talk/app/webrtc/peerconnectionfactory.h - Issue 1670153003: Introduce struct MediaConfig, with construction-time settings.

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Issue 1670153003: Introduce struct MediaConfig, with construction-time settings. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Added new peerconnection test for dscp constraint. Created 4 years, 10 months ago

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1 /*	1 /*

2 * libjingle	2 * libjingle

3 * Copyright 2011 Google Inc.	3 * Copyright 2011 Google Inc.

4 *	4 *

5 * Redistribution and use in source and binary forms, with or without	5 * Redistribution and use in source and binary forms, with or without

6 * modification, are permitted provided that the following conditions are met:	6 * modification, are permitted provided that the following conditions are met:

7 *	7 *

8 * 1. Redistributions of source code must retain the above copyright notice,	8 * 1. Redistributions of source code must retain the above copyright notice,

9 * this list of conditions and the following disclaimer.	9 * this list of conditions and the following disclaimer.

10 * 2. Redistributions in binary form must reproduce the above copyright notice,	10 * 2. Redistributions in binary form must reproduce the above copyright notice,

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80	80

81 rtc::scoped_refptr<AudioTrackInterface>	81 rtc::scoped_refptr<AudioTrackInterface>

82 CreateAudioTrack(const std::string& id,	82 CreateAudioTrack(const std::string& id,

83 AudioSourceInterface* audio_source) override;	83 AudioSourceInterface* audio_source) override;

84	84

85 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;	85 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;

86 void StopAecDump() override;	86 void StopAecDump() override;

87 bool StartRtcEventLog(rtc::PlatformFile file) override;	87 bool StartRtcEventLog(rtc::PlatformFile file) override;

88 void StopRtcEventLog() override;	88 void StopRtcEventLog() override;

89	89

90 virtual webrtc::MediaControllerInterface* CreateMediaController() const;	90 virtual webrtc::MediaControllerInterface* CreateMediaController(

	91 const cricket::MediaConfig &config) const;

91 virtual rtc::Thread* signaling_thread();	92 virtual rtc::Thread* signaling_thread();

92 virtual rtc::Thread* worker_thread();	93 virtual rtc::Thread* worker_thread();

93 const Options& options() const { return options_; }	94 const Options& options() const { return options_; }

94	95

95 protected:	96 protected:

96 PeerConnectionFactory();	97 PeerConnectionFactory();

97 PeerConnectionFactory(	98 PeerConnectionFactory(

98 rtc::Thread* worker_thread,	99 rtc::Thread* worker_thread,

99 rtc::Thread* signaling_thread,	100 rtc::Thread* signaling_thread,

100 AudioDeviceModule* default_adm,	101 AudioDeviceModule* default_adm,

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123 video_decoder_factory_;	124 video_decoder_factory_;

124 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;	125 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;

125 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;	126 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;

126	127

127 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;	128 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;

128 };	129 };

129	130

130 } // namespace webrtc	131 } // namespace webrtc

131	132

132 #endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_	133 #endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_

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