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//
// 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 "third_party/absl/types/optional.h"
namespace webrtc {
class Transport;
// 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 id 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.
enum class ChannelId : int {};
class VoipBase {
public:
// Creates a channel.
// Each channel handle maps into one audio media session where each has
// its own separate module for send/receive rtp packet with one peer.
// Caller must set |transport|, 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. |local_ssrc| is optional and when local_ssrc is not
// set, some random value will be used by voip engine.
// Returns value is optional as to indicate the failure to create channel.
virtual absl::optional<ChannelId> CreateChannel(
Transport* transport,
absl::optional<uint32_t> local_ssrc) = 0;
// Releases |channel_id| that has served the purpose.
// Released channel will be re-allocated again that invoking operations
// on released |channel_id| will lead to undefined behavior.
virtual void ReleaseChannel(ChannelId channel_id) = 0;
// Starts sending on |channel_id|. This will start microphone if first to
// start. Returns false if initialization has failed on selected microphone
// device. API is subject to expand to reflect error condition to application
// later.
virtual bool StartSend(ChannelId channel_id) = 0;
// Stops sending on |channel_id|. If this is the last active channel, it will
// stop microphone input from underlying audio platform layer.
// Returns false if termination logic has failed on selected microphone
// device. API is subject to expand to reflect error condition to application
// later.
virtual bool StopSend(ChannelId channel_id) = 0;
// Starts playing on speaker device for |channel_id|.
// This will start underlying platform speaker device if not started.
// Returns false if initialization has failed
// on selected speaker device. API is subject to expand to reflect error
// condition to application later.
virtual bool StartPlayout(ChannelId channel_id) = 0;
// Stops playing on speaker device for |channel_id|.
// If this is the last active channel playing, then it will stop speaker
// from the platform layer.
// Returns false if termination logic has failed on selected speaker device.
// API is subject to expand to reflect error condition to application later.
virtual bool StopPlayout(ChannelId channel_id) = 0;
protected:
virtual ~VoipBase() = default;
};
} // namespace webrtc
#endif // API_VOIP_VOIP_BASE_H_