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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h" | 11 #include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h" |
12 | 12 |
13 #include <string> | 13 #include <string> |
14 | 14 |
15 #include "webrtc/base/byteorder.h" | 15 #include "webrtc/base/byteorder.h" |
16 #include "webrtc/base/timeutils.h" | 16 #include "webrtc/base/timeutils.h" |
17 #include "webrtc/test/testsupport/fileutils.h" | |
18 #include "webrtc/system_wrappers/interface/sleep.h" | 17 #include "webrtc/system_wrappers/interface/sleep.h" |
19 | 18 |
20 namespace { | 19 namespace { |
21 static const unsigned int kReflectorSsrc = 0x0000; | 20 static const unsigned int kReflectorSsrc = 0x0000; |
22 static const unsigned int kLocalSsrc = 0x0001; | 21 static const unsigned int kLocalSsrc = 0x0001; |
23 static const unsigned int kFirstRemoteSsrc = 0x0002; | 22 static const unsigned int kFirstRemoteSsrc = 0x0002; |
24 static const webrtc::CodecInst kCodecInst = | 23 static const webrtc::CodecInst kCodecInst = |
25 {120, "opus", 48000, 960, 2, 64000}; | 24 {120, "opus", 48000, 960, 2, 64000}; |
| 25 static const int kAudioLevelHeaderId = 1; |
26 | 26 |
27 static unsigned int ParseSsrc(const void* data, size_t len, bool rtcp) { | 27 static unsigned int ParseRtcpSsrc(const void* data, size_t len) { |
28 const size_t ssrc_pos = (!rtcp) ? 8 : 4; | 28 const size_t ssrc_pos = 4; |
29 unsigned int ssrc = 0; | 29 unsigned int ssrc = 0; |
30 if (len >= (ssrc_pos + sizeof(ssrc))) { | 30 if (len >= (ssrc_pos + sizeof(ssrc))) { |
31 ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos); | 31 ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos); |
32 } | 32 } |
33 return ssrc; | 33 return ssrc; |
34 } | 34 } |
35 } // namespace | 35 } // namespace |
36 | 36 |
37 namespace voetest { | 37 namespace voetest { |
38 | 38 |
39 ConferenceTransport::ConferenceTransport() | 39 ConferenceTransport::ConferenceTransport() |
40 : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), | 40 : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), |
41 stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), | 41 stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), |
42 packet_event_(webrtc::EventWrapper::Create()), | 42 packet_event_(webrtc::EventWrapper::Create()), |
43 thread_(webrtc::ThreadWrapper::CreateThread(Run, | 43 thread_(webrtc::ThreadWrapper::CreateThread(Run, |
44 this, | 44 this, |
45 "ConferenceTransport")), | 45 "ConferenceTransport")), |
46 rtt_ms_(0), | 46 rtt_ms_(0), |
47 stream_count_(0) { | 47 stream_count_(0), |
| 48 rtp_header_parser_(webrtc::RtpHeaderParser::Create()) { |
| 49 rtp_header_parser_-> |
| 50 RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, |
| 51 kAudioLevelHeaderId); |
| 52 |
48 local_voe_ = webrtc::VoiceEngine::Create(); | 53 local_voe_ = webrtc::VoiceEngine::Create(); |
49 local_base_ = webrtc::VoEBase::GetInterface(local_voe_); | 54 local_base_ = webrtc::VoEBase::GetInterface(local_voe_); |
50 local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); | 55 local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); |
51 local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_); | 56 local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_); |
52 | 57 |
53 // In principle, we can use one VoiceEngine to achieve the same goal. Well, in | 58 // In principle, we can use one VoiceEngine to achieve the same goal. Well, in |
54 // here, we use two engines to make it more like reality. | 59 // here, we use two engines to make it more like reality. |
55 remote_voe_ = webrtc::VoiceEngine::Create(); | 60 remote_voe_ = webrtc::VoiceEngine::Create(); |
56 remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_); | 61 remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_); |
57 remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_); | 62 remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_); |
58 remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_); | 63 remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_); |
59 remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_); | 64 remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_); |
60 remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_); | 65 remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_); |
61 | 66 |
62 EXPECT_EQ(0, local_base_->Init()); | 67 EXPECT_EQ(0, local_base_->Init()); |
63 local_sender_ = local_base_->CreateChannel(); | 68 local_sender_ = local_base_->CreateChannel(); |
64 EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this)); | 69 EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this)); |
65 EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc)); | 70 EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc)); |
| 71 EXPECT_EQ(0, local_rtp_rtcp_-> |
| 72 SetSendAudioLevelIndicationStatus(local_sender_, true, |
| 73 kAudioLevelHeaderId)); |
| 74 |
66 EXPECT_EQ(0, local_base_->StartSend(local_sender_)); | 75 EXPECT_EQ(0, local_base_->StartSend(local_sender_)); |
67 | 76 |
68 EXPECT_EQ(0, remote_base_->Init()); | 77 EXPECT_EQ(0, remote_base_->Init()); |
69 reflector_ = remote_base_->CreateChannel(); | 78 reflector_ = remote_base_->CreateChannel(); |
70 EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this)); | 79 EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this)); |
71 EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc)); | 80 EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc)); |
72 | 81 |
73 thread_->Start(); | 82 thread_->Start(); |
74 thread_->SetPriority(webrtc::kHighPriority); | 83 thread_->SetPriority(webrtc::kHighPriority); |
75 } | 84 } |
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126 webrtc::CriticalSectionScoped lock(pq_crit_.get()); | 135 webrtc::CriticalSectionScoped lock(pq_crit_.get()); |
127 packet_queue_.push_back(Packet(type, channel, data, len, rtc::Time())); | 136 packet_queue_.push_back(Packet(type, channel, data, len, rtc::Time())); |
128 } | 137 } |
129 packet_event_->Set(); | 138 packet_event_->Set(); |
130 } | 139 } |
131 | 140 |
132 // This simulates the flow of RTP and RTCP packets. Complications like that | 141 // This simulates the flow of RTP and RTCP packets. Complications like that |
133 // a packet is first sent to the reflector, and then forwarded to the receiver | 142 // a packet is first sent to the reflector, and then forwarded to the receiver |
134 // are simplified, in this particular case, to a direct link between the sender | 143 // are simplified, in this particular case, to a direct link between the sender |
135 // and the receiver. | 144 // and the receiver. |
136 void ConferenceTransport::SendPacket(const Packet& packet) const { | 145 void ConferenceTransport::SendPacket(const Packet& packet) { |
137 unsigned int sender_ssrc; | |
138 int destination = -1; | 146 int destination = -1; |
| 147 |
139 switch (packet.type_) { | 148 switch (packet.type_) { |
140 case Packet::Rtp: | 149 case Packet::Rtp: { |
141 sender_ssrc = ParseSsrc(packet.data_, packet.len_, false); | 150 webrtc::RTPHeader rtp_header; |
142 if (sender_ssrc == kLocalSsrc) { | 151 rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header); |
| 152 if (rtp_header.ssrc == kLocalSsrc) { |
143 remote_network_->ReceivedRTPPacket(reflector_, packet.data_, | 153 remote_network_->ReceivedRTPPacket(reflector_, packet.data_, |
144 packet.len_, webrtc::PacketTime()); | 154 packet.len_, webrtc::PacketTime()); |
145 } else { | 155 } else { |
146 destination = GetReceiverChannelForSsrc(sender_ssrc); | 156 if (loudest_filter_.ForwardThisPacket(rtp_header)) { |
147 if (destination != -1) { | 157 destination = GetReceiverChannelForSsrc(rtp_header.ssrc); |
148 local_network_->ReceivedRTPPacket(destination, packet.data_, | 158 if (destination != -1) { |
149 packet.len_, | 159 local_network_->ReceivedRTPPacket(destination, packet.data_, |
150 webrtc::PacketTime()); | 160 packet.len_, |
| 161 webrtc::PacketTime()); |
| 162 } |
151 } | 163 } |
152 } | 164 } |
153 break; | 165 break; |
154 case Packet::Rtcp: | 166 } |
155 sender_ssrc = ParseSsrc(packet.data_, packet.len_, true); | 167 case Packet::Rtcp: { |
| 168 unsigned int sender_ssrc = ParseRtcpSsrc(packet.data_, packet.len_); |
156 if (sender_ssrc == kLocalSsrc) { | 169 if (sender_ssrc == kLocalSsrc) { |
157 remote_network_->ReceivedRTCPPacket(reflector_, packet.data_, | 170 remote_network_->ReceivedRTCPPacket(reflector_, packet.data_, |
158 packet.len_); | 171 packet.len_); |
159 } else if (sender_ssrc == kReflectorSsrc) { | 172 } else if (sender_ssrc == kReflectorSsrc) { |
160 local_network_->ReceivedRTCPPacket(local_sender_, packet.data_, | 173 local_network_->ReceivedRTCPPacket(local_sender_, packet.data_, |
161 packet.len_); | 174 packet.len_); |
162 } else { | 175 } else { |
163 destination = GetReceiverChannelForSsrc(sender_ssrc); | 176 destination = GetReceiverChannelForSsrc(sender_ssrc); |
164 if (destination != -1) { | 177 if (destination != -1) { |
165 local_network_->ReceivedRTCPPacket(destination, packet.data_, | 178 local_network_->ReceivedRTCPPacket(destination, packet.data_, |
166 packet.len_); | 179 packet.len_); |
167 } | 180 } |
168 } | 181 } |
169 break; | 182 break; |
| 183 } |
170 } | 184 } |
171 } | 185 } |
172 | 186 |
173 bool ConferenceTransport::DispatchPackets() { | 187 bool ConferenceTransport::DispatchPackets() { |
174 switch (packet_event_->Wait(1000)) { | 188 switch (packet_event_->Wait(1000)) { |
175 case webrtc::kEventSignaled: | 189 case webrtc::kEventSignaled: |
176 break; | 190 break; |
177 case webrtc::kEventTimeout: | 191 case webrtc::kEventTimeout: |
178 return true; | 192 return true; |
179 case webrtc::kEventError: | 193 case webrtc::kEventError: |
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200 | 214 |
201 SendPacket(packet); | 215 SendPacket(packet); |
202 } | 216 } |
203 return true; | 217 return true; |
204 } | 218 } |
205 | 219 |
206 void ConferenceTransport::SetRtt(unsigned int rtt_ms) { | 220 void ConferenceTransport::SetRtt(unsigned int rtt_ms) { |
207 rtt_ms_ = rtt_ms; | 221 rtt_ms_ = rtt_ms; |
208 } | 222 } |
209 | 223 |
210 unsigned int ConferenceTransport::AddStream() { | 224 unsigned int ConferenceTransport::AddStream(std::string file_name, |
211 const std::string kInputFileName = | 225 webrtc::FileFormats format) { |
212 webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); | |
213 | |
214 const int new_sender = remote_base_->CreateChannel(); | 226 const int new_sender = remote_base_->CreateChannel(); |
215 EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this)); | 227 EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this)); |
216 | 228 |
217 const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++; | 229 const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++; |
218 EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc)); | 230 EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc)); |
| 231 EXPECT_EQ(0, remote_rtp_rtcp_-> |
| 232 SetSendAudioLevelIndicationStatus(new_sender, true, kAudioLevelHeaderId)); |
219 | 233 |
220 EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst)); | 234 EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst)); |
221 EXPECT_EQ(0, remote_base_->StartSend(new_sender)); | 235 EXPECT_EQ(0, remote_base_->StartSend(new_sender)); |
222 EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone( | 236 EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone( |
223 new_sender, kInputFileName.c_str(), true, false, | 237 new_sender, file_name.c_str(), true, false, format, 1.0)); |
224 webrtc::kFileFormatPcm32kHzFile, 1.0)); | |
225 | 238 |
226 const int new_receiver = local_base_->CreateChannel(); | 239 const int new_receiver = local_base_->CreateChannel(); |
227 EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_)); | 240 EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_)); |
228 | 241 |
229 EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this)); | 242 EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this)); |
230 // Receive channels have to have the same SSRC in order to send receiver | 243 // Receive channels have to have the same SSRC in order to send receiver |
231 // reports with this SSRC. | 244 // reports with this SSRC. |
232 EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc)); | 245 EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc)); |
233 | 246 |
234 { | 247 { |
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266 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, | 279 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, |
267 webrtc::CallStatistics* stats) { | 280 webrtc::CallStatistics* stats) { |
268 int dst = GetReceiverChannelForSsrc(id); | 281 int dst = GetReceiverChannelForSsrc(id); |
269 if (dst == -1) { | 282 if (dst == -1) { |
270 return false; | 283 return false; |
271 } | 284 } |
272 EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); | 285 EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); |
273 return true; | 286 return true; |
274 } | 287 } |
275 } // namespace voetest | 288 } // namespace voetest |
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