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/*
* Copyright (c) 2013 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.
*/
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
namespace webrtc {
namespace test {
uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
size_t payload_length_samples,
RTPHeader* rtp_header) {
RTC_DCHECK(rtp_header);
if (!rtp_header) {
return 0;
}
rtp_header->sequenceNumber = seq_number_++;
rtp_header->timestamp = timestamp_;
timestamp_ += static_cast<uint32_t>(payload_length_samples);
rtp_header->payloadType = payload_type;
rtp_header->markerBit = false;
rtp_header->ssrc = ssrc_;
rtp_header->numCSRCs = 0;
uint32_t this_send_time = next_send_time_ms_;
RTC_DCHECK_GT(samples_per_ms_, 0);
next_send_time_ms_ +=
((1.0 + drift_factor_) * payload_length_samples) / samples_per_ms_;
return this_send_time;
}
void RtpGenerator::set_drift_factor(double factor) {
if (factor > -1.0) {
drift_factor_ = factor;
}
}
uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
size_t payload_length_samples,
RTPHeader* rtp_header) {
uint32_t ret = RtpGenerator::GetRtpHeader(payload_type,
payload_length_samples, rtp_header);
if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <=
jump_from_timestamp_ &&
timestamp_ > jump_from_timestamp_) {
// We just moved across the `jump_from_timestamp_` timestamp. Do the jump.
timestamp_ = jump_to_timestamp_;
}
return ret;
}
} // namespace test
} // namespace webrtc