f7a1937e70
Ensures that frames are decoded instantly when in low-latency render mode. This also tests the max queue size behaviour. Adds a new test suite for FrameBufferProxy that sets the appropriate field trials. * Fixes FrameDecodeTiming to never use negative wait times for decode timestamps. R=kron@webrtc.org Change-Id: I06cbec52e1e866e21aa964b24c4fd0163c26961b Bug: webrtc:13658 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251601 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35999}
58 lines
2.3 KiB
C++
58 lines
2.3 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,
|
|
bool too_many_frames_queued) {
|
|
const Timestamp now = clock_->CurrentTime();
|
|
Timestamp render_time = Timestamp::Millis(
|
|
timing_->RenderTimeMs(next_temporal_unit_rtp, now.ms()));
|
|
TimeDelta max_wait = TimeDelta::Millis(timing_->MaxWaitingTime(
|
|
render_time.ms(), now.ms(), 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";
|
|
|
|
Timestamp latest_decode_time = now + std::max(max_wait, TimeDelta::Zero());
|
|
return FrameSchedule{.latest_decode_time = latest_decode_time,
|
|
.render_time = render_time};
|
|
}
|
|
|
|
} // namespace webrtc
|