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
|
/*
* Copyright (c) 2014 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/audio_coding/neteq/tools/rtp_file_source.h"
#include <assert.h>
#include <string.h>
#ifdef WIN32
#include <winsock2.h>
#else
#include <netinet/in.h>
#endif
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/test/rtp_file_reader.h"
namespace webrtc {
namespace test {
RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
RtpFileSource* source = new RtpFileSource();
CHECK(source->OpenFile(file_name));
return source;
}
RtpFileSource::~RtpFileSource() {
}
bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
assert(parser_.get());
return parser_->RegisterRtpHeaderExtension(type, id);
}
Packet* RtpFileSource::NextPacket() {
while (true) {
RtpPacket temp_packet;
if (!rtp_reader_->NextPacket(&temp_packet)) {
return NULL;
}
if (temp_packet.length == 0) {
// May be an RTCP packet.
// Read the next one.
continue;
}
scoped_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
scoped_ptr<Packet> packet(new Packet(packet_memory.release(),
temp_packet.length,
temp_packet.original_length,
temp_packet.time_ms,
*parser_.get()));
if (!packet->valid_header()) {
assert(false);
return NULL;
}
if (filter_.test(packet->header().payloadType) ||
(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
// This payload type should be filtered out. Continue to the next packet.
continue;
}
return packet.release();
}
}
RtpFileSource::RtpFileSource()
: PacketSource(),
parser_(RtpHeaderParser::Create()) {}
bool RtpFileSource::OpenFile(const std::string& file_name) {
rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
if (rtp_reader_)
return true;
rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
if (!rtp_reader_) {
FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
"that .pcapng is not supported.";
}
return true;
}
} // namespace test
} // namespace webrtc
|
