blob: b99b08eefbfeb0275101d55349a70d5c62ce5561 [file] [log] [blame]
/*
* 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 <utility>
#include "api/array_view.h"
#include "api/task_queue/task_queue_base.h"
#include "api/video/video_stream_encoder_settings.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/clock.h"
#include "video/adaptation/overuse_frame_detector.h"
#include "video/frame_cadence_adapter.h"
#include "video/video_stream_encoder.h"
namespace webrtc {
namespace {
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<VideoStreamEncoder> 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,
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,
field_trials),
std::move(encoder_queue), bitrate_allocation_callback_type, field_trials,
encoder_selector);
}
} // namespace
namespace internal {
VideoSendStream::VideoSendStream(
Clock* clock,
int num_cpu_cores,
TaskQueueFactory* task_queue_factory,
TaskQueueBase* network_queue,
RtcpRttStats* call_stats,
RtpTransportControllerSendInterface* transport,
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)
: 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_(CreateVideoStreamEncoder(
clock,
num_cpu_cores,
task_queue_factory,
&stats_proxy_,
config_.encoder_settings,
GetBitrateAllocationCallbackType(config_, field_trials),
field_trials,
config_.encoder_selector)),
encoder_feedback_(
clock,
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)),
send_stream_(clock,
&stats_proxy_,
transport,
bitrate_allocator,
video_stream_encoder_.get(),
&config_,
encoder_config.max_bitrate_bps,
encoder_config.bitrate_priority,
encoder_config.content_type,
rtp_video_sender_,
field_trials) {
RTC_DCHECK(config_.encoder_settings.encoder_factory);
RTC_DCHECK(config_.encoder_settings.bitrate_allocator_factory);
video_stream_encoder_->SetFecControllerOverride(rtp_video_sender_);
ReconfigureVideoEncoder(std::move(encoder_config));
}
VideoSendStream::~VideoSendStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(!running_);
transport_->DestroyRtpVideoSender(rtp_video_sender_);
}
void VideoSendStream::Start() {
const std::vector<bool> active_layers(config_.rtp.ssrcs.size(), true);
StartPerRtpStream(active_layers);
}
void VideoSendStream::StartPerRtpStream(const std::vector<bool> active_layers) {
RTC_DCHECK_RUN_ON(&thread_checker_);
// Keep our `running_` flag expected state in sync with active layers since
// the `send_stream_` will be implicitly stopped/started depending on the
// state of the layers.
bool running = false;
rtc::StringBuilder active_layers_string;
active_layers_string << "{";
for (size_t i = 0; i < active_layers.size(); ++i) {
if (active_layers[i]) {
running = true;
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();
send_stream_.StartPerRtpStream(active_layers);
running_ = running;
}
void VideoSendStream::Stop() {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!running_)
return;
RTC_DLOG(LS_INFO) << "VideoSendStream::Stop";
running_ = false;
send_stream_.Stop();
}
bool VideoSendStream::started() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return running_;
}
void VideoSendStream::AddAdaptationResource(
rtc::scoped_refptr<Resource> resource) {
RTC_DCHECK_RUN_ON(&thread_checker_);
video_stream_encoder_->AddAdaptationResource(resource);
}
std::vector<rtc::scoped_refptr<Resource>>
VideoSendStream::GetAdaptationResources() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return video_stream_encoder_->GetAdaptationResources();
}
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) {
ReconfigureVideoEncoder(std::move(config), nullptr);
}
void VideoSendStream::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 VideoSendStream::GetStats() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return stats_proxy_.GetStats();
}
absl::optional<float> VideoSendStream::GetPacingFactorOverride() const {
return send_stream_.configured_pacing_factor();
}
void VideoSendStream::StopPermanentlyAndGetRtpStates(
VideoSendStream::RtpStateMap* rtp_state_map,
VideoSendStream::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.
send_stream_.Stop();
*rtp_state_map = send_stream_.GetRtpStates();
*payload_state_map = send_stream_.GetRtpPayloadStates();
}
void VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
RTC_DCHECK_RUN_ON(&thread_checker_);
send_stream_.DeliverRtcp(packet, length);
}
void VideoSendStream::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);
}
}
} // namespace internal
} // namespace webrtc