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}
121 lines
4.2 KiB
C++
121 lines
4.2 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_interpolator.h"
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#include "system_wrappers/include/ntp_time.h"
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namespace webrtc {
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static_assert(
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AbsoluteCaptureTimeInterpolator::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) {}
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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 AbsoluteCaptureTimeInterpolator::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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return OnSendPacket(source, rtp_timestamp, rtp_clock_frequency,
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NtpTime(absolute_capture_timestamp),
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estimated_capture_clock_offset, /*force=*/false);
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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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int rtp_clock_frequency_hz,
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NtpTime absolute_capture_time,
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absl::optional<int64_t> estimated_capture_clock_offset,
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bool force) {
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Timestamp send_time = clock_->CurrentTime();
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if (!(force || ShouldSendExtension(
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send_time, source, rtp_timestamp, rtp_clock_frequency_hz,
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absolute_capture_time, 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_hz_ = rtp_clock_frequency_hz;
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last_absolute_capture_time_ = absolute_capture_time;
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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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return AbsoluteCaptureTime{
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.absolute_capture_timestamp = uint64_t{absolute_capture_time},
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.estimated_capture_clock_offset = estimated_capture_clock_offset,
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};
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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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int rtp_clock_frequency_hz,
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NtpTime absolute_capture_time,
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absl::optional<int64_t> estimated_capture_clock_offset) const {
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// Should if the last sent extension is too old, in particular if we've never
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// sent anything before.
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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_hz_ != rtp_clock_frequency_hz) {
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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_hz <= 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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AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
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rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_,
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uint64_t{last_absolute_capture_time_});
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const uint64_t absolute_capture_timestamp = uint64_t{absolute_capture_time};
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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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