a8eb756a34
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
96 lines
3.3 KiB
C++
96 lines
3.3 KiB
C++
/*
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* Copyright (c) 2011 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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// This sub-API supports the following functionalities:
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//
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// - External protocol support.
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// - Packet timeout notification.
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// - Dead-or-Alive connection observations.
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//
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// Usage example, omitting error checking:
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//
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// using namespace webrtc;
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// VoiceEngine* voe = VoiceEngine::Create();
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// VoEBase* base = VoEBase::GetInterface(voe);
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// VoENetwork* netw = VoENetwork::GetInterface(voe);
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// base->Init();
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// int ch = base->CreateChannel();
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// ...
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// netw->SetPeriodicDeadOrAliveStatus(ch, true);
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// ...
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// base->DeleteChannel(ch);
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// base->Terminate();
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// base->Release();
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// netw->Release();
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// VoiceEngine::Delete(voe);
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//
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#ifndef WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
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#define WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
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#include "webrtc/api/call/transport.h"
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#include "webrtc/common_types.h"
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namespace webrtc {
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class VoiceEngine;
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// VoENetwork
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class WEBRTC_DLLEXPORT VoENetwork {
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public:
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// Factory for the VoENetwork sub-API. Increases an internal
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// reference counter if successful. Returns NULL if the API is not
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// supported or if construction fails.
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static VoENetwork* GetInterface(VoiceEngine* voiceEngine);
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// Releases the VoENetwork sub-API and decreases an internal
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// reference counter. Returns the new reference count. This value should
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// be zero for all sub-API:s before the VoiceEngine object can be safely
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// deleted.
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virtual int Release() = 0;
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// Installs and enables a user-defined external transport protocol for a
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// specified |channel|. Returns -1 in case of an error, 0 otherwise.
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virtual int RegisterExternalTransport(int channel, Transport& transport) = 0;
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// Removes and disables a user-defined external transport protocol for a
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// specified |channel|. Returns -1 in case of an error, 0 otherwise.
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virtual int DeRegisterExternalTransport(int channel) = 0;
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// The packets received from the network should be passed to this
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// function when external transport is enabled. Note that the data
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// including the RTP-header must also be given to the VoiceEngine.
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// Returns -1 in case of an error, 0 otherwise.
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virtual int ReceivedRTPPacket(int channel,
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const void* data,
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size_t length) = 0;
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virtual int ReceivedRTPPacket(int channel,
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const void* data,
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size_t length,
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const PacketTime& packet_time) {
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return 0;
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}
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// The packets received from the network should be passed to this
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// function when external transport is enabled. Note that the data
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// including the RTCP-header must also be given to the VoiceEngine.
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// Returns -1 in case of an error, 0 otherwise.
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virtual int ReceivedRTCPPacket(int channel,
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const void* data,
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size_t length) = 0;
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protected:
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VoENetwork() {}
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virtual ~VoENetwork() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
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