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e7c015e112
Bug: webrtc:11567 Change-Id: I4c71f3a28ef875af2c232b1b553840d6e21649d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178804 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31645}
124 lines
4 KiB
C++
124 lines
4 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_sender.h"
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#include <limits>
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#include "modules/rtp_rtcp/source/absolute_capture_time_receiver.h"
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#include "system_wrappers/include/ntp_time.h"
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namespace webrtc {
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namespace {
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constexpr Timestamp kInvalidLastSendTime = Timestamp::MinusInfinity();
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} // namespace
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constexpr TimeDelta AbsoluteCaptureTimeSender::kInterpolationMaxInterval;
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constexpr TimeDelta AbsoluteCaptureTimeSender::kInterpolationMaxError;
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static_assert(
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AbsoluteCaptureTimeReceiver::kInterpolationMaxInterval >=
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AbsoluteCaptureTimeSender::kInterpolationMaxInterval,
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"Receivers should be as willing to interpolate timestamps as senders.");
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AbsoluteCaptureTimeSender::AbsoluteCaptureTimeSender(Clock* clock)
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: clock_(clock), last_send_time_(kInvalidLastSendTime) {}
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uint32_t AbsoluteCaptureTimeSender::GetSource(
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uint32_t ssrc,
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rtc::ArrayView<const uint32_t> csrcs) {
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return AbsoluteCaptureTimeReceiver::GetSource(ssrc, csrcs);
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}
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absl::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
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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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uint64_t absolute_capture_timestamp,
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absl::optional<int64_t> estimated_capture_clock_offset) {
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const Timestamp send_time = clock_->CurrentTime();
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MutexLock lock(&mutex_);
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if (!ShouldSendExtension(send_time, source, rtp_timestamp,
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rtp_clock_frequency, absolute_capture_timestamp,
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estimated_capture_clock_offset)) {
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return absl::nullopt;
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}
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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_ = absolute_capture_timestamp;
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last_estimated_capture_clock_offset_ = estimated_capture_clock_offset;
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last_send_time_ = send_time;
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AbsoluteCaptureTime extension;
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extension.absolute_capture_timestamp = absolute_capture_timestamp;
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extension.estimated_capture_clock_offset = estimated_capture_clock_offset;
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return extension;
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}
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bool AbsoluteCaptureTimeSender::ShouldSendExtension(
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Timestamp send_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,
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uint64_t absolute_capture_timestamp,
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absl::optional<int64_t> estimated_capture_clock_offset) const {
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// Should if we've never sent anything before.
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if (last_send_time_ == kInvalidLastSendTime) {
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return true;
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}
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// Should if the last sent extension is too old.
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if ((send_time - last_send_time_) > kInterpolationMaxInterval) {
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return true;
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}
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// Should if the source has changed.
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if (last_source_ != source) {
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return true;
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}
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// Should if the RTP clock frequency has changed.
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if (last_rtp_clock_frequency_ != rtp_clock_frequency) {
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return true;
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}
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// Should if the RTP clock frequency is invalid.
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if (rtp_clock_frequency <= 0) {
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return true;
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}
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// Should if the estimated capture clock offset has changed.
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if (last_estimated_capture_clock_offset_ != estimated_capture_clock_offset) {
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return true;
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}
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// Should if interpolation would introduce too much error.
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const uint64_t interpolated_absolute_capture_timestamp =
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AbsoluteCaptureTimeReceiver::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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const int64_t interpolation_error_ms = UQ32x32ToInt64Ms(std::min(
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interpolated_absolute_capture_timestamp - absolute_capture_timestamp,
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absolute_capture_timestamp - interpolated_absolute_capture_timestamp));
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if (interpolation_error_ms > kInterpolationMaxError.ms()) {
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return true;
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}
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return false;
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}
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} // namespace webrtc
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