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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 <cstdint>
#include <optional>
#include "absl/base/attributes.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 {};
enum class ABSL_MUST_USE_RESULT VoipResult {
// kOk indicates the function was successfully invoked with no error.
kOk,
// kInvalidArgument indicates the caller specified an invalid argument, such
// as an invalid ChannelId.
kInvalidArgument,
// kFailedPrecondition indicates that the operation was failed due to not
// satisfying prerequisite such as not setting codec type before sending.
kFailedPrecondition,
// kInternal is used to indicate various internal failures that are not the
// caller's fault. Further detail is commented on each function that uses this
// return value.
kInternal,
};
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 a ChannelId created for caller to handle subsequent Channel
// operations.
virtual ChannelId CreateChannel(Transport* transport,
std::optional<uint32_t> local_ssrc) = 0;
// Releases `channel_id` that no longer has any use.
// Returns following VoipResult;
// kOk - `channel_id` is released.
// kInvalidArgument - `channel_id` is invalid.
// kInternal - Fails to stop audio output device.
virtual VoipResult ReleaseChannel(ChannelId channel_id) = 0;
// Starts sending on `channel_id`. This starts microphone if not started yet.
// Returns following VoipResult;
// kOk - Channel successfully started to send.
// kInvalidArgument - `channel_id` is invalid.
// kFailedPrecondition - Missing prerequisite on VoipCodec::SetSendCodec.
// kInternal - initialization has failed on selected microphone.
virtual VoipResult 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 following VoipResult;
// kOk - Channel successfully stopped to send.
// kInvalidArgument - `channel_id` is invalid.
// kInternal - Failed to stop the active microphone device.
virtual VoipResult StopSend(ChannelId channel_id) = 0;
// Starts playing on speaker device for `channel_id`.
// This will start underlying platform speaker device if not started.
// Returns following VoipResult;
// kOk - Channel successfully started to play out.
// kInvalidArgument - `channel_id` is invalid.
// kFailedPrecondition - Missing prerequisite on VoipCodec::SetReceiveCodecs.
// kInternal - Failed to initializate the selected speaker device.
virtual VoipResult StartPlayout(ChannelId channel_id) = 0;
// Stops playing on speaker device for `channel_id`.
// Returns following VoipResult;
// kOk - Channel successfully stopped t play out.
// kInvalidArgument - `channel_id` is invalid.
virtual VoipResult StopPlayout(ChannelId channel_id) = 0;
protected:
virtual ~VoipBase() = default;
};
} // namespace webrtc
#endif // API_VOIP_VOIP_BASE_H_