacketSocketFactory; |
| 28 } | 27 } |
| 29 | 28 |
| 30 namespace webrtc { | 29 namespace webrtc { |
| 31 | 30 |
| 32 class RtcEventLog; | 31 class RtcEventLog; |
| 33 | 32 |
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| 89 // TODO(ivoc) Remove after Chrome is updated. | 88 // TODO(ivoc) Remove after Chrome is updated. |
| 90 bool StartRtcEventLog(rtc::PlatformFile file) override { return false; } | 89 bool StartRtcEventLog(rtc::PlatformFile file) override { return false; } |
| 91 // TODO(ivoc) Remove after Chrome is updated. | 90 // TODO(ivoc) Remove after Chrome is updated. |
| 92 bool StartRtcEventLog(rtc::PlatformFile file, | 91 bool StartRtcEventLog(rtc::PlatformFile file, |
| 93 int64_t max_size_bytes) override { | 92 int64_t max_size_bytes) override { |
| 94 return false; | 93 return false; |
| 95 } | 94 } |
| 96 // TODO(ivoc) Remove after Chrome is updated. | 95 // TODO(ivoc) Remove after Chrome is updated. |
| 97 void StopRtcEventLog() override {} | 96 void StopRtcEventLog() override {} |
| 98 | 97 |
| 99 virtual webrtc::MediaControllerInterface* CreateMediaController( | |
| 100 const cricket::MediaConfig& config, | |
| 101 RtcEventLog* event_log) const; | |
| 102 virtual cricket::TransportController* CreateTransportController( | 98 virtual cricket::TransportController* CreateTransportController( |
| 103 cricket::PortAllocator* port_allocator, | 99 cricket::PortAllocator* port_allocator, |
| 104 bool redetermine_role_on_ice_restart); | 100 bool redetermine_role_on_ice_restart); |
| 101 virtual cricket::ChannelManager* channel_manager(); |
| 105 virtual rtc::Thread* signaling_thread(); | 102 virtual rtc::Thread* signaling_thread(); |
| 106 virtual rtc::Thread* worker_thread(); | 103 virtual rtc::Thread* worker_thread(); |
| 107 virtual rtc::Thread* network_thread(); | 104 virtual rtc::Thread* network_thread(); |
| 108 const Options& options() const { return options_; } | 105 const Options& options() const { return options_; } |
| 109 | 106 |
| 110 protected: | 107 protected: |
| 111 PeerConnectionFactory( | 108 PeerConnectionFactory( |
| 112 rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, | 109 rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, |
| 113 rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory); | 110 rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory); |
| 114 PeerConnectionFactory( | 111 PeerConnectionFactory( |
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| 145 std::unique_ptr<rtc::BasicNetworkManager> default_network_manager_; | 142 std::unique_ptr<rtc::BasicNetworkManager> default_network_manager_; |
| 146 std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_; | 143 std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_; |
| 147 // External audio mixer. This can be NULL. In that case, internal audio mixer | 144 // External audio mixer. This can be NULL. In that case, internal audio mixer |
| 148 // will be created and used. | 145 // will be created and used. |
| 149 rtc::scoped_refptr<AudioMixer> external_audio_mixer_; | 146 rtc::scoped_refptr<AudioMixer> external_audio_mixer_; |
| 150 }; | 147 }; |
| 151 | 148 |
| 152 } // namespace webrtc | 149 } // namespace webrtc |
| 153 | 150 |
| 154 #endif // WEBRTC_PC_PEERCONNECTIONFACTORY_H_ | 151 #endif // WEBRTC_PC_PEERCONNECTIONFACTORY_H_ |
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