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
|
/*
* 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.
*/
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
#include <bitset>
#include <cstdint>
#include <memory>
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
namespace webrtc {
namespace test {
// Interface class for an object delivering RTP packets to test applications.
class PacketSource {
public:
PacketSource();
virtual ~PacketSource();
PacketSource(const PacketSource&) = delete;
PacketSource& operator=(const PacketSource&) = delete;
// Returns next packet. Returns nullptr if the source is depleted, or if an
// error occurred.
virtual std::unique_ptr<RtpPacketReceived> NextPacket() = 0;
virtual void FilterOutPayloadType(uint8_t payload_type);
protected:
std::bitset<128> filter_; // Payload type is 7 bits in the RFC.
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
|
