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/include/module_common_types.h"
16 #include "webrtc/typedefs.h"
17 
18 namespace webrtc {
19 namespace test {
20 
21 // Class for generating RTP headers.
22 class 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)
seq_number_(start_seq_number)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 
~RtpGenerator()37   virtual ~RtpGenerator() {}
38 
39   // Writes the next RTP header to |rtp_header|, which will be of type
40   // |payload_type|. Returns the send time for this packet (in ms). The value of
41   // |payload_length_samples| determines the send time for the next packet.
42   virtual uint32_t GetRtpHeader(uint8_t payload_type,
43                                 size_t payload_length_samples,
44                                 WebRtcRTPHeader* rtp_header);
45 
46   void set_drift_factor(double factor);
47 
48  protected:
49   uint16_t seq_number_;
50   uint32_t timestamp_;
51   uint32_t next_send_time_ms_;
52   const uint32_t ssrc_;
53   const int samples_per_ms_;
54   double drift_factor_;
55 
56  private:
57   RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
58 };
59 
60 class TimestampJumpRtpGenerator : public RtpGenerator {
61  public:
TimestampJumpRtpGenerator(int samples_per_ms,uint16_t start_seq_number,uint32_t start_timestamp,uint32_t jump_from_timestamp,uint32_t jump_to_timestamp)62   TimestampJumpRtpGenerator(int samples_per_ms,
63                             uint16_t start_seq_number,
64                             uint32_t start_timestamp,
65                             uint32_t jump_from_timestamp,
66                             uint32_t jump_to_timestamp)
67       : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp),
68         jump_from_timestamp_(jump_from_timestamp),
69         jump_to_timestamp_(jump_to_timestamp) {}
70 
71   uint32_t GetRtpHeader(uint8_t payload_type,
72                         size_t payload_length_samples,
73                         WebRtcRTPHeader* rtp_header) override;
74 
75  private:
76   uint32_t jump_from_timestamp_;
77   uint32_t jump_to_timestamp_;
78   RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
79 };
80 
81 }  // namespace test
82 }  // namespace webrtc
83 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
84