webrtc/modules/video_coding/receiver.cc
Niels Möller 2377588c82 Add accessor methods for RTP timestamp of EncodedImage.
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.

Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
2018-08-21 09:15:51 +00:00

280 lines
10 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/receiver.h"
#include <assert.h>
#include <cstdlib>
#include <utility>
#include <vector>
#include "modules/video_coding/encoded_frame.h"
#include "modules/video_coding/internal_defines.h"
#include "modules/video_coding/media_opt_util.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,
EventFactory* event_factory)
: VCMReceiver::VCMReceiver(timing,
clock,
event_factory,
nullptr, // NackSender
nullptr) // KeyframeRequestSender
{}
VCMReceiver::VCMReceiver(VCMTiming* timing,
Clock* clock,
EventFactory* event_factory,
NackSender* nack_sender,
KeyFrameRequestSender* keyframe_request_sender)
: VCMReceiver(
timing,
clock,
std::unique_ptr<EventWrapper>(event_factory
? event_factory->CreateEvent()
: EventWrapper::Create()),
std::unique_ptr<EventWrapper>(event_factory
? event_factory->CreateEvent()
: EventWrapper::Create()),
nack_sender,
keyframe_request_sender) {}
VCMReceiver::VCMReceiver(VCMTiming* timing,
Clock* clock,
std::unique_ptr<EventWrapper> receiver_event,
std::unique_ptr<EventWrapper> jitter_buffer_event)
: VCMReceiver::VCMReceiver(timing,
clock,
std::move(receiver_event),
std::move(jitter_buffer_event),
nullptr, // NackSender
nullptr) // KeyframeRequestSender
{}
VCMReceiver::VCMReceiver(VCMTiming* timing,
Clock* clock,
std::unique_ptr<EventWrapper> receiver_event,
std::unique_ptr<EventWrapper> jitter_buffer_event,
NackSender* nack_sender,
KeyFrameRequestSender* keyframe_request_sender)
: clock_(clock),
jitter_buffer_(clock_,
std::move(jitter_buffer_event),
nack_sender,
keyframe_request_sender),
timing_(timing),
render_wait_event_(std::move(receiver_event)),
max_video_delay_ms_(kMaxVideoDelayMs) {
Reset();
}
VCMReceiver::~VCMReceiver() {
render_wait_event_->Set();
}
void VCMReceiver::Reset() {
rtc::CritScope cs(&crit_sect_);
if (!jitter_buffer_.Running()) {
jitter_buffer_.Start();
} else {
jitter_buffer_.Flush();
}
}
void VCMReceiver::UpdateRtt(int64_t rtt) {
jitter_buffer_.UpdateRtt(rtt);
}
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_->TimeInMilliseconds());
}
return VCM_OK;
}
void VCMReceiver::TriggerDecoderShutdown() {
jitter_buffer_.Stop();
render_wait_event_->Set();
}
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 {
if (!jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp))
return nullptr;
}
if (min_playout_delay_ms >= 0)
timing_->set_min_playout_delay(min_playout_delay_ms);
if (max_playout_delay_ms >= 0)
timing_->set_max_playout_delay(max_playout_delay_ms);
// We have a frame - Set timing and render timestamp.
timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
const int64_t now_ms = clock_->TimeInMilliseconds();
timing_->UpdateCurrentDelay(frame_timestamp);
render_time_ms = timing_->RenderTimeMs(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()) >
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(render_time_ms, clock_->TimeInMilliseconds()));
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());
if (!frame->Complete()) {
// Update stats for incomplete frames.
bool retransmitted = false;
const int64_t last_packet_time_ms =
jitter_buffer_.LastPacketTime(frame, &retransmitted);
if (last_packet_time_ms >= 0 && !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(frame_timestamp, last_packet_time_ms);
}
}
return frame;
}
void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
jitter_buffer_.ReleaseFrame(frame);
}
void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate) {
assert(bitrate);
assert(framerate);
jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
}
void VCMReceiver::SetNackMode(VCMNackMode nackMode,
int64_t low_rtt_nack_threshold_ms,
int64_t high_rtt_nack_threshold_ms) {
rtc::CritScope cs(&crit_sect_);
// Default to always having NACK enabled in hybrid mode.
jitter_buffer_.SetNackMode(nackMode, low_rtt_nack_threshold_ms,
high_rtt_nack_threshold_ms);
}
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);
}
VCMNackMode VCMReceiver::NackMode() const {
rtc::CritScope cs(&crit_sect_);
return jitter_buffer_.nack_mode();
}
std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
return jitter_buffer_.GetNackList(request_key_frame);
}
void VCMReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
jitter_buffer_.SetDecodeErrorMode(decode_error_mode);
}
VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
return jitter_buffer_.decode_error_mode();
}
void VCMReceiver::RegisterStatsCallback(
VCMReceiveStatisticsCallback* callback) {
jitter_buffer_.RegisterStatsCallback(callback);
}
} // namespace webrtc