Files
platform-external-webrtc/video/frame_decode_timing.cc
Evan Shrubsole 3fa9a66f22 Cap max decode delay for FrameBuffer3
When a large queue of frames builds up due to a lost frame, the decode
delay can sometimes become quite large. In this case the stream may
signal as timed out when in fact it is not. Instead, the delay should
be capped at the timeout limit.

Bug: webrtc:14168
Change-Id: I5b4e8851b2c6d7d27a698627dc1633931d7fc00e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265404
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37199}
2022-06-13 14:52:46 +00:00

60 lines
2.4 KiB
C++

/*
* Copyright (c) 2022 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/frame_decode_timing.h"
#include <algorithm>
#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "rtc_base/logging.h"
namespace webrtc {
FrameDecodeTiming::FrameDecodeTiming(Clock* clock,
webrtc::VCMTiming const* timing)
: clock_(clock), timing_(timing) {
RTC_DCHECK(clock_);
RTC_DCHECK(timing_);
}
absl::optional<FrameDecodeTiming::FrameSchedule>
FrameDecodeTiming::OnFrameBufferUpdated(uint32_t next_temporal_unit_rtp,
uint32_t last_temporal_unit_rtp,
TimeDelta max_wait_for_frame,
bool too_many_frames_queued) {
RTC_DCHECK_GT(max_wait_for_frame, TimeDelta::Zero());
const Timestamp now = clock_->CurrentTime();
Timestamp render_time = timing_->RenderTime(next_temporal_unit_rtp, now);
TimeDelta max_wait =
timing_->MaxWaitingTime(render_time, now, too_many_frames_queued);
// If the delay is not too far in the past, or this is the last decodable
// frame then it is the best frame to be decoded. Otherwise, fast-forward
// to the next frame in the buffer.
if (max_wait <= -kMaxAllowedFrameDelay &&
next_temporal_unit_rtp != last_temporal_unit_rtp) {
RTC_DLOG(LS_VERBOSE) << "Fast-forwarded frame " << next_temporal_unit_rtp
<< " render time " << render_time.ms()
<< " with delay " << max_wait.ms() << "ms";
return absl::nullopt;
}
RTC_DLOG(LS_VERBOSE) << "Selected frame with rtp " << next_temporal_unit_rtp
<< " render time " << render_time.ms()
<< " with a max wait of " << max_wait.ms() << "ms";
max_wait.Clamp(TimeDelta::Zero(), max_wait_for_frame);
Timestamp latest_decode_time = now + max_wait;
return FrameSchedule{.latest_decode_time = latest_decode_time,
.render_time = render_time};
}
} // namespace webrtc