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 "api/rtp_headers.h" 15 16 namespace webrtc { 17 namespace test { 18 19 // Class for generating RTP headers. 20 class RtpGenerator { 21 public: 22 RtpGenerator(int samples_per_ms, 23 uint16_t start_seq_number = 0, 24 uint32_t start_timestamp = 0, 25 uint32_t start_send_time_ms = 0, 26 uint32_t ssrc = 0x12345678) seq_number_(start_seq_number)27 : seq_number_(start_seq_number), 28 timestamp_(start_timestamp), 29 next_send_time_ms_(start_send_time_ms), 30 ssrc_(ssrc), 31 samples_per_ms_(samples_per_ms), 32 drift_factor_(0.0) {} 33 ~RtpGenerator()34 virtual ~RtpGenerator() {} 35 36 RtpGenerator(const RtpGenerator&) = delete; 37 RtpGenerator& operator=(const RtpGenerator&) = delete; 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 RTPHeader* 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 57 class TimestampJumpRtpGenerator : public RtpGenerator { 58 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)59 TimestampJumpRtpGenerator(int samples_per_ms, 60 uint16_t start_seq_number, 61 uint32_t start_timestamp, 62 uint32_t jump_from_timestamp, 63 uint32_t jump_to_timestamp) 64 : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp), 65 jump_from_timestamp_(jump_from_timestamp), 66 jump_to_timestamp_(jump_to_timestamp) {} 67 68 TimestampJumpRtpGenerator(const TimestampJumpRtpGenerator&) = delete; 69 TimestampJumpRtpGenerator& operator=(const TimestampJumpRtpGenerator&) = 70 delete; 71 72 uint32_t GetRtpHeader(uint8_t payload_type, 73 size_t payload_length_samples, 74 RTPHeader* rtp_header) override; 75 76 private: 77 uint32_t jump_from_timestamp_; 78 uint32_t jump_to_timestamp_; 79 }; 80 81 } // namespace test 82 } // namespace webrtc 83 #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ 84