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1 /* | |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | |
3 * | |
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 | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | |
12 #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | |
13 | |
14 #include <deque> | |
15 #include <map> | |
16 #include <memory> | |
17 #include <utility> | |
18 | |
19 #include "webrtc/common_types.h" | |
20 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | |
21 #include "webrtc/rtc_base/basictypes.h" | |
22 #include "webrtc/rtc_base/criticalsection.h" | |
23 #include "webrtc/rtc_base/platform_thread.h" | |
24 #include "webrtc/system_wrappers/include/event_wrapper.h" | |
25 #include "webrtc/test/gtest.h" | |
26 #include "webrtc/voice_engine/include/voe_base.h" | |
27 #include "webrtc/voice_engine/include/voe_codec.h" | |
28 #include "webrtc/voice_engine/include/voe_file.h" | |
29 #include "webrtc/voice_engine/include/voe_network.h" | |
30 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | |
31 #include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h" | |
32 | |
33 namespace webrtc { | |
34 namespace voetest { | |
35 | |
36 static const size_t kMaxPacketSizeByte = 1500; | |
37 | |
38 // This class is to simulate a conference call. There are two Voice Engines, one | |
39 // for local channels and the other for remote channels. There is a simulated | |
40 // reflector, which exchanges RTCP with local channels. For simplicity, it | |
41 // also uses the Voice Engine for remote channels. One can add streams by | |
42 // calling AddStream(), which creates a remote sender channel and a local | |
43 // receive channel. The remote sender channel plays a file as microphone in a | |
44 // looped fashion. Received streams are mixed and played. | |
45 | |
46 class ConferenceTransport: public webrtc::Transport { | |
47 public: | |
48 ConferenceTransport(); | |
49 virtual ~ConferenceTransport(); | |
50 | |
51 /* SetRtt() | |
52 * Set RTT between local channels and reflector. | |
53 * | |
54 * Input: | |
55 * rtt_ms : RTT in milliseconds. | |
56 */ | |
57 void SetRtt(unsigned int rtt_ms); | |
58 | |
59 /* AddStream() | |
60 * Adds a stream in the conference. | |
61 * | |
62 * Input: | |
63 * file_name : name of the file to be added as microphone input. | |
64 * format : format of the input file. | |
65 * | |
66 * Returns stream id. | |
67 */ | |
68 unsigned int AddStream(std::string file_name, webrtc::FileFormats format); | |
69 | |
70 /* RemoveStream() | |
71 * Removes a stream with specified ID from the conference. | |
72 * | |
73 * Input: | |
74 * id : stream id. | |
75 * | |
76 * Returns false if the specified stream does not exist, true if succeeds. | |
77 */ | |
78 bool RemoveStream(unsigned int id); | |
79 | |
80 /* StartPlayout() | |
81 * Starts playing out the stream with specified ID, using the default device. | |
82 * | |
83 * Input: | |
84 * id : stream id. | |
85 * | |
86 * Returns false if the specified stream does not exist, true if succeeds. | |
87 */ | |
88 bool StartPlayout(unsigned int id); | |
89 | |
90 /* GetReceiverStatistics() | |
91 * Gets RTCP statistics of the stream with specified ID. | |
92 * | |
93 * Input: | |
94 * id : stream id; | |
95 * stats : pointer to a CallStatistics to store the result. | |
96 * | |
97 * Returns false if the specified stream does not exist, true if succeeds. | |
98 */ | |
99 bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats); | |
100 | |
101 // Inherit from class webrtc::Transport. | |
102 bool SendRtp(const uint8_t* data, | |
103 size_t len, | |
104 const webrtc::PacketOptions& options) override; | |
105 bool SendRtcp(const uint8_t *data, size_t len) override; | |
106 | |
107 private: | |
108 struct Packet { | |
109 enum Type { Rtp, Rtcp, } type_; | |
110 | |
111 Packet() : len_(0) {} | |
112 Packet(Type type, const void* data, size_t len, int64_t time_ms) | |
113 : type_(type), len_(len), send_time_ms_(time_ms) { | |
114 EXPECT_LE(len_, kMaxPacketSizeByte); | |
115 memcpy(data_, data, len_); | |
116 } | |
117 | |
118 uint8_t data_[kMaxPacketSizeByte]; | |
119 size_t len_; | |
120 int64_t send_time_ms_; | |
121 }; | |
122 | |
123 static bool Run(void* transport) { | |
124 return static_cast<ConferenceTransport*>(transport)->DispatchPackets(); | |
125 } | |
126 | |
127 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const; | |
128 void StorePacket(Packet::Type type, const void* data, size_t len); | |
129 void SendPacket(const Packet& packet); | |
130 bool DispatchPackets(); | |
131 | |
132 rtc::CriticalSection pq_crit_; | |
133 rtc::CriticalSection stream_crit_; | |
134 const std::unique_ptr<webrtc::EventWrapper> packet_event_; | |
135 rtc::PlatformThread thread_; | |
136 | |
137 unsigned int rtt_ms_; | |
138 unsigned int stream_count_; | |
139 | |
140 std::map<unsigned int, std::pair<int, int>> streams_ | |
141 RTC_GUARDED_BY(stream_crit_); | |
142 std::deque<Packet> packet_queue_ RTC_GUARDED_BY(pq_crit_); | |
143 | |
144 int local_sender_; // Channel Id of local sender | |
145 int reflector_; | |
146 | |
147 webrtc::VoiceEngine* local_voe_; | |
148 webrtc::VoEBase* local_base_; | |
149 webrtc::VoERTP_RTCP* local_rtp_rtcp_; | |
150 webrtc::VoENetwork* local_network_; | |
151 rtc::scoped_refptr<webrtc::AudioProcessing> local_apm_; | |
152 | |
153 webrtc::VoiceEngine* remote_voe_; | |
154 webrtc::VoEBase* remote_base_; | |
155 webrtc::VoECodec* remote_codec_; | |
156 webrtc::VoERTP_RTCP* remote_rtp_rtcp_; | |
157 webrtc::VoENetwork* remote_network_; | |
158 webrtc::VoEFile* remote_file_; | |
159 rtc::scoped_refptr<webrtc::AudioProcessing> remote_apm_; | |
160 LoudestFilter loudest_filter_; | |
161 | |
162 const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; | |
163 }; | |
164 | |
165 } // namespace voetest | |
166 } // namespace webrtc | |
167 | |
168 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | |
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