mirror of
https://github.com/mollyim/webrtc.git
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6634c91194
Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized. Added a feature to force producing extension as requested by downstream. Cleanup and document api: Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t Documented all the parameters. Cleanup tests. Bug: b/307553606 Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/main@{#41023}
110 lines
3.3 KiB
C++
110 lines
3.3 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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AbsoluteCaptureTimeInterpolator::AbsoluteCaptureTimeInterpolator(Clock* clock)
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: clock_(clock) {}
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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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int rtp_clock_frequency_hz,
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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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if (received_extension == absl::nullopt) {
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if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp,
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rtp_clock_frequency_hz)) {
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last_receive_time_ = Timestamp::MinusInfinity();
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return absl::nullopt;
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}
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return AbsoluteCaptureTime{
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.absolute_capture_timestamp = InterpolateAbsoluteCaptureTimestamp(
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rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_,
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last_received_extension_.absolute_capture_timestamp),
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.estimated_capture_clock_offset =
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last_received_extension_.estimated_capture_clock_offset,
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};
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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_hz_ = rtp_clock_frequency_hz;
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last_received_extension_ = *received_extension;
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last_receive_time_ = receive_time;
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return received_extension;
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}
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}
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uint64_t AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
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uint32_t rtp_timestamp,
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int rtp_clock_frequency_hz,
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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_hz, 0);
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return last_absolute_capture_timestamp +
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static_cast<int64_t>(uint64_t{rtp_timestamp - last_rtp_timestamp}
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<< 32) /
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rtp_clock_frequency_hz;
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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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int rtp_clock_frequency_hz) const {
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// Shouldn't if the last received extension is not eligible for interpolation,
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// in particular if we don't have a previously received extension stored.
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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_hz_ != rtp_clock_frequency_hz) {
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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_hz <= 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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