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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 #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ |
12 #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | 12 #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ |
13 | 13 |
14 #include <deque> | 14 #include <deque> |
15 #include <map> | 15 #include <map> |
16 #include <utility> | 16 #include <utility> |
17 | 17 |
18 #include "testing/gtest/include/gtest/gtest.h" | 18 #include "testing/gtest/include/gtest/gtest.h" |
19 #include "webrtc/base/basictypes.h" | 19 #include "webrtc/base/basictypes.h" |
20 #include "webrtc/base/scoped_ptr.h" | 20 #include "webrtc/base/scoped_ptr.h" |
21 #include "webrtc/common_types.h" | 21 #include "webrtc/common_types.h" |
| 22 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
22 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 23 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
23 #include "webrtc/system_wrappers/interface/event_wrapper.h" | 24 #include "webrtc/system_wrappers/interface/event_wrapper.h" |
24 #include "webrtc/system_wrappers/interface/thread_wrapper.h" | 25 #include "webrtc/system_wrappers/interface/thread_wrapper.h" |
25 #include "webrtc/voice_engine/include/voe_base.h" | 26 #include "webrtc/voice_engine/include/voe_base.h" |
26 #include "webrtc/voice_engine/include/voe_codec.h" | 27 #include "webrtc/voice_engine/include/voe_codec.h" |
27 #include "webrtc/voice_engine/include/voe_file.h" | 28 #include "webrtc/voice_engine/include/voe_file.h" |
28 #include "webrtc/voice_engine/include/voe_network.h" | 29 #include "webrtc/voice_engine/include/voe_network.h" |
29 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 30 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
30 | 31 #include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h" |
31 | 32 |
32 static const size_t kMaxPacketSizeByte = 1500; | 33 static const size_t kMaxPacketSizeByte = 1500; |
33 | 34 |
34 namespace voetest { | 35 namespace voetest { |
35 | 36 |
36 // This class is to simulate a conference call. There are two Voice Engines, one | 37 // This class is to simulate a conference call. There are two Voice Engines, one |
37 // for local channels and the other for remote channels. There is a simulated | 38 // for local channels and the other for remote channels. There is a simulated |
38 // reflector, which exchanges RTCP with local channels. For simplicity, it | 39 // reflector, which exchanges RTCP with local channels. For simplicity, it |
39 // also uses the Voice Engine for remote channels. One can add streams by | 40 // also uses the Voice Engine for remote channels. One can add streams by |
40 // calling AddStream(), which creates a remote sender channel and a local | 41 // calling AddStream(), which creates a remote sender channel and a local |
41 // receive channel. The remote sender channel plays a file as microphone in a | 42 // receive channel. The remote sender channel plays a file as microphone in a |
42 // looped fashion. Received streams are mixed and played. | 43 // looped fashion. Received streams are mixed and played. |
43 | 44 |
44 class ConferenceTransport: public webrtc::Transport { | 45 class ConferenceTransport: public webrtc::Transport { |
45 public: | 46 public: |
46 ConferenceTransport(); | 47 ConferenceTransport(); |
47 virtual ~ConferenceTransport(); | 48 virtual ~ConferenceTransport(); |
48 | 49 |
49 /* SetRtt() | 50 /* SetRtt() |
50 * Set RTT between local channels and reflector. | 51 * Set RTT between local channels and reflector. |
51 * | 52 * |
52 * Input: | 53 * Input: |
53 * rtt_ms : RTT in milliseconds. | 54 * rtt_ms : RTT in milliseconds. |
54 */ | 55 */ |
55 void SetRtt(unsigned int rtt_ms); | 56 void SetRtt(unsigned int rtt_ms); |
56 | 57 |
57 /* AddStream() | 58 /* AddStream() |
58 * Adds a stream in the conference. | 59 * Adds a stream in the conference. |
59 * | 60 * |
| 61 * Input: |
| 62 * file_name : name of the file to be added as microphone input. |
| 63 * format : format of the input file. |
| 64 * |
60 * Returns stream id. | 65 * Returns stream id. |
61 */ | 66 */ |
62 unsigned int AddStream(); | 67 unsigned int AddStream(std::string file_name, webrtc::FileFormats format); |
63 | 68 |
64 /* RemoveStream() | 69 /* RemoveStream() |
65 * Removes a stream with specified ID from the conference. | 70 * Removes a stream with specified ID from the conference. |
66 * | 71 * |
67 * Input: | 72 * Input: |
68 * id : stream id. | 73 * id : stream id. |
69 * | 74 * |
70 * Returns false if the specified stream does not exist, true if succeeds. | 75 * Returns false if the specified stream does not exist, true if succeeds. |
71 */ | 76 */ |
72 bool RemoveStream(unsigned int id); | 77 bool RemoveStream(unsigned int id); |
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116 uint32 send_time_ms_; | 121 uint32 send_time_ms_; |
117 }; | 122 }; |
118 | 123 |
119 static bool Run(void* transport) { | 124 static bool Run(void* transport) { |
120 return static_cast<ConferenceTransport*>(transport)->DispatchPackets(); | 125 return static_cast<ConferenceTransport*>(transport)->DispatchPackets(); |
121 } | 126 } |
122 | 127 |
123 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const; | 128 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const; |
124 void StorePacket(Packet::Type type, int channel, const void* data, | 129 void StorePacket(Packet::Type type, int channel, const void* data, |
125 size_t len); | 130 size_t len); |
126 void SendPacket(const Packet& packet) const; | 131 void SendPacket(const Packet& packet); |
127 bool DispatchPackets(); | 132 bool DispatchPackets(); |
128 | 133 |
129 const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_; | 134 const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_; |
130 const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_; | 135 const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_; |
131 const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_; | 136 const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_; |
132 const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_; | 137 const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_; |
133 | 138 |
134 unsigned int rtt_ms_; | 139 unsigned int rtt_ms_; |
135 unsigned int stream_count_; | 140 unsigned int stream_count_; |
136 | 141 |
137 std::map<unsigned int, std::pair<int, int>> streams_ | 142 std::map<unsigned int, std::pair<int, int>> streams_ |
138 GUARDED_BY(stream_crit_.get()); | 143 GUARDED_BY(stream_crit_.get()); |
139 std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get()); | 144 std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get()); |
140 | 145 |
141 int local_sender_; // Channel Id of local sender | 146 int local_sender_; // Channel Id of local sender |
142 int reflector_; | 147 int reflector_; |
143 | 148 |
144 webrtc::VoiceEngine* local_voe_; | 149 webrtc::VoiceEngine* local_voe_; |
145 webrtc::VoEBase* local_base_; | 150 webrtc::VoEBase* local_base_; |
146 webrtc::VoERTP_RTCP* local_rtp_rtcp_; | 151 webrtc::VoERTP_RTCP* local_rtp_rtcp_; |
147 webrtc::VoENetwork* local_network_; | 152 webrtc::VoENetwork* local_network_; |
148 | 153 |
149 webrtc::VoiceEngine* remote_voe_; | 154 webrtc::VoiceEngine* remote_voe_; |
150 webrtc::VoEBase* remote_base_; | 155 webrtc::VoEBase* remote_base_; |
151 webrtc::VoECodec* remote_codec_; | 156 webrtc::VoECodec* remote_codec_; |
152 webrtc::VoERTP_RTCP* remote_rtp_rtcp_; | 157 webrtc::VoERTP_RTCP* remote_rtp_rtcp_; |
153 webrtc::VoENetwork* remote_network_; | 158 webrtc::VoENetwork* remote_network_; |
154 webrtc::VoEFile* remote_file_; | 159 webrtc::VoEFile* remote_file_; |
| 160 |
| 161 LoudestFilter loudest_filter_; |
| 162 |
| 163 const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; |
155 }; | 164 }; |
156 } // namespace voetest | 165 } // namespace voetest |
157 | 166 |
158 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | 167 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ |
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