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webrtc_m130/video/video_send_stream_impl.cc
Jiawei Ou 55718120e6 Adding rtcp report interval into RTCConfiguration. 
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201.

Issue 43201 didn't do the job properly.
1. The audio rtcp report interval is not properly hooked up.
2. We don't need to propagate audio rtcp interval into video send stream or vice versa.
3. We don't need to propagate rtcp report interval to any receiving streams.

Bug: webrtc:8789
Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f
Reviewed-on: https://webrtc-review.googlesource.com/c/110105
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25610}
2018-11-12 20:00:00 +00:00

633 lines
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/*
* 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/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 {
// Max positive size difference to treat allocations as "similar".
static constexpr int kMaxVbaSizeDifferencePercent = 10;
// Max time we will throttle similar video bitrate allocations.
static constexpr int64_t kMaxVbaThrottleTimeMs = 500;
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);
}
// Calculate max padding bitrate for a multi layer codec.
int CalculateMaxPadBitrateBps(const std::vector<VideoStream>& streams,
int min_transmit_bitrate_bps,
bool pad_to_min_bitrate,
bool alr_probing) {
int pad_up_to_bitrate_bps = 0;
// Filter out only the active streams;
std::vector<VideoStream> active_streams;
for (const VideoStream& stream : streams) {
if (stream.active)
active_streams.emplace_back(stream);
}
if (active_streams.size() > 1) {
if (alr_probing) {
// With alr probing, just pad to the min bitrate of the lowest stream,
// probing will handle the rest of the rampup.
pad_up_to_bitrate_bps = active_streams[0].min_bitrate_bps;
} else {
// Pad to min bitrate of the highest layer.
pad_up_to_bitrate_bps =
active_streams[active_streams.size() - 1].min_bitrate_bps;
// Add target_bitrate_bps of the lower layers.
for (size_t i = 0; i < active_streams.size() - 1; ++i)
pad_up_to_bitrate_bps += active_streams[i].target_bitrate_bps;
}
} else if (!active_streams.empty() && pad_to_min_bitrate) {
pad_up_to_bitrate_bps = active_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;
}
RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig(
const VideoSendStream::Config* config) {
RtpSenderFrameEncryptionConfig frame_encryption_config;
frame_encryption_config.frame_encryptor = config->frame_encryptor;
frame_encryption_config.crypto_options = config->crypto_options;
return frame_encryption_config;
}
RtpSenderObservers CreateObservers(CallStats* call_stats,
EncoderRtcpFeedback* encoder_feedback,
SendStatisticsProxy* stats_proxy,
SendDelayStats* send_delay_stats) {
RtpSenderObservers observers;
observers.rtcp_rtt_stats = call_stats;
observers.intra_frame_callback = encoder_feedback;
observers.rtcp_stats = stats_proxy;
observers.rtp_stats = stats_proxy;
observers.bitrate_observer = stats_proxy;
observers.frame_count_observer = stats_proxy;
observers.rtcp_type_observer = stats_proxy;
observers.send_delay_observer = stats_proxy;
observers.send_packet_observer = send_delay_stats;
return observers;
}
absl::optional<AlrExperimentSettings> GetAlrSettings(
VideoEncoderConfig::ContentType content_type) {
if (content_type == VideoEncoderConfig::ContentType::kScreen) {
return AlrExperimentSettings::CreateFromFieldTrial(
AlrExperimentSettings::kScreenshareProbingBweExperimentName);
}
return AlrExperimentSettings::CreateFromFieldTrial(
AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
}
bool SameStreamsEnabled(const VideoBitrateAllocation& lhs,
const VideoBitrateAllocation& rhs) {
for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
if (lhs.HasBitrate(si, ti) != rhs.HasBitrate(si, ti)) {
return false;
}
}
}
return true;
}
} // 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)
: has_alr_probing_(config->periodic_alr_bandwidth_probing ||
GetAlrSettings(content_type)),
stats_proxy_(stats_proxy),
config_(config),
worker_queue_(worker_queue),
check_encoder_activity_task_(nullptr),
call_stats_(call_stats),
transport_(transport),
bitrate_allocator_(bitrate_allocator),
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_video_sender_(transport_->CreateRtpVideoSender(
config_->rtp.ssrcs,
suspended_ssrcs,
suspended_payload_states,
config_->rtp,
config_->rtcp_report_interval_ms,
config_->send_transport,
CreateObservers(call_stats,
&encoder_feedback_,
stats_proxy_,
send_delay_stats),
event_log,
std::move(fec_controller),
CreateFrameEncryptionConfig(config_))),
weak_ptr_factory_(this) {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
weak_ptr_ = weak_ptr_factory_.GetWeakPtr();
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 =
GetAlrSettings(content_type);
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);
}
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);
}
VideoSendStreamImpl::~VideoSendStreamImpl() {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK(!rtp_video_sender_->IsActive())
<< "VideoSendStreamImpl::Stop not called";
RTC_LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString();
transport_->DestroyRtpVideoSender(rtp_video_sender_);
}
void VideoSendStreamImpl::RegisterProcessThread(
ProcessThread* module_process_thread) {
rtp_video_sender_->RegisterProcessThread(module_process_thread);
}
void VideoSendStreamImpl::DeRegisterProcessThread() {
rtp_video_sender_->DeRegisterProcessThread();
}
bool VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
// Runs on a network thread.
RTC_DCHECK(!worker_queue_->IsCurrent());
rtp_video_sender_->DeliverRtcp(packet, length);
return true;
}
void VideoSendStreamImpl::UpdateActiveSimulcastLayers(
const std::vector<bool> active_layers) {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_LOG(LS_INFO) << "VideoSendStream::UpdateActiveSimulcastLayers";
bool previously_active = rtp_video_sender_->IsActive();
rtp_video_sender_->SetActiveModules(active_layers);
if (!rtp_video_sender_->IsActive() && previously_active) {
// Payload router switched from active to inactive.
StopVideoSendStream();
} else if (rtp_video_sender_->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 (rtp_video_sender_->IsActive())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
rtp_video_sender_->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 (!rtp_video_sender_->IsActive())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
rtp_video_sender_->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) {
if (!worker_queue_->IsCurrent()) {
auto ptr = weak_ptr_;
worker_queue_->PostTask([=] {
if (!ptr.get())
return;
ptr->OnBitrateAllocationUpdated(allocation);
});
return;
}
RTC_DCHECK_RUN_ON(worker_queue_);
int64_t now_ms = rtc::TimeMillis();
if (encoder_target_rate_bps_ != 0) {
if (video_bitrate_allocation_context_) {
// If new allocation is within kMaxVbaSizeDifferencePercent larger than
// the previously sent allocation and the same streams are still enabled,
// it is considered "similar". We do not want send similar allocations
// more once per kMaxVbaThrottleTimeMs.
const VideoBitrateAllocation& last =
video_bitrate_allocation_context_->last_sent_allocation;
const bool is_similar =
allocation.get_sum_bps() >= last.get_sum_bps() &&
allocation.get_sum_bps() <
(last.get_sum_bps() * (100 + kMaxVbaSizeDifferencePercent)) /
100 &&
SameStreamsEnabled(allocation, last);
if (is_similar &&
(now_ms - video_bitrate_allocation_context_->last_send_time_ms) <
kMaxVbaThrottleTimeMs) {
// This allocation is too similar, cache it and return.
video_bitrate_allocation_context_->throttled_allocation = allocation;
return;
}
} else {
video_bitrate_allocation_context_.emplace();
}
video_bitrate_allocation_context_->last_sent_allocation = allocation;
video_bitrate_allocation_context_->throttled_allocation.reset();
video_bitrate_allocation_context_->last_send_time_ms = now_ms;
// Send bitrate allocation metadata only if encoder is not paused.
rtp_video_sender_->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,
has_alr_probing_);
}
// 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);
rtp_video_sender_->SetEncodingData(streams[0].width, streams[0].height,
num_temporal_layers);
if (rtp_video_sender_->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.
{
rtc::CritScope lock(&encoder_activity_crit_sect_);
if (check_encoder_activity_task_)
check_encoder_activity_task_->UpdateEncoderActivity();
}
EncodedImageCallback::Result result = rtp_video_sender_->OnEncodedImage(
encoded_image, codec_specific_info, fragmentation);
// Check if there's a throttled VideoBitrateAllocation that we should try
// sending.
rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_;
auto update_task = [send_stream]() {
if (send_stream) {
RTC_DCHECK_RUN_ON(send_stream->worker_queue_);
auto& context = send_stream->video_bitrate_allocation_context_;
if (context && context->throttled_allocation) {
send_stream->OnBitrateAllocationUpdated(*context->throttled_allocation);
}
}
};
if (!worker_queue_->IsCurrent()) {
worker_queue_->PostTask(update_task);
} else {
update_task();
}
return result;
}
std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const {
return rtp_video_sender_->GetRtpStates();
}
std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates()
const {
return rtp_video_sender_->GetRtpPayloadStates();
}
uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK(rtp_video_sender_->IsActive())
<< "VideoSendStream::Start has not been called.";
rtp_video_sender_->OnBitrateUpdated(update.bitrate_bps, update.fraction_loss,
update.rtt,
stats_proxy_->GetSendFrameRate());
encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps();
encoder_target_rate_bps_ =
std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
video_stream_encoder_->OnBitrateUpdated(encoder_target_rate_bps_,
update.fraction_loss, update.rtt);
stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
return rtp_video_sender_->GetProtectionBitrateBps();
}
} // namespace internal
} // namespace webrtc
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