zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
58c79f66dd3afef0ec2d77053e44f26b998b84b1
platform-external-webrtc/video/video_send_stream_impl.cc
Karl Wiberg 918f50c5d1 Use absl::make_unique and absl::WrapUnique directly 
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00

1047 lines
40 KiB
C++
Raw Blame History

/*
* Copyright 2018 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/video_send_stream_impl.h"
#include <algorithm>
#include <string>
#include <utility>
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/alr_experiment.h"
#include "rtc_base/file.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace internal {
namespace {
static const int kMinSendSidePacketHistorySize = 600;
// Assume an average video stream has around 3 packets per frame (1 mbps / 30
// fps / 1400B) A sequence number set with size 5500 will be able to store
// packet sequence number for at least last 60 seconds.
static const int kSendSideSeqNumSetMaxSize = 5500;
// We don't do MTU discovery, so assume that we have the standard ethernet MTU.
const size_t kPathMTU = 1500;
std::vector<RtpRtcp*> CreateRtpRtcpModules(
const VideoSendStream::Config& config,
RtcpIntraFrameObserver* intra_frame_callback,
RtcpBandwidthObserver* bandwidth_callback,
RtpTransportControllerSendInterface* transport,
RtcpRttStats* rtt_stats,
FlexfecSender* flexfec_sender,
SendStatisticsProxy* stats_proxy,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer,
RtpKeepAliveConfig keepalive_config) {
RTC_DCHECK_GT(config.rtp.ssrcs.size(), 0);
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.receiver_only = false;
configuration.outgoing_transport = config.send_transport;
configuration.intra_frame_callback = intra_frame_callback;
configuration.bandwidth_callback = bandwidth_callback;
configuration.transport_feedback_callback =
transport->transport_feedback_observer();
configuration.rtt_stats = rtt_stats;
configuration.rtcp_packet_type_counter_observer = stats_proxy;
configuration.paced_sender = transport->packet_sender();
configuration.transport_sequence_number_allocator =
transport->packet_router();
configuration.send_bitrate_observer = stats_proxy;
configuration.send_frame_count_observer = stats_proxy;
configuration.send_side_delay_observer = stats_proxy;
configuration.send_packet_observer = send_delay_stats;
configuration.event_log = event_log;
configuration.retransmission_rate_limiter = retransmission_rate_limiter;
configuration.overhead_observer = overhead_observer;
configuration.keepalive_config = keepalive_config;
configuration.rtcp_interval_config.video_interval_ms =
config.rtcp.video_report_interval_ms;
configuration.rtcp_interval_config.audio_interval_ms =
config.rtcp.audio_report_interval_ms;
std::vector<RtpRtcp*> modules;
const std::vector<uint32_t>& flexfec_protected_ssrcs =
config.rtp.flexfec.protected_media_ssrcs;
for (uint32_t ssrc : config.rtp.ssrcs) {
bool enable_flexfec = flexfec_sender != nullptr &&
std::find(flexfec_protected_ssrcs.begin(),
flexfec_protected_ssrcs.end(),
ssrc) != flexfec_protected_ssrcs.end();
configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
modules.push_back(rtp_rtcp);
}
return modules;
}
// TODO(brandtr): Update this function when we support multistream protection.
std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender(
const VideoSendStream::Config& config,
const std::map<uint32_t, RtpState>& suspended_ssrcs) {
if (config.rtp.flexfec.payload_type < 0) {
return nullptr;
}
RTC_DCHECK_GE(config.rtp.flexfec.payload_type, 0);
RTC_DCHECK_LE(config.rtp.flexfec.payload_type, 127);
if (config.rtp.flexfec.ssrc == 0) {
RTC_LOG(LS_WARNING) << "FlexFEC is enabled, but no FlexFEC SSRC given. "
"Therefore disabling FlexFEC.";
return nullptr;
}
if (config.rtp.flexfec.protected_media_ssrcs.empty()) {
RTC_LOG(LS_WARNING)
<< "FlexFEC is enabled, but no protected media SSRC given. "
"Therefore disabling FlexFEC.";
return nullptr;
}
if (config.rtp.flexfec.protected_media_ssrcs.size() > 1) {
RTC_LOG(LS_WARNING)
<< "The supplied FlexfecConfig contained multiple protected "
"media streams, but our implementation currently only "
"supports protecting a single media stream. "
"To avoid confusion, disabling FlexFEC completely.";
return nullptr;
}
const RtpState* rtp_state = nullptr;
auto it = suspended_ssrcs.find(config.rtp.flexfec.ssrc);
if (it != suspended_ssrcs.end()) {
rtp_state = &it->second;
}
RTC_DCHECK_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size());
return absl::make_unique<FlexfecSender>(
config.rtp.flexfec.payload_type, config.rtp.flexfec.ssrc,
config.rtp.flexfec.protected_media_ssrcs[0], config.rtp.mid,
config.rtp.extensions, RTPSender::FecExtensionSizes(), rtp_state,
Clock::GetRealTimeClock());
}
bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) {
const std::vector<RtpExtension>& extensions = config.rtp.extensions;
return std::find_if(
extensions.begin(), extensions.end(), [](const RtpExtension& ext) {
return ext.uri == RtpExtension::kTransportSequenceNumberUri;
}) != extensions.end();
}
const char kForcedFallbackFieldTrial[] =
"WebRTC-VP8-Forced-Fallback-Encoder-v2";
absl::optional<int> GetFallbackMinBpsFromFieldTrial() {
if (!webrtc::field_trial::IsEnabled(kForcedFallbackFieldTrial))
return absl::nullopt;
std::string group =
webrtc::field_trial::FindFullName(kForcedFallbackFieldTrial);
if (group.empty())
return absl::nullopt;
int min_pixels;
int max_pixels;
int min_bps;
if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &min_pixels, &max_pixels,
&min_bps) != 3) {
return absl::nullopt;
}
if (min_bps <= 0)
return absl::nullopt;
return min_bps;
}
int GetEncoderMinBitrateBps() {
const int kDefaultEncoderMinBitrateBps = 30000;
return GetFallbackMinBpsFromFieldTrial().value_or(
kDefaultEncoderMinBitrateBps);
}
bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
const VideoCodecType codecType = PayloadStringToCodecType(payload_name);
if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
return true;
}
return false;
}
int CalculateMaxPadBitrateBps(std::vector<VideoStream> streams,
int min_transmit_bitrate_bps,
bool pad_to_min_bitrate) {
int pad_up_to_bitrate_bps = 0;
// Calculate max padding bitrate for a multi layer codec.
if (streams.size() > 1) {
// Pad to min bitrate of the highest layer.
pad_up_to_bitrate_bps = streams[streams.size() - 1].min_bitrate_bps;
// Add target_bitrate_bps of the lower layers.
for (size_t i = 0; i < streams.size() - 1; ++i)
pad_up_to_bitrate_bps += streams[i].target_bitrate_bps;
} else if (pad_to_min_bitrate) {
pad_up_to_bitrate_bps = streams[0].min_bitrate_bps;
}
pad_up_to_bitrate_bps =
std::max(pad_up_to_bitrate_bps, min_transmit_bitrate_bps);
return pad_up_to_bitrate_bps;
}
uint32_t CalculateOverheadRateBps(int packets_per_second,
size_t overhead_bytes_per_packet,
uint32_t max_overhead_bps) {
uint32_t overhead_bps =
static_cast<uint32_t>(8 * overhead_bytes_per_packet * packets_per_second);
return std::min(overhead_bps, max_overhead_bps);
}
int CalculatePacketRate(uint32_t bitrate_bps, size_t packet_size_bytes) {
size_t packet_size_bits = 8 * packet_size_bytes;
// Ceil for int value of bitrate_bps / packet_size_bits.
return static_cast<int>((bitrate_bps + packet_size_bits - 1) /
packet_size_bits);
}
} // namespace
// CheckEncoderActivityTask is used for tracking when the encoder last produced
// and encoded video frame. If the encoder has not produced anything the last
// kEncoderTimeOutMs we also want to stop sending padding.
class VideoSendStreamImpl::CheckEncoderActivityTask : public rtc::QueuedTask {
public:
static const int kEncoderTimeOutMs = 2000;
explicit CheckEncoderActivityTask(
const rtc::WeakPtr<VideoSendStreamImpl>& send_stream)
: activity_(0), send_stream_(std::move(send_stream)), timed_out_(false) {}
void Stop() {
RTC_CHECK(task_checker_.CalledSequentially());
send_stream_.reset();
}
void UpdateEncoderActivity() {
// UpdateEncoderActivity is called from VideoSendStreamImpl::Encoded on
// whatever thread the real encoder implementation run on. In the case of
// hardware encoders, there might be several encoders
// running in parallel on different threads.
rtc::AtomicOps::ReleaseStore(&activity_, 1);
}
private:
bool Run() override {
RTC_CHECK(task_checker_.CalledSequentially());
if (!send_stream_)
return true;
if (!rtc::AtomicOps::AcquireLoad(&activity_)) {
if (!timed_out_) {
send_stream_->SignalEncoderTimedOut();
}
timed_out_ = true;
} else if (timed_out_) {
send_stream_->SignalEncoderActive();
timed_out_ = false;
}
rtc::AtomicOps::ReleaseStore(&activity_, 0);
rtc::TaskQueue::Current()->PostDelayedTask(
std::unique_ptr<rtc::QueuedTask>(this), kEncoderTimeOutMs);
// Return false to prevent this task from being deleted. Ownership has been
// transferred to the task queue when PostDelayedTask was called.
return false;
}
volatile int activity_;
rtc::SequencedTaskChecker task_checker_;
rtc::WeakPtr<VideoSendStreamImpl> send_stream_;
bool timed_out_;
};
VideoSendStreamImpl::VideoSendStreamImpl(
SendStatisticsProxy* stats_proxy,
rtc::TaskQueue* worker_queue,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocatorInterface* bitrate_allocator,
SendDelayStats* send_delay_stats,
VideoStreamEncoderInterface* video_stream_encoder,
RtcEventLog* event_log,
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
double initial_encoder_bitrate_priority,
std::map<uint32_t, RtpState> suspended_ssrcs,
std::map<uint32_t, RtpPayloadState> suspended_payload_states,
VideoEncoderConfig::ContentType content_type,
std::unique_ptr<FecController> fec_controller,
RateLimiter* retransmission_limiter)
: send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
stats_proxy_(stats_proxy),
config_(config),
suspended_ssrcs_(std::move(suspended_ssrcs)),
fec_controller_(std::move(fec_controller)),
module_process_thread_(nullptr),
worker_queue_(worker_queue),
check_encoder_activity_task_(nullptr),
call_stats_(call_stats),
transport_(transport),
bitrate_allocator_(bitrate_allocator),
flexfec_sender_(MaybeCreateFlexfecSender(*config_, suspended_ssrcs_)),
max_padding_bitrate_(0),
encoder_min_bitrate_bps_(0),
encoder_target_rate_bps_(0),
encoder_bitrate_priority_(initial_encoder_bitrate_priority),
has_packet_feedback_(false),
video_stream_encoder_(video_stream_encoder),
encoder_feedback_(Clock::GetRealTimeClock(),
config_->rtp.ssrcs,
video_stream_encoder),
bandwidth_observer_(transport->GetBandwidthObserver()),
rtp_rtcp_modules_(CreateRtpRtcpModules(*config_,
&encoder_feedback_,
bandwidth_observer_,
transport,
call_stats,
flexfec_sender_.get(),
stats_proxy_,
send_delay_stats,
event_log,
retransmission_limiter,
this,
transport->keepalive_config())),
payload_router_(rtp_rtcp_modules_,
config_->rtp.ssrcs,
config_->rtp.payload_type,
suspended_payload_states),
weak_ptr_factory_(this),
overhead_bytes_per_packet_(0),
transport_overhead_bytes_per_packet_(0) {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
weak_ptr_ = weak_ptr_factory_.GetWeakPtr();
module_process_thread_checker_.DetachFromThread();
RTC_DCHECK(!config_->rtp.ssrcs.empty());
RTC_DCHECK(call_stats_);
RTC_DCHECK(transport_);
RTC_DCHECK_NE(initial_encoder_max_bitrate, 0);
if (initial_encoder_max_bitrate > 0) {
encoder_max_bitrate_bps_ =
rtc::dchecked_cast<uint32_t>(initial_encoder_max_bitrate);
} else {
// TODO(srte): Make sure max bitrate is not set to negative values. We don't
// have any way to handle unset values in downstream code, such as the
// bitrate allocator. Previously -1 was implicitly casted to UINT32_MAX, a
// behaviour that is not safe. Converting to 10 Mbps should be safe for
// reasonable use cases as it allows adding the max of multiple streams
// without wrappping around.
const int kFallbackMaxBitrateBps = 10000000;
RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = "
<< initial_encoder_max_bitrate << " which is <= 0!";
RTC_DLOG(LS_INFO) << "Using default encoder max bitrate = 10 Mbps";
encoder_max_bitrate_bps_ = kFallbackMaxBitrateBps;
}
RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
// If send-side BWE is enabled, check if we should apply updated probing and
// pacing settings.
if (TransportSeqNumExtensionConfigured(*config_)) {
has_packet_feedback_ = true;
absl::optional<AlrExperimentSettings> alr_settings;
if (content_type == VideoEncoderConfig::ContentType::kScreen) {
alr_settings = AlrExperimentSettings::CreateFromFieldTrial(
AlrExperimentSettings::kScreenshareProbingBweExperimentName);
} else {
alr_settings = AlrExperimentSettings::CreateFromFieldTrial(
AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
}
if (alr_settings) {
transport->EnablePeriodicAlrProbing(true);
transport->SetPacingFactor(alr_settings->pacing_factor);
configured_pacing_factor_ = alr_settings->pacing_factor;
transport->SetQueueTimeLimit(alr_settings->max_paced_queue_time);
} else {
transport->EnablePeriodicAlrProbing(false);
transport->SetPacingFactor(PacedSender::kDefaultPaceMultiplier);
configured_pacing_factor_ = PacedSender::kDefaultPaceMultiplier;
transport->SetQueueTimeLimit(PacedSender::kMaxQueueLengthMs);
}
}
if (config_->periodic_alr_bandwidth_probing) {
transport->EnablePeriodicAlrProbing(true);
}
// RTP/RTCP initialization.
// We add the highest spatial layer first to ensure it'll be prioritized
// when sending padding, with the hope that the packet rate will be smaller,
// and that it's more important to protect than the lower layers.
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
constexpr bool remb_candidate = true;
transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate);
}
for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) {
const std::string& extension = config_->rtp.extensions[i].uri;
int id = config_->rtp.extensions[i].id;
// One-byte-extension local identifiers are in the range 1-14 inclusive.
RTC_DCHECK_GE(id, 1);
RTC_DCHECK_LE(id, 14);
RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
StringToRtpExtensionType(extension), id));
}
}
ConfigureProtection();
ConfigureSsrcs();
if (!config_->rtp.mid.empty()) {
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetMid(config_->rtp.mid);
}
}
// TODO(pbos): Should we set CNAME on all RTP modules?
rtp_rtcp_modules_.front()->SetCNAME(config_->rtp.c_name.c_str());
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->RegisterRtcpStatisticsCallback(stats_proxy_);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(stats_proxy_);
rtp_rtcp->SetMaxRtpPacketSize(config_->rtp.max_packet_size);
rtp_rtcp->RegisterVideoSendPayload(config_->rtp.payload_type,
config_->rtp.payload_name.c_str());
}
fec_controller_->SetProtectionCallback(this);
// Signal congestion controller this object is ready for OnPacket* callbacks.
if (fec_controller_->UseLossVectorMask()) {
transport_->RegisterPacketFeedbackObserver(this);
}
RTC_DCHECK_GE(config_->rtp.payload_type, 0);
RTC_DCHECK_LE(config_->rtp.payload_type, 127);
video_stream_encoder_->SetStartBitrate(
bitrate_allocator_->GetStartBitrate(this));
// Only request rotation at the source when we positively know that the remote
// side doesn't support the rotation extension. This allows us to prepare the
// encoder in the expectation that rotation is supported - which is the common
// case.
bool rotation_applied =
std::find_if(config_->rtp.extensions.begin(),
config_->rtp.extensions.end(),
[](const RtpExtension& extension) {
return extension.uri == RtpExtension::kVideoRotationUri;
}) == config_->rtp.extensions.end();
video_stream_encoder_->SetSink(this, rotation_applied);
}
void VideoSendStreamImpl::RegisterProcessThread(
ProcessThread* module_process_thread) {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
RTC_DCHECK(!module_process_thread_);
module_process_thread_ = module_process_thread;
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
module_process_thread_->RegisterModule(rtp_rtcp, RTC_FROM_HERE);
}
void VideoSendStreamImpl::DeRegisterProcessThread() {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
module_process_thread_->DeRegisterModule(rtp_rtcp);
}
VideoSendStreamImpl::~VideoSendStreamImpl() {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK(!payload_router_.IsActive())
<< "VideoSendStreamImpl::Stop not called";
RTC_LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString();
if (fec_controller_->UseLossVectorMask()) {
transport_->DeRegisterPacketFeedbackObserver(this);
}
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
transport_->packet_router()->RemoveSendRtpModule(rtp_rtcp);
delete rtp_rtcp;
}
}
bool VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
// Runs on a network thread.
RTC_DCHECK(!worker_queue_->IsCurrent());
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
rtp_rtcp->IncomingRtcpPacket(packet, length);
return true;
}
void VideoSendStreamImpl::UpdateActiveSimulcastLayers(
const std::vector<bool> active_layers) {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK_EQ(rtp_rtcp_modules_.size(), active_layers.size());
RTC_LOG(LS_INFO) << "VideoSendStream::UpdateActiveSimulcastLayers";
bool previously_active = payload_router_.IsActive();
payload_router_.SetActiveModules(active_layers);
if (!payload_router_.IsActive() && previously_active) {
// Payload router switched from active to inactive.
StopVideoSendStream();
} else if (payload_router_.IsActive() && !previously_active) {
// Payload router switched from inactive to active.
StartupVideoSendStream();
}
}
void VideoSendStreamImpl::Start() {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_LOG(LS_INFO) << "VideoSendStream::Start";
if (payload_router_.IsActive())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
payload_router_.SetActive(true);
StartupVideoSendStream();
}
void VideoSendStreamImpl::StartupVideoSendStream() {
RTC_DCHECK_RUN_ON(worker_queue_);
bitrate_allocator_->AddObserver(
this,
MediaStreamAllocationConfig{
static_cast<uint32_t>(encoder_min_bitrate_bps_),
encoder_max_bitrate_bps_, static_cast<uint32_t>(max_padding_bitrate_),
!config_->suspend_below_min_bitrate, config_->track_id,
encoder_bitrate_priority_, has_packet_feedback_});
// Start monitoring encoder activity.
{
rtc::CritScope lock(&encoder_activity_crit_sect_);
RTC_DCHECK(!check_encoder_activity_task_);
check_encoder_activity_task_ = new CheckEncoderActivityTask(weak_ptr_);
worker_queue_->PostDelayedTask(
std::unique_ptr<rtc::QueuedTask>(check_encoder_activity_task_),
CheckEncoderActivityTask::kEncoderTimeOutMs);
}
video_stream_encoder_->SendKeyFrame();
}
void VideoSendStreamImpl::Stop() {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_LOG(LS_INFO) << "VideoSendStream::Stop";
if (!payload_router_.IsActive())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
payload_router_.SetActive(false);
StopVideoSendStream();
}
void VideoSendStreamImpl::StopVideoSendStream() {
bitrate_allocator_->RemoveObserver(this);
{
rtc::CritScope lock(&encoder_activity_crit_sect_);
check_encoder_activity_task_->Stop();
check_encoder_activity_task_ = nullptr;
}
video_stream_encoder_->OnBitrateUpdated(0, 0, 0);
stats_proxy_->OnSetEncoderTargetRate(0);
}
void VideoSendStreamImpl::SignalEncoderTimedOut() {
RTC_DCHECK_RUN_ON(worker_queue_);
// If the encoder has not produced anything the last kEncoderTimeOutMs and it
// is supposed to, deregister as BitrateAllocatorObserver. This can happen
// if a camera stops producing frames.
if (encoder_target_rate_bps_ > 0) {
RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out.";
bitrate_allocator_->RemoveObserver(this);
}
}
void VideoSendStreamImpl::OnBitrateAllocationUpdated(
const VideoBitrateAllocation& allocation) {
payload_router_.OnBitrateAllocationUpdated(allocation);
}
void VideoSendStreamImpl::SignalEncoderActive() {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active.";
bitrate_allocator_->AddObserver(
this,
MediaStreamAllocationConfig{
static_cast<uint32_t>(encoder_min_bitrate_bps_),
encoder_max_bitrate_bps_, static_cast<uint32_t>(max_padding_bitrate_),
!config_->suspend_below_min_bitrate, config_->track_id,
encoder_bitrate_priority_, has_packet_feedback_});
}
void VideoSendStreamImpl::OnEncoderConfigurationChanged(
std::vector<VideoStream> streams,
int min_transmit_bitrate_bps) {
if (!worker_queue_->IsCurrent()) {
rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_;
worker_queue_->PostTask([send_stream, streams, min_transmit_bitrate_bps]() {
if (send_stream)
send_stream->OnEncoderConfigurationChanged(std::move(streams),
min_transmit_bitrate_bps);
});
return;
}
RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size());
TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged");
RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size());
RTC_DCHECK_RUN_ON(worker_queue_);
encoder_min_bitrate_bps_ =
std::max(streams[0].min_bitrate_bps, GetEncoderMinBitrateBps());
encoder_max_bitrate_bps_ = 0;
double stream_bitrate_priority_sum = 0;
for (const auto& stream : streams) {
// We don't want to allocate more bitrate than needed to inactive streams.
encoder_max_bitrate_bps_ += stream.active ? stream.max_bitrate_bps : 0;
if (stream.bitrate_priority) {
RTC_DCHECK_GT(*stream.bitrate_priority, 0);
stream_bitrate_priority_sum += *stream.bitrate_priority;
}
}
RTC_DCHECK_GT(stream_bitrate_priority_sum, 0);
encoder_bitrate_priority_ = stream_bitrate_priority_sum;
encoder_max_bitrate_bps_ =
std::max(static_cast<uint32_t>(encoder_min_bitrate_bps_),
encoder_max_bitrate_bps_);
const VideoCodecType codec_type =
PayloadStringToCodecType(config_->rtp.payload_name);
if (codec_type == kVideoCodecVP9) {
max_padding_bitrate_ = streams[0].target_bitrate_bps;
} else {
max_padding_bitrate_ = CalculateMaxPadBitrateBps(
streams, min_transmit_bitrate_bps, config_->suspend_below_min_bitrate);
}
// Clear stats for disabled layers.
for (size_t i = streams.size(); i < config_->rtp.ssrcs.size(); ++i) {
stats_proxy_->OnInactiveSsrc(config_->rtp.ssrcs[i]);
}
const size_t num_temporal_layers =
streams.back().num_temporal_layers.value_or(1);
fec_controller_->SetEncodingData(streams[0].width, streams[0].height,
num_temporal_layers,
config_->rtp.max_packet_size);
if (payload_router_.IsActive()) {
// The send stream is started already. Update the allocator with new bitrate
// limits.
bitrate_allocator_->AddObserver(
this, MediaStreamAllocationConfig{
static_cast<uint32_t>(encoder_min_bitrate_bps_),
encoder_max_bitrate_bps_,
static_cast<uint32_t>(max_padding_bitrate_),
!config_->suspend_below_min_bitrate, config_->track_id,
encoder_bitrate_priority_, has_packet_feedback_});
}
}
EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation) {
// Encoded is called on whatever thread the real encoder implementation run
// on. In the case of hardware encoders, there might be several encoders
// running in parallel on different threads.
size_t simulcast_idx = 0;
if (codec_specific_info->codecType == kVideoCodecVP8) {
simulcast_idx = codec_specific_info->codecSpecific.VP8.simulcastIdx;
}
if (config_->post_encode_callback) {
config_->post_encode_callback->EncodedFrameCallback(EncodedFrame(
encoded_image._buffer, encoded_image._length, encoded_image._frameType,
simulcast_idx, encoded_image._timeStamp));
}
{
rtc::CritScope lock(&encoder_activity_crit_sect_);
if (check_encoder_activity_task_)
check_encoder_activity_task_->UpdateEncoderActivity();
}
fec_controller_->UpdateWithEncodedData(encoded_image._length,
encoded_image._frameType);
EncodedImageCallback::Result result = payload_router_.OnEncodedImage(
encoded_image, codec_specific_info, fragmentation);
RTC_DCHECK(codec_specific_info);
int layer = codec_specific_info->codecType == kVideoCodecVP8
? codec_specific_info->codecSpecific.VP8.simulcastIdx
: 0;
{
rtc::CritScope lock(&ivf_writers_crit_);
if (file_writers_[layer].get()) {
bool ok = file_writers_[layer]->WriteFrame(
encoded_image, codec_specific_info->codecType);
RTC_DCHECK(ok);
}
}
return result;
}
void VideoSendStreamImpl::ConfigureProtection() {
RTC_DCHECK_RUN_ON(worker_queue_);
// Consistency of FlexFEC parameters is checked in MaybeCreateFlexfecSender.
const bool flexfec_enabled = (flexfec_sender_ != nullptr);
// Consistency of NACK and RED+ULPFEC parameters is checked in this function.
const bool nack_enabled = config_->rtp.nack.rtp_history_ms > 0;
int red_payload_type = config_->rtp.ulpfec.red_payload_type;
int ulpfec_payload_type = config_->rtp.ulpfec.ulpfec_payload_type;
// Shorthands.
auto IsRedEnabled = [&]() { return red_payload_type >= 0; };
auto DisableRed = [&]() { red_payload_type = -1; };
auto IsUlpfecEnabled = [&]() { return ulpfec_payload_type >= 0; };
auto DisableUlpfec = [&]() { ulpfec_payload_type = -1; };
if (webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment")) {
RTC_LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled.";
DisableUlpfec();
}
// If enabled, FlexFEC takes priority over RED+ULPFEC.
if (flexfec_enabled) {
// We can safely disable RED here, because if the remote supports FlexFEC,
// we know that it has a receiver without the RED/RTX workaround.
// See http://crbug.com/webrtc/6650 for more information.
if (IsRedEnabled()) {
RTC_LOG(LS_INFO) << "Both FlexFEC and RED are configured. Disabling RED.";
DisableRed();
}
if (IsUlpfecEnabled()) {
RTC_LOG(LS_INFO)
<< "Both FlexFEC and ULPFEC are configured. Disabling ULPFEC.";
DisableUlpfec();
}
}
// Payload types without picture ID cannot determine that a stream is complete
// without retransmitting FEC, so using ULPFEC + NACK for H.264 (for instance)
// is a waste of bandwidth since FEC packets still have to be transmitted.
// Note that this is not the case with FlexFEC.
if (nack_enabled && IsUlpfecEnabled() &&
!PayloadTypeSupportsSkippingFecPackets(config_->rtp.payload_name)) {
RTC_LOG(LS_WARNING)
<< "Transmitting payload type without picture ID using "
"NACK+ULPFEC is a waste of bandwidth since ULPFEC packets "
"also have to be retransmitted. Disabling ULPFEC.";
DisableUlpfec();
}
// Verify payload types.
//
// Due to how old receivers work, we need to always send RED if it has been
// negotiated. This is a remnant of an old RED/RTX workaround, see
// https://codereview.webrtc.org/2469093003.
// TODO(brandtr): This change went into M56, so we can remove it in ~M59.
// At that time, we can disable RED whenever ULPFEC is disabled, as there is
// no point in using RED without ULPFEC.
if (IsRedEnabled()) {
RTC_DCHECK_GE(red_payload_type, 0);
RTC_DCHECK_LE(red_payload_type, 127);
}
if (IsUlpfecEnabled()) {
RTC_DCHECK_GE(ulpfec_payload_type, 0);
RTC_DCHECK_LE(ulpfec_payload_type, 127);
if (!IsRedEnabled()) {
RTC_LOG(LS_WARNING)
<< "ULPFEC is enabled but RED is disabled. Disabling ULPFEC.";
DisableUlpfec();
}
}
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
// Set NACK.
rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
// Set RED/ULPFEC information.
rtp_rtcp->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
}
// Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic,
// so enable that logic if either of those FEC schemes are enabled.
fec_controller_->SetProtectionMethod(flexfec_enabled || IsUlpfecEnabled(),
nack_enabled);
}
void VideoSendStreamImpl::ConfigureSsrcs() {
RTC_DCHECK_RUN_ON(worker_queue_);
// Configure regular SSRCs.
for (size_t i = 0; i < config_->rtp.ssrcs.size(); ++i) {
uint32_t ssrc = config_->rtp.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
rtp_rtcp->SetSSRC(ssrc);
// Restore RTP state if previous existed.
VideoSendStream::RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtpState(it->second);
}
// Set up RTX if available.
if (config_->rtp.rtx.ssrcs.empty())
return;
// Configure RTX SSRCs.
RTC_DCHECK_EQ(config_->rtp.rtx.ssrcs.size(), config_->rtp.ssrcs.size());
for (size_t i = 0; i < config_->rtp.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = config_->rtp.rtx.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
rtp_rtcp->SetRtxSsrc(ssrc);
VideoSendStream::RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtxState(it->second);
}
// Configure RTX payload types.
RTC_DCHECK_GE(config_->rtp.rtx.payload_type, 0);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetRtxSendPayloadType(config_->rtp.rtx.payload_type,
config_->rtp.payload_type);
rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
}
if (config_->rtp.ulpfec.red_payload_type != -1 &&
config_->rtp.ulpfec.red_rtx_payload_type != -1) {
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetRtxSendPayloadType(config_->rtp.ulpfec.red_rtx_payload_type,
config_->rtp.ulpfec.red_payload_type);
}
}
}
std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const {
RTC_DCHECK_RUN_ON(worker_queue_);
std::map<uint32_t, RtpState> rtp_states;
for (size_t i = 0; i < config_->rtp.ssrcs.size(); ++i) {
uint32_t ssrc = config_->rtp.ssrcs[i];
RTC_DCHECK_EQ(ssrc, rtp_rtcp_modules_[i]->SSRC());
rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtpState();
}
for (size_t i = 0; i < config_->rtp.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = config_->rtp.rtx.ssrcs[i];
rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtxState();
}
if (flexfec_sender_) {
uint32_t ssrc = config_->rtp.flexfec.ssrc;
rtp_states[ssrc] = flexfec_sender_->GetRtpState();
}
return rtp_states;
}
std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates()
const {
RTC_DCHECK_RUN_ON(worker_queue_);
return payload_router_.GetRtpPayloadStates();
}
void VideoSendStreamImpl::SignalNetworkState(NetworkState state) {
RTC_DCHECK_RUN_ON(worker_queue_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_->rtp.rtcp_mode
: RtcpMode::kOff);
}
}
uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt,
int64_t probing_interval_ms) {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK(payload_router_.IsActive())
<< "VideoSendStream::Start has not been called.";
// Substract overhead from bitrate.
rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
uint32_t payload_bitrate_bps = bitrate_bps;
if (send_side_bwe_with_overhead_) {
payload_bitrate_bps -= CalculateOverheadRateBps(
CalculatePacketRate(bitrate_bps,
config_->rtp.max_packet_size +
transport_overhead_bytes_per_packet_),
overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_,
bitrate_bps);
}
// Get the encoder target rate. It is the estimated network rate -
// protection overhead.
encoder_target_rate_bps_ = fec_controller_->UpdateFecRates(
payload_bitrate_bps, stats_proxy_->GetSendFrameRate(), fraction_loss,
loss_mask_vector_, rtt);
loss_mask_vector_.clear();
uint32_t encoder_overhead_rate_bps =
send_side_bwe_with_overhead_
? CalculateOverheadRateBps(
CalculatePacketRate(encoder_target_rate_bps_,
config_->rtp.max_packet_size +
transport_overhead_bytes_per_packet_ -
overhead_bytes_per_packet_),
overhead_bytes_per_packet_ +
transport_overhead_bytes_per_packet_,
bitrate_bps - encoder_target_rate_bps_)
: 0;
// When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled
// protection_bitrate includes overhead.
uint32_t protection_bitrate =
bitrate_bps - (encoder_target_rate_bps_ + encoder_overhead_rate_bps);
encoder_target_rate_bps_ =
std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
video_stream_encoder_->OnBitrateUpdated(encoder_target_rate_bps_,
fraction_loss, rtt);
stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
return protection_bitrate;
}
void VideoSendStreamImpl::EnableEncodedFrameRecording(
const std::vector<rtc::PlatformFile>& files,
size_t byte_limit) {
{
rtc::CritScope lock(&ivf_writers_crit_);
for (unsigned int i = 0; i < kMaxSimulcastStreams; ++i) {
if (i < files.size()) {
file_writers_[i] = IvfFileWriter::Wrap(rtc::File(files[i]), byte_limit);
} else {
file_writers_[i].reset();
}
}
}
if (!files.empty()) {
// Make a keyframe appear as early as possible in the logs, to give actually
// decodable output.
video_stream_encoder_->SendKeyFrame();
}
}
int VideoSendStreamImpl::ProtectionRequest(
const FecProtectionParams* delta_params,
const FecProtectionParams* key_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps) {
RTC_DCHECK_RUN_ON(worker_queue_);
*sent_video_rate_bps = 0;
*sent_nack_rate_bps = 0;
*sent_fec_rate_bps = 0;
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
uint32_t not_used = 0;
uint32_t module_video_rate = 0;
uint32_t module_fec_rate = 0;
uint32_t module_nack_rate = 0;
rtp_rtcp->SetFecParameters(*delta_params, *key_params);
rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
&module_nack_rate);
*sent_video_rate_bps += module_video_rate;
*sent_nack_rate_bps += module_nack_rate;
*sent_fec_rate_bps += module_fec_rate;
}
return 0;
}
void VideoSendStreamImpl::OnOverheadChanged(size_t overhead_bytes_per_packet) {
rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
overhead_bytes_per_packet_ = overhead_bytes_per_packet;
}
void VideoSendStreamImpl::SetTransportOverhead(
size_t transport_overhead_bytes_per_packet) {
if (transport_overhead_bytes_per_packet >= static_cast<int>(kPathMTU)) {
RTC_LOG(LS_ERROR) << "Transport overhead exceeds size of ethernet frame";
return;
}
transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet;
size_t rtp_packet_size =
std::min(config_->rtp.max_packet_size,
kPathMTU - transport_overhead_bytes_per_packet_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
}
}
void VideoSendStreamImpl::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
if (!worker_queue_->IsCurrent()) {
auto ptr = weak_ptr_;
worker_queue_->PostTask([=] {
if (!ptr.get())
return;
ptr->OnPacketAdded(ssrc, seq_num);
});
return;
}
const auto ssrcs = config_->rtp.ssrcs;
if (std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end()) {
feedback_packet_seq_num_set_.insert(seq_num);
if (feedback_packet_seq_num_set_.size() > kSendSideSeqNumSetMaxSize) {
RTC_LOG(LS_WARNING) << "Feedback packet sequence number set exceed it's "
"max size', will get reset.";
feedback_packet_seq_num_set_.clear();
}
}
}
void VideoSendStreamImpl::OnPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) {
if (!worker_queue_->IsCurrent()) {
auto ptr = weak_ptr_;
worker_queue_->PostTask([=] {
if (!ptr.get())
return;
ptr->OnPacketFeedbackVector(packet_feedback_vector);
});
return;
}
// Lost feedbacks are not considered to be lost packets.
for (const PacketFeedback& packet : packet_feedback_vector) {
if (auto it = feedback_packet_seq_num_set_.find(packet.sequence_number) !=
feedback_packet_seq_num_set_.end()) {
const bool lost = packet.arrival_time_ms == PacketFeedback::kNotReceived;
loss_mask_vector_.push_back(lost);
feedback_packet_seq_num_set_.erase(it);
}
}
}
} // namespace internal
} // namespace webrtc
Reference in New Issue View Git Blame Copy Permalink