| /* |
| * Copyright 2018 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef PC_CHANNEL_INTERFACE_H_ |
| #define PC_CHANNEL_INTERFACE_H_ |
| |
| #include <string> |
| #include <vector> |
| |
| #include "api/jsep.h" |
| #include "api/media_types.h" |
| #include "media/base/media_channel.h" |
| #include "pc/rtp_transport_internal.h" |
| |
| namespace cricket { |
| |
| class MediaContentDescription; |
| |
| // ChannelInterface contains methods common to voice, video and data channels. |
| // As more methods are added to BaseChannel, they should be included in the |
| // interface as well. |
| class ChannelInterface { |
| public: |
| virtual cricket::MediaType media_type() const = 0; |
| |
| virtual MediaChannel* media_channel() const = 0; |
| |
| // TODO(deadbeef): This is redundant; remove this. |
| virtual const std::string& transport_name() const = 0; |
| |
| virtual const std::string& content_name() const = 0; |
| |
| // Enables or disables this channel |
| virtual void Enable(bool enable) = 0; |
| |
| // Used for latency measurements. |
| virtual void SetFirstPacketReceivedCallback( |
| std::function<void()> callback) = 0; |
| |
| // Channel control |
| virtual bool SetLocalContent(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) = 0; |
| virtual bool SetRemoteContent(const MediaContentDescription* content, |
| webrtc::SdpType type, |
| std::string* error_desc) = 0; |
| virtual bool SetPayloadTypeDemuxingEnabled(bool enabled) = 0; |
| |
| // Access to the local and remote streams that were set on the channel. |
| virtual const std::vector<StreamParams>& local_streams() const = 0; |
| virtual const std::vector<StreamParams>& remote_streams() const = 0; |
| |
| // Set an RTP level transport. |
| // Some examples: |
| // * An RtpTransport without encryption. |
| // * An SrtpTransport for SDES. |
| // * A DtlsSrtpTransport for DTLS-SRTP. |
| virtual bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) = 0; |
| |
| protected: |
| virtual ~ChannelInterface() = default; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // PC_CHANNEL_INTERFACE_H_ |