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webrtc/video/video_send_stream_impl.cc
Philipp Hancke c14a2cb9cc Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs 
which needs to be added to the remote codecs a=fmtp:

This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes.

This parameter allows for large-scale experimentation and A/B testing
whether the new behavior has advantages. It is to be considered
transitional and may be removed again in the future.

BUG=webrtc:10107

Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41805}
2024-05-03 10:59:22 -04:00

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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 <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
#include "api/call/bitrate_allocation.h"
#include "api/crypto/crypto_options.h"
#include "api/fec_controller.h"
#include "api/field_trials_view.h"
#include "api/metronome/metronome.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/video/encoded_image.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_frame.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_layers_allocation.h"
#include "api/video/video_source_interface.h"
#include "api/video/video_stream_encoder_settings.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/bitrate_allocator.h"
#include "call/rtp_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/video_send_stream.h"
#include "media/base/media_constants.h"
#include "media/base/sdp_video_format_utils.h"
#include "modules/pacing/pacing_controller.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/video_coding/include/video_codec_interface.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/strings/string_builder.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "video/adaptation/overuse_frame_detector.h"
#include "video/config/video_encoder_config.h"
#include "video/encoder_rtcp_feedback.h"
#include "video/frame_cadence_adapter.h"
#include "video/send_delay_stats.h"
#include "video/send_statistics_proxy.h"
#include "video/video_stream_encoder.h"
#include "video/video_stream_encoder_interface.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);
constexpr double kVideoHysteresis = 1.2;
constexpr double kScreenshareHysteresis = 1.35;
constexpr int kMinDefaultAv1BitrateBps =
15000; // This value acts as an absolute minimum AV1 bitrate limit.
// 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 =
content_type == VideoEncoderConfig::ContentType::kScreen
? kScreenshareHysteresis
: kVideoHysteresis;
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(
const FieldTrialsView& field_trials,
VideoEncoderConfig::ContentType content_type) {
if (content_type == VideoEncoderConfig::ContentType::kScreen) {
return AlrExperimentSettings::CreateFromFieldTrial(
field_trials,
AlrExperimentSettings::kScreenshareProbingBweExperimentName);
}
return AlrExperimentSettings::CreateFromFieldTrial(
field_trials,
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,
const FieldTrialsView& field_trials) {
if (!TransportSeqNumExtensionConfigured(config))
return absl::nullopt;
absl::optional<AlrExperimentSettings> alr_settings =
GetAlrSettings(field_trials, content_type);
if (alr_settings)
return alr_settings->pacing_factor;
RateControlSettings rate_control_settings =
RateControlSettings::ParseFromKeyValueConfig(&field_trials);
return rate_control_settings.GetPacingFactor().value_or(
default_pacing_config.pacing_factor);
}
int GetEncoderPriorityBitrate(std::string codec_name,
const FieldTrialsView& field_trials) {
int priority_bitrate = 0;
if (PayloadStringToCodecType(codec_name) == VideoCodecType::kVideoCodecAV1) {
webrtc::FieldTrialParameter<int> av1_priority_bitrate("bitrate", 0);
webrtc::ParseFieldTrial(
{&av1_priority_bitrate},
field_trials.Lookup("WebRTC-AV1-OverridePriorityBitrate"));
priority_bitrate = av1_priority_bitrate;
}
return priority_bitrate;
}
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;
}
int GetDefaultMinVideoBitrateBps(VideoCodecType codec_type) {
if (codec_type == VideoCodecType::kVideoCodecAV1) {
return kMinDefaultAv1BitrateBps;
}
return kDefaultMinVideoBitrateBps;
}
size_t CalculateMaxHeaderSize(const RtpConfig& config) {
size_t header_size = kRtpHeaderSize;
size_t extensions_size = 0;
size_t fec_extensions_size = 0;
if (!config.extensions.empty()) {
RtpHeaderExtensionMap extensions_map(config.extensions);
extensions_size = RtpHeaderExtensionSize(RTPSender::VideoExtensionSizes(),
extensions_map);
fec_extensions_size =
RtpHeaderExtensionSize(RTPSender::FecExtensionSizes(), extensions_map);
}
header_size += extensions_size;
if (config.flexfec.payload_type >= 0) {
// All FEC extensions again plus maximum FlexFec overhead.
header_size += fec_extensions_size + 32;
} else {
if (config.ulpfec.ulpfec_payload_type >= 0) {
// Header with all the FEC extensions will be repeated plus maximum
// UlpFec overhead.
header_size += fec_extensions_size + 18;
}
if (config.ulpfec.red_payload_type >= 0) {
header_size += 1; // RED header.
}
}
// Additional room for Rtx.
if (config.rtx.payload_type >= 0)
header_size += kRtxHeaderSize;
return header_size;
}
VideoStreamEncoder::BitrateAllocationCallbackType
GetBitrateAllocationCallbackType(const VideoSendStream::Config& config,
const FieldTrialsView& field_trials) {
if (webrtc::RtpExtension::FindHeaderExtensionByUri(
config.rtp.extensions,
webrtc::RtpExtension::kVideoLayersAllocationUri,
config.crypto_options.srtp.enable_encrypted_rtp_header_extensions
? RtpExtension::Filter::kPreferEncryptedExtension
: RtpExtension::Filter::kDiscardEncryptedExtension)) {
return VideoStreamEncoder::BitrateAllocationCallbackType::
kVideoLayersAllocation;
}
if (field_trials.IsEnabled("WebRTC-Target-Bitrate-Rtcp")) {
return VideoStreamEncoder::BitrateAllocationCallbackType::
kVideoBitrateAllocation;
}
return VideoStreamEncoder::BitrateAllocationCallbackType::
kVideoBitrateAllocationWhenScreenSharing;
}
RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig(
const VideoSendStream::Config* config) {
RtpSenderFrameEncryptionConfig frame_encryption_config;
frame_encryption_config.frame_encryptor = config->frame_encryptor.get();
frame_encryption_config.crypto_options = config->crypto_options;
return frame_encryption_config;
}
RtpSenderObservers CreateObservers(RtcpRttStats* call_stats,
EncoderRtcpFeedback* encoder_feedback,
SendStatisticsProxy* stats_proxy,
SendPacketObserver* send_packet_observer) {
RtpSenderObservers observers;
observers.rtcp_rtt_stats = call_stats;
observers.intra_frame_callback = encoder_feedback;
observers.rtcp_loss_notification_observer = encoder_feedback;
observers.report_block_data_observer = 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_packet_observer = send_packet_observer;
return observers;
}
std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(
Clock* clock,
int num_cpu_cores,
TaskQueueFactory* task_queue_factory,
SendStatisticsProxy* stats_proxy,
const VideoStreamEncoderSettings& encoder_settings,
VideoStreamEncoder::BitrateAllocationCallbackType
bitrate_allocation_callback_type,
const FieldTrialsView& field_trials,
Metronome* metronome,
webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue =
task_queue_factory->CreateTaskQueue("EncoderQueue",
TaskQueueFactory::Priority::NORMAL);
TaskQueueBase* encoder_queue_ptr = encoder_queue.get();
return std::make_unique<VideoStreamEncoder>(
clock, num_cpu_cores, stats_proxy, encoder_settings,
std::make_unique<OveruseFrameDetector>(stats_proxy),
FrameCadenceAdapterInterface::Create(
clock, encoder_queue_ptr, metronome,
/*worker_queue=*/TaskQueueBase::Current(), field_trials),
std::move(encoder_queue), bitrate_allocation_callback_type, field_trials,
encoder_selector);
}
} // 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,
int num_cpu_cores,
TaskQueueFactory* task_queue_factory,
RtcpRttStats* call_stats,
RtpTransportControllerSendInterface* transport,
Metronome* metronome,
BitrateAllocatorInterface* bitrate_allocator,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
std::unique_ptr<FecController> fec_controller,
const FieldTrialsView& field_trials,
std::unique_ptr<VideoStreamEncoderInterface> video_stream_encoder_for_test)
: transport_(transport),
stats_proxy_(clock, config, encoder_config.content_type, field_trials),
send_packet_observer_(&stats_proxy_, send_delay_stats),
config_(std::move(config)),
content_type_(encoder_config.content_type),
video_stream_encoder_(
video_stream_encoder_for_test
? std::move(video_stream_encoder_for_test)
: CreateVideoStreamEncoder(
clock,
num_cpu_cores,
task_queue_factory,
&stats_proxy_,
config_.encoder_settings,
GetBitrateAllocationCallbackType(config_, field_trials),
field_trials,
metronome,
config_.encoder_selector)),
encoder_feedback_(
clock,
SupportsPerLayerPictureLossIndication(
encoder_config.video_format.parameters),
config_.rtp.ssrcs,
video_stream_encoder_.get(),
[this](uint32_t ssrc, const std::vector<uint16_t>& seq_nums) {
return rtp_video_sender_->GetSentRtpPacketInfos(ssrc, seq_nums);
}),
rtp_video_sender_(transport->CreateRtpVideoSender(
suspended_ssrcs,
suspended_payload_states,
config_.rtp,
config_.rtcp_report_interval_ms,
config_.send_transport,
CreateObservers(call_stats,
&encoder_feedback_,
&stats_proxy_,
&send_packet_observer_),
event_log,
std::move(fec_controller),
CreateFrameEncryptionConfig(&config_),
config_.frame_transformer)),
clock_(clock),
has_alr_probing_(
config_.periodic_alr_bandwidth_probing ||
GetAlrSettings(field_trials, encoder_config.content_type)),
pacing_config_(PacingConfig(field_trials)),
worker_queue_(TaskQueueBase::Current()),
timed_out_(false),
bitrate_allocator_(bitrate_allocator),
disable_padding_(true),
max_padding_bitrate_(0),
encoder_min_bitrate_bps_(0),
encoder_max_bitrate_bps_(
GetInitialEncoderMaxBitrate(encoder_config.max_bitrate_bps)),
encoder_target_rate_bps_(0),
encoder_bitrate_priority_(encoder_config.bitrate_priority),
encoder_av1_priority_bitrate_override_bps_(
GetEncoderPriorityBitrate(config_.rtp.payload_name, field_trials)),
configured_pacing_factor_(GetConfiguredPacingFactor(config_,
content_type_,
pacing_config_,
field_trials)) {
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(encoder_max_bitrate_bps_, 0);
RTC_LOG(LS_INFO) << "VideoSendStreamImpl: " << config_.ToString();
RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled(field_trials));
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(field_trials, 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::ParseFromKeyValueConfig(&field_trials);
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);
}
if (configured_pacing_factor_)
transport_->SetPacingFactor(*configured_pacing_factor_);
// 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);
video_stream_encoder_->SetStartBitrate(
bitrate_allocator_->GetStartBitrate(this));
video_stream_encoder_->SetFecControllerOverride(rtp_video_sender_);
ReconfigureVideoEncoder(std::move(encoder_config));
}
VideoSendStreamImpl::~VideoSendStreamImpl() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "~VideoSendStreamImpl: " << config_.ToString();
RTC_DCHECK(!started());
transport_->DestroyRtpVideoSender(rtp_video_sender_);
}
void VideoSendStreamImpl::AddAdaptationResource(
rtc::scoped_refptr<Resource> resource) {
RTC_DCHECK_RUN_ON(&thread_checker_);
video_stream_encoder_->AddAdaptationResource(resource);
}
std::vector<rtc::scoped_refptr<Resource>>
VideoSendStreamImpl::GetAdaptationResources() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return video_stream_encoder_->GetAdaptationResources();
}
void VideoSendStreamImpl::SetSource(
rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
const DegradationPreference& degradation_preference) {
RTC_DCHECK_RUN_ON(&thread_checker_);
video_stream_encoder_->SetSource(source, degradation_preference);
}
void VideoSendStreamImpl::ReconfigureVideoEncoder(VideoEncoderConfig config) {
ReconfigureVideoEncoder(std::move(config), nullptr);
}
void VideoSendStreamImpl::ReconfigureVideoEncoder(
VideoEncoderConfig config,
SetParametersCallback callback) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK_EQ(content_type_, config.content_type);
RTC_LOG(LS_VERBOSE) << "Encoder config: " << config.ToString()
<< " VideoSendStream config: " << config_.ToString();
video_stream_encoder_->ConfigureEncoder(
std::move(config),
config_.rtp.max_packet_size - CalculateMaxHeaderSize(config_.rtp),
std::move(callback));
}
VideoSendStream::Stats VideoSendStreamImpl::GetStats() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return stats_proxy_.GetStats();
}
absl::optional<float> VideoSendStreamImpl::GetPacingFactorOverride() const {
return configured_pacing_factor_;
}
void VideoSendStreamImpl::StopPermanentlyAndGetRtpStates(
VideoSendStreamImpl::RtpStateMap* rtp_state_map,
VideoSendStreamImpl::RtpPayloadStateMap* payload_state_map) {
RTC_DCHECK_RUN_ON(&thread_checker_);
video_stream_encoder_->Stop();
running_ = false;
// Always run these cleanup steps regardless of whether running_ was set
// or not. This will unregister callbacks before destruction.
// See `VideoSendStreamImpl::StopVideoSendStream` for more.
Stop();
*rtp_state_map = GetRtpStates();
*payload_state_map = GetRtpPayloadStates();
}
void VideoSendStreamImpl::GenerateKeyFrame(
const std::vector<std::string>& rids) {
RTC_DCHECK_RUN_ON(&thread_checker_);
// Map rids to layers. If rids is empty, generate a keyframe for all layers.
std::vector<VideoFrameType> next_frames(config_.rtp.ssrcs.size(),
VideoFrameType::kVideoFrameKey);
if (!config_.rtp.rids.empty() && !rids.empty()) {
std::fill(next_frames.begin(), next_frames.end(),
VideoFrameType::kVideoFrameDelta);
for (const auto& rid : rids) {
for (size_t i = 0; i < config_.rtp.rids.size(); i++) {
if (config_.rtp.rids[i] == rid) {
next_frames[i] = VideoFrameType::kVideoFrameKey;
break;
}
}
}
}
if (video_stream_encoder_) {
video_stream_encoder_->SendKeyFrame(next_frames);
}
}
void VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
RTC_DCHECK_RUN_ON(&thread_checker_);
rtp_video_sender_->DeliverRtcp(packet, length);
}
bool VideoSendStreamImpl::started() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return rtp_video_sender_->IsActive();
}
void VideoSendStreamImpl::Start() {
const std::vector<bool> active_layers(config_.rtp.ssrcs.size(), true);
StartPerRtpStream(active_layers);
}
void VideoSendStreamImpl::StartPerRtpStream(
const std::vector<bool> active_layers) {
RTC_DCHECK_RUN_ON(&thread_checker_);
rtc::StringBuilder active_layers_string;
active_layers_string << "{";
for (size_t i = 0; i < active_layers.size(); ++i) {
if (active_layers[i]) {
active_layers_string << "1";
} else {
active_layers_string << "0";
}
if (i < active_layers.size() - 1) {
active_layers_string << ", ";
}
}
active_layers_string << "}";
RTC_LOG(LS_INFO) << "StartPerRtpStream: " << active_layers_string.str();
bool previously_active = rtp_video_sender_->IsActive();
rtp_video_sender_->SetActiveModules(active_layers);
if (!rtp_video_sender_->IsActive() && previously_active) {
StopVideoSendStream();
} else if (rtp_video_sender_->IsActive() && !previously_active) {
StartupVideoSendStream();
}
}
void VideoSendStreamImpl::StartupVideoSendStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
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(
worker_queue_, kEncoderTimeOut, [this] {
RTC_DCHECK_RUN_ON(&thread_checker_);
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(&thread_checker_);
RTC_LOG(LS_INFO) << "VideoSendStreamImpl::Stop";
if (!rtp_video_sender_->IsActive())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
rtp_video_sender_->Stop();
StopVideoSendStream();
}
void VideoSendStreamImpl::StopVideoSendStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
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);
}
void VideoSendStreamImpl::SignalEncoderTimedOut() {
RTC_DCHECK_RUN_ON(&thread_checker_);
// 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) {
// OnBitrateAllocationUpdated is invoked from the encoder task queue or
// the worker_queue_.
auto task = [this, allocation] {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (encoder_target_rate_bps_ == 0) {
return;
}
int64_t now_ms = clock_->TimeInMilliseconds();
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);
};
if (!worker_queue_->IsCurrent()) {
worker_queue_->PostTask(
SafeTask(worker_queue_safety_.flag(), std::move(task)));
} else {
task();
}
}
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(&thread_checker_);
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_),
encoder_av1_priority_bitrate_override_bps_,
!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) {
// Currently called on the encoder TQ
RTC_DCHECK(!worker_queue_->IsCurrent());
auto closure = [this, streams = std::move(streams), is_svc, content_type,
min_transmit_bitrate_bps]() mutable {
RTC_DCHECK_GE(config_.rtp.ssrcs.size(), streams.size());
TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged");
RTC_DCHECK_RUN_ON(&thread_checker_);
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,
GetDefaultMinVideoBitrateBps(codec_type));
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());
}
};
worker_queue_->PostTask(
SafeTask(worker_queue_safety_.flag(), std::move(closure)));
}
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;
RTC_DCHECK(!worker_queue_->IsCurrent());
auto task_to_run_on_worker = [this]() {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (disable_padding_) {
disable_padding_ = false;
// To ensure that padding bitrate is propagated to the bitrate allocator.
SignalEncoderActive();
}
// Check if there's a throttled VideoBitrateAllocation that we should try
// sending.
auto& context = video_bitrate_allocation_context_;
if (context && context->throttled_allocation) {
OnBitrateAllocationUpdated(*context->throttled_allocation);
}
};
worker_queue_->PostTask(
SafeTask(worker_queue_safety_.flag(), std::move(task_to_run_on_worker)));
return rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info);
}
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(&thread_checker_);
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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