| /* |
| * 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 <functional> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/strings/string_view.h" |
| #include "api/jsep.h" |
| #include "api/media_types.h" |
| #include "media/base/media_channel.h" |
| #include "pc/rtp_transport_internal.h" |
| |
| namespace webrtc { |
| class Call; |
| class VideoBitrateAllocatorFactory; |
| } // namespace webrtc |
| |
| namespace cricket { |
| |
| class VoiceChannel; |
| class VideoChannel; |
| class MediaContentDescription; |
| struct MediaConfig; |
| |
| // A Channel is a construct that groups media streams of the same type |
| // (audio or video), both outgoing and incoming. |
| // When the PeerConnection API is used, a Channel corresponds one to one |
| // to an RtpTransceiver. |
| // When Unified Plan is used, there can only be at most one outgoing and |
| // one incoming stream. With Plan B, there can be more than one. |
| |
| // ChannelInterface contains methods common to voice and video channels. |
| // As more methods are added to BaseChannel, they should be included in the |
| // interface as well. |
| // TODO(bugs.webrtc.org/13931): Merge this class into RtpTransceiver. |
| class ChannelInterface { |
| public: |
| virtual ~ChannelInterface() = default; |
| virtual cricket::MediaType media_type() const = 0; |
| |
| virtual VideoChannel* AsVideoChannel() = 0; |
| virtual VoiceChannel* AsVoiceChannel() = 0; |
| |
| virtual MediaSendChannelInterface* media_send_channel() = 0; |
| // Typecasts of media_channel(). Will cause an exception if the |
| // channel is of the wrong type. |
| virtual VideoMediaSendChannelInterface* video_media_send_channel() = 0; |
| virtual VoiceMediaSendChannelInterface* voice_media_send_channel() = 0; |
| virtual MediaReceiveChannelInterface* media_receive_channel() = 0; |
| // Typecasts of media_channel(). Will cause an exception if the |
| // channel is of the wrong type. |
| virtual VideoMediaReceiveChannelInterface* video_media_receive_channel() = 0; |
| virtual VoiceMediaReceiveChannelInterface* voice_media_receive_channel() = 0; |
| |
| // Returns a string view for the transport name. Fetching the transport name |
| // must be done on the network thread only and note that the lifetime of |
| // the returned object should be assumed to only be the calling scope. |
| // TODO(deadbeef): This is redundant; remove this. |
| virtual absl::string_view transport_name() const = 0; |
| |
| // TODO(tommi): Change return type to string_view. |
| virtual const std::string& mid() 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; |
| virtual void SetFirstPacketSentCallback(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; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // PC_CHANNEL_INTERFACE_H_ |