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d00ce747c7
Bug: webrtc:12338 Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34621}
60 lines
2 KiB
C++
60 lines
2 KiB
C++
/*
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* Copyright (c) 2013 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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#include "modules/audio_coding/neteq/tools/rtp_generator.h"
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namespace webrtc {
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namespace test {
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uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
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size_t payload_length_samples,
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RTPHeader* rtp_header) {
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RTC_DCHECK(rtp_header);
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if (!rtp_header) {
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return 0;
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}
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rtp_header->sequenceNumber = seq_number_++;
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rtp_header->timestamp = timestamp_;
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timestamp_ += static_cast<uint32_t>(payload_length_samples);
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rtp_header->payloadType = payload_type;
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rtp_header->markerBit = false;
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rtp_header->ssrc = ssrc_;
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rtp_header->numCSRCs = 0;
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uint32_t this_send_time = next_send_time_ms_;
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RTC_DCHECK_GT(samples_per_ms_, 0);
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next_send_time_ms_ +=
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((1.0 + drift_factor_) * payload_length_samples) / samples_per_ms_;
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return this_send_time;
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}
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void RtpGenerator::set_drift_factor(double factor) {
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if (factor > -1.0) {
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drift_factor_ = factor;
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}
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}
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uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
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size_t payload_length_samples,
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RTPHeader* rtp_header) {
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uint32_t ret = RtpGenerator::GetRtpHeader(payload_type,
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payload_length_samples, rtp_header);
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if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <=
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jump_from_timestamp_ &&
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timestamp_ > jump_from_timestamp_) {
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// We just moved across the `jump_from_timestamp_` timestamp. Do the jump.
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timestamp_ = jump_to_timestamp_;
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}
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return ret;
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}
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} // namespace test
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} // namespace webrtc
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