blob: 059a6150af9b9ce3ed4f879ac4c97abf41223c67 [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/video/video_stream_encoder_create.h"
#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"
#include "video/video_send_stream_impl.h"
namespace webrtc {
namespace {
constexpr char kTargetBitrateRtcpFieldTrial[] = "WebRTC-Target-Bitrate-Rtcp";
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.size() > 0) {
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;
}
} // namespace
namespace internal {
VideoSendStream::VideoSendStream(
int num_cpu_cores,
ProcessThread* module_process_thread,
rtc::TaskQueue* worker_queue,
CallStats* 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)
: worker_queue_(worker_queue),
stats_proxy_(Clock::GetRealTimeClock(),
config,
encoder_config.content_type),
config_(std::move(config)),
content_type_(encoder_config.content_type) {
RTC_DCHECK(config_.encoder_settings.encoder_factory);
RTC_DCHECK(config_.encoder_settings.bitrate_allocator_factory);
video_stream_encoder_ = CreateVideoStreamEncoder(num_cpu_cores, &stats_proxy_,
config_.encoder_settings,
config_.pre_encode_callback);
// TODO(srte): Initialization should not be done posted on a task queue.
// Note that the posted task must not outlive this scope since the closure
// references local variables.
worker_queue_->PostTask(rtc::NewClosure(
[this, call_stats, transport, bitrate_allocator, send_delay_stats,
event_log, &suspended_ssrcs, &encoder_config, &suspended_payload_states,
&fec_controller]() {
send_stream_.reset(new VideoSendStreamImpl(
&stats_proxy_, worker_queue_, call_stats, transport,
bitrate_allocator, send_delay_stats, video_stream_encoder_.get(),
event_log, &config_, encoder_config.max_bitrate_bps,
encoder_config.bitrate_priority, suspended_ssrcs,
suspended_payload_states, encoder_config.content_type,
std::move(fec_controller)));
},
[this]() { thread_sync_event_.Set(); }));
// 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 by default, if it
// works well.
if (encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen ||
field_trial::IsEnabled(kTargetBitrateRtcpFieldTrial)) {
video_stream_encoder_->SetBitrateAllocationObserver(send_stream_.get());
}
ReconfigureVideoEncoder(std::move(encoder_config));
}
VideoSendStream::~VideoSendStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(!send_stream_);
}
void VideoSendStream::UpdateActiveSimulcastLayers(
const std::vector<bool> active_layers) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "VideoSendStream::UpdateActiveSimulcastLayers";
VideoSendStreamImpl* send_stream = send_stream_.get();
worker_queue_->PostTask([this, send_stream, active_layers] {
send_stream->UpdateActiveSimulcastLayers(active_layers);
thread_sync_event_.Set();
});
thread_sync_event_.Wait(rtc::Event::kForever);
}
void VideoSendStream::Start() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_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_);
RTC_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 - CalculateMaxHeaderSize(config_.rtp));
}
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();
}
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();
send_stream_->DeRegisterProcessThread();
worker_queue_->PostTask([this, rtp_state_map, payload_state_map]() {
send_stream_->Stop();
*rtp_state_map = send_stream_->GetRtpStates();
*payload_state_map = send_stream_->GetRtpPayloadStates();
send_stream_.reset();
thread_sync_event_.Set();
});
thread_sync_event_.Wait(rtc::Event::kForever);
}
bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// Called on a network thread.
return send_stream_->DeliverRtcp(packet, length);
}
} // namespace internal
} // namespace webrtc