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/*
* Copyright (c) 2012 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 "video/render/video_render_frames.h"
#include <type_traits>
#include <utility>
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
// Don't render frames with timestamp older than 500ms from now.
const int kOldRenderTimestampMS = 500;
// Don't render frames with timestamp more than 10s into the future.
const int kFutureRenderTimestampMS = 10000;
const uint32_t kEventMaxWaitTimeMs = 200;
const uint32_t kMinRenderDelayMs = 10;
const uint32_t kMaxRenderDelayMs = 500;
const size_t kMaxIncomingFramesBeforeLogged = 100;
uint32_t EnsureValidRenderDelay(uint32_t render_delay) {
return (render_delay < kMinRenderDelayMs || render_delay > kMaxRenderDelayMs)
? kMinRenderDelayMs
: render_delay;
}
} // namespace
VideoRenderFrames::VideoRenderFrames(uint32_t render_delay_ms)
: render_delay_ms_(EnsureValidRenderDelay(render_delay_ms)) {}
VideoRenderFrames::~VideoRenderFrames() {
frames_dropped_ += incoming_frames_.size();
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DroppedFrames.RenderQueue",
frames_dropped_);
RTC_LOG(LS_INFO) << "WebRTC.Video.DroppedFrames.RenderQueue "
<< frames_dropped_;
}
int32_t VideoRenderFrames::AddFrame(VideoFrame&& new_frame) {
const int64_t time_now = rtc::TimeMillis();
// Drop old frames only when there are other frames in the queue, otherwise, a
// really slow system never renders any frames.
if (!incoming_frames_.empty() &&
new_frame.render_time_ms() + kOldRenderTimestampMS < time_now) {
RTC_LOG(LS_WARNING) << "Too old frame, timestamp="
<< new_frame.rtp_timestamp();
++frames_dropped_;
return -1;
}
if (new_frame.render_time_ms() > time_now + kFutureRenderTimestampMS) {
RTC_LOG(LS_WARNING) << "Frame too long into the future, timestamp="
<< new_frame.rtp_timestamp();
++frames_dropped_;
return -1;
}
if (new_frame.render_time_ms() < last_render_time_ms_) {
RTC_LOG(LS_WARNING) << "Frame scheduled out of order, render_time="
<< new_frame.render_time_ms()
<< ", latest=" << last_render_time_ms_;
// For more details, see bug:
++frames_dropped_;
return -1;
}
last_render_time_ms_ = new_frame.render_time_ms();
incoming_frames_.emplace_back(std::move(new_frame));
if (incoming_frames_.size() > kMaxIncomingFramesBeforeLogged) {
RTC_LOG(LS_WARNING) << "Stored incoming frames: "
<< incoming_frames_.size();
}
return static_cast<int32_t>(incoming_frames_.size());
}
absl::optional<VideoFrame> VideoRenderFrames::FrameToRender() {
absl::optional<VideoFrame> render_frame;
// Get the newest frame that can be released for rendering.
while (!incoming_frames_.empty() && TimeToNextFrameRelease() <= 0) {
if (render_frame) {
++frames_dropped_;
}
render_frame = std::move(incoming_frames_.front());
incoming_frames_.pop_front();
}
return render_frame;
}
uint32_t VideoRenderFrames::TimeToNextFrameRelease() {
if (incoming_frames_.empty()) {
return kEventMaxWaitTimeMs;
}
const int64_t time_to_release = incoming_frames_.front().render_time_ms() -
render_delay_ms_ - rtc::TimeMillis();
return time_to_release < 0 ? 0u : static_cast<uint32_t>(time_to_release);
}
bool VideoRenderFrames::HasPendingFrames() const {
return !incoming_frames_.empty();
}
} // namespace webrtc