Danil Chapovalov 87b7c1aa6e Reduce warning logging when minimum playout delay exceed maximum
There can be error log each frame when maximum playout delay sent with the frame exceed delay derived from the av-sync.
In such scenario prefer to limit the playout delay by the one attached to the received frame.

Bug: b/390043766
Change-Id: Ia57969df72f7a649e5a9280d5bb29986f5ea14b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374284
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43814}
2025-01-28 03:34:18 -08:00

185 lines
6.8 KiB
C++

/*
* 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 "modules/video_coding/deprecated/receiver.h"
#include <cstdint>
#include <cstdlib>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/video/encoded_image.h"
#include "modules/video_coding/deprecated/jitter_buffer_common.h"
#include "modules/video_coding/encoded_frame.h"
#include "modules/video_coding/internal_defines.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
enum { kMaxReceiverDelayMs = 10000 };
VCMReceiver::VCMReceiver(VCMTiming* timing,
Clock* clock,
const FieldTrialsView& field_trials)
: VCMReceiver::VCMReceiver(timing,
clock,
absl::WrapUnique(EventWrapper::Create()),
absl::WrapUnique(EventWrapper::Create()),
field_trials) {}
VCMReceiver::VCMReceiver(VCMTiming* timing,
Clock* clock,
std::unique_ptr<EventWrapper> receiver_event,
std::unique_ptr<EventWrapper> jitter_buffer_event,
const FieldTrialsView& field_trials)
: clock_(clock),
jitter_buffer_(clock_, std::move(jitter_buffer_event), field_trials),
timing_(timing),
render_wait_event_(std::move(receiver_event)),
max_video_delay_ms_(kMaxVideoDelayMs) {
jitter_buffer_.Start();
}
VCMReceiver::~VCMReceiver() {
render_wait_event_->Set();
}
int32_t VCMReceiver::InsertPacket(const VCMPacket& packet) {
// Insert the packet into the jitter buffer. The packet can either be empty or
// contain media at this point.
bool retransmitted = false;
const VCMFrameBufferEnum ret =
jitter_buffer_.InsertPacket(packet, &retransmitted);
if (ret == kOldPacket) {
return VCM_OK;
} else if (ret == kFlushIndicator) {
return VCM_FLUSH_INDICATOR;
} else if (ret < 0) {
return VCM_JITTER_BUFFER_ERROR;
}
if (ret == kCompleteSession && !retransmitted) {
// We don't want to include timestamps which have suffered from
// retransmission here, since we compensate with extra retransmission
// delay within the jitter estimate.
timing_->IncomingTimestamp(packet.timestamp, clock_->CurrentTime());
}
return VCM_OK;
}
VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
bool prefer_late_decoding) {
const int64_t start_time_ms = clock_->TimeInMilliseconds();
int64_t render_time_ms = 0;
// Exhaust wait time to get a complete frame for decoding.
VCMEncodedFrame* found_frame =
jitter_buffer_.NextCompleteFrame(max_wait_time_ms);
if (found_frame == nullptr) {
return nullptr;
}
uint32_t frame_timestamp = found_frame->RtpTimestamp();
if (std::optional<VideoPlayoutDelay> playout_delay =
found_frame->EncodedImage().PlayoutDelay()) {
timing_->set_playout_delay(*playout_delay);
}
// We have a frame - Set timing and render timestamp.
timing_->SetJitterDelay(
TimeDelta::Millis(jitter_buffer_.EstimatedJitterMs()));
const Timestamp now = clock_->CurrentTime();
const int64_t now_ms = now.ms();
timing_->UpdateCurrentDelay(frame_timestamp);
render_time_ms = timing_->RenderTime(frame_timestamp, now).ms();
// Check render timing.
bool timing_error = false;
// Assume that render timing errors are due to changes in the video stream.
if (render_time_ms < 0) {
timing_error = true;
} else if (std::abs(render_time_ms - now_ms) > max_video_delay_ms_) {
int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
RTC_LOG(LS_WARNING)
<< "A frame about to be decoded is out of the configured "
"delay bounds ("
<< frame_delay << " > " << max_video_delay_ms_
<< "). Resetting the video jitter buffer.";
timing_error = true;
} else if (static_cast<int>(timing_->TargetVideoDelay().ms()) >
max_video_delay_ms_) {
RTC_LOG(LS_WARNING) << "The video target delay has grown larger than "
<< max_video_delay_ms_
<< " ms. Resetting jitter buffer.";
timing_error = true;
}
if (timing_error) {
// Timing error => reset timing and flush the jitter buffer.
jitter_buffer_.Flush();
timing_->Reset();
return NULL;
}
if (prefer_late_decoding) {
// Decode frame as close as possible to the render timestamp.
const int32_t available_wait_time =
max_wait_time_ms -
static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
uint16_t new_max_wait_time =
static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
uint32_t wait_time_ms = rtc::saturated_cast<uint32_t>(
timing_
->MaxWaitingTime(Timestamp::Millis(render_time_ms),
clock_->CurrentTime(),
/*too_many_frames_queued=*/false)
.ms());
if (new_max_wait_time < wait_time_ms) {
// We're not allowed to wait until the frame is supposed to be rendered,
// waiting as long as we're allowed to avoid busy looping, and then return
// NULL. Next call to this function might return the frame.
render_wait_event_->Wait(new_max_wait_time);
return NULL;
}
// Wait until it's time to render.
render_wait_event_->Wait(wait_time_ms);
}
// Extract the frame from the jitter buffer and set the render time.
VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
if (frame == NULL) {
return NULL;
}
frame->SetRenderTime(render_time_ms);
TRACE_EVENT_ASYNC_STEP_INTO1("webrtc", "Video", frame->RtpTimestamp(),
"SetRenderTS", "render_time",
frame->RenderTimeMs());
return frame;
}
void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
jitter_buffer_.ReleaseFrame(frame);
}
void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
int max_packet_age_to_nack,
int max_incomplete_time_ms) {
jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
max_incomplete_time_ms);
}
std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
return jitter_buffer_.GetNackList(request_key_frame);
}
} // namespace webrtc