isheriff 6b4b5f3770 Add sender controlled playout delay limits
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.

The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.

There are no tests at this time and most of testing is done with chromium
webrtc prototype.

On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.

BUG=webrtc:5895

Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
2016-06-08 07:24:30 +00:00

306 lines
11 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 "webrtc/modules/video_coding/receiver.h"
#include <assert.h>
#include <cstdlib>
#include <utility>
#include <vector>
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/modules/video_coding/encoded_frame.h"
#include "webrtc/modules/video_coding/internal_defines.h"
#include "webrtc/modules/video_coding/media_opt_util.h"
#include "webrtc/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)
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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();
delete crit_sect_;
}
void VCMReceiver::Reset() {
CriticalSectionScoped cs(crit_sect_);
if (!jitter_buffer_.Running()) {
jitter_buffer_.Start();
} else {
jitter_buffer_.Flush();
}
}
void VCMReceiver::UpdateRtt(int64_t rtt) {
jitter_buffer_.UpdateRtt(rtt);
}
int64_t VCMReceiver::TimeUntilNextProcess() {
return jitter_buffer_.TimeUntilNextProcess();
}
void VCMReceiver::Process() {
jitter_buffer_.Process();
}
int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
uint16_t frame_width,
uint16_t frame_height) {
// 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,
int64_t* next_render_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;
// 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);
*next_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 (*next_render_time_ms < 0) {
timing_error = true;
} else if (std::abs(*next_render_time_ms - now_ms) > max_video_delay_ms_) {
int frame_delay = static_cast<int>(std::abs(*next_render_time_ms - now_ms));
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_) {
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 = timing_->MaxWaitingTime(
*next_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(*next_render_time_ms);
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), "SetRenderTS",
"render_time", *next_render_time_ms);
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);
}
uint32_t VCMReceiver::DiscardedPackets() const {
return jitter_buffer_.num_discarded_packets();
}
void VCMReceiver::SetNackMode(VCMNackMode nackMode,
int64_t low_rtt_nack_threshold_ms,
int64_t high_rtt_nack_threshold_ms) {
CriticalSectionScoped 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 {
CriticalSectionScoped 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();
}
int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
CriticalSectionScoped cs(crit_sect_);
if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
return -1;
}
max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
// Initializing timing to the desired delay.
timing_->set_min_playout_delay(desired_delay_ms);
return 0;
}
void VCMReceiver::RegisterStatsCallback(
VCMReceiveStatisticsCallback* callback) {
jitter_buffer_.RegisterStatsCallback(callback);
}
} // namespace webrtc