rtp_generator.h revision a90f6d67f72359cf63b59480fa87a13aae808c03 
1/*
2 *  Copyright (c) 2012 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_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
12#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
13
14#include "webrtc/base/constructormagic.h"
15#include "webrtc/modules/interface/module_common_types.h"
16#include "webrtc/typedefs.h"
17
18namespace webrtc {
19namespace test {
20
21// Class for generating RTP headers.
22class RtpGenerator {
23 public:
24  RtpGenerator(int samples_per_ms,
25               uint16_t start_seq_number = 0,
26               uint32_t start_timestamp = 0,
27               uint32_t start_send_time_ms = 0,
28               uint32_t ssrc = 0x12345678)
29      : seq_number_(start_seq_number),
30        timestamp_(start_timestamp),
31        next_send_time_ms_(start_send_time_ms),
32        ssrc_(ssrc),
33        samples_per_ms_(samples_per_ms),
34        drift_factor_(0.0) {
35  }
36
37  // Writes the next RTP header to |rtp_header|, which will be of type
38  // |payload_type|. Returns the send time for this packet (in ms). The value of
39  // |payload_length_samples| determines the send time for the next packet.
40  uint32_t GetRtpHeader(uint8_t payload_type, size_t payload_length_samples,
41                        WebRtcRTPHeader* rtp_header);
42
43  void set_drift_factor(double factor);
44
45 private:
46  uint16_t seq_number_;
47  uint32_t timestamp_;
48  uint32_t next_send_time_ms_;
49  const uint32_t ssrc_;
50  const int samples_per_ms_;
51  double drift_factor_;
52  DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
53};
54
55}  // namespace test
56}  // namespace webrtc
57#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
58