| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ |
| |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| // Class for generating RTP headers. |
| class RtpGenerator { |
| public: |
| RtpGenerator(int samples_per_ms, |
| uint16_t start_seq_number = 0, |
| uint32_t start_timestamp = 0, |
| uint32_t start_send_time_ms = 0, |
| uint32_t ssrc = 0x12345678) |
| : seq_number_(start_seq_number), |
| timestamp_(start_timestamp), |
| next_send_time_ms_(start_send_time_ms), |
| ssrc_(ssrc), |
| samples_per_ms_(samples_per_ms), |
| drift_factor_(0.0) { |
| } |
| |
| virtual ~RtpGenerator() {} |
| |
| // Writes the next RTP header to |rtp_header|, which will be of type |
| // |payload_type|. Returns the send time for this packet (in ms). The value of |
| // |payload_length_samples| determines the send time for the next packet. |
| virtual uint32_t GetRtpHeader(uint8_t payload_type, |
| size_t payload_length_samples, |
| WebRtcRTPHeader* rtp_header); |
| |
| void set_drift_factor(double factor); |
| |
| protected: |
| uint16_t seq_number_; |
| uint32_t timestamp_; |
| uint32_t next_send_time_ms_; |
| const uint32_t ssrc_; |
| const int samples_per_ms_; |
| double drift_factor_; |
| |
| private: |
| RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator); |
| }; |
| |
| class TimestampJumpRtpGenerator : public RtpGenerator { |
| 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) |
| : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp), |
| jump_from_timestamp_(jump_from_timestamp), |
| jump_to_timestamp_(jump_to_timestamp) {} |
| |
| uint32_t GetRtpHeader(uint8_t payload_type, |
| size_t payload_length_samples, |
| WebRtcRTPHeader* rtp_header) override; |
| |
| private: |
| uint32_t jump_from_timestamp_; |
| uint32_t jump_to_timestamp_; |
| RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator); |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ |