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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc')
-rw-r--r-- | third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc | 110 |
1 files changed, 110 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc new file mode 100644 index 0000000000..f151084c7d --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc @@ -0,0 +1,110 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h" + +#include <limits> + +#include "rtc_base/checks.h" + +namespace webrtc { + +AbsoluteCaptureTimeInterpolator::AbsoluteCaptureTimeInterpolator(Clock* clock) + : clock_(clock) {} + +uint32_t AbsoluteCaptureTimeInterpolator::GetSource( + uint32_t ssrc, + rtc::ArrayView<const uint32_t> csrcs) { + if (csrcs.empty()) { + return ssrc; + } + + return csrcs[0]; +} + +absl::optional<AbsoluteCaptureTime> +AbsoluteCaptureTimeInterpolator::OnReceivePacket( + uint32_t source, + uint32_t rtp_timestamp, + int rtp_clock_frequency_hz, + const absl::optional<AbsoluteCaptureTime>& received_extension) { + const Timestamp receive_time = clock_->CurrentTime(); + + MutexLock lock(&mutex_); + + if (received_extension == absl::nullopt) { + if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp, + rtp_clock_frequency_hz)) { + last_receive_time_ = Timestamp::MinusInfinity(); + return absl::nullopt; + } + + return AbsoluteCaptureTime{ + .absolute_capture_timestamp = InterpolateAbsoluteCaptureTimestamp( + rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_, + last_received_extension_.absolute_capture_timestamp), + .estimated_capture_clock_offset = + last_received_extension_.estimated_capture_clock_offset, + }; + } else { + last_source_ = source; + last_rtp_timestamp_ = rtp_timestamp; + last_rtp_clock_frequency_hz_ = rtp_clock_frequency_hz; + last_received_extension_ = *received_extension; + + last_receive_time_ = receive_time; + + return received_extension; + } +} + +uint64_t AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp( + uint32_t rtp_timestamp, + int rtp_clock_frequency_hz, + uint32_t last_rtp_timestamp, + uint64_t last_absolute_capture_timestamp) { + RTC_DCHECK_GT(rtp_clock_frequency_hz, 0); + + return last_absolute_capture_timestamp + + static_cast<int64_t>(uint64_t{rtp_timestamp - last_rtp_timestamp} + << 32) / + rtp_clock_frequency_hz; +} + +bool AbsoluteCaptureTimeInterpolator::ShouldInterpolateExtension( + Timestamp receive_time, + uint32_t source, + uint32_t rtp_timestamp, + int rtp_clock_frequency_hz) const { + // Shouldn't if the last received extension is not eligible for interpolation, + // in particular if we don't have a previously received extension stored. + if (receive_time - last_receive_time_ > kInterpolationMaxInterval) { + return false; + } + + // Shouldn't if the source has changed. + if (last_source_ != source) { + return false; + } + + // Shouldn't if the RTP clock frequency has changed. + if (last_rtp_clock_frequency_hz_ != rtp_clock_frequency_hz) { + return false; + } + + // Shouldn't if the RTP clock frequency is invalid. + if (rtp_clock_frequency_hz <= 0) { + return false; + } + + return true; +} + +} // namespace webrtc |