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
|
/*
* Copyright (c) 2012 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.
*/
#include "webrtc/modules/video_coding/main/test/mt_test_common.h"
#include <math.h>
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
#include "webrtc/modules/utility/interface/rtp_dump.h"
#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
TransportCallback::TransportCallback(Clock* clock, const char* filename)
: RTPSendCompleteCallback(clock, filename) {
}
TransportCallback::~TransportCallback()
{
//
}
int
TransportCallback::SendPacket(int channel, const void *data, size_t len)
{
_sendCount++;
_totalSentLength += len;
if (_rtpDump != NULL)
{
if (_rtpDump->DumpPacket((const uint8_t*)data, len) != 0)
{
return -1;
}
}
bool transmitPacket = true;
// Off-line tests, don't drop first Key frame (approx.)
if (_sendCount > 20)
{
transmitPacket = PacketLoss();
}
Clock* clock = Clock::GetRealTimeClock();
int64_t now = clock->TimeInMilliseconds();
// Insert outgoing packet into list
if (transmitPacket)
{
RtpPacket* newPacket = new RtpPacket();
memcpy(newPacket->data, data, len);
newPacket->length = len;
// Simulate receive time = network delay + packet jitter
// simulated as a Normal distribution random variable with
// mean = networkDelay and variance = jitterVar
int32_t
simulatedDelay = (int32_t)NormalDist(_networkDelayMs,
sqrt(_jitterVar));
newPacket->receiveTime = now + simulatedDelay;
_rtpPackets.push_back(newPacket);
}
return 0;
}
int
TransportCallback::TransportPackets()
{
// Are we ready to send packets to the receiver?
RtpPacket* packet = NULL;
Clock* clock = Clock::GetRealTimeClock();
int64_t now = clock->TimeInMilliseconds();
while (!_rtpPackets.empty())
{
// Take first packet in list
packet = _rtpPackets.front();
int64_t timeToReceive = packet->receiveTime - now;
if (timeToReceive > 0)
{
// No available packets to send
break;
}
_rtpPackets.pop_front();
// Send to receive side
RTPHeader header;
rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
if (!parser->Parse(packet->data, packet->length, &header)) {
delete packet;
return -1;
}
PayloadUnion payload_specific;
if (!rtp_payload_registry_->GetPayloadSpecifics(
header.payloadType, &payload_specific)) {
return -1;
}
if (!rtp_receiver_->IncomingRtpPacket(header, packet->data,
packet->length, payload_specific,
true))
{
delete packet;
return -1;
}
delete packet;
packet = NULL;
}
return 0; // OK
}
bool VCMProcessingThread(void* obj)
{
SharedRTPState* state = static_cast<SharedRTPState*>(obj);
if (state->_vcm.TimeUntilNextProcess() <= 0)
{
if (state->_vcm.Process() < 0)
{
return false;
}
}
return true;
}
bool VCMDecodeThread(void* obj)
{
SharedRTPState* state = static_cast<SharedRTPState*>(obj);
state->_vcm.Decode();
return true;
}
bool TransportThread(void *obj)
{
SharedTransportState* state = static_cast<SharedTransportState*>(obj);
state->_transport.TransportPackets();
return true;
}
} // namespace webrtc
|
