1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
|
/*
* 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.
*/
#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
#include <deque>
#include <map>
#include <utility>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_file.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
static const size_t kMaxPacketSizeByte = 1500;
namespace voetest {
// This class is to simulate a conference call. There are two Voice Engines, one
// for local channels and the other for remote channels. There is a simulated
// reflector, which exchanges RTCP with local channels. For simplicity, it
// also uses the Voice Engine for remote channels. One can add streams by
// calling AddStream(), which creates a remote sender channel and a local
// receive channel. The remote sender channel plays a file as microphone in a
// looped fashion. Received streams are mixed and played.
class ConferenceTransport: public webrtc::Transport {
public:
ConferenceTransport();
virtual ~ConferenceTransport();
/* SetRtt()
* Set RTT between local channels and reflector.
*
* Input:
* rtt_ms : RTT in milliseconds.
*/
void SetRtt(unsigned int rtt_ms);
/* AddStream()
* Adds a stream in the conference.
*
* Input:
* file_name : name of the file to be added as microphone input.
* format : format of the input file.
*
* Returns stream id.
*/
unsigned int AddStream(std::string file_name, webrtc::FileFormats format);
/* RemoveStream()
* Removes a stream with specified ID from the conference.
*
* Input:
* id : stream id.
*
* Returns false if the specified stream does not exist, true if succeeds.
*/
bool RemoveStream(unsigned int id);
/* StartPlayout()
* Starts playing out the stream with specified ID, using the default device.
*
* Input:
* id : stream id.
*
* Returns false if the specified stream does not exist, true if succeeds.
*/
bool StartPlayout(unsigned int id);
/* GetReceiverStatistics()
* Gets RTCP statistics of the stream with specified ID.
*
* Input:
* id : stream id;
* stats : pointer to a CallStatistics to store the result.
*
* Returns false if the specified stream does not exist, true if succeeds.
*/
bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
// Inherit from class webrtc::Transport.
bool SendRtp(const uint8_t* data,
size_t len,
const webrtc::PacketOptions& options) override;
bool SendRtcp(const uint8_t *data, size_t len) override;
private:
struct Packet {
enum Type { Rtp, Rtcp, } type_;
Packet() : len_(0) {}
Packet(Type type, const void* data, size_t len, uint32_t time_ms)
: type_(type), len_(len), send_time_ms_(time_ms) {
EXPECT_LE(len_, kMaxPacketSizeByte);
memcpy(data_, data, len_);
}
uint8_t data_[kMaxPacketSizeByte];
size_t len_;
uint32_t send_time_ms_;
};
static bool Run(void* transport) {
return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
}
int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
void StorePacket(Packet::Type type, const void* data, size_t len);
void SendPacket(const Packet& packet);
bool DispatchPackets();
const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_;
const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
unsigned int rtt_ms_;
unsigned int stream_count_;
std::map<unsigned int, std::pair<int, int>> streams_
GUARDED_BY(stream_crit_.get());
std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get());
int local_sender_; // Channel Id of local sender
int reflector_;
webrtc::VoiceEngine* local_voe_;
webrtc::VoEBase* local_base_;
webrtc::VoERTP_RTCP* local_rtp_rtcp_;
webrtc::VoENetwork* local_network_;
webrtc::VoiceEngine* remote_voe_;
webrtc::VoEBase* remote_base_;
webrtc::VoECodec* remote_codec_;
webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
webrtc::VoENetwork* remote_network_;
webrtc::VoEFile* remote_file_;
LoudestFilter loudest_filter_;
const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
};
} // namespace voetest
#endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
|
