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 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 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