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d30f796d8c
Nagram/TMessagesProj/jni/voip/webrtc/video/video_send_stream_impl.cc
xaxtix d30f796d8c Update to 8.6.0
2022-03-11 19:49:54 +03:00

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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 <stdio.h>
#include <algorithm>
#include <cstdint>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/video_codecs/video_codec.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/video_send_stream.h"
#include "modules/pacing/paced_sender.h"
#include "rtc_base/atomic_ops.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/alr_experiment.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/min_video_bitrate_experiment.h"
#include "rtc_base/experiments/rate_control_settings.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.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;
constexpr TimeDelta kEncoderTimeOut = TimeDelta::Seconds(2);
// When send-side BWE is used a stricter 1.1x pacing factor is used, rather than
// the 2.5x which is used with receive-side BWE. Provides a more careful
// bandwidth rampup with less risk of overshoots causing adverse effects like
// packet loss. Not used for receive side BWE, since there we lack the probing
// feature and so may result in too slow initial rampup.
static constexpr double kStrictPacingMultiplier = 1.1;
bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) {
const std::vector<RtpExtension>& extensions = config.rtp.extensions;
return absl::c_any_of(extensions, [](const RtpExtension& ext) {
return ext.uri == RtpExtension::kTransportSequenceNumberUri;
});
}
// Calculate max padding bitrate for a multi layer codec.
int CalculateMaxPadBitrateBps(const std::vector<VideoStream>& streams,
bool is_svc,
VideoEncoderConfig::ContentType content_type,
int min_transmit_bitrate_bps,
bool pad_to_min_bitrate,
bool alr_probing) {
int pad_up_to_bitrate_bps = 0;
RTC_DCHECK(!is_svc || streams.size() <= 1) << "Only one stream is allowed in "
"SVC mode.";
// 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 || (!active_streams.empty() && is_svc)) {
// Simulcast or SVC is used.
// if SVC is used, stream bitrates should already encode svc bitrates:
// min_bitrate = min bitrate of a lowest svc layer.
// target_bitrate = sum of target bitrates of lower layers + min bitrate
// of the last one (as used in the calculations below).
// max_bitrate = sum of all active layers' max_bitrate.
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 {
// Without alr probing, pad up to start bitrate of the
// highest active stream.
const double hysteresis_factor =
RateControlSettings::ParseFromFieldTrials()
.GetSimulcastHysteresisFactor(content_type);
if (is_svc) {
// For SVC, since there is only one "stream", the padding bitrate
// needed to enable the top spatial layer is stored in the
// `target_bitrate_bps` field.
// TODO(sprang): This behavior needs to die.
pad_up_to_bitrate_bps = static_cast<int>(
hysteresis_factor * active_streams[0].target_bitrate_bps + 0.5);
} else {
const size_t top_active_stream_idx = active_streams.size() - 1;
pad_up_to_bitrate_bps = std::min(
static_cast<int>(
hysteresis_factor *
active_streams[top_active_stream_idx].min_bitrate_bps +
0.5),
active_streams[top_active_stream_idx].target_bitrate_bps);
// Add target_bitrate_bps of the lower active streams.
for (size_t i = 0; i < top_active_stream_idx; ++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;
}
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;
}
// Returns an optional that has value iff TransportSeqNumExtensionConfigured
// is `true` for the given video send stream config.
absl::optional<float> GetConfiguredPacingFactor(
const VideoSendStream::Config& config,
VideoEncoderConfig::ContentType content_type,
const PacingConfig& default_pacing_config) {
if (!TransportSeqNumExtensionConfigured(config))
return absl::nullopt;
absl::optional<AlrExperimentSettings> alr_settings =
GetAlrSettings(content_type);
if (alr_settings)
return alr_settings->pacing_factor;
RateControlSettings rate_control_settings =
RateControlSettings::ParseFromFieldTrials();
return rate_control_settings.GetPacingFactor().value_or(
default_pacing_config.pacing_factor);
}
uint32_t GetInitialEncoderMaxBitrate(int initial_encoder_max_bitrate) {
if (initial_encoder_max_bitrate > 0)
return rtc::dchecked_cast<uint32_t>(initial_encoder_max_bitrate);
// 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";
return kFallbackMaxBitrateBps;
}
} // namespace
PacingConfig::PacingConfig()
: pacing_factor("factor", kStrictPacingMultiplier),
max_pacing_delay("max_delay",
TimeDelta::Millis(PacedSender::kMaxQueueLengthMs)) {
ParseFieldTrial({&pacing_factor, &max_pacing_delay},
field_trial::FindFullName("WebRTC-Video-Pacing"));
}
PacingConfig::PacingConfig(const PacingConfig&) = default;
PacingConfig::~PacingConfig() = default;
VideoSendStreamImpl::VideoSendStreamImpl(
Clock* clock,
SendStatisticsProxy* stats_proxy,
rtc::TaskQueue* rtp_transport_queue,
RtpTransportControllerSendInterface* transport,
BitrateAllocatorInterface* bitrate_allocator,
VideoStreamEncoderInterface* video_stream_encoder,
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
double initial_encoder_bitrate_priority,
VideoEncoderConfig::ContentType content_type,
RtpVideoSenderInterface* rtp_video_sender)
: clock_(clock),
has_alr_probing_(config->periodic_alr_bandwidth_probing ||
GetAlrSettings(content_type)),
pacing_config_(PacingConfig()),
stats_proxy_(stats_proxy),
config_(config),
rtp_transport_queue_(rtp_transport_queue),
timed_out_(false),
transport_(transport),
bitrate_allocator_(bitrate_allocator),
disable_padding_(true),
max_padding_bitrate_(0),
encoder_min_bitrate_bps_(0),
encoder_max_bitrate_bps_(
GetInitialEncoderMaxBitrate(initial_encoder_max_bitrate)),
encoder_target_rate_bps_(0),
encoder_bitrate_priority_(initial_encoder_bitrate_priority),
video_stream_encoder_(video_stream_encoder),
bandwidth_observer_(transport->GetBandwidthObserver()),
rtp_video_sender_(rtp_video_sender),
configured_pacing_factor_(
GetConfiguredPacingFactor(*config_, content_type, pacing_config_)) {
RTC_DCHECK_GE(config_->rtp.payload_type, 0);
RTC_DCHECK_LE(config_->rtp.payload_type, 127);
RTC_DCHECK(!config_->rtp.ssrcs.empty());
RTC_DCHECK(transport_);
RTC_DCHECK_NE(initial_encoder_max_bitrate, 0);
RTC_LOG(LS_INFO) << "VideoSendStreamImpl: " << config_->ToString();
RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
// 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 = absl::c_none_of(
config_->rtp.extensions, [](const RtpExtension& extension) {
return extension.uri == RtpExtension::kVideoRotationUri;
});
video_stream_encoder_->SetSink(this, rotation_applied);
absl::optional<bool> enable_alr_bw_probing;
// If send-side BWE is enabled, check if we should apply updated probing and
// pacing settings.
if (configured_pacing_factor_) {
absl::optional<AlrExperimentSettings> alr_settings =
GetAlrSettings(content_type);
int queue_time_limit_ms;
if (alr_settings) {
enable_alr_bw_probing = true;
queue_time_limit_ms = alr_settings->max_paced_queue_time;
} else {
RateControlSettings rate_control_settings =
RateControlSettings::ParseFromFieldTrials();
enable_alr_bw_probing = rate_control_settings.UseAlrProbing();
queue_time_limit_ms = pacing_config_.max_pacing_delay.Get().ms();
}
transport->SetQueueTimeLimit(queue_time_limit_ms);
}
if (config_->periodic_alr_bandwidth_probing) {
enable_alr_bw_probing = config_->periodic_alr_bandwidth_probing;
}
if (enable_alr_bw_probing) {
transport->EnablePeriodicAlrProbing(*enable_alr_bw_probing);
}
rtp_transport_queue_->PostTask(ToQueuedTask(transport_queue_safety_, [this] {
if (configured_pacing_factor_)
transport_->SetPacingFactor(*configured_pacing_factor_);
video_stream_encoder_->SetStartBitrate(
bitrate_allocator_->GetStartBitrate(this));
}));
}
VideoSendStreamImpl::~VideoSendStreamImpl() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "~VideoSendStreamImpl: " << config_->ToString();
}
void VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
// Runs on a network thread.
RTC_DCHECK(!rtp_transport_queue_->IsCurrent());
rtp_video_sender_->DeliverRtcp(packet, length);
}
void VideoSendStreamImpl::UpdateActiveSimulcastLayers(
const std::vector<bool> active_layers) {
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
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(rtp_transport_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(rtp_transport_queue_);
transport_queue_safety_->SetAlive();
bitrate_allocator_->AddObserver(this, GetAllocationConfig());
// Start monitoring encoder activity.
{
RTC_DCHECK(!check_encoder_activity_task_.Running());
activity_ = false;
timed_out_ = false;
check_encoder_activity_task_ = RepeatingTaskHandle::DelayedStart(
rtp_transport_queue_->Get(), kEncoderTimeOut, [this] {
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
if (!activity_) {
if (!timed_out_) {
SignalEncoderTimedOut();
}
timed_out_ = true;
disable_padding_ = true;
} else if (timed_out_) {
SignalEncoderActive();
timed_out_ = false;
}
activity_ = false;
return kEncoderTimeOut;
});
}
video_stream_encoder_->SendKeyFrame();
}
void VideoSendStreamImpl::Stop() {
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
RTC_LOG(LS_INFO) << "VideoSendStreamImpl::Stop";
if (!rtp_video_sender_->IsActive())
return;
RTC_DCHECK(transport_queue_safety_->alive());
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
rtp_video_sender_->SetActive(false);
StopVideoSendStream();
}
// RTC_RUN_ON(rtp_transport_queue_)
void VideoSendStreamImpl::StopVideoSendStream() {
bitrate_allocator_->RemoveObserver(this);
check_encoder_activity_task_.Stop();
video_stream_encoder_->OnBitrateUpdated(DataRate::Zero(), DataRate::Zero(),
DataRate::Zero(), 0, 0, 0);
stats_proxy_->OnSetEncoderTargetRate(0);
transport_queue_safety_->SetNotAlive();
}
void VideoSendStreamImpl::SignalEncoderTimedOut() {
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
// If the encoder has not produced anything the last kEncoderTimeOut 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 (!rtp_transport_queue_->IsCurrent()) {
rtp_transport_queue_->PostTask(ToQueuedTask(transport_queue_safety_, [=] {
OnBitrateAllocationUpdated(allocation);
}));
return;
}
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
int64_t now_ms = clock_->TimeInMilliseconds();
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::OnVideoLayersAllocationUpdated(
VideoLayersAllocation allocation) {
// OnVideoLayersAllocationUpdated is handled on the encoder task queue in
// order to not race with OnEncodedImage callbacks.
rtp_video_sender_->OnVideoLayersAllocationUpdated(allocation);
}
void VideoSendStreamImpl::SignalEncoderActive() {
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
if (rtp_video_sender_->IsActive()) {
RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active.";
bitrate_allocator_->AddObserver(this, GetAllocationConfig());
}
}
MediaStreamAllocationConfig VideoSendStreamImpl::GetAllocationConfig() const {
return MediaStreamAllocationConfig{
static_cast<uint32_t>(encoder_min_bitrate_bps_),
encoder_max_bitrate_bps_,
static_cast<uint32_t>(disable_padding_ ? 0 : max_padding_bitrate_),
/* priority_bitrate */ 0,
!config_->suspend_below_min_bitrate,
encoder_bitrate_priority_};
}
void VideoSendStreamImpl::OnEncoderConfigurationChanged(
std::vector<VideoStream> streams,
bool is_svc,
VideoEncoderConfig::ContentType content_type,
int min_transmit_bitrate_bps) {
if (!rtp_transport_queue_->IsCurrent()) {
rtp_transport_queue_->PostTask(ToQueuedTask(
transport_queue_safety_,
[this, streams = std::move(streams), is_svc, content_type,
min_transmit_bitrate_bps]() mutable {
OnEncoderConfigurationChanged(std::move(streams), is_svc,
content_type, min_transmit_bitrate_bps);
}));
return;
}
RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size());
TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged");
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
const VideoCodecType codec_type =
PayloadStringToCodecType(config_->rtp.payload_name);
const absl::optional<DataRate> experimental_min_bitrate =
GetExperimentalMinVideoBitrate(codec_type);
encoder_min_bitrate_bps_ =
experimental_min_bitrate
? experimental_min_bitrate->bps()
: std::max(streams[0].min_bitrate_bps, kDefaultMinVideoBitrateBps);
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_);
// TODO(bugs.webrtc.org/10266): Query the VideoBitrateAllocator instead.
max_padding_bitrate_ = CalculateMaxPadBitrateBps(
streams, is_svc, content_type, 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, GetAllocationConfig());
}
}
EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info) {
// 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.
// Indicate that there still is activity going on.
activity_ = true;
auto enable_padding_task = [this]() {
if (disable_padding_) {
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
disable_padding_ = false;
// To ensure that padding bitrate is propagated to the bitrate allocator.
SignalEncoderActive();
}
};
if (!rtp_transport_queue_->IsCurrent()) {
rtp_transport_queue_->PostTask(
ToQueuedTask(transport_queue_safety_, std::move(enable_padding_task)));
} else {
enable_padding_task();
}
EncodedImageCallback::Result result(EncodedImageCallback::Result::OK);
result =
rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info);
// Check if there's a throttled VideoBitrateAllocation that we should try
// sending.
auto update_task = [this]() {
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
auto& context = video_bitrate_allocation_context_;
if (context && context->throttled_allocation) {
OnBitrateAllocationUpdated(*context->throttled_allocation);
}
};
if (!rtp_transport_queue_->IsCurrent()) {
rtp_transport_queue_->PostTask(
ToQueuedTask(transport_queue_safety_, std::move(update_task)));
} else {
update_task();
}
return result;
}
void VideoSendStreamImpl::OnDroppedFrame(
EncodedImageCallback::DropReason reason) {
activity_ = true;
}
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(rtp_transport_queue_);
RTC_DCHECK(rtp_video_sender_->IsActive())
<< "VideoSendStream::Start has not been called.";
// When the BWE algorithm doesn't pass a stable estimate, we'll use the
// unstable one instead.
if (update.stable_target_bitrate.IsZero()) {
update.stable_target_bitrate = update.target_bitrate;
}
rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_->GetSendFrameRate());
encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps();
const uint32_t protection_bitrate_bps =
rtp_video_sender_->GetProtectionBitrateBps();
DataRate link_allocation = DataRate::Zero();
if (encoder_target_rate_bps_ > protection_bitrate_bps) {
link_allocation =
DataRate::BitsPerSec(encoder_target_rate_bps_ - protection_bitrate_bps);
}
DataRate overhead =
update.target_bitrate - DataRate::BitsPerSec(encoder_target_rate_bps_);
DataRate encoder_stable_target_rate = update.stable_target_bitrate;
if (encoder_stable_target_rate > overhead) {
encoder_stable_target_rate = encoder_stable_target_rate - overhead;
} else {
encoder_stable_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
}
encoder_target_rate_bps_ =
std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
encoder_stable_target_rate =
std::min(DataRate::BitsPerSec(encoder_max_bitrate_bps_),
encoder_stable_target_rate);
DataRate encoder_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
link_allocation = std::max(encoder_target_rate, link_allocation);
video_stream_encoder_->OnBitrateUpdated(
encoder_target_rate, encoder_stable_target_rate, link_allocation,
rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
update.round_trip_time.ms(), update.cwnd_reduce_ratio);
stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
return protection_bitrate_bps;
}
} // namespace internal
} // namespace webrtc
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