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/*
* 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/interface/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_
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