c63bf10790
Bug: webrtc:11251 Change-Id: I26b6915d3ad6bb5a50f9898a6866889867fd53f5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169000 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30594}
82 lines
3.2 KiB
C++
82 lines
3.2 KiB
C++
//
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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 "api/call/transport.h"
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namespace webrtc {
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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 handle 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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class VoipBase {
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public:
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// This config enables application to set webrtc::Transport callback pointer
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// to receive rtp/rtcp packets from corresponding media session in VoIP
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// engine. VoipEngine framework expects applications to handle network I/O
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// directly and injection for incoming RTP from remote endpoint is handled
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// via VoipNetwork interface.
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struct Config {
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Transport* transport = nullptr;
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uint32_t local_ssrc = 0;
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};
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// Create a channel handle.
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// Valid handle value is zero or greater integer whereas -1 represents error
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// during media session construction. Each channel handle maps into one
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// audio media session where each has its own separate module for
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// send/receive rtp packet with one peer.
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virtual int CreateChannel(const Config& config) = 0;
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// Following methods return boolean to indicate if the operation is succeeded.
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// API is subject to expand to reflect error condition to application later.
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// Release |channel| that has served the purpose.
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// Released channel handle will be re-allocated again. Invoking
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// an operation on released channel will lead to undefined behavior.
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virtual bool ReleaseChannel(int channel) = 0;
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// Start sending on |channel|. This will start microphone if first to start.
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virtual bool StartSend(int channel) = 0;
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// Stop sending on |channel|. 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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virtual bool StopSend(int channel) = 0;
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// Start playing on speaker device for |channel|.
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// This will start underlying platform speaker device if not started.
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virtual bool StartPlayout(int channel) = 0;
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// Stop playing on speaker device for |channel|. If this is the last
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// active channel playing, then it will stop speaker from the platform layer.
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virtual bool StopPlayout(int channel) = 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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