05e7bfeeea
Review URL: https://webrtc-codereview.appspot.com/1052004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3405 4adac7df-926f-26a2-2b94-8c16560cd09d
54 lines
1.8 KiB
C++
54 lines
1.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/system_wrappers/interface/constructor_magic.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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namespace test {
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// Class for generating RTP headers.
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class RtpGenerator {
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public:
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RtpGenerator(int samples_per_ms,
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uint16_t start_seq_number = 0,
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uint32_t start_timestamp = 0,
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uint32_t start_send_time_ms = 0,
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uint32_t ssrc = 0x12345678)
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: seq_number_(start_seq_number),
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timestamp_(start_timestamp),
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next_send_time_ms_(start_send_time_ms),
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ssrc_(ssrc),
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samples_per_ms_(samples_per_ms) {
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}
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// Writes the next RTP header to |rtp_header|, wich will be of type
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// |payload_type|. Returns the send time for this packet (in ms). The value of
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// |payload_length_samples| determines the send time for the next packet.
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uint32_t GetRtpHeader(uint8_t payload_type, size_t payload_length_samples,
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WebRtcRTPHeader* rtp_header);
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private:
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uint16_t seq_number_;
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uint32_t timestamp_;
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uint32_t next_send_time_ms_;
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const uint32_t ssrc_;
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const int samples_per_ms_;
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DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
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