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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 "modules/video_coding/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/encoded_frame.h"
#include "modules/video_coding/internal_defines.h"
#include "modules/video_coding/jitter_buffer_common.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();
uint32_t frame_timestamp = 0;
int min_playout_delay_ms = -1;
int max_playout_delay_ms = -1;
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) {
frame_timestamp = found_frame->Timestamp();
min_playout_delay_ms = found_frame->EncodedImage().playout_delay_.min_ms;
max_playout_delay_ms = found_frame->EncodedImage().playout_delay_.max_ms;
} else {
return nullptr;
}
if (min_playout_delay_ms >= 0)
timing_->set_min_playout_delay(TimeDelta::Millis(min_playout_delay_ms));
if (max_playout_delay_ms >= 0)
timing_->set_max_playout_delay(TimeDelta::Millis(max_playout_delay_ms));
// 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_STEP1("webrtc", "Video", frame->Timestamp(), "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
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