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webrtc_m130/video/video_send_stream.cc
Alex Narest b3944f021d Media track ID visibility at BWE level 
Track IDs are assigned by application during track creation. 
Track IDs are used by custom bitrate allocation strategies to identify tracks. 
Track ID can be empty, in that case bitrate allocation strategies will not be able to handle
these tracks specifically and will handle them as a default.

Bug: webrtc:8243
Change-Id: I89987e33328320bfd0539ad532342df6da144c98
Reviewed-on: https://webrtc-review.googlesource.com/4820
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20285}
2017-10-13 13:47:07 +00:00

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/*
* Copyright (c) 2013 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.h"
#include <algorithm>
#include <cmath>
#include <sstream>
#include <string>
#include <utility>
#include <vector>
#include "call/rtp_transport_controller_send_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "common_video/include/video_bitrate_allocator.h"
#include "modules/bitrate_controller/include/bitrate_controller.h"
#include "modules/congestion_controller/include/send_side_congestion_controller.h"
#include "modules/pacing/alr_detector.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/utility/ivf_file_writer.h"
#include "rtc_base/checks.h"
#include "rtc_base/file.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
#include "rtc_base/weak_ptr.h"
#include "system_wrappers/include/field_trial.h"
#include "video/call_stats.h"
#include "video/payload_router.h"
#include "call/video_send_stream.h"
namespace webrtc {
static const int kMinSendSidePacketHistorySize = 600;
namespace {
// We don't do MTU discovery, so assume that we have the standard ethernet MTU.
const size_t kPathMTU = 1500;
std::vector<RtpRtcp*> CreateRtpRtcpModules(
Transport* outgoing_transport,
RtcpIntraFrameObserver* intra_frame_callback,
RtcpBandwidthObserver* bandwidth_callback,
RtpTransportControllerSendInterface* transport,
RtcpRttStats* rtt_stats,
FlexfecSender* flexfec_sender,
SendStatisticsProxy* stats_proxy,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer,
size_t num_modules,
RtpKeepAliveConfig keepalive_config) {
RTC_DCHECK_GT(num_modules, 0);
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.receiver_only = false;
configuration.flexfec_sender = flexfec_sender;
configuration.outgoing_transport = outgoing_transport;
configuration.intra_frame_callback = intra_frame_callback;
configuration.bandwidth_callback = bandwidth_callback;
configuration.transport_feedback_callback =
transport->transport_feedback_observer();
configuration.rtt_stats = rtt_stats;
configuration.rtcp_packet_type_counter_observer = stats_proxy;
configuration.paced_sender = transport->packet_sender();
configuration.transport_sequence_number_allocator =
transport->packet_router();
configuration.send_bitrate_observer = stats_proxy;
configuration.send_frame_count_observer = stats_proxy;
configuration.send_side_delay_observer = stats_proxy;
configuration.send_packet_observer = send_delay_stats;
configuration.event_log = event_log;
configuration.retransmission_rate_limiter = retransmission_rate_limiter;
configuration.overhead_observer = overhead_observer;
configuration.keepalive_config = keepalive_config;
std::vector<RtpRtcp*> modules;
for (size_t i = 0; i < num_modules; ++i) {
RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
modules.push_back(rtp_rtcp);
}
return modules;
}
// TODO(brandtr): Update this function when we support multistream protection.
std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender(
const VideoSendStream::Config& config,
const std::map<uint32_t, RtpState>& suspended_ssrcs) {
if (config.rtp.flexfec.payload_type < 0) {
return nullptr;
}
RTC_DCHECK_GE(config.rtp.flexfec.payload_type, 0);
RTC_DCHECK_LE(config.rtp.flexfec.payload_type, 127);
if (config.rtp.flexfec.ssrc == 0) {
LOG(LS_WARNING) << "FlexFEC is enabled, but no FlexFEC SSRC given. "
"Therefore disabling FlexFEC.";
return nullptr;
}
if (config.rtp.flexfec.protected_media_ssrcs.empty()) {
LOG(LS_WARNING) << "FlexFEC is enabled, but no protected media SSRC given. "
"Therefore disabling FlexFEC.";
return nullptr;
}
if (config.rtp.ssrcs.size() > 1) {
LOG(LS_WARNING) << "Both FlexFEC and simulcast are enabled. This "
"combination is however not supported by our current "
"FlexFEC implementation. Therefore disabling FlexFEC.";
return nullptr;
}
if (config.rtp.flexfec.protected_media_ssrcs.size() > 1) {
LOG(LS_WARNING)
<< "The supplied FlexfecConfig contained multiple protected "
"media streams, but our implementation currently only "
"supports protecting a single media stream. "
"To avoid confusion, disabling FlexFEC completely.";
return nullptr;
}
const RtpState* rtp_state = nullptr;
auto it = suspended_ssrcs.find(config.rtp.flexfec.ssrc);
if (it != suspended_ssrcs.end()) {
rtp_state = &it->second;
}
RTC_DCHECK_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size());
return std::unique_ptr<FlexfecSender>(new FlexfecSender(
config.rtp.flexfec.payload_type, config.rtp.flexfec.ssrc,
config.rtp.flexfec.protected_media_ssrcs[0], config.rtp.extensions,
RTPSender::FecExtensionSizes(), rtp_state, Clock::GetRealTimeClock()));
}
} // namespace
namespace {
bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
const VideoCodecType codecType = PayloadStringToCodecType(payload_name);
if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
return true;
}
return false;
}
int CalculateMaxPadBitrateBps(std::vector<VideoStream> streams,
int min_transmit_bitrate_bps,
bool pad_to_min_bitrate) {
int pad_up_to_bitrate_bps = 0;
// Calculate max padding bitrate for a multi layer codec.
if (streams.size() > 1) {
// Pad to min bitrate of the highest layer.
pad_up_to_bitrate_bps = streams[streams.size() - 1].min_bitrate_bps;
// Add target_bitrate_bps of the lower layers.
for (size_t i = 0; i < streams.size() - 1; ++i)
pad_up_to_bitrate_bps += streams[i].target_bitrate_bps;
} else if (pad_to_min_bitrate) {
pad_up_to_bitrate_bps = streams[0].min_bitrate_bps;
}
pad_up_to_bitrate_bps =
std::max(pad_up_to_bitrate_bps, min_transmit_bitrate_bps);
return pad_up_to_bitrate_bps;
}
uint32_t CalculateOverheadRateBps(int packets_per_second,
size_t overhead_bytes_per_packet,
uint32_t max_overhead_bps) {
uint32_t overhead_bps =
static_cast<uint32_t>(8 * overhead_bytes_per_packet * packets_per_second);
return std::min(overhead_bps, max_overhead_bps);
}
int CalculatePacketRate(uint32_t bitrate_bps, size_t packet_size_bytes) {
size_t packet_size_bits = 8 * packet_size_bytes;
// Ceil for int value of bitrate_bps / packet_size_bits.
return static_cast<int>((bitrate_bps + packet_size_bits - 1) /
packet_size_bits);
}
} // namespace
namespace internal {
// VideoSendStreamImpl implements internal::VideoSendStream.
// It is created and destroyed on |worker_queue|. The intent is to decrease the
// need for locking and to ensure methods are called in sequence.
// Public methods except |DeliverRtcp| must be called on |worker_queue|.
// DeliverRtcp is called on the libjingle worker thread or a network thread.
// An encoder may deliver frames through the EncodedImageCallback on an
// arbitrary thread.
class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver,
public webrtc::OverheadObserver,
public webrtc::VCMProtectionCallback,
public VideoStreamEncoder::EncoderSink,
public VideoBitrateAllocationObserver {
public:
VideoSendStreamImpl(
SendStatisticsProxy* stats_proxy,
rtc::TaskQueue* worker_queue,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocator* bitrate_allocator,
SendDelayStats* send_delay_stats,
VideoStreamEncoder* video_stream_encoder,
RtcEventLog* event_log,
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
std::map<uint32_t, RtpState> suspended_ssrcs,
std::map<uint32_t, RtpPayloadState> suspended_payload_states,
VideoEncoderConfig::ContentType content_type);
~VideoSendStreamImpl() override;
// RegisterProcessThread register |module_process_thread| with those objects
// that use it. Registration has to happen on the thread were
// |module_process_thread| was created (libjingle's worker thread).
// TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue,
// maybe |worker_queue|.
void RegisterProcessThread(ProcessThread* module_process_thread);
void DeRegisterProcessThread();
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
void Start();
void Stop();
VideoSendStream::RtpStateMap GetRtpStates() const;
VideoSendStream::RtpPayloadStateMap GetRtpPayloadStates() const;
void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
size_t byte_limit);
void SetTransportOverhead(size_t transport_overhead_per_packet);
private:
class CheckEncoderActivityTask;
class EncoderReconfiguredTask;
// Implements BitrateAllocatorObserver.
uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt,
int64_t probing_interval_ms) override;
// Implements webrtc::VCMProtectionCallback.
int ProtectionRequest(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps) override;
// Implements OverheadObserver.
void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
void OnEncoderConfigurationChanged(std::vector<VideoStream> streams,
int min_transmit_bitrate_bps) override;
// Implements EncodedImageCallback. The implementation routes encoded frames
// to the |payload_router_| and |config.pre_encode_callback| if set.
// Called on an arbitrary encoder callback thread.
EncodedImageCallback::Result OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation) override;
// Implements VideoBitrateAllocationObserver.
void OnBitrateAllocationUpdated(const BitrateAllocation& allocation) override;
void ConfigureProtection();
void ConfigureSsrcs();
void SignalEncoderTimedOut();
void SignalEncoderActive();
const bool send_side_bwe_with_overhead_;
SendStatisticsProxy* const stats_proxy_;
const VideoSendStream::Config* const config_;
std::map<uint32_t, RtpState> suspended_ssrcs_;
ProcessThread* module_process_thread_;
rtc::ThreadChecker module_process_thread_checker_;
rtc::TaskQueue* const worker_queue_;
rtc::CriticalSection encoder_activity_crit_sect_;
CheckEncoderActivityTask* check_encoder_activity_task_
RTC_GUARDED_BY(encoder_activity_crit_sect_);
CallStats* const call_stats_;
RtpTransportControllerSendInterface* const transport_;
BitrateAllocator* const bitrate_allocator_;
// TODO(brandtr): Move ownership to PayloadRouter.
std::unique_ptr<FlexfecSender> flexfec_sender_;
rtc::CriticalSection ivf_writers_crit_;
std::unique_ptr<IvfFileWriter>
file_writers_[kMaxSimulcastStreams] RTC_GUARDED_BY(ivf_writers_crit_);
int max_padding_bitrate_;
int encoder_min_bitrate_bps_;
uint32_t encoder_max_bitrate_bps_;
uint32_t encoder_target_rate_bps_;
VideoStreamEncoder* const video_stream_encoder_;
EncoderRtcpFeedback encoder_feedback_;
ProtectionBitrateCalculator protection_bitrate_calculator_;
const std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
// RtpRtcp modules, declared here as they use other members on construction.
const std::vector<RtpRtcp*> rtp_rtcp_modules_;
PayloadRouter payload_router_;
// |weak_ptr_| to our self. This is used since we can not call
// |weak_ptr_factory_.GetWeakPtr| from multiple sequences but it is ok to copy
// an existing WeakPtr.
rtc::WeakPtr<VideoSendStreamImpl> weak_ptr_;
// |weak_ptr_factory_| must be declared last to make sure all WeakPtr's are
// invalidated before any other members are destroyed.
rtc::WeakPtrFactory<VideoSendStreamImpl> weak_ptr_factory_;
rtc::CriticalSection overhead_bytes_per_packet_crit_;
size_t overhead_bytes_per_packet_
RTC_GUARDED_BY(overhead_bytes_per_packet_crit_);
size_t transport_overhead_bytes_per_packet_;
};
// TODO(tommi): See if there's a more elegant way to create a task that creates
// an object on the correct task queue.
class VideoSendStream::ConstructionTask : public rtc::QueuedTask {
public:
ConstructionTask(
std::unique_ptr<VideoSendStreamImpl>* send_stream,
rtc::Event* done_event,
SendStatisticsProxy* stats_proxy,
VideoStreamEncoder* video_stream_encoder,
ProcessThread* module_process_thread,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocator* bitrate_allocator,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
VideoEncoderConfig::ContentType content_type)
: send_stream_(send_stream),
done_event_(done_event),
stats_proxy_(stats_proxy),
video_stream_encoder_(video_stream_encoder),
call_stats_(call_stats),
transport_(transport),
bitrate_allocator_(bitrate_allocator),
send_delay_stats_(send_delay_stats),
event_log_(event_log),
config_(config),
initial_encoder_max_bitrate_(initial_encoder_max_bitrate),
suspended_ssrcs_(suspended_ssrcs),
suspended_payload_states_(suspended_payload_states),
content_type_(content_type) {}
~ConstructionTask() override { done_event_->Set(); }
private:
bool Run() override {
send_stream_->reset(new VideoSendStreamImpl(
stats_proxy_, rtc::TaskQueue::Current(), call_stats_, transport_,
bitrate_allocator_, send_delay_stats_, video_stream_encoder_,
event_log_, config_, initial_encoder_max_bitrate_,
std::move(suspended_ssrcs_), std::move(suspended_payload_states_),
content_type_));
return true;
}
std::unique_ptr<VideoSendStreamImpl>* const send_stream_;
rtc::Event* const done_event_;
SendStatisticsProxy* const stats_proxy_;
VideoStreamEncoder* const video_stream_encoder_;
CallStats* const call_stats_;
RtpTransportControllerSendInterface* const transport_;
BitrateAllocator* const bitrate_allocator_;
SendDelayStats* const send_delay_stats_;
RtcEventLog* const event_log_;
const VideoSendStream::Config* config_;
int initial_encoder_max_bitrate_;
std::map<uint32_t, RtpState> suspended_ssrcs_;
std::map<uint32_t, RtpPayloadState> suspended_payload_states_;
const VideoEncoderConfig::ContentType content_type_;
};
class VideoSendStream::DestructAndGetRtpStateTask : public rtc::QueuedTask {
public:
DestructAndGetRtpStateTask(
VideoSendStream::RtpStateMap* state_map,
VideoSendStream::RtpPayloadStateMap* payload_state_map,
std::unique_ptr<VideoSendStreamImpl> send_stream,
rtc::Event* done_event)
: state_map_(state_map),
payload_state_map_(payload_state_map),
send_stream_(std::move(send_stream)),
done_event_(done_event) {}
~DestructAndGetRtpStateTask() override { RTC_CHECK(!send_stream_); }
private:
bool Run() override {
send_stream_->Stop();
*state_map_ = send_stream_->GetRtpStates();
*payload_state_map_ = send_stream_->GetRtpPayloadStates();
send_stream_.reset();
done_event_->Set();
return true;
}
VideoSendStream::RtpStateMap* state_map_;
VideoSendStream::RtpPayloadStateMap* payload_state_map_;
std::unique_ptr<VideoSendStreamImpl> send_stream_;
rtc::Event* done_event_;
};
// 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_;
};
class VideoSendStreamImpl::EncoderReconfiguredTask : public rtc::QueuedTask {
public:
EncoderReconfiguredTask(const rtc::WeakPtr<VideoSendStreamImpl>& send_stream,
std::vector<VideoStream> streams,
int min_transmit_bitrate_bps)
: send_stream_(std::move(send_stream)),
streams_(std::move(streams)),
min_transmit_bitrate_bps_(min_transmit_bitrate_bps) {}
private:
bool Run() override {
if (send_stream_)
send_stream_->OnEncoderConfigurationChanged(std::move(streams_),
min_transmit_bitrate_bps_);
return true;
}
rtc::WeakPtr<VideoSendStreamImpl> send_stream_;
std::vector<VideoStream> streams_;
int min_transmit_bitrate_bps_;
};
VideoSendStream::VideoSendStream(
int num_cpu_cores,
ProcessThread* module_process_thread,
rtc::TaskQueue* worker_queue,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocator* 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)
: worker_queue_(worker_queue),
thread_sync_event_(false /* manual_reset */, false),
stats_proxy_(Clock::GetRealTimeClock(),
config,
encoder_config.content_type),
config_(std::move(config)),
content_type_(encoder_config.content_type) {
video_stream_encoder_.reset(
new VideoStreamEncoder(num_cpu_cores, &stats_proxy_,
config_.encoder_settings,
config_.pre_encode_callback,
config_.post_encode_callback,
std::unique_ptr<OveruseFrameDetector>()));
worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(new ConstructionTask(
&send_stream_, &thread_sync_event_, &stats_proxy_,
video_stream_encoder_.get(), module_process_thread, call_stats, transport,
bitrate_allocator, send_delay_stats, event_log, &config_,
encoder_config.max_bitrate_bps, suspended_ssrcs, suspended_payload_states,
encoder_config.content_type)));
// Wait for ConstructionTask to complete so that |send_stream_| can be used.
// |module_process_thread| must be registered and deregistered on the thread
// it was created on.
thread_sync_event_.Wait(rtc::Event::kForever);
send_stream_->RegisterProcessThread(module_process_thread);
// TODO(sprang): Enable this also for regular video calls if it works well.
if (encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen) {
// Only signal target bitrate for screenshare streams, for now.
video_stream_encoder_->SetBitrateObserver(send_stream_.get());
}
video_stream_encoder_->RegisterProcessThread(module_process_thread);
ReconfigureVideoEncoder(std::move(encoder_config));
}
VideoSendStream::~VideoSendStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(!send_stream_);
}
void VideoSendStream::Start() {
RTC_DCHECK_RUN_ON(&thread_checker_);
LOG(LS_INFO) << "VideoSendStream::Start";
VideoSendStreamImpl* send_stream = send_stream_.get();
worker_queue_->PostTask([this, send_stream] {
send_stream->Start();
thread_sync_event_.Set();
});
// It is expected that after VideoSendStream::Start has been called, incoming
// frames are not dropped in VideoStreamEncoder. To ensure this, Start has to
// be synchronized.
thread_sync_event_.Wait(rtc::Event::kForever);
}
void VideoSendStream::Stop() {
RTC_DCHECK_RUN_ON(&thread_checker_);
LOG(LS_INFO) << "VideoSendStream::Stop";
VideoSendStreamImpl* send_stream = send_stream_.get();
worker_queue_->PostTask([send_stream] { send_stream->Stop(); });
}
void VideoSendStream::SetSource(
rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
const DegradationPreference& degradation_preference) {
RTC_DCHECK_RUN_ON(&thread_checker_);
video_stream_encoder_->SetSource(source, degradation_preference);
}
void VideoSendStream::ReconfigureVideoEncoder(VideoEncoderConfig config) {
// TODO(perkj): Some test cases in VideoSendStreamTest call
// ReconfigureVideoEncoder from the network thread.
// RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(content_type_ == config.content_type);
video_stream_encoder_->ConfigureEncoder(std::move(config),
config_.rtp.max_packet_size,
config_.rtp.nack.rtp_history_ms > 0);
}
VideoSendStream::Stats VideoSendStream::GetStats() {
// TODO(perkj, solenberg): Some test cases in EndToEndTest call GetStats from
// a network thread. See comment in Call::GetStats().
// RTC_DCHECK_RUN_ON(&thread_checker_);
return stats_proxy_.GetStats();
}
void VideoSendStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK_RUN_ON(&thread_checker_);
VideoSendStreamImpl* send_stream = send_stream_.get();
worker_queue_->PostTask(
[send_stream, state] { send_stream->SignalNetworkState(state); });
}
void VideoSendStream::StopPermanentlyAndGetRtpStates(
VideoSendStream::RtpStateMap* rtp_state_map,
VideoSendStream::RtpPayloadStateMap* payload_state_map) {
RTC_DCHECK_RUN_ON(&thread_checker_);
video_stream_encoder_->Stop();
video_stream_encoder_->DeRegisterProcessThread();
send_stream_->DeRegisterProcessThread();
worker_queue_->PostTask(
std::unique_ptr<rtc::QueuedTask>(new DestructAndGetRtpStateTask(
rtp_state_map, payload_state_map, std::move(send_stream_),
&thread_sync_event_)));
thread_sync_event_.Wait(rtc::Event::kForever);
}
void VideoSendStream::SetTransportOverhead(
size_t transport_overhead_per_packet) {
RTC_DCHECK_RUN_ON(&thread_checker_);
VideoSendStreamImpl* send_stream = send_stream_.get();
worker_queue_->PostTask([send_stream, transport_overhead_per_packet] {
send_stream->SetTransportOverhead(transport_overhead_per_packet);
});
}
bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// Called on a network thread.
return send_stream_->DeliverRtcp(packet, length);
}
void VideoSendStream::EnableEncodedFrameRecording(
const std::vector<rtc::PlatformFile>& files,
size_t byte_limit) {
send_stream_->EnableEncodedFrameRecording(files, byte_limit);
}
VideoSendStreamImpl::VideoSendStreamImpl(
SendStatisticsProxy* stats_proxy,
rtc::TaskQueue* worker_queue,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocator* bitrate_allocator,
SendDelayStats* send_delay_stats,
VideoStreamEncoder* video_stream_encoder,
RtcEventLog* event_log,
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
std::map<uint32_t, RtpState> suspended_ssrcs,
std::map<uint32_t, RtpPayloadState> suspended_payload_states,
VideoEncoderConfig::ContentType content_type)
: send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
stats_proxy_(stats_proxy),
config_(config),
suspended_ssrcs_(std::move(suspended_ssrcs)),
module_process_thread_(nullptr),
worker_queue_(worker_queue),
check_encoder_activity_task_(nullptr),
call_stats_(call_stats),
transport_(transport),
bitrate_allocator_(bitrate_allocator),
flexfec_sender_(MaybeCreateFlexfecSender(*config_, suspended_ssrcs_)),
max_padding_bitrate_(0),
encoder_min_bitrate_bps_(0),
encoder_max_bitrate_bps_(initial_encoder_max_bitrate),
encoder_target_rate_bps_(0),
video_stream_encoder_(video_stream_encoder),
encoder_feedback_(Clock::GetRealTimeClock(),
config_->rtp.ssrcs,
video_stream_encoder),
protection_bitrate_calculator_(Clock::GetRealTimeClock(), this),
bandwidth_observer_(transport->send_side_cc()
->GetBitrateController()
->CreateRtcpBandwidthObserver()),
rtp_rtcp_modules_(CreateRtpRtcpModules(
config_->send_transport,
&encoder_feedback_,
bandwidth_observer_.get(),
transport,
call_stats_->rtcp_rtt_stats(),
flexfec_sender_.get(),
stats_proxy_,
send_delay_stats,
event_log,
transport->send_side_cc()->GetRetransmissionRateLimiter(),
this,
config_->rtp.ssrcs.size(),
transport->keepalive_config())),
payload_router_(rtp_rtcp_modules_,
config_->rtp.ssrcs,
config_->encoder_settings.payload_type,
suspended_payload_states),
weak_ptr_factory_(this),
overhead_bytes_per_packet_(0),
transport_overhead_bytes_per_packet_(0) {
RTC_DCHECK_RUN_ON(worker_queue_);
LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
weak_ptr_ = weak_ptr_factory_.GetWeakPtr();
module_process_thread_checker_.DetachFromThread();
RTC_DCHECK(!config_->rtp.ssrcs.empty());
RTC_DCHECK(call_stats_);
RTC_DCHECK(transport_);
RTC_DCHECK(transport_->send_side_cc());
RTC_CHECK(field_trial::FindFullName(
AlrDetector::kStrictPacingAndProbingExperimentName)
.empty() ||
field_trial::FindFullName(
AlrDetector::kScreenshareProbingBweExperimentName)
.empty());
rtc::Optional<AlrDetector::AlrExperimentSettings> alr_settings;
if (content_type == VideoEncoderConfig::ContentType::kScreen) {
alr_settings = AlrDetector::ParseAlrSettingsFromFieldTrial(
AlrDetector::kScreenshareProbingBweExperimentName);
} else {
alr_settings = AlrDetector::ParseAlrSettingsFromFieldTrial(
AlrDetector::kStrictPacingAndProbingExperimentName);
}
if (alr_settings) {
transport->send_side_cc()->EnablePeriodicAlrProbing(true);
transport->pacer()->SetPacingFactor(alr_settings->pacing_factor);
transport->pacer()->SetQueueTimeLimit(alr_settings->max_paced_queue_time);
}
if (config_->periodic_alr_bandwidth_probing) {
transport->send_side_cc()->EnablePeriodicAlrProbing(true);
}
// RTP/RTCP initialization.
// We add the highest spatial layer first to ensure it'll be prioritized
// when sending padding, with the hope that the packet rate will be smaller,
// and that it's more important to protect than the lower layers.
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
constexpr bool remb_candidate = true;
transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate);
}
for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) {
const std::string& extension = config_->rtp.extensions[i].uri;
int id = config_->rtp.extensions[i].id;
// One-byte-extension local identifiers are in the range 1-14 inclusive.
RTC_DCHECK_GE(id, 1);
RTC_DCHECK_LE(id, 14);
RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
StringToRtpExtensionType(extension), id));
}
}
ConfigureProtection();
ConfigureSsrcs();
// TODO(pbos): Should we set CNAME on all RTP modules?
rtp_rtcp_modules_.front()->SetCNAME(config_->rtp.c_name.c_str());
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->RegisterRtcpStatisticsCallback(stats_proxy_);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(stats_proxy_);
rtp_rtcp->SetMaxRtpPacketSize(config_->rtp.max_packet_size);
rtp_rtcp->RegisterVideoSendPayload(
config_->encoder_settings.payload_type,
config_->encoder_settings.payload_name.c_str());
}
RTC_DCHECK(config_->encoder_settings.encoder);
RTC_DCHECK_GE(config_->encoder_settings.payload_type, 0);
RTC_DCHECK_LE(config_->encoder_settings.payload_type, 127);
video_stream_encoder_->SetStartBitrate(
bitrate_allocator_->GetStartBitrate(this));
// Only request rotation at the source when we positively know that the remote
// side doesn't support the rotation extension. This allows us to prepare the
// encoder in the expectation that rotation is supported - which is the common
// case.
bool rotation_applied =
std::find_if(config_->rtp.extensions.begin(),
config_->rtp.extensions.end(),
[](const RtpExtension& extension) {
return extension.uri == RtpExtension::kVideoRotationUri;
}) == config_->rtp.extensions.end();
video_stream_encoder_->SetSink(this, rotation_applied);
}
void VideoSendStreamImpl::RegisterProcessThread(
ProcessThread* module_process_thread) {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
RTC_DCHECK(!module_process_thread_);
module_process_thread_ = module_process_thread;
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
module_process_thread_->RegisterModule(rtp_rtcp, RTC_FROM_HERE);
}
void VideoSendStreamImpl::DeRegisterProcessThread() {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
module_process_thread_->DeRegisterModule(rtp_rtcp);
}
VideoSendStreamImpl::~VideoSendStreamImpl() {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK(!payload_router_.IsActive())
<< "VideoSendStreamImpl::Stop not called";
LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString();
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
transport_->packet_router()->RemoveSendRtpModule(rtp_rtcp);
delete rtp_rtcp;
}
}
bool VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
// Runs on a network thread.
RTC_DCHECK(!worker_queue_->IsCurrent());
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
rtp_rtcp->IncomingRtcpPacket(packet, length);
return true;
}
void VideoSendStreamImpl::Start() {
RTC_DCHECK_RUN_ON(worker_queue_);
LOG(LS_INFO) << "VideoSendStream::Start";
if (payload_router_.IsActive())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
payload_router_.SetActive(true);
bitrate_allocator_->AddObserver(
this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_,
max_padding_bitrate_, !config_->suspend_below_min_bitrate,
config_->track_id);
// 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_);
LOG(LS_INFO) << "VideoSendStream::Stop";
if (!payload_router_.IsActive())
return;
TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
payload_router_.SetActive(false);
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) {
LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out.";
bitrate_allocator_->RemoveObserver(this);
}
}
void VideoSendStreamImpl::OnBitrateAllocationUpdated(
const BitrateAllocation& allocation) {
payload_router_.OnBitrateAllocationUpdated(allocation);
}
void VideoSendStreamImpl::SignalEncoderActive() {
RTC_DCHECK_RUN_ON(worker_queue_);
LOG(LS_INFO) << "SignalEncoderActive, Encoder is active.";
bitrate_allocator_->AddObserver(
this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_,
max_padding_bitrate_, !config_->suspend_below_min_bitrate,
config_->track_id);
}
void VideoSendStreamImpl::OnEncoderConfigurationChanged(
std::vector<VideoStream> streams,
int min_transmit_bitrate_bps) {
if (!worker_queue_->IsCurrent()) {
worker_queue_->PostTask(
std::unique_ptr<rtc::QueuedTask>(new EncoderReconfiguredTask(
weak_ptr_, 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_);
const int kEncoderMinBitrateBps = 30000;
encoder_min_bitrate_bps_ =
std::max(streams[0].min_bitrate_bps, kEncoderMinBitrateBps);
encoder_max_bitrate_bps_ = 0;
for (const auto& stream : streams)
encoder_max_bitrate_bps_ += stream.max_bitrate_bps;
max_padding_bitrate_ = CalculateMaxPadBitrateBps(
streams, min_transmit_bitrate_bps, config_->suspend_below_min_bitrate);
// Clear stats for disabled layers.
for (size_t i = streams.size(); i < config_->rtp.ssrcs.size(); ++i) {
stats_proxy_->OnInactiveSsrc(config_->rtp.ssrcs[i]);
}
size_t number_of_temporal_layers =
streams.back().temporal_layer_thresholds_bps.size() + 1;
protection_bitrate_calculator_.SetEncodingData(
streams[0].width, streams[0].height, number_of_temporal_layers,
config_->rtp.max_packet_size);
if (payload_router_.IsActive()) {
// The send stream is started already. Update the allocator with new bitrate
// limits.
bitrate_allocator_->AddObserver(
this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_,
max_padding_bitrate_, !config_->suspend_below_min_bitrate,
config_->track_id);
}
}
EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation) {
// Encoded is called on whatever thread the real encoder implementation run
// on. In the case of hardware encoders, there might be several encoders
// running in parallel on different threads.
size_t simulcast_idx = 0;
if (codec_specific_info->codecType == kVideoCodecVP8) {
simulcast_idx = codec_specific_info->codecSpecific.VP8.simulcastIdx;
}
if (config_->post_encode_callback) {
config_->post_encode_callback->EncodedFrameCallback(EncodedFrame(
encoded_image._buffer, encoded_image._length, encoded_image._frameType,
simulcast_idx, encoded_image._timeStamp));
}
{
rtc::CritScope lock(&encoder_activity_crit_sect_);
if (check_encoder_activity_task_)
check_encoder_activity_task_->UpdateEncoderActivity();
}
protection_bitrate_calculator_.UpdateWithEncodedData(encoded_image);
EncodedImageCallback::Result result = payload_router_.OnEncodedImage(
encoded_image, codec_specific_info, fragmentation);
RTC_DCHECK(codec_specific_info);
int layer = codec_specific_info->codecType == kVideoCodecVP8
? codec_specific_info->codecSpecific.VP8.simulcastIdx
: 0;
{
rtc::CritScope lock(&ivf_writers_crit_);
if (file_writers_[layer].get()) {
bool ok = file_writers_[layer]->WriteFrame(
encoded_image, codec_specific_info->codecType);
RTC_DCHECK(ok);
}
}
return result;
}
void VideoSendStreamImpl::ConfigureProtection() {
RTC_DCHECK_RUN_ON(worker_queue_);
// Consistency of FlexFEC parameters is checked in MaybeCreateFlexfecSender.
const bool flexfec_enabled = (flexfec_sender_ != nullptr);
// Consistency of NACK and RED+ULPFEC parameters is checked in this function.
const bool nack_enabled = config_->rtp.nack.rtp_history_ms > 0;
int red_payload_type = config_->rtp.ulpfec.red_payload_type;
int ulpfec_payload_type = config_->rtp.ulpfec.ulpfec_payload_type;
// Shorthands.
auto IsRedEnabled = [&]() { return red_payload_type >= 0; };
auto DisableRed = [&]() { red_payload_type = -1; };
auto IsUlpfecEnabled = [&]() { return ulpfec_payload_type >= 0; };
auto DisableUlpfec = [&]() { ulpfec_payload_type = -1; };
if (webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment")) {
LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled.";
DisableUlpfec();
}
// If enabled, FlexFEC takes priority over RED+ULPFEC.
if (flexfec_enabled) {
// We can safely disable RED here, because if the remote supports FlexFEC,
// we know that it has a receiver without the RED/RTX workaround.
// See http://crbug.com/webrtc/6650 for more information.
if (IsRedEnabled()) {
LOG(LS_INFO) << "Both FlexFEC and RED are configured. Disabling RED.";
DisableRed();
}
if (IsUlpfecEnabled()) {
LOG(LS_INFO)
<< "Both FlexFEC and ULPFEC are configured. Disabling ULPFEC.";
DisableUlpfec();
}
}
// Payload types without picture ID cannot determine that a stream is complete
// without retransmitting FEC, so using ULPFEC + NACK for H.264 (for instance)
// is a waste of bandwidth since FEC packets still have to be transmitted.
// Note that this is not the case with FlexFEC.
if (nack_enabled && IsUlpfecEnabled() &&
!PayloadTypeSupportsSkippingFecPackets(
config_->encoder_settings.payload_name)) {
LOG(LS_WARNING)
<< "Transmitting payload type without picture ID using "
"NACK+ULPFEC is a waste of bandwidth since ULPFEC packets "
"also have to be retransmitted. Disabling ULPFEC.";
DisableUlpfec();
}
// Verify payload types.
//
// Due to how old receivers work, we need to always send RED if it has been
// negotiated. This is a remnant of an old RED/RTX workaround, see
// https://codereview.webrtc.org/2469093003.
// TODO(brandtr): This change went into M56, so we can remove it in ~M59.
// At that time, we can disable RED whenever ULPFEC is disabled, as there is
// no point in using RED without ULPFEC.
if (IsRedEnabled()) {
RTC_DCHECK_GE(red_payload_type, 0);
RTC_DCHECK_LE(red_payload_type, 127);
}
if (IsUlpfecEnabled()) {
RTC_DCHECK_GE(ulpfec_payload_type, 0);
RTC_DCHECK_LE(ulpfec_payload_type, 127);
if (!IsRedEnabled()) {
LOG(LS_WARNING)
<< "ULPFEC is enabled but RED is disabled. Disabling ULPFEC.";
DisableUlpfec();
}
}
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
// Set NACK.
rtp_rtcp->SetStorePacketsStatus(
true,
kMinSendSidePacketHistorySize);
// Set RED/ULPFEC information.
rtp_rtcp->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
}
// Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic,
// so enable that logic if either of those FEC schemes are enabled.
protection_bitrate_calculator_.SetProtectionMethod(
flexfec_enabled || IsUlpfecEnabled(), nack_enabled);
}
void VideoSendStreamImpl::ConfigureSsrcs() {
RTC_DCHECK_RUN_ON(worker_queue_);
// Configure regular SSRCs.
for (size_t i = 0; i < config_->rtp.ssrcs.size(); ++i) {
uint32_t ssrc = config_->rtp.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
rtp_rtcp->SetSSRC(ssrc);
// Restore RTP state if previous existed.
VideoSendStream::RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtpState(it->second);
}
// Set up RTX if available.
if (config_->rtp.rtx.ssrcs.empty())
return;
// Configure RTX SSRCs.
RTC_DCHECK_EQ(config_->rtp.rtx.ssrcs.size(), config_->rtp.ssrcs.size());
for (size_t i = 0; i < config_->rtp.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = config_->rtp.rtx.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
rtp_rtcp->SetRtxSsrc(ssrc);
VideoSendStream::RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtxState(it->second);
}
// Configure RTX payload types.
RTC_DCHECK_GE(config_->rtp.rtx.payload_type, 0);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetRtxSendPayloadType(config_->rtp.rtx.payload_type,
config_->encoder_settings.payload_type);
rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
}
if (config_->rtp.ulpfec.red_payload_type != -1 &&
config_->rtp.ulpfec.red_rtx_payload_type != -1) {
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetRtxSendPayloadType(config_->rtp.ulpfec.red_rtx_payload_type,
config_->rtp.ulpfec.red_payload_type);
}
}
}
std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const {
RTC_DCHECK_RUN_ON(worker_queue_);
std::map<uint32_t, RtpState> rtp_states;
for (size_t i = 0; i < config_->rtp.ssrcs.size(); ++i) {
uint32_t ssrc = config_->rtp.ssrcs[i];
RTC_DCHECK_EQ(ssrc, rtp_rtcp_modules_[i]->SSRC());
rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtpState();
}
for (size_t i = 0; i < config_->rtp.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = config_->rtp.rtx.ssrcs[i];
rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtxState();
}
if (flexfec_sender_) {
uint32_t ssrc = config_->rtp.flexfec.ssrc;
rtp_states[ssrc] = flexfec_sender_->GetRtpState();
}
return rtp_states;
}
std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates()
const {
RTC_DCHECK_RUN_ON(worker_queue_);
return payload_router_.GetRtpPayloadStates();
}
void VideoSendStreamImpl::SignalNetworkState(NetworkState state) {
RTC_DCHECK_RUN_ON(worker_queue_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_->rtp.rtcp_mode
: RtcpMode::kOff);
}
}
uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt,
int64_t probing_interval_ms) {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK(payload_router_.IsActive())
<< "VideoSendStream::Start has not been called.";
// Substract overhead from bitrate.
rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
uint32_t payload_bitrate_bps = bitrate_bps;
if (send_side_bwe_with_overhead_) {
payload_bitrate_bps -= CalculateOverheadRateBps(
CalculatePacketRate(bitrate_bps,
config_->rtp.max_packet_size +
transport_overhead_bytes_per_packet_),
overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_,
bitrate_bps);
}
// Get the encoder target rate. It is the estimated network rate -
// protection overhead.
encoder_target_rate_bps_ = protection_bitrate_calculator_.SetTargetRates(
payload_bitrate_bps, stats_proxy_->GetSendFrameRate(), fraction_loss,
rtt);
uint32_t encoder_overhead_rate_bps =
send_side_bwe_with_overhead_
? CalculateOverheadRateBps(
CalculatePacketRate(encoder_target_rate_bps_,
config_->rtp.max_packet_size +
transport_overhead_bytes_per_packet_ -
overhead_bytes_per_packet_),
overhead_bytes_per_packet_ +
transport_overhead_bytes_per_packet_,
bitrate_bps - encoder_target_rate_bps_)
: 0;
// When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled
// protection_bitrate includes overhead.
uint32_t protection_bitrate =
bitrate_bps - (encoder_target_rate_bps_ + encoder_overhead_rate_bps);
encoder_target_rate_bps_ =
std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
video_stream_encoder_->OnBitrateUpdated(encoder_target_rate_bps_,
fraction_loss, rtt);
stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
return protection_bitrate;
}
void VideoSendStreamImpl::EnableEncodedFrameRecording(
const std::vector<rtc::PlatformFile>& files,
size_t byte_limit) {
{
rtc::CritScope lock(&ivf_writers_crit_);
for (unsigned int i = 0; i < kMaxSimulcastStreams; ++i) {
if (i < files.size()) {
file_writers_[i] = IvfFileWriter::Wrap(rtc::File(files[i]), byte_limit);
} else {
file_writers_[i].reset();
}
}
}
if (!files.empty()) {
// Make a keyframe appear as early as possible in the logs, to give actually
// decodable output.
video_stream_encoder_->SendKeyFrame();
}
}
int VideoSendStreamImpl::ProtectionRequest(
const FecProtectionParams* delta_params,
const FecProtectionParams* key_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps) {
RTC_DCHECK_RUN_ON(worker_queue_);
*sent_video_rate_bps = 0;
*sent_nack_rate_bps = 0;
*sent_fec_rate_bps = 0;
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
uint32_t not_used = 0;
uint32_t module_video_rate = 0;
uint32_t module_fec_rate = 0;
uint32_t module_nack_rate = 0;
rtp_rtcp->SetFecParameters(*delta_params, *key_params);
rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
&module_nack_rate);
*sent_video_rate_bps += module_video_rate;
*sent_nack_rate_bps += module_nack_rate;
*sent_fec_rate_bps += module_fec_rate;
}
return 0;
}
void VideoSendStreamImpl::OnOverheadChanged(size_t overhead_bytes_per_packet) {
rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
overhead_bytes_per_packet_ = overhead_bytes_per_packet;
}
void VideoSendStreamImpl::SetTransportOverhead(
size_t transport_overhead_bytes_per_packet) {
if (transport_overhead_bytes_per_packet >= static_cast<int>(kPathMTU)) {
LOG(LS_ERROR) << "Transport overhead exceeds size of ethernet frame";
return;
}
transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet;
transport_->send_side_cc()->SetTransportOverhead(
transport_overhead_bytes_per_packet_);
size_t rtp_packet_size =
std::min(config_->rtp.max_packet_size,
kPathMTU - transport_overhead_bytes_per_packet_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
}
}
} // namespace internal
} // namespace webrtc
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