| // |
| // Copyright (c) 2020 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 API_VOIP_VOIP_BASE_H_ |
| #define API_VOIP_VOIP_BASE_H_ |
| |
| #include "api/call/transport.h" |
| |
| namespace webrtc { |
| |
| // VoipBase interface |
| // |
| // VoipBase provides a management interface on a media session using a |
| // concept called 'channel'. A channel represents an interface handle |
| // for application to request various media session operations. This |
| // notion of channel is used throughout other interfaces as well. |
| // |
| // Underneath the interface, a channel handle is mapped into an audio session |
| // object that is capable of sending and receiving a single RTP stream with |
| // another media endpoint. It's possible to create and use multiple active |
| // channels simultaneously which would mean that particular application |
| // session has RTP streams with multiple remote endpoints. |
| // |
| // A typical example for the usage context is outlined in VoipEngine |
| // header file. |
| class VoipBase { |
| public: |
| // This config enables application to set webrtc::Transport callback pointer |
| // to receive rtp/rtcp packets from corresponding media session in VoIP |
| // engine. VoipEngine framework expects applications to handle network I/O |
| // directly and injection for incoming RTP from remote endpoint is handled |
| // via VoipNetwork interface. |
| struct Config { |
| Transport* transport = nullptr; |
| uint32_t local_ssrc = 0; |
| }; |
| |
| // Create a channel handle. |
| // Valid handle value is zero or greater integer whereas -1 represents error |
| // during media session construction. Each channel handle maps into one |
| // audio media session where each has its own separate module for |
| // send/receive rtp packet with one peer. |
| virtual int CreateChannel(const Config& config) = 0; |
| |
| // Following methods return boolean to indicate if the operation is succeeded. |
| // API is subject to expand to reflect error condition to application later. |
| |
| // Release |channel| that has served the purpose. |
| // Released channel handle will be re-allocated again. Invoking |
| // an operation on released channel will lead to undefined behavior. |
| virtual bool ReleaseChannel(int channel) = 0; |
| |
| // Start sending on |channel|. This will start microphone if first to start. |
| virtual bool StartSend(int channel) = 0; |
| |
| // Stop sending on |channel|. If this is the last active channel, it will |
| // stop microphone input from underlying audio platform layer. |
| virtual bool StopSend(int channel) = 0; |
| |
| // Start playing on speaker device for |channel|. |
| // This will start underlying platform speaker device if not started. |
| virtual bool StartPlayout(int channel) = 0; |
| |
| // Stop playing on speaker device for |channel|. If this is the last |
| // active channel playing, then it will stop speaker from the platform layer. |
| virtual bool StopPlayout(int channel) = 0; |
| |
| protected: |
| virtual ~VoipBase() = default; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_VOIP_VOIP_BASE_H_ |