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webrtc_m130/video/video_send_stream_impl.cc
Danil Chapovalov 6e7c2685e3 Allow recursive check for RTC_DCHECK_RUN_ON macro 
instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.

Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue

Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
2022-07-26 09:27:23 +00:00

646 lines
24 KiB
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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/pacing_controller.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/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(const FieldTrialsView& field_trials)
: pacing_factor("factor", kStrictPacingMultiplier),
max_pacing_delay("max_delay", PacingController::kMaxExpectedQueueLength) {
ParseFieldTrial({&pacing_factor, &max_pacing_delay},
field_trials.Lookup("WebRTC-Video-Pacing"));
}
PacingConfig::PacingConfig(const PacingConfig&) = default;
PacingConfig::~PacingConfig() = default;
VideoSendStreamImpl::VideoSendStreamImpl(
Clock* clock,
SendStatisticsProxy* stats_proxy,
TaskQueueBase* 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,
const FieldTrialsView& field_trials)
: clock_(clock),
has_alr_probing_(config->periodic_alr_bandwidth_probing ||
GetAlrSettings(content_type)),
pacing_config_(PacingConfig(field_trials)),
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(SafeTask(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_, 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();
}
void VideoSendStreamImpl::StopVideoSendStream() {
RTC_DCHECK_RUN_ON(rtp_transport_queue_);
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(SafeTask(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(SafeTask(
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(
SafeTask(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(
SafeTask(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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