91 lines
3.7 KiB
C
91 lines
3.7 KiB
C
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/*
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* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_VOIP_VOIP_BASE_H_
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#define API_VOIP_VOIP_BASE_H_
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#include "absl/types/optional.h"
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namespace webrtc {
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class Transport;
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// VoipBase interface
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//
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// VoipBase provides a management interface on a media session using a
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// concept called 'channel'. A channel represents an interface handle
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// for application to request various media session operations. This
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// notion of channel is used throughout other interfaces as well.
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//
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// Underneath the interface, a channel id is mapped into an audio session
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// object that is capable of sending and receiving a single RTP stream with
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// another media endpoint. It's possible to create and use multiple active
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// channels simultaneously which would mean that particular application
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// session has RTP streams with multiple remote endpoints.
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//
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// A typical example for the usage context is outlined in VoipEngine
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// header file.
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enum class ChannelId : int {};
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class VoipBase {
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public:
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// Creates a channel.
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// Each channel handle maps into one audio media session where each has
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// its own separate module for send/receive rtp packet with one peer.
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// Caller must set |transport|, webrtc::Transport callback pointer to
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// receive rtp/rtcp packets from corresponding media session in VoIP engine.
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// VoipEngine framework expects applications to handle network I/O directly
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// and injection for incoming RTP from remote endpoint is handled via
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// VoipNetwork interface. |local_ssrc| is optional and when local_ssrc is not
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// set, some random value will be used by voip engine.
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// Returns value is optional as to indicate the failure to create channel.
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virtual absl::optional<ChannelId> CreateChannel(
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Transport* transport,
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absl::optional<uint32_t> local_ssrc) = 0;
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// Releases |channel_id| that no longer has any use.
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virtual void ReleaseChannel(ChannelId channel_id) = 0;
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// Starts sending on |channel_id|. This will start microphone if not started
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// yet. Returns false if initialization has failed on selected microphone
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// device. API is subject to expand to reflect error condition to application
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// later.
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virtual bool StartSend(ChannelId channel_id) = 0;
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// Stops sending on |channel_id|. If this is the last active channel, it will
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// stop microphone input from underlying audio platform layer.
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// Returns false if termination logic has failed on selected microphone
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// device. API is subject to expand to reflect error condition to application
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// later.
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virtual bool StopSend(ChannelId channel_id) = 0;
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// Starts playing on speaker device for |channel_id|.
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// This will start underlying platform speaker device if not started.
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// Returns false if initialization has failed
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// on selected speaker device. API is subject to expand to reflect error
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// condition to application later.
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virtual bool StartPlayout(ChannelId channel_id) = 0;
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// Stops playing on speaker device for |channel_id|.
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// If this is the last active channel playing, then it will stop speaker
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// from the platform layer.
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// Returns false if termination logic has failed on selected speaker device.
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// API is subject to expand to reflect error condition to application later.
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virtual bool StopPlayout(ChannelId channel_id) = 0;
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protected:
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virtual ~VoipBase() = default;
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};
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} // namespace webrtc
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#endif // API_VOIP_VOIP_BASE_H_
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