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63b3095d2b
This also changes the meaning of |estimated_capture_clock_offset| in |absolute_capture_time_| to become a remote to capturer clock offset. Bug: chromium:1056230, webrtc:10739 Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#34067}
125 lines
3.7 KiB
C++
125 lines
3.7 KiB
C++
/*
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* Copyright (c) 2019 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/rtp_rtcp/source/absolute_capture_time_interpolator.h"
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#include <limits>
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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constexpr Timestamp kInvalidLastReceiveTime = Timestamp::MinusInfinity();
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} // namespace
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constexpr TimeDelta AbsoluteCaptureTimeInterpolator::kInterpolationMaxInterval;
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AbsoluteCaptureTimeInterpolator::AbsoluteCaptureTimeInterpolator(Clock* clock)
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: clock_(clock), last_receive_time_(kInvalidLastReceiveTime) {}
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uint32_t AbsoluteCaptureTimeInterpolator::GetSource(
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uint32_t ssrc,
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rtc::ArrayView<const uint32_t> csrcs) {
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if (csrcs.empty()) {
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return ssrc;
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}
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return csrcs[0];
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}
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absl::optional<AbsoluteCaptureTime>
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AbsoluteCaptureTimeInterpolator::OnReceivePacket(
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uint32_t source,
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uint32_t rtp_timestamp,
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uint32_t rtp_clock_frequency,
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const absl::optional<AbsoluteCaptureTime>& received_extension) {
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const Timestamp receive_time = clock_->CurrentTime();
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MutexLock lock(&mutex_);
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AbsoluteCaptureTime extension;
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if (received_extension == absl::nullopt) {
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if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp,
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rtp_clock_frequency)) {
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last_receive_time_ = kInvalidLastReceiveTime;
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return absl::nullopt;
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}
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extension.absolute_capture_timestamp = InterpolateAbsoluteCaptureTimestamp(
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rtp_timestamp, rtp_clock_frequency, last_rtp_timestamp_,
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last_absolute_capture_timestamp_);
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extension.estimated_capture_clock_offset =
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last_estimated_capture_clock_offset_;
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} else {
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last_source_ = source;
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last_rtp_timestamp_ = rtp_timestamp;
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last_rtp_clock_frequency_ = rtp_clock_frequency;
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last_absolute_capture_timestamp_ =
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received_extension->absolute_capture_timestamp;
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last_estimated_capture_clock_offset_ =
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received_extension->estimated_capture_clock_offset;
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last_receive_time_ = receive_time;
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extension = *received_extension;
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}
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return extension;
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}
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uint64_t AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
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uint32_t rtp_timestamp,
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uint32_t rtp_clock_frequency,
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uint32_t last_rtp_timestamp,
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uint64_t last_absolute_capture_timestamp) {
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RTC_DCHECK_GT(rtp_clock_frequency, 0);
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return last_absolute_capture_timestamp +
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static_cast<int64_t>(
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rtc::dchecked_cast<uint64_t>(rtp_timestamp - last_rtp_timestamp)
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<< 32) /
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rtp_clock_frequency;
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}
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bool AbsoluteCaptureTimeInterpolator::ShouldInterpolateExtension(
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Timestamp receive_time,
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uint32_t source,
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uint32_t rtp_timestamp,
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uint32_t rtp_clock_frequency) const {
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// Shouldn't if we don't have a previously received extension stored.
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if (last_receive_time_ == kInvalidLastReceiveTime) {
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return false;
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}
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// Shouldn't if the last received extension is too old.
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if ((receive_time - last_receive_time_) > kInterpolationMaxInterval) {
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return false;
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}
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// Shouldn't if the source has changed.
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if (last_source_ != source) {
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return false;
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}
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// Shouldn't if the RTP clock frequency has changed.
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if (last_rtp_clock_frequency_ != rtp_clock_frequency) {
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return false;
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}
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// Shouldn't if the RTP clock frequency is invalid.
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if (rtp_clock_frequency <= 0) {
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return false;
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}
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return true;
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}
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} // namespace webrtc
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