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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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 |
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900 ERROR_PLAY_SRTP_ERROR, // Generic SRTP failure. | 900 ERROR_PLAY_SRTP_ERROR, // Generic SRTP failure. |
901 ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets. | 901 ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets. |
902 ERROR_PLAY_SRTP_REPLAY, // Packet replay detected. | 902 ERROR_PLAY_SRTP_REPLAY, // Packet replay detected. |
903 }; | 903 }; |
904 | 904 |
905 VoiceMediaChannel() {} | 905 VoiceMediaChannel() {} |
906 VoiceMediaChannel(const MediaConfig& config) : MediaChannel(config) {} | 906 VoiceMediaChannel(const MediaConfig& config) : MediaChannel(config) {} |
907 virtual ~VoiceMediaChannel() {} | 907 virtual ~VoiceMediaChannel() {} |
908 virtual bool SetSendParameters(const AudioSendParameters& params) = 0; | 908 virtual bool SetSendParameters(const AudioSendParameters& params) = 0; |
909 virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0; | 909 virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0; |
910 virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const = 0; | 910 virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0; |
911 virtual bool SetRtpParameters(uint32_t ssrc, | 911 virtual bool SetRtpSendParameters( |
912 const webrtc::RtpParameters& parameters) = 0; | 912 uint32_t ssrc, |
| 913 const webrtc::RtpParameters& parameters) = 0; |
| 914 virtual webrtc::RtpParameters GetRtpReceiveParameters( |
| 915 uint32_t ssrc) const = 0; |
| 916 virtual bool SetRtpReceiveParameters( |
| 917 uint32_t ssrc, |
| 918 const webrtc::RtpParameters& parameters) = 0; |
913 // Starts or stops playout of received audio. | 919 // Starts or stops playout of received audio. |
914 virtual bool SetPlayout(bool playout) = 0; | 920 virtual bool SetPlayout(bool playout) = 0; |
915 // Starts or stops sending (and potentially capture) of local audio. | 921 // Starts or stops sending (and potentially capture) of local audio. |
916 virtual void SetSend(bool send) = 0; | 922 virtual void SetSend(bool send) = 0; |
917 // Configure stream for sending. | 923 // Configure stream for sending. |
918 virtual bool SetAudioSend(uint32_t ssrc, | 924 virtual bool SetAudioSend(uint32_t ssrc, |
919 bool enable, | 925 bool enable, |
920 const AudioOptions* options, | 926 const AudioOptions* options, |
921 AudioSource* source) = 0; | 927 AudioSource* source) = 0; |
922 // Gets current energy levels for all incoming streams. | 928 // Gets current energy levels for all incoming streams. |
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979 ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets. | 985 ERROR_PLAY_SRTP_AUTH_FAILED, // Failed to authenticate packets. |
980 ERROR_PLAY_SRTP_REPLAY, // Packet replay detected. | 986 ERROR_PLAY_SRTP_REPLAY, // Packet replay detected. |
981 }; | 987 }; |
982 | 988 |
983 VideoMediaChannel() {} | 989 VideoMediaChannel() {} |
984 VideoMediaChannel(const MediaConfig& config) : MediaChannel(config) {} | 990 VideoMediaChannel(const MediaConfig& config) : MediaChannel(config) {} |
985 virtual ~VideoMediaChannel() {} | 991 virtual ~VideoMediaChannel() {} |
986 | 992 |
987 virtual bool SetSendParameters(const VideoSendParameters& params) = 0; | 993 virtual bool SetSendParameters(const VideoSendParameters& params) = 0; |
988 virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0; | 994 virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0; |
989 virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const = 0; | 995 virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0; |
990 virtual bool SetRtpParameters(uint32_t ssrc, | 996 virtual bool SetRtpSendParameters( |
991 const webrtc::RtpParameters& parameters) = 0; | 997 uint32_t ssrc, |
| 998 const webrtc::RtpParameters& parameters) = 0; |
| 999 virtual webrtc::RtpParameters GetRtpReceiveParameters( |
| 1000 uint32_t ssrc) const = 0; |
| 1001 virtual bool SetRtpReceiveParameters( |
| 1002 uint32_t ssrc, |
| 1003 const webrtc::RtpParameters& parameters) = 0; |
992 // Gets the currently set codecs/payload types to be used for outgoing media. | 1004 // Gets the currently set codecs/payload types to be used for outgoing media. |
993 virtual bool GetSendCodec(VideoCodec* send_codec) = 0; | 1005 virtual bool GetSendCodec(VideoCodec* send_codec) = 0; |
994 // Starts or stops transmission (and potentially capture) of local video. | 1006 // Starts or stops transmission (and potentially capture) of local video. |
995 virtual bool SetSend(bool send) = 0; | 1007 virtual bool SetSend(bool send) = 0; |
996 // Configure stream for sending. | 1008 // Configure stream for sending. |
997 virtual bool SetVideoSend(uint32_t ssrc, | 1009 virtual bool SetVideoSend(uint32_t ssrc, |
998 bool enable, | 1010 bool enable, |
999 const VideoOptions* options) = 0; | 1011 const VideoOptions* options) = 0; |
1000 // Sets the sink object to be used for the specified stream. | 1012 // Sets the sink object to be used for the specified stream. |
1001 // If SSRC is 0, the renderer is used for the 'default' stream. | 1013 // If SSRC is 0, the renderer is used for the 'default' stream. |
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1127 // Signal when the media channel is ready to send the stream. Arguments are: | 1139 // Signal when the media channel is ready to send the stream. Arguments are: |
1128 // writable(bool) | 1140 // writable(bool) |
1129 sigslot::signal1<bool> SignalReadyToSend; | 1141 sigslot::signal1<bool> SignalReadyToSend; |
1130 // Signal for notifying that the remote side has closed the DataChannel. | 1142 // Signal for notifying that the remote side has closed the DataChannel. |
1131 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1143 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1132 }; | 1144 }; |
1133 | 1145 |
1134 } // namespace cricket | 1146 } // namespace cricket |
1135 | 1147 |
1136 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1148 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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