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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}
83 lines
2.3 KiB
C++
83 lines
2.3 KiB
C++
/*
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* Copyright (c) 2018 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/video_coding/utility/framerate_controller.h"
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#include <stddef.h>
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#include <cstdint>
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namespace webrtc {
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FramerateController::FramerateController(float target_framerate_fps)
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: min_frame_interval_ms_(0), framerate_estimator_(1000.0, 1000.0) {
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SetTargetRate(target_framerate_fps);
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}
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void FramerateController::SetTargetRate(float target_framerate_fps) {
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if (target_framerate_fps_ != target_framerate_fps) {
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framerate_estimator_.Reset();
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if (last_timestamp_ms_) {
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framerate_estimator_.Update(1, *last_timestamp_ms_);
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}
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const size_t target_frame_interval_ms = 1000 / target_framerate_fps;
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target_framerate_fps_ = target_framerate_fps;
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min_frame_interval_ms_ = 85 * target_frame_interval_ms / 100;
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}
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}
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float FramerateController::GetTargetRate() {
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return *target_framerate_fps_;
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}
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void FramerateController::Reset() {
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framerate_estimator_.Reset();
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last_timestamp_ms_.reset();
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}
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bool FramerateController::DropFrame(uint32_t timestamp_ms) const {
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if (timestamp_ms < last_timestamp_ms_) {
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// Timestamp jumps backward. We can't make adequate drop decision. Don't
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// drop this frame. Stats will be reset in AddFrame().
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return false;
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}
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if (Rate(timestamp_ms).value_or(*target_framerate_fps_) >
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target_framerate_fps_) {
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return true;
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}
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if (last_timestamp_ms_) {
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const int64_t diff_ms =
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static_cast<int64_t>(timestamp_ms) - *last_timestamp_ms_;
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if (diff_ms < min_frame_interval_ms_) {
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return true;
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}
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}
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return false;
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}
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void FramerateController::AddFrame(uint32_t timestamp_ms) {
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if (timestamp_ms < last_timestamp_ms_) {
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// Timestamp jumps backward.
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Reset();
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}
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framerate_estimator_.Update(1, timestamp_ms);
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last_timestamp_ms_ = timestamp_ms;
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}
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absl::optional<float> FramerateController::Rate(uint32_t timestamp_ms) const {
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return framerate_estimator_.Rate(timestamp_ms);
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}
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} // namespace webrtc
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