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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
61 lines
2 KiB
C++
61 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 <assert.h>
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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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assert(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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assert(samples_per_ms_ > 0);
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next_send_time_ms_ += ((1.0 + drift_factor_) * payload_length_samples) /
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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(
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payload_type, 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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