Adding upstream version 126.0.upstream/126.0

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 66 insertions, 84 deletions

diff --git a/third_party/libwebrtc/call/call.cc b/third_party/libwebrtc/call/call.cc
index 63dc370f1a..71511b2559 100644
--- a/third_party/libwebrtc/call/call.cc
+++ b/third_party/libwebrtc/call/call.cc
@@ -70,7 +70,7 @@
 #include "video/send_delay_stats.h"
 #include "video/stats_counter.h"
 #include "video/video_receive_stream2.h"
-#include "video/video_send_stream.h"
+#include "video/video_send_stream_impl.h"
 
 namespace webrtc {
 
@@ -183,10 +183,8 @@ class Call final : public webrtc::Call,
                    public TargetTransferRateObserver,
                    public BitrateAllocator::LimitObserver {
  public:
-  Call(Clock* clock,
-       const CallConfig& config,
-       std::unique_ptr<RtpTransportControllerSendInterface> transport_send,
-       TaskQueueFactory* task_queue_factory);
+  Call(const CallConfig& config,
+       std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
   ~Call() override;
 
   Call(const Call&) = delete;
@@ -345,8 +343,7 @@ class Call final : public webrtc::Call,
   // callbacks have been registered.
   void EnsureStarted() RTC_RUN_ON(worker_thread_);
 
-  Clock* const clock_;
-  TaskQueueFactory* const task_queue_factory_;
+  const Environment env_;
   TaskQueueBase* const worker_thread_;
   TaskQueueBase* const network_thread_;
   const std::unique_ptr<DecodeSynchronizer> decode_sync_;
@@ -356,8 +353,6 @@ class Call final : public webrtc::Call,
   const std::unique_ptr<CallStats> call_stats_;
   const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
   const CallConfig config_ RTC_GUARDED_BY(worker_thread_);
-  // Maps to config_.trials, can be used from any thread via `trials()`.
-  const FieldTrialsView& trials_;
 
   NetworkState audio_network_state_ RTC_GUARDED_BY(worker_thread_);
   NetworkState video_network_state_ RTC_GUARDED_BY(worker_thread_);
@@ -393,9 +388,10 @@ class Call final : public webrtc::Call,
   // should be accessed on the network thread.
   std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_
       RTC_GUARDED_BY(worker_thread_);
-  std::map<uint32_t, VideoSendStream*> video_send_ssrcs_
+  std::map<uint32_t, VideoSendStreamImpl*> video_send_ssrcs_
+      RTC_GUARDED_BY(worker_thread_);
+  std::set<VideoSendStreamImpl*> video_send_streams_
       RTC_GUARDED_BY(worker_thread_);
-  std::set<VideoSendStream*> video_send_streams_ RTC_GUARDED_BY(worker_thread_);
   // True if `video_send_streams_` is empty, false if not. The atomic variable
   // is used to decide UMA send statistics behavior and enables avoiding a
   // PostTask().
@@ -413,8 +409,6 @@ class Call final : public webrtc::Call,
   RtpPayloadStateMap suspended_video_payload_states_
       RTC_GUARDED_BY(worker_thread_);
 
-  webrtc::RtcEventLog* const event_log_;
-
   // TODO(bugs.webrtc.org/11993) ready to move stats access to the network
   // thread.
   ReceiveStats receive_stats_ RTC_GUARDED_BY(worker_thread_);
@@ -460,25 +454,17 @@ class Call final : public webrtc::Call,
 };
 }  // namespace internal
 
-/* Mozilla: Avoid this since it could use GetRealTimeClock().
 std::unique_ptr<Call> Call::Create(const CallConfig& config) {
-  Clock* clock =
-      config.env.has_value() ? &config.env->clock() : Clock::GetRealTimeClock();
-  return Create(config, clock,
-                RtpTransportControllerSendFactory().Create(
-                    config.ExtractTransportConfig(), clock));
-}
- */
+  std::unique_ptr<RtpTransportControllerSendInterface> transport_send;
+  if (config.rtp_transport_controller_send_factory != nullptr) {
+    transport_send = config.rtp_transport_controller_send_factory->Create(
+        config.ExtractTransportConfig());
+  } else {
+    transport_send = RtpTransportControllerSendFactory().Create(
+        config.ExtractTransportConfig());
+  }
 
-std::unique_ptr<Call> Call::Create(
-    const CallConfig& config,
-    Clock* clock,
-    std::unique_ptr<RtpTransportControllerSendInterface>
-        transportControllerSend) {
-  RTC_DCHECK(config.task_queue_factory);
-  return std::make_unique<internal::Call>(clock, config,
-                                          std::move(transportControllerSend),
-                                          config.task_queue_factory);
+  return std::make_unique<internal::Call>(config, std::move(transport_send));
 }
 
 // This method here to avoid subclasses has to implement this method.
@@ -644,47 +630,41 @@ void Call::SendStats::SetMinAllocatableRate(BitrateAllocationLimits limits) {
   min_allocated_send_bitrate_bps_ = limits.min_allocatable_rate.bps();
 }
 
-Call::Call(Clock* clock,
-           const CallConfig& config,
-           std::unique_ptr<RtpTransportControllerSendInterface> transport_send,
-           TaskQueueFactory* task_queue_factory)
-    : clock_(clock),
-      task_queue_factory_(task_queue_factory),
+Call::Call(const CallConfig& config,
+           std::unique_ptr<RtpTransportControllerSendInterface> transport_send)
+    : env_(config.env),
       worker_thread_(GetCurrentTaskQueueOrThread()),
       // If `network_task_queue_` was set to nullptr, network related calls
       // must be made on `worker_thread_` (i.e. they're one and the same).
       network_thread_(config.network_task_queue_ ? config.network_task_queue_
                                                  : worker_thread_),
-      decode_sync_(config.metronome
-                       ? std::make_unique<DecodeSynchronizer>(clock_,
-                                                              config.metronome,
-                                                              worker_thread_)
-                       : nullptr),
+      decode_sync_(
+          config.decode_metronome
+              ? std::make_unique<DecodeSynchronizer>(&env_.clock(),
+                                                     config.decode_metronome,
+                                                     worker_thread_)
+              : nullptr),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
-      call_stats_(new CallStats(clock_, worker_thread_)),
+      call_stats_(new CallStats(&env_.clock(), worker_thread_)),
       bitrate_allocator_(new BitrateAllocator(this)),
       config_(config),
-      trials_(*config.trials),
       audio_network_state_(kNetworkDown),
       video_network_state_(kNetworkDown),
       aggregate_network_up_(false),
-      event_log_(config.event_log),
-      receive_stats_(clock_),
-      send_stats_(clock_),
-      receive_side_cc_(clock,
+      receive_stats_(&env_.clock()),
+      send_stats_(&env_.clock()),
+      receive_side_cc_(&env_.clock(),
                        absl::bind_front(&PacketRouter::SendCombinedRtcpPacket,
                                         transport_send->packet_router()),
                        absl::bind_front(&PacketRouter::SendRemb,
                                         transport_send->packet_router()),
                        /*network_state_estimator=*/nullptr),
       receive_time_calculator_(
-          ReceiveTimeCalculator::CreateFromFieldTrial(*config.trials)),
-      video_send_delay_stats_(new SendDelayStats(clock_)),
-      start_of_call_(clock_->CurrentTime()),
+          ReceiveTimeCalculator::CreateFromFieldTrial(env_.field_trials())),
+      video_send_delay_stats_(new SendDelayStats(&env_.clock())),
+      start_of_call_(env_.clock().CurrentTime()),
       transport_send_ptr_(transport_send.get()),
       transport_send_(std::move(transport_send)) {
-  RTC_DCHECK(config.event_log != nullptr);
-  RTC_DCHECK(config.trials != nullptr);
   RTC_DCHECK(network_thread_);
   RTC_DCHECK(worker_thread_->IsCurrent());
 
@@ -702,7 +682,7 @@ Call::Call(Clock* clock,
   receive_side_cc_periodic_task_ = RepeatingTaskHandle::Start(
       worker_thread_,
       [receive_side_cc] { return receive_side_cc->MaybeProcess(); },
-      TaskQueueBase::DelayPrecision::kLow, clock_);
+      TaskQueueBase::DelayPrecision::kLow, &env_.clock());
 }
 
 Call::~Call() {
@@ -720,7 +700,7 @@ Call::~Call() {
 
   RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Call.LifetimeInSeconds",
-      (clock_->CurrentTime() - start_of_call_).seconds());
+      (env_.clock().CurrentTime() - start_of_call_).seconds());
 }
 
 void Call::EnsureStarted() {
@@ -765,8 +745,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
   }
 
   AudioSendStream* send_stream = new AudioSendStream(
-      clock_, config, config_.audio_state, task_queue_factory_,
-      transport_send_.get(), bitrate_allocator_.get(), event_log_,
+      &env_.clock(), config, config_.audio_state, &env_.task_queue_factory(),
+      transport_send_.get(), bitrate_allocator_.get(), &env_.event_log(),
       call_stats_->AsRtcpRttStats(), suspended_rtp_state, trials());
   RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
              audio_send_ssrcs_.end());
@@ -818,12 +798,12 @@ webrtc::AudioReceiveStreamInterface* Call::CreateAudioReceiveStream(
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK_RUN_ON(worker_thread_);
   EnsureStarted();
-  event_log_->Log(std::make_unique<RtcEventAudioReceiveStreamConfig>(
+  env_.event_log().Log(std::make_unique<RtcEventAudioReceiveStreamConfig>(
       CreateRtcLogStreamConfig(config)));
 
   AudioReceiveStreamImpl* receive_stream = new AudioReceiveStreamImpl(
-      clock_, transport_send_->packet_router(), config_.neteq_factory, config,
-      config_.audio_state, event_log_);
+      &env_.clock(), transport_send_->packet_router(), config_.neteq_factory,
+      config, config_.audio_state, &env_.event_log());
   audio_receive_streams_.insert(receive_stream);
 
   // TODO(bugs.webrtc.org/11993): Make the registration on the network thread
@@ -885,7 +865,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
   video_send_delay_stats_->AddSsrcs(config);
   for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
        ++ssrc_index) {
-    event_log_->Log(std::make_unique<RtcEventVideoSendStreamConfig>(
+    env_.event_log().Log(std::make_unique<RtcEventVideoSendStreamConfig>(
         CreateRtcLogStreamConfig(config, ssrc_index)));
   }
 
@@ -894,13 +874,14 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
   // Copy ssrcs from `config` since `config` is moved.
   std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
 
-  VideoSendStream* send_stream = new VideoSendStream(
-      clock_, num_cpu_cores_, task_queue_factory_, network_thread_,
+  VideoSendStreamImpl* send_stream = new VideoSendStreamImpl(
+      &env_.clock(), num_cpu_cores_, &env_.task_queue_factory(),
       call_stats_->AsRtcpRttStats(), transport_send_.get(),
-      bitrate_allocator_.get(), video_send_delay_stats_.get(), event_log_,
-      std::move(config), std::move(encoder_config), suspended_video_send_ssrcs_,
+      config_.encode_metronome, bitrate_allocator_.get(),
+      video_send_delay_stats_.get(), &env_.event_log(), std::move(config),
+      std::move(encoder_config), suspended_video_send_ssrcs_,
       suspended_video_payload_states_, std::move(fec_controller),
-      *config_.trials);
+      env_.field_trials());
 
   for (uint32_t ssrc : ssrcs) {
     RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
@@ -928,8 +909,8 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
   }
   std::unique_ptr<FecController> fec_controller =
       config_.fec_controller_factory
-          ? config_.fec_controller_factory->CreateFecController()
-          : std::make_unique<FecControllerDefault>(clock_);
+          ? config_.fec_controller_factory->CreateFecController(env_)
+          : std::make_unique<FecControllerDefault>(env_);
   return CreateVideoSendStream(std::move(config), std::move(encoder_config),
                                std::move(fec_controller));
 }
@@ -939,12 +920,12 @@ void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
   RTC_DCHECK(send_stream != nullptr);
   RTC_DCHECK_RUN_ON(worker_thread_);
 
-  VideoSendStream* send_stream_impl =
-      static_cast<VideoSendStream*>(send_stream);
+  VideoSendStreamImpl* send_stream_impl =
+      static_cast<VideoSendStreamImpl*>(send_stream);
 
   auto it = video_send_ssrcs_.begin();
   while (it != video_send_ssrcs_.end()) {
-    if (it->second == static_cast<VideoSendStream*>(send_stream)) {
+    if (it->second == static_cast<VideoSendStreamImpl*>(send_stream)) {
       send_stream_impl = it->second;
       video_send_ssrcs_.erase(it++);
     } else {
@@ -960,8 +941,8 @@ void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
   if (video_send_streams_.empty())
     video_send_streams_empty_.store(true, std::memory_order_relaxed);
 
-  VideoSendStream::RtpStateMap rtp_states;
-  VideoSendStream::RtpPayloadStateMap rtp_payload_states;
+  VideoSendStreamImpl::RtpStateMap rtp_states;
+  VideoSendStreamImpl::RtpPayloadStateMap rtp_payload_states;
   send_stream_impl->StopPermanentlyAndGetRtpStates(&rtp_states,
                                                    &rtp_payload_states);
   for (const auto& kv : rtp_states) {
@@ -984,7 +965,7 @@ webrtc::VideoReceiveStreamInterface* Call::CreateVideoReceiveStream(
 
   EnsureStarted();
 
-  event_log_->Log(std::make_unique<RtcEventVideoReceiveStreamConfig>(
+  env_.event_log().Log(std::make_unique<RtcEventVideoReceiveStreamConfig>(
       CreateRtcLogStreamConfig(configuration)));
 
   // TODO(bugs.webrtc.org/11993): Move the registration between `receive_stream`
@@ -994,10 +975,10 @@ webrtc::VideoReceiveStreamInterface* Call::CreateVideoReceiveStream(
   // TODO(crbug.com/1381982): Re-enable decode synchronizer once the Chromium
   // API has adapted to the new Metronome interface.
   VideoReceiveStream2* receive_stream = new VideoReceiveStream2(
-      task_queue_factory_, this, num_cpu_cores_,
-      transport_send_->packet_router(), std::move(configuration),
-      call_stats_.get(), clock_, std::make_unique<VCMTiming>(clock_, trials()),
-      &nack_periodic_processor_, decode_sync_.get(), event_log_);
+      env_, this, num_cpu_cores_, transport_send_->packet_router(),
+      std::move(configuration), call_stats_.get(),
+      std::make_unique<VCMTiming>(&env_.clock(), trials()),
+      &nack_periodic_processor_, decode_sync_.get());
   // TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
   // thread.
   receive_stream->RegisterWithTransport(&video_receiver_controller_);
@@ -1040,7 +1021,7 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
   // OnRtpPacket until the constructor is finished and the object is
   // in a valid state, since OnRtpPacket runs on the same thread.
   FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl(
-      clock_, std::move(config), &video_receiver_controller_,
+      &env_.clock(), std::move(config), &video_receiver_controller_,
       call_stats_->AsRtcpRttStats());
 
   // TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
@@ -1104,7 +1085,7 @@ Call::Stats Call::GetStats() const {
 }
 
 const FieldTrialsView& Call::trials() const {
-  return trials_;
+  return env_.field_trials();
 }
 
 TaskQueueBase* Call::network_thread() const {
@@ -1244,7 +1225,7 @@ void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   // on a ProcessThread. This is alright as is since we forward the call to
   // implementations that either just do a PostTask or use locking.
   video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
-                                        clock_->CurrentTime());
+                                        env_.clock().CurrentTime());
   transport_send_->OnSentPacket(sent_packet);
 }
 
@@ -1341,7 +1322,7 @@ void Call::DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
     rtcp_delivered = true;
   }
 
-  for (VideoSendStream* stream : video_send_streams_) {
+  for (VideoSendStreamImpl* stream : video_send_streams_) {
     stream->DeliverRtcp(packet.cdata(), packet.size());
     rtcp_delivered = true;
   }
@@ -1352,7 +1333,7 @@ void Call::DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
   }
 
   if (rtcp_delivered) {
-    event_log_->Log(std::make_unique<RtcEventRtcpPacketIncoming>(packet));
+    env_.event_log().Log(std::make_unique<RtcEventRtcpPacketIncoming>(packet));
   }
 }
 
@@ -1368,13 +1349,14 @@ void Call::DeliverRtpPacket(
     // Repair packet_time_us for clock resets by comparing a new read of
     // the same clock (TimeUTCMicros) to a monotonic clock reading.
     packet_time_us = receive_time_calculator_->ReconcileReceiveTimes(
-        packet_time_us, rtc::TimeUTCMicros(), clock_->TimeInMicroseconds());
+        packet_time_us, rtc::TimeUTCMicros(),
+        env_.clock().TimeInMicroseconds());
     packet.set_arrival_time(Timestamp::Micros(packet_time_us));
   }
 
   NotifyBweOfReceivedPacket(packet, media_type);
 
-  event_log_->Log(std::make_unique<RtcEventRtpPacketIncoming>(packet));
+  env_.event_log().Log(std::make_unique<RtcEventRtpPacketIncoming>(packet));
   if (media_type != MediaType::AUDIO && media_type != MediaType::VIDEO) {
     return;
   }