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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 |
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101 remote_base_->Release(); | 101 remote_base_->Release(); |
102 | 102 |
103 local_rtp_rtcp_->Release(); | 103 local_rtp_rtcp_->Release(); |
104 local_network_->Release(); | 104 local_network_->Release(); |
105 local_base_->Release(); | 105 local_base_->Release(); |
106 | 106 |
107 EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_)); | 107 EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_)); |
108 EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_)); | 108 EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_)); |
109 } | 109 } |
110 | 110 |
111 int ConferenceTransport::SendPacket(int channel, const void* data, size_t len) { | 111 int ConferenceTransport::SendPacket(const void* data, size_t len) { |
112 StorePacket(Packet::Rtp, channel, data, len); | 112 StorePacket(Packet::Rtp, data, len); |
113 return static_cast<int>(len); | 113 return static_cast<int>(len); |
114 } | 114 } |
115 | 115 |
116 int ConferenceTransport::SendRTCPPacket(int channel, const void* data, | 116 int ConferenceTransport::SendRTCPPacket(const void* data, size_t len) { |
117 size_t len) { | 117 StorePacket(Packet::Rtcp, data, len); |
118 StorePacket(Packet::Rtcp, channel, data, len); | |
119 return static_cast<int>(len); | 118 return static_cast<int>(len); |
120 } | 119 } |
121 | 120 |
122 int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc) | 121 int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc) |
123 const { | 122 const { |
124 webrtc::CriticalSectionScoped lock(stream_crit_.get()); | 123 webrtc::CriticalSectionScoped lock(stream_crit_.get()); |
125 auto it = streams_.find(sender_ssrc); | 124 auto it = streams_.find(sender_ssrc); |
126 if (it != streams_.end()) { | 125 if (it != streams_.end()) { |
127 return it->second.second; | 126 return it->second.second; |
128 } | 127 } |
129 return -1; | 128 return -1; |
130 } | 129 } |
131 | 130 |
132 void ConferenceTransport::StorePacket(Packet::Type type, int channel, | 131 void ConferenceTransport::StorePacket(Packet::Type type, |
133 const void* data, size_t len) { | 132 const void* data, |
| 133 size_t len) { |
134 { | 134 { |
135 webrtc::CriticalSectionScoped lock(pq_crit_.get()); | 135 webrtc::CriticalSectionScoped lock(pq_crit_.get()); |
136 packet_queue_.push_back(Packet(type, channel, data, len, rtc::Time())); | 136 packet_queue_.push_back(Packet(type, data, len, rtc::Time())); |
137 } | 137 } |
138 packet_event_->Set(); | 138 packet_event_->Set(); |
139 } | 139 } |
140 | 140 |
141 // 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 |
142 // 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 |
143 // 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 |
144 // and the receiver. | 144 // and the receiver. |
145 void ConferenceTransport::SendPacket(const Packet& packet) { | 145 void ConferenceTransport::SendPacket(const Packet& packet) { |
146 int destination = -1; | 146 int destination = -1; |
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279 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, | 279 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, |
280 webrtc::CallStatistics* stats) { | 280 webrtc::CallStatistics* stats) { |
281 int dst = GetReceiverChannelForSsrc(id); | 281 int dst = GetReceiverChannelForSsrc(id); |
282 if (dst == -1) { | 282 if (dst == -1) { |
283 return false; | 283 return false; |
284 } | 284 } |
285 EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); | 285 EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); |
286 return true; | 286 return true; |
287 } | 287 } |
288 } // namespace voetest | 288 } // namespace voetest |
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