| Index: webrtc/pc/channelmanager.h
|
| diff --git a/webrtc/pc/channelmanager.h b/webrtc/pc/channelmanager.h
|
| index 879ea4d39b8194e27166b4379fbf20b87c8c68cd..61a732da8928398af1a1b069497ad2abaaaa7a5f 100644
|
| --- a/webrtc/pc/channelmanager.h
|
| +++ b/webrtc/pc/channelmanager.h
|
| @@ -39,12 +39,12 @@ class ChannelManager {
|
| public:
|
| // For testing purposes. Allows the media engine and data media
|
| // engine and dev manager to be mocks. The ChannelManager takes
|
| - // ownership of these objects.
|
| - ChannelManager(MediaEngineInterface* me,
|
| - DataEngineInterface* dme,
|
| + // ownership of the objects passed as unique_ptrs.
|
| + ChannelManager(std::unique_ptr<MediaEngineInterface> me,
|
| + std::unique_ptr<DataEngineInterface> dme,
|
| rtc::Thread* worker_and_network);
|
| // Same as above, but gives an easier default DataEngine.
|
| - ChannelManager(MediaEngineInterface* me,
|
| + ChannelManager(std::unique_ptr<MediaEngineInterface> me,
|
| rtc::Thread* worker,
|
| rtc::Thread* network);
|
| ~ChannelManager();
|
| @@ -94,8 +94,15 @@ class ChannelManager {
|
| DtlsTransportInternal* rtcp_transport,
|
| rtc::Thread* signaling_thread,
|
| const std::string& content_name,
|
| - const std::string* bundle_transport_name,
|
| - bool rtcp_mux_required,
|
| + bool srtp_required,
|
| + const AudioOptions& options);
|
| + // Version of the above that takes PacketTransportInternal.
|
| + VoiceChannel* CreateVoiceChannel(
|
| + webrtc::MediaControllerInterface* media_controller,
|
| + rtc::PacketTransportInternal* rtp_transport,
|
| + rtc::PacketTransportInternal* rtcp_transport,
|
| + rtc::Thread* signaling_thread,
|
| + const std::string& content_name,
|
| bool srtp_required,
|
| const AudioOptions& options);
|
| // Destroys a voice channel created with the Create API.
|
| @@ -108,8 +115,15 @@ class ChannelManager {
|
| DtlsTransportInternal* rtcp_transport,
|
| rtc::Thread* signaling_thread,
|
| const std::string& content_name,
|
| - const std::string* bundle_transport_name,
|
| - bool rtcp_mux_required,
|
| + bool srtp_required,
|
| + const VideoOptions& options);
|
| + // Version of the above that takes PacketTransportInternal.
|
| + VideoChannel* CreateVideoChannel(
|
| + webrtc::MediaControllerInterface* media_controller,
|
| + rtc::PacketTransportInternal* rtp_transport,
|
| + rtc::PacketTransportInternal* rtcp_transport,
|
| + rtc::Thread* signaling_thread,
|
| + const std::string& content_name,
|
| bool srtp_required,
|
| const VideoOptions& options);
|
| // Destroys a video channel created with the Create API.
|
| @@ -120,8 +134,6 @@ class ChannelManager {
|
| DtlsTransportInternal* rtcp_transport,
|
| rtc::Thread* signaling_thread,
|
| const std::string& content_name,
|
| - const std::string* bundle_transport_name,
|
| - bool rtcp_mux_required,
|
| bool srtp_required);
|
| // Destroys a data channel created with the Create API.
|
| void DestroyRtpDataChannel(RtpDataChannel* data_channel);
|
| @@ -157,8 +169,8 @@ class ChannelManager {
|
| typedef std::vector<VideoChannel*> VideoChannels;
|
| typedef std::vector<RtpDataChannel*> RtpDataChannels;
|
|
|
| - void Construct(MediaEngineInterface* me,
|
| - DataEngineInterface* dme,
|
| + void Construct(std::unique_ptr<MediaEngineInterface> me,
|
| + std::unique_ptr<DataEngineInterface> dme,
|
| rtc::Thread* worker_thread,
|
| rtc::Thread* network_thread);
|
| bool InitMediaEngine_w();
|
| @@ -167,23 +179,23 @@ class ChannelManager {
|
| bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options);
|
| VoiceChannel* CreateVoiceChannel_w(
|
| webrtc::MediaControllerInterface* media_controller,
|
| - DtlsTransportInternal* rtp_transport,
|
| - DtlsTransportInternal* rtcp_transport,
|
| + DtlsTransportInternal* rtp_dtls_transport,
|
| + DtlsTransportInternal* rtcp_dtls_transport,
|
| + rtc::PacketTransportInternal* rtp_packet_transport,
|
| + rtc::PacketTransportInternal* rtcp_packet_transport,
|
| rtc::Thread* signaling_thread,
|
| const std::string& content_name,
|
| - const std::string* bundle_transport_name,
|
| - bool rtcp_mux_required,
|
| bool srtp_required,
|
| const AudioOptions& options);
|
| void DestroyVoiceChannel_w(VoiceChannel* voice_channel);
|
| VideoChannel* CreateVideoChannel_w(
|
| webrtc::MediaControllerInterface* media_controller,
|
| - DtlsTransportInternal* rtp_transport,
|
| - DtlsTransportInternal* rtcp_transport,
|
| + DtlsTransportInternal* rtp_dtls_transport,
|
| + DtlsTransportInternal* rtcp_dtls_transport,
|
| + rtc::PacketTransportInternal* rtp_packet_transport,
|
| + rtc::PacketTransportInternal* rtcp_packet_transport,
|
| rtc::Thread* signaling_thread,
|
| const std::string& content_name,
|
| - const std::string* bundle_transport_name,
|
| - bool rtcp_mux_required,
|
| bool srtp_required,
|
| const VideoOptions& options);
|
| void DestroyVideoChannel_w(VideoChannel* video_channel);
|
| @@ -193,8 +205,6 @@ class ChannelManager {
|
| DtlsTransportInternal* rtcp_transport,
|
| rtc::Thread* signaling_thread,
|
| const std::string& content_name,
|
| - const std::string* bundle_transport_name,
|
| - bool rtcp_mux_required,
|
| bool srtp_required);
|
| void DestroyRtpDataChannel_w(RtpDataChannel* data_channel);
|
|
|
|
|