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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
90 lines
3 KiB
C++
90 lines
3 KiB
C++
/*
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* Copyright (c) 2016 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/playout_delay_oracle.h"
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#include <algorithm>
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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PlayoutDelayOracle::PlayoutDelayOracle() = default;
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PlayoutDelayOracle::~PlayoutDelayOracle() = default;
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absl::optional<PlayoutDelay> PlayoutDelayOracle::PlayoutDelayToSend(
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PlayoutDelay requested_delay) const {
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rtc::CritScope lock(&crit_sect_);
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if (requested_delay.min_ms > PlayoutDelayLimits::kMaxMs ||
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requested_delay.max_ms > PlayoutDelayLimits::kMaxMs) {
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RTC_DLOG(LS_ERROR)
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<< "Requested playout delay values out of range, ignored";
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return absl::nullopt;
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}
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if (requested_delay.max_ms != -1 &&
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requested_delay.min_ms > requested_delay.max_ms) {
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RTC_DLOG(LS_ERROR) << "Requested playout delay values out of order";
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return absl::nullopt;
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}
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if ((requested_delay.min_ms == -1 ||
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requested_delay.min_ms == latest_delay_.min_ms) &&
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(requested_delay.max_ms == -1 ||
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requested_delay.max_ms == latest_delay_.max_ms)) {
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// Unchanged.
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return unacked_sequence_number_ ? absl::make_optional(latest_delay_)
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: absl::nullopt;
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}
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if (requested_delay.min_ms == -1) {
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RTC_DCHECK_GE(requested_delay.max_ms, 0);
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requested_delay.min_ms =
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std::min(latest_delay_.min_ms, requested_delay.max_ms);
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}
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if (requested_delay.max_ms == -1) {
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requested_delay.max_ms =
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std::max(latest_delay_.max_ms, requested_delay.min_ms);
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}
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return requested_delay;
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}
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void PlayoutDelayOracle::OnSentPacket(uint16_t sequence_number,
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absl::optional<PlayoutDelay> delay) {
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rtc::CritScope lock(&crit_sect_);
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int64_t unwrapped_sequence_number = unwrapper_.Unwrap(sequence_number);
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if (!delay) {
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return;
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}
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RTC_DCHECK_LE(0, delay->min_ms);
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RTC_DCHECK_LE(delay->max_ms, PlayoutDelayLimits::kMaxMs);
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RTC_DCHECK_LE(delay->min_ms, delay->max_ms);
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if (delay->min_ms != latest_delay_.min_ms ||
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delay->max_ms != latest_delay_.max_ms) {
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latest_delay_ = *delay;
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unacked_sequence_number_ = unwrapped_sequence_number;
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}
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}
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// If an ACK is received on the packet containing the playout delay extension,
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// we stop sending the extension on future packets.
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void PlayoutDelayOracle::OnReceivedAck(
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int64_t extended_highest_sequence_number) {
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rtc::CritScope lock(&crit_sect_);
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if (unacked_sequence_number_ &&
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extended_highest_sequence_number > *unacked_sequence_number_) {
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unacked_sequence_number_ = absl::nullopt;
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}
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}
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} // namespace webrtc
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