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