webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc - src - Git at Google

 /*
  *  Copyright (c) 2015 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.
  */

 #include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"

 #include <string>

 #include "webrtc/rtc_base/byteorder.h"
 #include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"

 namespace webrtc {
 namespace voetest {

 namespace {

 static const unsigned int kReflectorSsrc = 0x0000;
 static const unsigned int kLocalSsrc = 0x0001;
 static const unsigned int kFirstRemoteSsrc = 0x0002;
 static const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
 static const int kAudioLevelHeaderId = 1;

 static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
   const size_t ssrc_pos = 4;
   unsigned int ssrc = 0;
   if (len >= (ssrc_pos + sizeof(ssrc))) {
     ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
   }
   return ssrc;
 }

 }  // namespace

 ConferenceTransport::ConferenceTransport()
     : packet_event_(webrtc::EventWrapper::Create()),
       thread_(Run, this, "ConferenceTransport"),
       rtt_ms_(0),
       stream_count_(0),
       rtp_header_parser_(webrtc::RtpHeaderParser::Create()) {
   rtp_header_parser_->
       RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
                                  kAudioLevelHeaderId);

   local_voe_ = webrtc::VoiceEngine::Create();
   local_base_ = webrtc::VoEBase::GetInterface(local_voe_);
   local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
   local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);

   local_apm_ = webrtc::AudioProcessing::Create();
   local_base_->Init(nullptr, local_apm_.get(), nullptr);

   // In principle, we can use one VoiceEngine to achieve the same goal. Well, in
   // here, we use two engines to make it more like reality.
   remote_voe_ = webrtc::VoiceEngine::Create();
   remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_);
   remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_);
   remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_);
   remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
   remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);

   remote_apm_ = webrtc::AudioProcessing::Create();
   remote_base_->Init(nullptr, remote_apm_.get(), nullptr);

   local_sender_ = local_base_->CreateChannel();
   static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
       ->GetChannelProxy(local_sender_)
       ->RegisterLegacyReceiveCodecs();
   EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
   EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
   EXPECT_EQ(0, local_rtp_rtcp_->
       SetSendAudioLevelIndicationStatus(local_sender_, true,
                                         kAudioLevelHeaderId));
   EXPECT_EQ(0, local_base_->StartSend(local_sender_));

   reflector_ = remote_base_->CreateChannel();
   static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
       ->GetChannelProxy(reflector_)
       ->RegisterLegacyReceiveCodecs();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
   EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));

   thread_.Start();
   thread_.SetPriority(rtc::kHighPriority);
 }

 ConferenceTransport::~ConferenceTransport() {
   // Must stop sending, otherwise DispatchPackets() cannot quit.
   EXPECT_EQ(0, remote_network_->DeRegisterExternalTransport(reflector_));
   EXPECT_EQ(0, local_network_->DeRegisterExternalTransport(local_sender_));

   while (!streams_.empty()) {
     auto stream = streams_.begin();
     RemoveStream(stream->first);
   }

   thread_.Stop();

   remote_file_->Release();
   remote_rtp_rtcp_->Release();
   remote_network_->Release();
   remote_base_->Release();

   local_rtp_rtcp_->Release();
   local_network_->Release();
   local_base_->Release();

   EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_));
   EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
 }

 bool ConferenceTransport::SendRtp(const uint8_t* data,
                                   size_t len,
                                   const webrtc::PacketOptions& options) {
   StorePacket(Packet::Rtp, data, len);
   return true;
 }

 bool ConferenceTransport::SendRtcp(const uint8_t* data, size_t len) {
   StorePacket(Packet::Rtcp, data, len);
   return true;
 }

 int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc)
     const {
   rtc::CritScope lock(&stream_crit_);
   auto it = streams_.find(sender_ssrc);
   if (it != streams_.end()) {
     return it->second.second;
   }
   return -1;
 }

 void ConferenceTransport::StorePacket(Packet::Type type,
                                       const void* data,
                                       size_t len) {
   {
     rtc::CritScope lock(&pq_crit_);
     packet_queue_.push_back(Packet(type, data, len, rtc::TimeMillis()));
   }
   packet_event_->Set();
 }

 // This simulates the flow of RTP and RTCP packets. Complications like that
 // a packet is first sent to the reflector, and then forwarded to the receiver
 // are simplified, in this particular case, to a direct link between the sender
 // and the receiver.
 void ConferenceTransport::SendPacket(const Packet& packet) {
   int destination = -1;

   switch (packet.type_) {
     case Packet::Rtp: {
       webrtc::RTPHeader rtp_header;
       rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header);
       if (rtp_header.ssrc == kLocalSsrc) {
         remote_network_->ReceivedRTPPacket(reflector_, packet.data_,
                                            packet.len_, webrtc::PacketTime());
       } else {
         if (loudest_filter_.ForwardThisPacket(rtp_header)) {
           destination = GetReceiverChannelForSsrc(rtp_header.ssrc);
           if (destination != -1) {
             local_network_->ReceivedRTPPacket(destination, packet.data_,
                                               packet.len_,
                                               webrtc::PacketTime());
           }
         }
       }
       break;
     }
     case Packet::Rtcp: {
       unsigned int sender_ssrc = ParseRtcpSsrc(packet.data_, packet.len_);
       if (sender_ssrc == kLocalSsrc) {
         remote_network_->ReceivedRTCPPacket(reflector_, packet.data_,
                                             packet.len_);
       } else if (sender_ssrc == kReflectorSsrc) {
         local_network_->ReceivedRTCPPacket(local_sender_, packet.data_,
                                            packet.len_);
       } else {
         destination = GetReceiverChannelForSsrc(sender_ssrc);
         if (destination != -1) {
           local_network_->ReceivedRTCPPacket(destination, packet.data_,
                                              packet.len_);
         }
       }
       break;
     }
   }
 }

 bool ConferenceTransport::DispatchPackets() {
   switch (packet_event_->Wait(1000)) {
     case webrtc::kEventSignaled:
       break;
     case webrtc::kEventTimeout:
       return true;
     case webrtc::kEventError:
       ADD_FAILURE() << "kEventError encountered.";
       return true;
   }

   while (true) {
     Packet packet;
     {
       rtc::CritScope lock(&pq_crit_);
       if (packet_queue_.empty())
         break;
       packet = packet_queue_.front();
       packet_queue_.pop_front();
     }

     int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);
     int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
     if (sleep_ms > 0) {
       // Every packet should be delayed by half of RTT.
       webrtc::SleepMs(sleep_ms);
     }

     SendPacket(packet);
   }
   return true;
 }

 void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
   rtt_ms_ = rtt_ms;
 }

 unsigned int ConferenceTransport::AddStream(std::string file_name,
                                             webrtc::FileFormats format) {
   const int new_sender = remote_base_->CreateChannel();
   static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
       ->GetChannelProxy(new_sender)
       ->RegisterLegacyReceiveCodecs();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));

   const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
   EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc));
   EXPECT_EQ(0, remote_rtp_rtcp_->
       SetSendAudioLevelIndicationStatus(new_sender, true, kAudioLevelHeaderId));

   EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst));
   EXPECT_EQ(0, remote_base_->StartSend(new_sender));
   EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone(
       new_sender, file_name.c_str(), true, false, format, 1.0));

   const int new_receiver = local_base_->CreateChannel();
   static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
       ->GetChannelProxy(new_receiver)
       ->RegisterLegacyReceiveCodecs();
   EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));

   EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
   // Receive channels have to have the same SSRC in order to send receiver
   // reports with this SSRC.
   EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc));

   {
     rtc::CritScope lock(&stream_crit_);
     streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver);
   }
   return remote_ssrc;  // remote ssrc used as stream id.
 }

 bool ConferenceTransport::RemoveStream(unsigned int id) {
   rtc::CritScope lock(&stream_crit_);
   auto it = streams_.find(id);
   if (it == streams_.end()) {
     return false;
   }
   EXPECT_EQ(0, remote_network_->
       DeRegisterExternalTransport(it->second.second));
   EXPECT_EQ(0, local_network_->
       DeRegisterExternalTransport(it->second.first));
   EXPECT_EQ(0, remote_base_->DeleteChannel(it->second.second));
   EXPECT_EQ(0, local_base_->DeleteChannel(it->second.first));
   streams_.erase(it);
   return true;
 }

 bool ConferenceTransport::StartPlayout(unsigned int id) {
   int dst = GetReceiverChannelForSsrc(id);
   if (dst == -1) {
     return false;
   }
   EXPECT_EQ(0, local_base_->StartPlayout(dst));
   return true;
 }

 bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
                                                 webrtc::CallStatistics* stats) {
   int dst = GetReceiverChannelForSsrc(id);
   if (dst == -1) {
     return false;
   }
   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
   return true;
 }

 }  // namespace voetest
 }  // namespace webrtc
	/*
	* Copyright (c) 2015 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.
	*/

	#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"

	#include <string>

	#include "webrtc/rtc_base/byteorder.h"
	#include "webrtc/rtc_base/timeutils.h"
	#include "webrtc/system_wrappers/include/sleep.h"
	#include "webrtc/voice_engine/channel_proxy.h"
	#include "webrtc/voice_engine/voice_engine_impl.h"

	namespace webrtc {
	namespace voetest {

	namespace {

	static const unsigned int kReflectorSsrc = 0x0000;
	static const unsigned int kLocalSsrc = 0x0001;
	static const unsigned int kFirstRemoteSsrc = 0x0002;
	static const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
	static const int kAudioLevelHeaderId = 1;

	static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
	const size_t ssrc_pos = 4;
	unsigned int ssrc = 0;
	if (len >= (ssrc_pos + sizeof(ssrc))) {
	ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
	}
	return ssrc;
	}

	} // namespace

	ConferenceTransport::ConferenceTransport()
	: packet_event_(webrtc::EventWrapper::Create()),
	thread_(Run, this, "ConferenceTransport"),
	rtt_ms_(0),
	stream_count_(0),
	rtp_header_parser_(webrtc::RtpHeaderParser::Create()) {
	rtp_header_parser_->
	RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
	kAudioLevelHeaderId);

	local_voe_ = webrtc::VoiceEngine::Create();
	local_base_ = webrtc::VoEBase::GetInterface(local_voe_);
	local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
	local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);

	local_apm_ = webrtc::AudioProcessing::Create();
	local_base_->Init(nullptr, local_apm_.get(), nullptr);

	// In principle, we can use one VoiceEngine to achieve the same goal. Well, in
	// here, we use two engines to make it more like reality.
	remote_voe_ = webrtc::VoiceEngine::Create();
	remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_);
	remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_);
	remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_);
	remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
	remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);

	remote_apm_ = webrtc::AudioProcessing::Create();
	remote_base_->Init(nullptr, remote_apm_.get(), nullptr);

	local_sender_ = local_base_->CreateChannel();
	static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
	->GetChannelProxy(local_sender_)
	->RegisterLegacyReceiveCodecs();
	EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
	EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
	EXPECT_EQ(0, local_rtp_rtcp_->
	SetSendAudioLevelIndicationStatus(local_sender_, true,
	kAudioLevelHeaderId));
	EXPECT_EQ(0, local_base_->StartSend(local_sender_));

	reflector_ = remote_base_->CreateChannel();
	static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
	->GetChannelProxy(reflector_)
	->RegisterLegacyReceiveCodecs();
	EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
	EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));

	thread_.Start();
	thread_.SetPriority(rtc::kHighPriority);
	}

	ConferenceTransport::~ConferenceTransport() {
	// Must stop sending, otherwise DispatchPackets() cannot quit.
	EXPECT_EQ(0, remote_network_->DeRegisterExternalTransport(reflector_));
	EXPECT_EQ(0, local_network_->DeRegisterExternalTransport(local_sender_));

	while (!streams_.empty()) {
	auto stream = streams_.begin();
	RemoveStream(stream->first);
	}

	thread_.Stop();

	remote_file_->Release();
	remote_rtp_rtcp_->Release();
	remote_network_->Release();
	remote_base_->Release();

	local_rtp_rtcp_->Release();
	local_network_->Release();
	local_base_->Release();

	EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_));
	EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
	}

	bool ConferenceTransport::SendRtp(const uint8_t* data,
	size_t len,
	const webrtc::PacketOptions& options) {
	StorePacket(Packet::Rtp, data, len);
	return true;
	}

	bool ConferenceTransport::SendRtcp(const uint8_t* data, size_t len) {
	StorePacket(Packet::Rtcp, data, len);
	return true;
	}

	int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc)
	const {
	rtc::CritScope lock(&stream_crit_);
	auto it = streams_.find(sender_ssrc);
	if (it != streams_.end()) {
	return it->second.second;
	}
	return -1;
	}

	void ConferenceTransport::StorePacket(Packet::Type type,
	const void* data,
	size_t len) {
	{
	rtc::CritScope lock(&pq_crit_);
	packet_queue_.push_back(Packet(type, data, len, rtc::TimeMillis()));
	}
	packet_event_->Set();
	}

	// This simulates the flow of RTP and RTCP packets. Complications like that
	// a packet is first sent to the reflector, and then forwarded to the receiver
	// are simplified, in this particular case, to a direct link between the sender
	// and the receiver.
	void ConferenceTransport::SendPacket(const Packet& packet) {
	int destination = -1;

	switch (packet.type_) {
	case Packet::Rtp: {
	webrtc::RTPHeader rtp_header;
	rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header);
	if (rtp_header.ssrc == kLocalSsrc) {
	remote_network_->ReceivedRTPPacket(reflector_, packet.data_,
	packet.len_, webrtc::PacketTime());
	} else {
	if (loudest_filter_.ForwardThisPacket(rtp_header)) {
	destination = GetReceiverChannelForSsrc(rtp_header.ssrc);
	if (destination != -1) {
	local_network_->ReceivedRTPPacket(destination, packet.data_,
	packet.len_,
	webrtc::PacketTime());
	}
	}
	}
	break;
	}
	case Packet::Rtcp: {
	unsigned int sender_ssrc = ParseRtcpSsrc(packet.data_, packet.len_);
	if (sender_ssrc == kLocalSsrc) {
	remote_network_->ReceivedRTCPPacket(reflector_, packet.data_,
	packet.len_);
	} else if (sender_ssrc == kReflectorSsrc) {
	local_network_->ReceivedRTCPPacket(local_sender_, packet.data_,
	packet.len_);
	} else {
	destination = GetReceiverChannelForSsrc(sender_ssrc);
	if (destination != -1) {
	local_network_->ReceivedRTCPPacket(destination, packet.data_,
	packet.len_);
	}
	}
	break;
	}
	}
	}

	bool ConferenceTransport::DispatchPackets() {
	switch (packet_event_->Wait(1000)) {
	case webrtc::kEventSignaled:
	break;
	case webrtc::kEventTimeout:
	return true;
	case webrtc::kEventError:
	ADD_FAILURE() << "kEventError encountered.";
	return true;
	}

	while (true) {
	Packet packet;
	{
	rtc::CritScope lock(&pq_crit_);
	if (packet_queue_.empty())
	break;
	packet = packet_queue_.front();
	packet_queue_.pop_front();
	}

	int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);
	int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
	if (sleep_ms > 0) {
	// Every packet should be delayed by half of RTT.
	webrtc::SleepMs(sleep_ms);
	}

	SendPacket(packet);
	}
	return true;
	}

	void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
	rtt_ms_ = rtt_ms;
	}

	unsigned int ConferenceTransport::AddStream(std::string file_name,
	webrtc::FileFormats format) {
	const int new_sender = remote_base_->CreateChannel();
	static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
	->GetChannelProxy(new_sender)
	->RegisterLegacyReceiveCodecs();
	EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));

	const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
	EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc));
	EXPECT_EQ(0, remote_rtp_rtcp_->
	SetSendAudioLevelIndicationStatus(new_sender, true, kAudioLevelHeaderId));

	EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst));
	EXPECT_EQ(0, remote_base_->StartSend(new_sender));
	EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone(
	new_sender, file_name.c_str(), true, false, format, 1.0));

	const int new_receiver = local_base_->CreateChannel();
	static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
	->GetChannelProxy(new_receiver)
	->RegisterLegacyReceiveCodecs();
	EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));

	EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
	// Receive channels have to have the same SSRC in order to send receiver
	// reports with this SSRC.
	EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc));

	{
	rtc::CritScope lock(&stream_crit_);
	streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver);
	}
	return remote_ssrc; // remote ssrc used as stream id.
	}

	bool ConferenceTransport::RemoveStream(unsigned int id) {
	rtc::CritScope lock(&stream_crit_);
	auto it = streams_.find(id);
	if (it == streams_.end()) {
	return false;
	}
	EXPECT_EQ(0, remote_network_->
	DeRegisterExternalTransport(it->second.second));
	EXPECT_EQ(0, local_network_->
	DeRegisterExternalTransport(it->second.first));
	EXPECT_EQ(0, remote_base_->DeleteChannel(it->second.second));
	EXPECT_EQ(0, local_base_->DeleteChannel(it->second.first));
	streams_.erase(it);
	return true;
	}

	bool ConferenceTransport::StartPlayout(unsigned int id) {
	int dst = GetReceiverChannelForSsrc(id);
	if (dst == -1) {
	return false;
	}
	EXPECT_EQ(0, local_base_->StartPlayout(dst));
	return true;
	}

	bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
	webrtc::CallStatistics* stats) {
	int dst = GetReceiverChannelForSsrc(id);
	if (dst == -1) {
	return false;
	}
	EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
	return true;
	}

	} // namespace voetest
	} // namespace webrtc