rtp_generator.h (2900B)
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 MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ 12 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ 13 14 #include <cstddef> 15 #include <cstdint> 16 17 #include "api/rtp_headers.h" 18 19 namespace webrtc { 20 namespace test { 21 22 // Class for generating RTP headers. 23 class RtpGenerator { 24 public: 25 RtpGenerator(int samples_per_ms, 26 uint16_t start_seq_number = 0, 27 uint32_t start_timestamp = 0, 28 uint32_t start_send_time_ms = 0, 29 uint32_t ssrc = 0x12345678) 30 : seq_number_(start_seq_number), 31 timestamp_(start_timestamp), 32 next_send_time_ms_(start_send_time_ms), 33 ssrc_(ssrc), 34 samples_per_ms_(samples_per_ms), 35 drift_factor_(0.0) {} 36 37 virtual ~RtpGenerator() {} 38 39 RtpGenerator(const RtpGenerator&) = delete; 40 RtpGenerator& operator=(const RtpGenerator&) = delete; 41 42 // Writes the next RTP header to `rtp_header`, which will be of type 43 // `payload_type`. Returns the send time for this packet (in ms). The value of 44 // `payload_length_samples` determines the send time for the next packet. 45 virtual uint32_t GetRtpHeader(uint8_t payload_type, 46 size_t payload_length_samples, 47 RTPHeader* rtp_header); 48 49 void set_drift_factor(double factor); 50 51 protected: 52 uint16_t seq_number_; 53 uint32_t timestamp_; 54 uint32_t next_send_time_ms_; 55 const uint32_t ssrc_; 56 const int samples_per_ms_; 57 double drift_factor_; 58 }; 59 60 class TimestampJumpRtpGenerator : public RtpGenerator { 61 public: 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 TimestampJumpRtpGenerator(const TimestampJumpRtpGenerator&) = delete; 72 TimestampJumpRtpGenerator& operator=(const TimestampJumpRtpGenerator&) = 73 delete; 74 75 uint32_t GetRtpHeader(uint8_t payload_type, 76 size_t payload_length_samples, 77 RTPHeader* rtp_header) override; 78 79 private: 80 uint32_t jump_from_timestamp_; 81 uint32_t jump_to_timestamp_; 82 }; 83 84 } // namespace test 85 } // namespace webrtc 86 #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_