Replace Thread::Invoke with Thread::BlockingCall

BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed

Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}

audio/voip/BUILD.gn

@@ -46,7 +46,6 @@ rtc_library("audio_channel") {
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_rtcp_format",
     "../../rtc_base:criticalsection",
-    "../../rtc_base:location",
     "../../rtc_base:logging",
     "../../rtc_base:refcount",
   ]

audio/voip/audio_channel.cc

@@ -17,7 +17,6 @@
 #include "api/task_queue/task_queue_factory.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 
 namespace webrtc {

call/BUILD.gn

@@ -209,7 +209,6 @@ rtc_library("rtp_sender") {
     "../rtc_base",
     "../rtc_base:checks",
     "../rtc_base:event_tracer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:race_checker",
@@ -318,7 +317,6 @@ rtc_library("call") {
     "../rtc_base:checks",
     "../rtc_base:copy_on_write_buffer",
     "../rtc_base:event_tracer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:rate_limiter",

call/call.cc

@@ -52,7 +52,6 @@
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "modules/video_coding/fec_controller_default.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/system/no_unique_address.h"

call/degraded_call.cc

@@ -14,7 +14,6 @@
 #include <utility>
 
 #include "absl/strings/string_view.h"
-#include "rtc_base/location.h"
 
 namespace webrtc {
 

call/flexfec_receive_stream_impl.cc

@@ -24,7 +24,6 @@
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 #include "system_wrappers/include/clock.h"

call/rtp_video_sender.cc

@@ -28,7 +28,6 @@
 #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/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/trace_event.h"

examples/androidvoip/jni/android_voip_client.cc

@@ -135,7 +135,7 @@ void AndroidVoipClient::Init(
   // Due to consistent thread requirement on
   // modules/audio_device/android/audio_device_template.h,
   // code is invoked in the context of voip_thread_.
-  voip_thread_->Invoke<void>(RTC_FROM_HERE, [this, &config] {
+  voip_thread_->BlockingCall([this, &config] {
     RTC_DCHECK_RUN_ON(voip_thread_.get());
 
     supported_codecs_ = config.encoder_factory->GetSupportedEncoders();
@@ -145,7 +145,7 @@ void AndroidVoipClient::Init(
 }
 
 AndroidVoipClient::~AndroidVoipClient() {
-  voip_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
+  voip_thread_->BlockingCall([this] {
     RTC_DCHECK_RUN_ON(voip_thread_.get());
 
     JavaVM* jvm = nullptr;

modules/pacing/BUILD.gn

@@ -46,7 +46,6 @@ rtc_library("pacing") {
     "../../logging:rtc_event_pacing",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
-    "../../rtc_base:location",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
     "../../rtc_base:rtc_numerics",

modules/video_coding/BUILD.gn

@@ -381,7 +381,6 @@ rtc_library("video_coding_legacy") {
     "../../modules/rtp_rtcp:rtp_video_header",
     "../../rtc_base:checks",
     "../../rtc_base:event_tracer",
-    "../../rtc_base:location",
     "../../rtc_base:logging",
     "../../rtc_base:macromagic",
     "../../rtc_base:one_time_event",
@@ -1220,7 +1219,6 @@ if (rtc_include_tests) {
       "../../rtc_base",
       "../../rtc_base:checks",
       "../../rtc_base:histogram_percentile_counter",
-      "../../rtc_base:location",
       "../../rtc_base:platform_thread",
       "../../rtc_base:random",
       "../../rtc_base:refcount",

modules/video_coding/jitter_buffer_unittest.cc

@@ -21,7 +21,6 @@
 #include "modules/video_coding/media_opt_util.h"
 #include "modules/video_coding/packet.h"
 #include "modules/video_coding/test/stream_generator.h"
-#include "rtc_base/location.h"
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/metrics.h"
 #include "test/gmock.h"

modules/video_coding/video_receiver.cc

@@ -30,7 +30,6 @@
 #include "modules/video_coding/timing/timing.h"
 #include "modules/video_coding/video_coding_impl.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/one_time_event.h"
 #include "rtc_base/trace_event.h"

pc/BUILD.gn

@@ -87,7 +87,6 @@ rtc_source_set("channel") {
     "../rtc_base:checks",
     "../rtc_base:copy_on_write_buffer",
     "../rtc_base:event_tracer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:socket",
@@ -294,7 +293,6 @@ rtc_source_set("jsep_transport_controller") {
     "../rtc_base:checks",
     "../rtc_base:copy_on_write_buffer",
     "../rtc_base:event_tracer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
@@ -482,7 +480,6 @@ rtc_source_set("sctp_transport") {
     "../p2p:rtc_p2p",
     "../rtc_base",
     "../rtc_base:checks",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
@@ -854,7 +851,6 @@ rtc_library("sctp_data_channel") {
     "../media:rtc_media_base",
     "../rtc_base:checks",
     "../rtc_base:copy_on_write_buffer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:rtc_base",
@@ -935,7 +931,6 @@ rtc_source_set("data_channel_controller") {
     "../rtc_base",
     "../rtc_base:checks",
     "../rtc_base:copy_on_write_buffer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
@@ -1021,7 +1016,6 @@ rtc_source_set("rtc_stats_collector") {
     "../rtc_base:checks",
     "../rtc_base:event_tracer",
     "../rtc_base:ip_address",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:network_constants",
     "../rtc_base:refcount",
@@ -1107,7 +1101,6 @@ rtc_source_set("sdp_offer_answer") {
     "../rtc_base",
     "../rtc_base:checks",
     "../rtc_base:event_tracer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:rtc_operations_chain",
@@ -1209,7 +1202,6 @@ rtc_source_set("peer_connection") {
     "../rtc_base:copy_on_write_buffer",
     "../rtc_base:event_tracer",
     "../rtc_base:ip_address",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:network_constants",
@@ -1294,7 +1286,6 @@ rtc_source_set("legacy_stats_collector") {
     "../rtc_base:checks",
     "../rtc_base:event_tracer",
     "../rtc_base:ip_address",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:network_constants",
@@ -1484,7 +1475,6 @@ rtc_source_set("peer_connection_factory") {
     "../pc:session_description",
     "../pc:video_track",
     "../rtc_base:checks",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:rtc_base",
@@ -1561,7 +1551,6 @@ rtc_library("rtp_transceiver") {
     "../api/video:video_bitrate_allocator_factory",
     "../media:rtc_media_base",
     "../rtc_base:checks",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
@@ -1693,7 +1682,6 @@ rtc_library("audio_rtp_receiver") {
     "../media:rtc_media_base",
     "../rtc_base",
     "../rtc_base:checks",
-    "../rtc_base:location",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
     "../rtc_base/system:no_unique_address",
@@ -1732,7 +1720,6 @@ rtc_library("video_rtp_receiver") {
     "../media:rtc_media_base",
     "../rtc_base",
     "../rtc_base:checks",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
@@ -1795,7 +1782,6 @@ rtc_library("video_track") {
     "../media:rtc_media_base",
     "../rtc_base",
     "../rtc_base:checks",
-    "../rtc_base:location",
     "../rtc_base:macromagic",
     "../rtc_base:threading",
     "../rtc_base/system:no_unique_address",
@@ -1877,7 +1863,6 @@ rtc_library("rtp_sender") {
     "../media:rtc_media_base",
     "../rtc_base:checks",
     "../rtc_base:event_tracer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:rtc_base",
@@ -1932,7 +1917,6 @@ rtc_library("dtmf_sender") {
     "../api/task_queue:pending_task_safety_flag",
     "../api/units:time_delta",
     "../rtc_base:checks",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:refcount",

pc/audio_rtp_receiver.cc

@@ -20,7 +20,6 @@
 #include "pc/audio_track.h"
 #include "pc/media_stream_track_proxy.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 
 namespace webrtc {
 
@@ -101,7 +100,7 @@ void AudioRtpReceiver::OnSetVolume(double volume) {
   RTC_DCHECK_LE(volume, 10);
 
   bool track_enabled = track_->internal()->enabled();
-  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
+  worker_thread_->BlockingCall([&]() {
     RTC_DCHECK_RUN_ON(worker_thread_);
     // Update the cached_volume_ even when stopped, to allow clients to set
     // the volume before starting/restarting, eg see crbug.com/1272566.
@@ -168,7 +167,7 @@ void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   bool enabled = track_->internal()->enabled();
   MediaSourceInterface::SourceState state = source_->state();
-  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
+  worker_thread_->BlockingCall([&]() {
     RTC_DCHECK_RUN_ON(worker_thread_);
     RestartMediaChannel_w(std::move(ssrc), enabled, state);
   });

pc/channel.cc

@@ -29,7 +29,6 @@
 #include "pc/rtp_media_utils.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/strings/string_format.h"
@@ -269,8 +268,8 @@ bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) {
   // network thread. At the moment there's a workaround for inconsistent state
   // between the worker and network thread because of this (see
   // OnDemuxerCriteriaUpdatePending elsewhere in this file) and
-  // SetPayloadTypeDemuxingEnabled_w has an Invoke over to the network thread
+  // SetPayloadTypeDemuxingEnabled_w has a BlockingCall over to the network
-  // to apply state updates.
+  // thread to apply state updates.
   RTC_DCHECK_RUN_ON(worker_thread());
   TRACE_EVENT0("webrtc", "BaseChannel::SetPayloadTypeDemuxingEnabled");
   return SetPayloadTypeDemuxingEnabled_w(enabled);
@@ -461,7 +460,7 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w(
   if (update_demuxer)
     media_channel()->OnDemuxerCriteriaUpdatePending();
 
-  bool success = network_thread()->Invoke<bool>(RTC_FROM_HERE, [&]() mutable {
+  bool success = network_thread()->BlockingCall([&]() mutable {
     RTC_DCHECK_RUN_ON(network_thread());
     // NOTE: This doesn't take the BUNDLE case in account meaning the RTP header
     // extension maps are not merged when BUNDLE is enabled. This is fine
@@ -491,8 +490,8 @@ bool BaseChannel::RegisterRtpDemuxerSink_w() {
   media_channel_->OnDemuxerCriteriaUpdatePending();
   // Copy demuxer criteria, since they're a worker-thread variable
   // and we want to pass them to the network thread
-  bool ret = network_thread_->Invoke<bool>(
+  bool ret = network_thread_->BlockingCall(
-      RTC_FROM_HERE, [this, demuxer_criteria = demuxer_criteria_] {
+      [this, demuxer_criteria = demuxer_criteria_] {
         RTC_DCHECK_RUN_ON(network_thread());
         if (!rtp_transport_) {
           // Transport was disconnected before attempting to update the

pc/connection_context.cc

@@ -167,13 +167,13 @@ ConnectionContext::ConnectionContext(
   if (media_engine_) {
     // TODO(tommi): Change VoiceEngine to do ctor time initialization so that
     // this isn't necessary.
-    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] { media_engine_->Init(); });
+    worker_thread_->BlockingCall([&] { media_engine_->Init(); });
   }
 }
 
 ConnectionContext::~ConnectionContext() {
   RTC_DCHECK_RUN_ON(signaling_thread_);
-  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+  worker_thread_->BlockingCall([&] {
     RTC_DCHECK_RUN_ON(worker_thread());
     // While `media_engine_` is const throughout the ConnectionContext's
     // lifetime, it requires destruction to happen on the worker thread. Instead

pc/data_channel_controller.cc

@@ -16,7 +16,6 @@
 #include "api/rtc_error.h"
 #include "pc/peer_connection_internal.h"
 #include "pc/sctp_utils.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 
 namespace webrtc {
@@ -81,7 +80,7 @@ void DataChannelController::DisconnectDataChannel(
 
 void DataChannelController::AddSctpDataStream(int sid) {
   if (data_channel_transport()) {
-    network_thread()->Invoke<void>(RTC_FROM_HERE, [this, sid] {
+    network_thread()->BlockingCall([this, sid] {
       if (data_channel_transport()) {
         data_channel_transport()->OpenChannel(sid);
       }
@@ -91,7 +90,7 @@ void DataChannelController::AddSctpDataStream(int sid) {
 
 void DataChannelController::RemoveSctpDataStream(int sid) {
   if (data_channel_transport()) {
-    network_thread()->Invoke<void>(RTC_FROM_HERE, [this, sid] {
+    network_thread()->BlockingCall([this, sid] {
       if (data_channel_transport()) {
         data_channel_transport()->CloseChannel(sid);
       }
@@ -382,15 +381,14 @@ bool DataChannelController::DataChannelSendData(
     const rtc::CopyOnWriteBuffer& payload,
     cricket::SendDataResult* result) {
   // TODO(bugs.webrtc.org/11547): Expect method to be called on the network
-  // thread instead. Remove the Invoke() below and move assocated state to
+  // thread instead. Remove the BlockingCall() below and move assocated state to
   // the network thread.
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(data_channel_transport());
 
-  RTCError error = network_thread()->Invoke<RTCError>(
+  RTCError error = network_thread()->BlockingCall([this, sid, params, payload] {
-      RTC_FROM_HERE, [this, sid, params, payload] {
+    return data_channel_transport()->SendData(sid, params, payload);
-        return data_channel_transport()->SendData(sid, params, payload);
+  });
-      });
 
   if (error.ok()) {
     *result = cricket::SendDataResult::SDR_SUCCESS;

pc/dtmf_sender.cc

@@ -142,7 +142,7 @@ bool DtmfSender::InsertDtmf(const std::string& tones,
   }
   safety_flag_ = PendingTaskSafetyFlag::Create();
   // Kick off a new DTMF task.
-  QueueInsertDtmf(RTC_FROM_HERE, 1 /*ms*/);
+  QueueInsertDtmf(1 /*ms*/);
   return true;
 }
 
@@ -166,8 +166,7 @@ int DtmfSender::comma_delay() const {
   return comma_delay_;
 }
 
-void DtmfSender::QueueInsertDtmf(const rtc::Location& posted_from,
+void DtmfSender::QueueInsertDtmf(uint32_t delay_ms) {
-                                 uint32_t delay_ms) {
   signaling_thread_->PostDelayedHighPrecisionTask(
       SafeTask(safety_flag_,
                [this] {
@@ -232,7 +231,7 @@ void DtmfSender::DoInsertDtmf() {
   tones_.erase(0, first_tone_pos + 1);
 
   // Continue with the next tone.
-  QueueInsertDtmf(RTC_FROM_HERE, tone_gap);
+  QueueInsertDtmf(tone_gap);
 }
 
 void DtmfSender::StopSending() {

pc/dtmf_sender.h

@@ -21,7 +21,6 @@
 #include "api/task_queue/pending_task_safety_flag.h"
 #include "api/task_queue/task_queue_base.h"
 #include "pc/proxy.h"
-#include "rtc_base/location.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/thread_annotations.h"
 
@@ -77,8 +76,7 @@ class DtmfSender : public DtmfSenderInterface {
  private:
   DtmfSender();
 
-  void QueueInsertDtmf(const rtc::Location& posted_from, uint32_t delay_ms)
+  void QueueInsertDtmf(uint32_t delay_ms) RTC_RUN_ON(signaling_thread_);
-      RTC_RUN_ON(signaling_thread_);
 
   // The DTMF sending task.
   void DoInsertDtmf() RTC_RUN_ON(signaling_thread_);

pc/jsep_transport_controller.cc

@@ -29,7 +29,6 @@
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/port.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/trace_event.h"
@@ -77,8 +76,8 @@ RTCError JsepTransportController::SetLocalDescription(
     const cricket::SessionDescription* description) {
   TRACE_EVENT0("webrtc", "JsepTransportController::SetLocalDescription");
   if (!network_thread_->IsCurrent()) {
-    return network_thread_->Invoke<RTCError>(
+    return network_thread_->BlockingCall(
-        RTC_FROM_HERE, [=] { return SetLocalDescription(type, description); });
+        [=] { return SetLocalDescription(type, description); });
   }
 
   RTC_DCHECK_RUN_ON(network_thread_);
@@ -98,8 +97,8 @@ RTCError JsepTransportController::SetRemoteDescription(
     const cricket::SessionDescription* description) {
   TRACE_EVENT0("webrtc", "JsepTransportController::SetRemoteDescription");
   if (!network_thread_->IsCurrent()) {
-    return network_thread_->Invoke<RTCError>(
+    return network_thread_->BlockingCall(
-        RTC_FROM_HERE, [=] { return SetRemoteDescription(type, description); });
+        [=] { return SetRemoteDescription(type, description); });
   }
 
   RTC_DCHECK_RUN_ON(network_thread_);
@@ -199,8 +198,7 @@ absl::optional<rtc::SSLRole> JsepTransportController::GetDtlsRole(
   // thread during negotiations, potentially multiple times.
   // WebRtcSessionDescriptionFactory::InternalCreateAnswer is one example.
   if (!network_thread_->IsCurrent()) {
-    return network_thread_->Invoke<absl::optional<rtc::SSLRole>>(
+    return network_thread_->BlockingCall([&] { return GetDtlsRole(mid); });
-        RTC_FROM_HERE, [&] { return GetDtlsRole(mid); });
   }
 
   RTC_DCHECK_RUN_ON(network_thread_);
@@ -215,8 +213,8 @@ absl::optional<rtc::SSLRole> JsepTransportController::GetDtlsRole(
 bool JsepTransportController::SetLocalCertificate(
     const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
   if (!network_thread_->IsCurrent()) {
-    return network_thread_->Invoke<bool>(
+    return network_thread_->BlockingCall(
-        RTC_FROM_HERE, [&] { return SetLocalCertificate(certificate); });
+        [&] { return SetLocalCertificate(certificate); });
   }
 
   RTC_DCHECK_RUN_ON(network_thread_);
@@ -274,8 +272,7 @@ JsepTransportController::GetRemoteSSLCertChain(
 
 void JsepTransportController::MaybeStartGathering() {
   if (!network_thread_->IsCurrent()) {
-    network_thread_->Invoke<void>(RTC_FROM_HERE,
+    network_thread_->BlockingCall([&] { MaybeStartGathering(); });
-                                  [&] { MaybeStartGathering(); });
     return;
   }
 
@@ -301,8 +298,8 @@ RTCError JsepTransportController::AddRemoteCandidates(
 RTCError JsepTransportController::RemoveRemoteCandidates(
     const cricket::Candidates& candidates) {
   if (!network_thread_->IsCurrent()) {
-    return network_thread_->Invoke<RTCError>(
+    return network_thread_->BlockingCall(
-        RTC_FROM_HERE, [&] { return RemoveRemoteCandidates(candidates); });
+        [&] { return RemoveRemoteCandidates(candidates); });
   }
 
   RTC_DCHECK_RUN_ON(network_thread_);
@@ -361,9 +358,8 @@ bool JsepTransportController::GetStats(const std::string& transport_name,
 void JsepTransportController::SetActiveResetSrtpParams(
     bool active_reset_srtp_params) {
   if (!network_thread_->IsCurrent()) {
-    network_thread_->Invoke<void>(RTC_FROM_HERE, [=] {
+    network_thread_->BlockingCall(
-      SetActiveResetSrtpParams(active_reset_srtp_params);
+        [=] { SetActiveResetSrtpParams(active_reset_srtp_params); });
-    });
     return;
   }
   RTC_DCHECK_RUN_ON(network_thread_);
@@ -378,8 +374,7 @@ void JsepTransportController::SetActiveResetSrtpParams(
 
 RTCError JsepTransportController::RollbackTransports() {
   if (!network_thread_->IsCurrent()) {
-    return network_thread_->Invoke<RTCError>(
+    return network_thread_->BlockingCall([=] { return RollbackTransports(); });
-        RTC_FROM_HERE, [=] { return RollbackTransports(); });
   }
   RTC_DCHECK_RUN_ON(network_thread_);
   bundles_.Rollback();

pc/legacy_stats_collector.cc

@@ -46,7 +46,6 @@
 #include "pc/transport_stats.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/ip_address.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/socket_address.h"
@@ -861,9 +860,9 @@ std::map<std::string, std::string> LegacyStatsCollector::ExtractSessionInfo() {
 
   SessionStats stats;
   auto transceivers = pc_->GetTransceiversInternal();
-  pc_->network_thread()->Invoke<void>(
+  pc_->network_thread()->BlockingCall(
-      RTC_FROM_HERE, [&, sctp_transport_name = pc_->sctp_transport_name(),
+      [&, sctp_transport_name = pc_->sctp_transport_name(),
-                      sctp_mid = pc_->sctp_mid()]() mutable {
+       sctp_mid = pc_->sctp_mid()]() mutable {
         stats = ExtractSessionInfo_n(
             transceivers, std::move(sctp_transport_name), std::move(sctp_mid));
       });
@@ -1049,7 +1048,7 @@ void LegacyStatsCollector::ExtractBweInfo() {
   }
 
   if (!video_media_channels.empty()) {
-    pc_->worker_thread()->Invoke<void>(RTC_FROM_HERE, [&] {
+    pc_->worker_thread()->BlockingCall([&] {
       for (const auto& channel : video_media_channels) {
         channel->FillBitrateInfo(&bwe_info);
       }
@@ -1200,7 +1199,7 @@ void LegacyStatsCollector::ExtractMediaInfo(
     }
   }
 
-  pc_->worker_thread()->Invoke<void>(RTC_FROM_HERE, [&] {
+  pc_->worker_thread()->BlockingCall([&] {
     rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
     // Populate `receiver_track_id_by_ssrc` for the gatherers.
     int i = 0;

pc/peer_connection.cc

@@ -52,7 +52,6 @@
 #include "pc/webrtc_session_description_factory.h"
 #include "rtc_base/helpers.h"
 #include "rtc_base/ip_address.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/net_helper.h"
 #include "rtc_base/network.h"
@@ -525,7 +524,7 @@ PeerConnection::PeerConnection(
       data_channel_controller_(this),
       message_handler_(signaling_thread()),
       weak_factory_(this) {
-  worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
+  worker_thread()->BlockingCall([this] {
     RTC_DCHECK_RUN_ON(worker_thread());
     worker_thread_safety_ = PendingTaskSafetyFlag::Create();
     if (!call_)
@@ -569,7 +568,7 @@ PeerConnection::~PeerConnection() {
   // port_allocator_ and transport_controller_ live on the network thread and
   // should be destroyed there.
   transport_controller_copy_ = nullptr;
-  network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
+  network_thread()->BlockingCall([this] {
     RTC_DCHECK_RUN_ON(network_thread());
     TeardownDataChannelTransport_n();
     transport_controller_.reset();
@@ -579,7 +578,7 @@ PeerConnection::~PeerConnection() {
   });
 
   // call_ and event_log_ must be destroyed on the worker thread.
-  worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
+  worker_thread()->BlockingCall([this] {
     RTC_DCHECK_RUN_ON(worker_thread());
     worker_thread_safety_->SetNotAlive();
     call_.reset();
@@ -618,7 +617,7 @@ RTCError PeerConnection::Initialize(
 
   // Network thread initialization.
   transport_controller_copy_ =
-      network_thread()->Invoke<JsepTransportController*>(RTC_FROM_HERE, [&] {
+      network_thread()->BlockingCall([&] {
         RTC_DCHECK_RUN_ON(network_thread());
         network_thread_safety_ = PendingTaskSafetyFlag::Create();
         InitializePortAllocatorResult pa_result = InitializePortAllocator_n(
@@ -940,13 +939,12 @@ RtpTransportInternal* PeerConnection::GetRtpTransport(const std::string& mid) {
   // TODO(bugs.webrtc.org/9987): Avoid the thread jump.
   // This might be done by caching the value on the signaling thread.
   RTC_DCHECK_RUN_ON(signaling_thread());
-  return network_thread()->Invoke<RtpTransportInternal*>(
+  return network_thread()->BlockingCall([this, &mid] {
-      RTC_FROM_HERE, [this, &mid] {
+    RTC_DCHECK_RUN_ON(network_thread());
-        RTC_DCHECK_RUN_ON(network_thread());
+    auto rtp_transport = transport_controller_->GetRtpTransport(mid);
-        auto rtp_transport = transport_controller_->GetRtpTransport(mid);
+    RTC_DCHECK(rtp_transport);
-        RTC_DCHECK(rtp_transport);
+    return rtp_transport;
-        return rtp_transport;
+  });
-      });
 }
 
 RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
@@ -1556,8 +1554,7 @@ RTCError PeerConnection::SetConfiguration(
 
   // Apply part of the configuration on the network thread.  In theory this
   // shouldn't fail.
-  if (!network_thread()->Invoke<bool>(
+  if (!network_thread()->BlockingCall(
-          RTC_FROM_HERE,
           [this, needs_ice_restart, &ice_config, &stun_servers, &turn_servers,
            &modified_config, has_local_description] {
             RTC_DCHECK_RUN_ON(network_thread());
@@ -1583,8 +1580,8 @@ RTCError PeerConnection::SetConfiguration(
 
   if (configuration_.active_reset_srtp_params !=
       modified_config.active_reset_srtp_params) {
-    // TODO(tommi): merge invokes
+    // TODO(tommi): merge BlockingCalls
-    network_thread()->Invoke<void>(RTC_FROM_HERE, [this, &modified_config] {
+    network_thread()->BlockingCall([this, &modified_config] {
       RTC_DCHECK_RUN_ON(network_thread());
       transport_controller_->SetActiveResetSrtpParams(
           modified_config.active_reset_srtp_params);
@@ -1603,8 +1600,7 @@ RTCError PeerConnection::SetConfiguration(
             video_channel->media_channel()));
     }
 
-    worker_thread()->Invoke<void>(
+    worker_thread()->BlockingCall(
-        RTC_FROM_HERE,
         [channels = std::move(channels),
          allow_codec_switching = *modified_config.allow_codec_switching]() {
           for (auto* ch : channels)
@@ -1643,8 +1639,7 @@ bool PeerConnection::RemoveIceCandidates(
 
 RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
   if (!worker_thread()->IsCurrent()) {
-    return worker_thread()->Invoke<RTCError>(
+    return worker_thread()->BlockingCall([&]() { return SetBitrate(bitrate); });
-        RTC_FROM_HERE, [&]() { return SetBitrate(bitrate); });
   }
   RTC_DCHECK_RUN_ON(worker_thread());
 
@@ -1685,8 +1680,8 @@ RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
 
 void PeerConnection::SetAudioPlayout(bool playout) {
   if (!worker_thread()->IsCurrent()) {
-    worker_thread()->Invoke<void>(
+    worker_thread()->BlockingCall(
-        RTC_FROM_HERE, [this, playout] { SetAudioPlayout(playout); });
+        [this, playout] { SetAudioPlayout(playout); });
     return;
   }
   auto audio_state = context_->media_engine()->voice().GetAudioState();
@@ -1695,8 +1690,8 @@ void PeerConnection::SetAudioPlayout(bool playout) {
 
 void PeerConnection::SetAudioRecording(bool recording) {
   if (!worker_thread()->IsCurrent()) {
-    worker_thread()->Invoke<void>(
+    worker_thread()->BlockingCall(
-        RTC_FROM_HERE, [this, recording] { SetAudioRecording(recording); });
+        [this, recording] { SetAudioRecording(recording); });
     return;
   }
   auto audio_state = context_->media_engine()->voice().GetAudioState();
@@ -1706,9 +1701,8 @@ void PeerConnection::SetAudioRecording(bool recording) {
 void PeerConnection::AddAdaptationResource(
     rtc::scoped_refptr<Resource> resource) {
   if (!worker_thread()->IsCurrent()) {
-    return worker_thread()->Invoke<void>(RTC_FROM_HERE, [this, resource]() {
+    return worker_thread()->BlockingCall(
-      return AddAdaptationResource(resource);
+        [this, resource]() { return AddAdaptationResource(resource); });
-    });
   }
   RTC_DCHECK_RUN_ON(worker_thread());
   if (!call_) {
@@ -1724,8 +1718,7 @@ bool PeerConnection::ConfiguredForMedia() const {
 
 bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
                                       int64_t output_period_ms) {
-  return worker_thread()->Invoke<bool>(
+  return worker_thread()->BlockingCall(
-      RTC_FROM_HERE,
       [this, output = std::move(output), output_period_ms]() mutable {
         return StartRtcEventLog_w(std::move(output), output_period_ms);
       });
@@ -1741,7 +1734,7 @@ bool PeerConnection::StartRtcEventLog(
 }
 
 void PeerConnection::StopRtcEventLog() {
-  worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] { StopRtcEventLog_w(); });
+  worker_thread()->BlockingCall([this] { StopRtcEventLog_w(); });
 }
 
 rtc::scoped_refptr<DtlsTransportInterface>
@@ -1755,11 +1748,10 @@ PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   // TODO(bugs.webrtc.org/9987): Avoid the thread jump.
   // This might be done by caching the value on the signaling thread.
-  return network_thread()->Invoke<rtc::scoped_refptr<DtlsTransport>>(
+  return network_thread()->BlockingCall([this, mid]() {
-      RTC_FROM_HERE, [this, mid]() {
+    RTC_DCHECK_RUN_ON(network_thread());
-        RTC_DCHECK_RUN_ON(network_thread());
+    return transport_controller_->LookupDtlsTransportByMid(mid);
-        return transport_controller_->LookupDtlsTransportByMid(mid);
+  });
-      });
 }
 
 rtc::scoped_refptr<SctpTransportInterface> PeerConnection::GetSctpTransport()
@@ -1856,7 +1848,7 @@ void PeerConnection::Close() {
     rtp_manager_->Close();
   }
 
-  network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
+  network_thread()->BlockingCall([this] {
     // Data channels will already have been unset via the DestroyAllChannels()
     // call above, which triggers a call to TeardownDataChannelTransport_n().
     // TODO(tommi): ^^ That's not exactly optimal since this is yet another
@@ -1870,7 +1862,7 @@ void PeerConnection::Close() {
     }
   });
 
-  worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
+  worker_thread()->BlockingCall([this] {
     RTC_DCHECK_RUN_ON(worker_thread());
     worker_thread_safety_->SetNotAlive();
     call_.reset();
@@ -2225,11 +2217,10 @@ bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
 
   absl::optional<rtc::SSLRole> dtls_role;
   if (sctp_mid_s_) {
-    dtls_role = network_thread()->Invoke<absl::optional<rtc::SSLRole>>(
+    dtls_role = network_thread()->BlockingCall([this] {
-        RTC_FROM_HERE, [this] {
+      RTC_DCHECK_RUN_ON(network_thread());
-          RTC_DCHECK_RUN_ON(network_thread());
+      return transport_controller_->GetDtlsRole(*sctp_mid_n_);
-          return transport_controller_->GetDtlsRole(*sctp_mid_n_);
+    });
-        });
     if (!dtls_role && sdp_handler_->is_caller().has_value()) {
       // This works fine if we are the offerer, but can be a mistake if
       // we are the answerer and the remote offer is ACTIVE. In that
@@ -2259,11 +2250,10 @@ bool PeerConnection::GetSslRole(const std::string& content_name,
     return false;
   }
 
-  auto dtls_role = network_thread()->Invoke<absl::optional<rtc::SSLRole>>(
+  auto dtls_role = network_thread()->BlockingCall([this, content_name]() {
-      RTC_FROM_HERE, [this, content_name]() {
+    RTC_DCHECK_RUN_ON(network_thread());
-        RTC_DCHECK_RUN_ON(network_thread());
+    return transport_controller_->GetDtlsRole(content_name);
-        return transport_controller_->GetDtlsRole(content_name);
+  });
-      });
   if (dtls_role) {
     *role = *dtls_role;
     return true;
@@ -2360,7 +2350,7 @@ bool PeerConnection::IceRestartPending(const std::string& content_name) const {
 }
 
 bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
-  return network_thread()->Invoke<bool>(RTC_FROM_HERE, [this, &content_name] {
+  return network_thread()->BlockingCall([this, &content_name] {
     RTC_DCHECK_RUN_ON(network_thread());
     return transport_controller_->NeedsIceRestart(content_name);
   });
@@ -2489,8 +2479,7 @@ bool PeerConnection::GetLocalCandidateMediaIndex(
 
 Call::Stats PeerConnection::GetCallStats() {
   if (!worker_thread()->IsCurrent()) {
-    return worker_thread()->Invoke<Call::Stats>(
+    return worker_thread()->BlockingCall([this] { return GetCallStats(); });
-        RTC_FROM_HERE, [this] { return GetCallStats(); });
   }
   RTC_DCHECK_RUN_ON(worker_thread());
   rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;

pc/peer_connection_factory.cc

@@ -46,7 +46,6 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/experiments/field_trial_parser.h"
 #include "rtc_base/experiments/field_trial_units.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/rtc_certificate_generator.h"
@@ -60,12 +59,9 @@ CreateModularPeerConnectionFactory(
   // The PeerConnectionFactory must be created on the signaling thread.
   if (dependencies.signaling_thread &&
       !dependencies.signaling_thread->IsCurrent()) {
-    return dependencies.signaling_thread
+    return dependencies.signaling_thread->BlockingCall([&dependencies] {
-        ->Invoke<rtc::scoped_refptr<PeerConnectionFactoryInterface>>(
+      return CreateModularPeerConnectionFactory(std::move(dependencies));
-            RTC_FROM_HERE, [&dependencies] {
+    });
-              return CreateModularPeerConnectionFactory(
-                  std::move(dependencies));
-            });
   }
 
   auto pc_factory = PeerConnectionFactory::Create(std::move(dependencies));
@@ -238,13 +234,12 @@ PeerConnectionFactory::CreatePeerConnectionOrError(
   dependencies.allocator->SetVpnList(configuration.vpn_list);
 
   std::unique_ptr<RtcEventLog> event_log =
-      worker_thread()->Invoke<std::unique_ptr<RtcEventLog>>(
+      worker_thread()->BlockingCall([this] { return CreateRtcEventLog_w(); });
-          RTC_FROM_HERE, [this] { return CreateRtcEventLog_w(); });
 
   const FieldTrialsView* trials =
       dependencies.trials ? dependencies.trials.get() : &field_trials();
-  std::unique_ptr<Call> call = worker_thread()->Invoke<std::unique_ptr<Call>>(
+  std::unique_ptr<Call> call =
-      RTC_FROM_HERE, [this, &event_log, trials] {
+      worker_thread()->BlockingCall([this, &event_log, trials] {
         return CreateCall_w(event_log.get(), *trials);
       });
 

pc/rtc_stats_collector.cc

@@ -50,7 +50,6 @@
 #include "pc/webrtc_sdp.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/ip_address.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/network_constants.h"
 #include "rtc_base/rtc_certificate.h"
@@ -2316,9 +2315,9 @@ void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() {
   auto transceivers = pc_->GetTransceiversInternal();
 
   // TODO(tommi): See if we can avoid synchronously blocking the signaling
-  // thread while we do this (or avoid the Invoke at all).
+  // thread while we do this (or avoid the BlockingCall at all).
-  network_thread_->Invoke<void>(RTC_FROM_HERE, [this, &transceivers,
+  network_thread_->BlockingCall([this, &transceivers, &voice_stats,
-                                                &voice_stats, &video_stats] {
+                                 &video_stats] {
     rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
     for (const auto& transceiver_proxy : transceivers) {
@@ -2363,7 +2362,7 @@ void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() {
   // well as GetCallStats(). At the same time we construct the
   // TrackMediaInfoMaps, which also needs info from the worker thread. This
   // minimizes the number of thread jumps.
-  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+  worker_thread_->BlockingCall([&] {
     rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
     for (auto& pair : voice_stats) {

pc/rtp_sender.cc

@@ -24,7 +24,6 @@
 #include "pc/legacy_stats_collector_interface.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/helpers.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/trace_event.h"
 
@@ -125,9 +124,8 @@ void RtpSenderBase::SetFrameEncryptor(
   frame_encryptor_ = std::move(frame_encryptor);
   // Special Case: Set the frame encryptor to any value on any existing channel.
   if (media_channel_ && ssrc_ && !stopped_) {
-    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+    worker_thread_->BlockingCall(
-      media_channel_->SetFrameEncryptor(ssrc_, frame_encryptor_);
+        [&] { media_channel_->SetFrameEncryptor(ssrc_, frame_encryptor_); });
-    });
   }
 }
 
@@ -142,7 +140,7 @@ void RtpSenderBase::SetEncoderSelector(
 void RtpSenderBase::SetEncoderSelectorOnChannel() {
   RTC_DCHECK_RUN_ON(signaling_thread_);
   if (media_channel_ && ssrc_ && !stopped_) {
-    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+    worker_thread_->BlockingCall([&] {
       media_channel_->SetEncoderSelector(ssrc_, encoder_selector_.get());
     });
   }
@@ -162,7 +160,7 @@ RtpParameters RtpSenderBase::GetParametersInternal() const {
   if (!media_channel_ || !ssrc_) {
     return init_parameters_;
   }
-  return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
+  return worker_thread_->BlockingCall([&] {
     RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_);
     RemoveEncodingLayers(disabled_rids_, &result.encodings);
     return result;
@@ -177,7 +175,7 @@ RtpParameters RtpSenderBase::GetParametersInternalWithAllLayers() const {
   if (!media_channel_ || !ssrc_) {
     return init_parameters_;
   }
-  return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
+  return worker_thread_->BlockingCall([&] {
     RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_);
     return result;
   });
@@ -208,7 +206,7 @@ RTCError RtpSenderBase::SetParametersInternal(const RtpParameters& parameters) {
     }
     return result;
   }
-  return worker_thread_->Invoke<RTCError>(RTC_FROM_HERE, [&] {
+  return worker_thread_->BlockingCall([&] {
     RtpParameters rtp_parameters = parameters;
     if (!disabled_rids_.empty()) {
       // Need to add the inactive layers.
@@ -239,7 +237,7 @@ RTCError RtpSenderBase::SetParametersInternalWithAllLayers(
     }
     return result;
   }
-  return worker_thread_->Invoke<RTCError>(RTC_FROM_HERE, [&] {
+  return worker_thread_->BlockingCall([&] {
     RtpParameters rtp_parameters = parameters;
     return media_channel_->SetRtpSendParameters(ssrc_, rtp_parameters);
   });
@@ -345,7 +343,7 @@ void RtpSenderBase::SetSsrc(uint32_t ssrc) {
   }
   if (!init_parameters_.encodings.empty() ||
       init_parameters_.degradation_preference.has_value()) {
-    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+    worker_thread_->BlockingCall([&] {
       RTC_DCHECK(media_channel_);
       // Get the current parameters, which are constructed from the SDP.
       // The number of layers in the SDP is currently authoritative to support
@@ -454,7 +452,7 @@ void RtpSenderBase::SetEncoderToPacketizerFrameTransformer(
   RTC_DCHECK_RUN_ON(signaling_thread_);
   frame_transformer_ = std::move(frame_transformer);
   if (media_channel_ && ssrc_ && !stopped_) {
-    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+    worker_thread_->BlockingCall([&] {
       media_channel_->SetEncoderToPacketizerFrameTransformer(
           ssrc_, frame_transformer_);
     });
@@ -526,8 +524,8 @@ bool AudioRtpSender::CanInsertDtmf() {
     RTC_LOG(LS_ERROR) << "CanInsertDtmf: Sender does not have SSRC.";
     return false;
   }
-  return worker_thread_->Invoke<bool>(
+  return worker_thread_->BlockingCall(
-      RTC_FROM_HERE, [&] { return voice_media_channel()->CanInsertDtmf(); });
+      [&] { return voice_media_channel()->CanInsertDtmf(); });
 }
 
 bool AudioRtpSender::InsertDtmf(int code, int duration) {
@@ -539,9 +537,8 @@ bool AudioRtpSender::InsertDtmf(int code, int duration) {
     RTC_LOG(LS_ERROR) << "InsertDtmf: Sender does not have SSRC.";
     return false;
   }
-  bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
+  bool success = worker_thread_->BlockingCall(
-    return voice_media_channel()->InsertDtmf(ssrc_, code, duration);
+      [&] { return voice_media_channel()->InsertDtmf(ssrc_, code, duration); });
-  });
   if (!success) {
     RTC_LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
   }
@@ -610,7 +607,7 @@ void AudioRtpSender::SetSend() {
   // `track_->enabled()` hops to the signaling thread, so call it before we hop
   // to the worker thread or else it will deadlock.
   bool track_enabled = track_->enabled();
-  bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
+  bool success = worker_thread_->BlockingCall([&] {
     return voice_media_channel()->SetAudioSend(ssrc_, track_enabled, &options,
                                                sink_adapter_.get());
   });
@@ -628,7 +625,7 @@ void AudioRtpSender::ClearSend() {
     return;
   }
   cricket::AudioOptions options;
-  bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
+  bool success = worker_thread_->BlockingCall([&] {
     return voice_media_channel()->SetAudioSend(ssrc_, false, &options, nullptr);
   });
   if (!success) {
@@ -704,7 +701,7 @@ void VideoRtpSender::SetSend() {
       options.is_screencast = true;
       break;
   }
-  bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
+  bool success = worker_thread_->BlockingCall([&] {
     return video_media_channel()->SetVideoSend(ssrc_, &options,
                                                video_track().get());
   });
@@ -722,9 +719,8 @@ void VideoRtpSender::ClearSend() {
   // Allow SetVideoSend to fail since `enable` is false and `source` is null.
   // This the normal case when the underlying media channel has already been
   // deleted.
-  worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
+  worker_thread_->BlockingCall(
-    return video_media_channel()->SetVideoSend(ssrc_, nullptr, nullptr);
+      [&] { video_media_channel()->SetVideoSend(ssrc_, nullptr, nullptr); });
-  });
 }
 
 }  // namespace webrtc

pc/rtp_transceiver.cc

@@ -28,7 +28,6 @@
 #include "pc/rtp_media_utils.h"
 #include "pc/session_description.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/thread.h"
 
@@ -185,57 +184,42 @@ RTCError RtpTransceiver::CreateChannel(
     // TODO(bugs.webrtc.org/11992): Remove this workaround after updates in
     // PeerConnection and add the expectation that we're already on the right
     // thread.
-    new_channel =
+    context()->worker_thread()->BlockingCall([&] {
-        context()
+      RTC_DCHECK_RUN_ON(context()->worker_thread());
-            ->worker_thread()
-            ->Invoke<std::unique_ptr<cricket::VoiceChannel>>(
-                RTC_FROM_HERE, [&]() -> std::unique_ptr<cricket::VoiceChannel> {
-                  RTC_DCHECK_RUN_ON(context()->worker_thread());
 
-                  cricket::VoiceMediaChannel* media_channel =
+      cricket::VoiceMediaChannel* media_channel =
-                      media_engine()->voice().CreateMediaChannel(
+          media_engine()->voice().CreateMediaChannel(
-                          call_ptr, media_config, audio_options,
+              call_ptr, media_config, audio_options, crypto_options);
-                          crypto_options);
+      if (!media_channel) {
-                  if (!media_channel) {
+        return;
-                    return nullptr;
+      }
-                  }
 
-                  auto voice_channel = std::make_unique<cricket::VoiceChannel>(
+      new_channel = std::make_unique<cricket::VoiceChannel>(
-                      context()->worker_thread(), context()->network_thread(),
+          context()->worker_thread(), context()->network_thread(),
-                      context()->signaling_thread(),
+          context()->signaling_thread(), absl::WrapUnique(media_channel), mid,
-                      absl::WrapUnique(media_channel), mid, srtp_required,
+          srtp_required, crypto_options, context()->ssrc_generator());
-                      crypto_options, context()->ssrc_generator());
+    });
-
-                  return voice_channel;
-                });
   } else {
     RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, media_type());
 
     // TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
     // the worker thread. We shouldn't be using the `call_ptr_` hack here but
     // simply be on the worker thread and use `call_` (update upstream code).
-    new_channel =
+    context()->worker_thread()->BlockingCall([&] {
-        context()
+      RTC_DCHECK_RUN_ON(context()->worker_thread());
-            ->worker_thread()
+      cricket::VideoMediaChannel* media_channel =
-            ->Invoke<std::unique_ptr<cricket::VideoChannel>>(
+          media_engine()->video().CreateMediaChannel(
-                RTC_FROM_HERE, [&]() -> std::unique_ptr<cricket::VideoChannel> {
+              call_ptr, media_config, video_options, crypto_options,
-                  RTC_DCHECK_RUN_ON(context()->worker_thread());
+              video_bitrate_allocator_factory);
-                  cricket::VideoMediaChannel* media_channel =
+      if (!media_channel) {
-                      media_engine()->video().CreateMediaChannel(
+        return;
-                          call_ptr, media_config, video_options, crypto_options,
+      }
-                          video_bitrate_allocator_factory);
-                  if (!media_channel) {
-                    return nullptr;
-                  }
 
-                  auto video_channel = std::make_unique<cricket::VideoChannel>(
+      new_channel = std::make_unique<cricket::VideoChannel>(
-                      context()->worker_thread(), context()->network_thread(),
+          context()->worker_thread(), context()->network_thread(),
-                      context()->signaling_thread(),
+          context()->signaling_thread(), absl::WrapUnique(media_channel), mid,
-                      absl::WrapUnique(media_channel), mid, srtp_required,
+          srtp_required, crypto_options, context()->ssrc_generator());
-                      crypto_options, context()->ssrc_generator());
+    });
-
-                  return video_channel;
-                });
   }
   if (!new_channel) {
     // TODO(hta): Must be a better way
@@ -274,7 +258,7 @@ void RtpTransceiver::SetChannel(
   // Similarly, if the channel() accessor is limited to the network thread, that
   // helps with keeping the channel implementation requirements being met and
   // avoids synchronization for accessing the pointer or network related state.
-  context()->network_thread()->Invoke<void>(RTC_FROM_HERE, [&]() {
+  context()->network_thread()->BlockingCall([&]() {
     if (channel_) {
       channel_->SetFirstPacketReceivedCallback(nullptr);
       channel_->SetRtpTransport(nullptr);
@@ -310,7 +294,7 @@ void RtpTransceiver::ClearChannel() {
   }
   std::unique_ptr<cricket::ChannelInterface> channel_to_delete;
 
-  context()->network_thread()->Invoke<void>(RTC_FROM_HERE, [&]() {
+  context()->network_thread()->BlockingCall([&]() {
     if (channel_) {
       channel_->SetFirstPacketReceivedCallback(nullptr);
       channel_->SetRtpTransport(nullptr);
@@ -331,7 +315,7 @@ void RtpTransceiver::PushNewMediaChannelAndDeleteChannel(
   if (!channel_to_delete && senders_.empty() && receivers_.empty()) {
     return;
   }
-  context()->worker_thread()->Invoke<void>(RTC_FROM_HERE, [&]() {
+  context()->worker_thread()->BlockingCall([&]() {
     // Push down the new media_channel, if any, otherwise clear it.
     auto* media_channel = channel_ ? channel_->media_channel() : nullptr;
     for (const auto& sender : senders_) {
@@ -399,7 +383,7 @@ bool RtpTransceiver::RemoveReceiver(RtpReceiverInterface* receiver) {
   }
 
   (*it)->internal()->Stop();
-  context()->worker_thread()->Invoke<void>(RTC_FROM_HERE, [&]() {
+  context()->worker_thread()->BlockingCall([&]() {
     // `Stop()` will clear the receiver's pointer to the media channel.
     (*it)->internal()->SetMediaChannel(nullptr);
   });
@@ -533,7 +517,7 @@ void RtpTransceiver::StopSendingAndReceiving() {
   for (const auto& receiver : receivers_)
     receiver->internal()->Stop();
 
-  context()->worker_thread()->Invoke<void>(RTC_FROM_HERE, [&]() {
+  context()->worker_thread()->BlockingCall([&]() {
     // 5 Stop receiving media with receiver.
     for (const auto& receiver : receivers_)
       receiver->internal()->SetMediaChannel(nullptr);

pc/sctp_data_channel.cc

@@ -20,7 +20,6 @@
 #include "pc/proxy.h"
 #include "pc/sctp_utils.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/system/unused.h"
 #include "rtc_base/thread.h"

pc/sctp_transport.cc

@@ -17,7 +17,6 @@
 #include "api/dtls_transport_interface.h"
 #include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 
 namespace webrtc {
@@ -50,8 +49,7 @@ SctpTransportInformation SctpTransport::Information() const {
   // expected thread. Chromium currently calls this method from
   // TransceiverStateSurfacer.
   if (!owner_thread_->IsCurrent()) {
-    return owner_thread_->Invoke<SctpTransportInformation>(
+    return owner_thread_->BlockingCall([this] { return Information(); });
-        RTC_FROM_HERE, [this] { return Information(); });
   }
   RTC_DCHECK_RUN_ON(owner_thread_);
   return info_;

pc/sdp_offer_answer.cc

@@ -55,7 +55,6 @@
 #include "pc/usage_pattern.h"
 #include "pc/webrtc_session_description_factory.h"
 #include "rtc_base/helpers.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/ssl_stream_adapter.h"
@@ -691,13 +690,12 @@ rtc::scoped_refptr<webrtc::DtlsTransport> LookupDtlsTransportByMid(
     JsepTransportController* controller,
     const std::string& mid) {
   // TODO(tommi): Can we post this (and associated operations where this
-  // function is called) to the network thread and avoid this Invoke?
+  // function is called) to the network thread and avoid this BlockingCall?
   // We might be able to simplify a few things if we set the transport on
   // the network thread and then update the implementation to check that
   // the set_ and relevant get methods are always called on the network
   // thread (we'll need to update proxy maps).
-  return network_thread->Invoke<rtc::scoped_refptr<webrtc::DtlsTransport>>(
+  return network_thread->BlockingCall(
-      RTC_FROM_HERE,
       [controller, &mid] { return controller->LookupDtlsTransportByMid(mid); });
 }
 
@@ -1813,7 +1811,7 @@ RTCError SdpOfferAnswerHandler::ReplaceRemoteDescription(
   ReportSimulcastApiVersion(kSimulcastVersionApplyRemoteDescription,
                             *session_desc);
 
-  // NOTE: This will perform an Invoke() to the network thread.
+  // NOTE: This will perform a BlockingCall() to the network thread.
   return transport_controller_s()->SetRemoteDescription(sdp_type, session_desc);
 }
 
@@ -2200,8 +2198,8 @@ void SdpOfferAnswerHandler::DoSetLocalDescription(
 
     // TODO(deadbeef): We already had to hop to the network thread for
     // MaybeStartGathering...
-    context_->network_thread()->Invoke<void>(
+    context_->network_thread()->BlockingCall(
-        RTC_FROM_HERE, [this] { port_allocator()->DiscardCandidatePool(); });
+        [this] { port_allocator()->DiscardCandidatePool(); });
     // Make UMA notes about what was agreed to.
     ReportNegotiatedSdpSemantics(*local_description());
   }
@@ -2404,8 +2402,8 @@ void SdpOfferAnswerHandler::SetRemoteDescriptionPostProcess(bool was_answer) {
   if (was_answer) {
     // TODO(deadbeef): We already had to hop to the network thread for
     // MaybeStartGathering...
-    context_->network_thread()->Invoke<void>(
+    context_->network_thread()->BlockingCall(
-        RTC_FROM_HERE, [this] { port_allocator()->DiscardCandidatePool(); });
+        [this] { port_allocator()->DiscardCandidatePool(); });
     // Make UMA notes about what was agreed to.
     ReportNegotiatedSdpSemantics(*remote_description());
   }
@@ -3831,8 +3829,7 @@ void SdpOfferAnswerHandler::GetOptionsForOffer(
   session_options->rtcp_cname = rtcp_cname_;
   session_options->crypto_options = pc_->GetCryptoOptions();
   session_options->pooled_ice_credentials =
-      context_->network_thread()->Invoke<std::vector<cricket::IceParameters>>(
+      context_->network_thread()->BlockingCall(
-          RTC_FROM_HERE,
           [this] { return port_allocator()->GetPooledIceCredentials(); });
   session_options->offer_extmap_allow_mixed =
       pc_->configuration()->offer_extmap_allow_mixed;
@@ -4095,8 +4092,7 @@ void SdpOfferAnswerHandler::GetOptionsForAnswer(
   session_options->rtcp_cname = rtcp_cname_;
   session_options->crypto_options = pc_->GetCryptoOptions();
   session_options->pooled_ice_credentials =
-      context_->network_thread()->Invoke<std::vector<cricket::IceParameters>>(
+      context_->network_thread()->BlockingCall(
-          RTC_FROM_HERE,
           [this] { return port_allocator()->GetPooledIceCredentials(); });
 }
 
@@ -4525,8 +4521,8 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
   RTC_DCHECK(sdesc);
 
   if (ConfiguredForMedia()) {
-    // Note: This will perform an Invoke over to the worker thread, which we'll
+    // Note: This will perform a BlockingCall over to the worker thread, which
-    // also do in a loop below.
+    // we'll also do in a loop below.
     if (!UpdatePayloadTypeDemuxingState(source, bundle_groups_by_mid)) {
       // Note that this is never expected to fail, since RtpDemuxer doesn't
       // return an error when changing payload type demux criteria, which is all
@@ -4570,7 +4566,7 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
     for (const auto& entry : channels) {
       std::string error;
       bool success =
-          context_->worker_thread()->Invoke<bool>(RTC_FROM_HERE, [&]() {
+          context_->worker_thread()->BlockingCall([&]() {
             return (source == cricket::CS_LOCAL)
                        ? entry.first->SetLocalContent(entry.second, type, error)
                        : entry.first->SetRemoteContent(entry.second, type,
@@ -4918,7 +4914,7 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
 
 bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
   RTC_DCHECK_RUN_ON(signaling_thread());
-  if (!context_->network_thread()->Invoke<bool>(RTC_FROM_HERE, [this, &mid] {
+  if (!context_->network_thread()->BlockingCall([this, &mid] {
         RTC_DCHECK_RUN_ON(context_->network_thread());
         return pc_->SetupDataChannelTransport_n(mid);
       })) {
@@ -4940,7 +4936,7 @@ void SdpOfferAnswerHandler::DestroyDataChannelTransport(RTCError error) {
   if (has_sctp)
     data_channel_controller()->OnTransportChannelClosed(error);
 
-  context_->network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
+  context_->network_thread()->BlockingCall([this] {
     RTC_DCHECK_RUN_ON(context_->network_thread());
     pc_->TeardownDataChannelTransport_n();
   });
@@ -5174,7 +5170,7 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
                                         pt_demuxing_has_been_used_video_;
 
   // Gather all updates ahead of time so that all channels can be updated in a
-  // single Invoke; necessary due to thread guards.
+  // single BlockingCall; necessary due to thread guards.
   std::vector<std::pair<bool, cricket::ChannelInterface*>> channels_to_update;
   for (const auto& transceiver : transceivers()->ListInternal()) {
     cricket::ChannelInterface* channel = transceiver->channel();
@@ -5226,22 +5222,21 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
     return true;
   }
 
-  // TODO(bugs.webrtc.org/11993): This Invoke() will also invoke on the network
+  // TODO(bugs.webrtc.org/11993): This BlockingCall() will also block on the
-  // thread for every demuxer sink that needs to be updated. The demuxer state
+  // network thread for every demuxer sink that needs to be updated. The demuxer
-  // needs to be fully (and only) managed on the network thread and once that's
+  // state needs to be fully (and only) managed on the network thread and once
-  // the case, there's no need to stop by on the worker. Ideally we could also
+  // that's the case, there's no need to stop by on the worker. Ideally we could
-  // do this without blocking.
+  // also do this without blocking.
-  return context_->worker_thread()->Invoke<bool>(
+  return context_->worker_thread()->BlockingCall([&channels_to_update]() {
-      RTC_FROM_HERE, [&channels_to_update]() {
+    for (const auto& it : channels_to_update) {
-        for (const auto& it : channels_to_update) {
+      if (!it.second->SetPayloadTypeDemuxingEnabled(it.first)) {
-          if (!it.second->SetPayloadTypeDemuxingEnabled(it.first)) {
+        // Note that the state has already been irrevocably changed at this
-            // Note that the state has already been irrevocably changed at this
+        // point. Is it useful to stop the loop?
-            // point. Is it useful to stop the loop?
+        return false;
-            return false;
+      }
-          }
+    }
-        }
+    return true;
-        return true;
+  });
-      });
 }
 
 bool SdpOfferAnswerHandler::ConfiguredForMedia() const {

pc/video_rtp_receiver.cc

@@ -19,7 +19,6 @@
 #include "api/video/recordable_encoded_frame.h"
 #include "pc/video_track.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 
 namespace webrtc {
@@ -117,7 +116,7 @@ void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   MediaSourceInterface::SourceState state = source_->state();
   // TODO(tommi): Can we restart the media channel without blocking?
-  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+  worker_thread_->BlockingCall([&] {
     RTC_DCHECK_RUN_ON(worker_thread_);
     RestartMediaChannel_w(std::move(ssrc), state);
   });
@@ -316,7 +315,7 @@ void VideoRtpReceiver::SetupMediaChannel(absl::optional<uint32_t> ssrc,
   RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
   RTC_DCHECK(media_channel);
   MediaSourceInterface::SourceState state = source_->state();
-  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+  worker_thread_->BlockingCall([&] {
     RTC_DCHECK_RUN_ON(worker_thread_);
     SetMediaChannel_w(media_channel);
     RestartMediaChannel_w(std::move(ssrc), state);

pc/video_track.cc

@@ -16,7 +16,6 @@
 #include "api/notifier.h"
 #include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 
 namespace webrtc {
 
@@ -95,7 +94,7 @@ bool VideoTrack::set_enabled(bool enable) {
 
   bool ret = MediaStreamTrack<VideoTrackInterface>::set_enabled(enable);
 
-  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
+  worker_thread_->BlockingCall([&]() {
     RTC_DCHECK_RUN_ON(worker_thread_);
     enabled_w_ = enable;
     for (auto& sink_pair : sink_pairs()) {

rtc_base/BUILD.gn

@@ -1492,7 +1492,6 @@ if (rtc_include_tests) {
         ":checks",
         ":gunit_helpers",
         ":ip_address",
-        ":location",
         ":logging",
         ":macromagic",
         ":net_helpers",
@@ -1581,7 +1580,6 @@ if (rtc_include_tests) {
         ":gunit_helpers",
         ":histogram_percentile_counter",
         ":ip_address",
-        ":location",
         ":logging",
         ":macromagic",
         ":mod_ops",

rtc_base/fake_mdns_responder.h

@@ -17,7 +17,6 @@
 
 #include "absl/strings/string_view.h"
 #include "rtc_base/ip_address.h"
-#include "rtc_base/location.h"
 #include "rtc_base/mdns_responder_interface.h"
 #include "rtc_base/thread.h"
 

rtc_base/network.cc

@@ -1253,7 +1253,7 @@ void BasicNetworkManager::set_vpn_list(const std::vector<NetworkMask>& vpn) {
   if (thread_ == nullptr) {
     vpn_ = vpn;
   } else {
-    thread_->Invoke<void>(RTC_FROM_HERE, [this, vpn] { vpn_ = vpn; });
+    thread_->BlockingCall([this, vpn] { vpn_ = vpn; });
   }
 }
 

rtc_base/rtc_certificate_generator.cc

@@ -17,7 +17,6 @@
 #include <utility>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/message_handler.h"
 #include "rtc_base/ssl_identity.h"
 

rtc_base/socket_unittest.cc

@@ -23,7 +23,6 @@
 #include "rtc_base/async_udp_socket.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/gunit.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/net_helpers.h"
 #include "rtc_base/socket_address.h"
@@ -730,7 +729,7 @@ void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) {
   // Shouldn't signal when blocked in a thread Send, where process_io is false.
   std::unique_ptr<Thread> thread(Thread::CreateWithSocketServer());
   thread->Start();
-  thread->Invoke<void>(RTC_FROM_HERE, [] { Thread::SleepMs(500); });
+  thread->BlockingCall([] { Thread::SleepMs(500); });
   EXPECT_FALSE(sink.Check(accepted.get(), SSE_READ));
 
   // But should signal when process_io is true.

rtc_base/stream.cc

@@ -16,7 +16,6 @@
 #include <string>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/thread.h"
 
 namespace rtc {

rtc_base/synchronization/BUILD.gn

@@ -86,7 +86,6 @@ if (rtc_include_tests) {
         ":yield",
         ":yield_policy",
         "..:checks",
-        "..:location",
         "..:macromagic",
         "..:platform_thread",
         "..:rtc_base",

rtc_base/task_queue.h

@@ -90,7 +90,6 @@ class RTC_LOCKABLE RTC_EXPORT TaskQueue {
   // Returns non-owning pointer to the task queue implementation.
   webrtc::TaskQueueBase* Get() { return impl_; }
 
-  // TODO(tommi): For better debuggability, implement RTC_FROM_HERE.
   void PostTask(absl::AnyInvocable<void() &&> task) {
     impl_->PostTask(std::move(task));
   }

rtc_base/thread.h

@@ -349,7 +349,8 @@ class RTC_LOCKABLE RTC_EXPORT Thread : public webrtc::TaskQueueBase {
 
   // Deprecated, use `BlockingCall` instead.
   template <typename ReturnT>
-  ReturnT Invoke(const Location& posted_from, FunctionView<ReturnT()> functor) {
+  [[deprecated]] ReturnT Invoke(const Location& /*posted_from*/,
+                                FunctionView<ReturnT()> functor) {
     return BlockingCall(functor);
   }
 

rtc_base/time_utils_unittest.cc

@@ -16,7 +16,6 @@
 #include "rtc_base/event.h"
 #include "rtc_base/fake_clock.h"
 #include "rtc_base/helpers.h"
-#include "rtc_base/location.h"
 #include "rtc_base/message_handler.h"
 #include "rtc_base/thread.h"
 #include "test/gtest.h"

rtc_tools/network_tester/test_controller.cc

@@ -38,7 +38,7 @@ TestController::TestController(int min_port,
   send_data_.fill(42);
   packet_sender_thread_->SetName("PacketSender", nullptr);
   packet_sender_thread_->Start();
-  packet_sender_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+  packet_sender_thread_->BlockingCall([&] {
     RTC_DCHECK_RUN_ON(packet_sender_thread_.get());
     udp_socket_ =
         std::unique_ptr<rtc::AsyncPacketSocket>(socket_factory_.CreateUdpSocket(
@@ -49,8 +49,8 @@ TestController::TestController(int min_port,
 
 TestController::~TestController() {
   RTC_DCHECK_RUN_ON(&test_controller_thread_checker_);
-  packet_sender_thread_->Invoke<void>(
+  packet_sender_thread_->BlockingCall(
-      RTC_FROM_HERE, [this]() { task_safety_flag_->SetNotAlive(); });
+      [this]() { task_safety_flag_->SetNotAlive(); });
 }
 
 void TestController::SendConnectTo(const std::string& hostname, int port) {

sdk/android/src/jni/pc/peer_connection.cc

@@ -516,7 +516,7 @@ static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetLocalDescription(
   // must do this odd dance.
   std::string sdp;
   std::string type;
-  pc->signaling_thread()->Invoke<void>(RTC_FROM_HERE, [pc, &sdp, &type] {
+  pc->signaling_thread()->BlockingCall([pc, &sdp, &type] {
     const SessionDescriptionInterface* desc = pc->local_description();
     if (desc) {
       RTC_CHECK(desc->ToString(&sdp)) << "got so far: " << sdp;
@@ -535,7 +535,7 @@ static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetRemoteDescription(
   // must do this odd dance.
   std::string sdp;
   std::string type;
-  pc->signaling_thread()->Invoke<void>(RTC_FROM_HERE, [pc, &sdp, &type] {
+  pc->signaling_thread()->BlockingCall([pc, &sdp, &type] {
     const SessionDescriptionInterface* desc = pc->remote_description();
     if (desc) {
       RTC_CHECK(desc->ToString(&sdp)) << "got so far: " << sdp;

sdk/objc/api/peerconnection/RTCMediaStream.mm

@@ -36,16 +36,14 @@
 
 - (NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *)audioTracks {
   if (!_signalingThread->IsCurrent()) {
-    return _signalingThread->Invoke<NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *>(
+    return _signalingThread->BlockingCall([self]() { return self.audioTracks; });
-        RTC_FROM_HERE, [self]() { return self.audioTracks; });
   }
   return [_audioTracks copy];
 }
 
 - (NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *)videoTracks {
   if (!_signalingThread->IsCurrent()) {
-    return _signalingThread->Invoke<NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *>(
+    return _signalingThread->BlockingCall([self]() { return self.videoTracks; });
-        RTC_FROM_HERE, [self]() { return self.videoTracks; });
   }
   return [_videoTracks copy];
 }
@@ -56,8 +54,8 @@
 
 - (void)addAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack {
   if (!_signalingThread->IsCurrent()) {
-    return _signalingThread->Invoke<void>(
+    return _signalingThread->BlockingCall(
-        RTC_FROM_HERE, [audioTrack, self]() { return [self addAudioTrack:audioTrack]; });
+        [audioTrack, self]() { return [self addAudioTrack:audioTrack]; });
   }
   if (_nativeMediaStream->AddTrack(audioTrack.nativeAudioTrack)) {
     [_audioTracks addObject:audioTrack];
@@ -66,8 +64,8 @@
 
 - (void)addVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack {
   if (!_signalingThread->IsCurrent()) {
-    return _signalingThread->Invoke<void>(
+    return _signalingThread->BlockingCall(
-        RTC_FROM_HERE, [videoTrack, self]() { return [self addVideoTrack:videoTrack]; });
+        [videoTrack, self]() { return [self addVideoTrack:videoTrack]; });
   }
   if (_nativeMediaStream->AddTrack(videoTrack.nativeVideoTrack)) {
     [_videoTracks addObject:videoTrack];
@@ -76,8 +74,8 @@
 
 - (void)removeAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack {
   if (!_signalingThread->IsCurrent()) {
-    return _signalingThread->Invoke<void>(
+    return _signalingThread->BlockingCall(
-        RTC_FROM_HERE, [audioTrack, self]() { return [self removeAudioTrack:audioTrack]; });
+        [audioTrack, self]() { return [self removeAudioTrack:audioTrack]; });
   }
   NSUInteger index = [_audioTracks indexOfObjectIdenticalTo:audioTrack];
   if (index == NSNotFound) {
@@ -91,8 +89,8 @@
 
 - (void)removeVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack {
   if (!_signalingThread->IsCurrent()) {
-    return _signalingThread->Invoke<void>(
+    return _signalingThread->BlockingCall(
-        RTC_FROM_HERE, [videoTrack, self]() { return [self removeVideoTrack:videoTrack]; });
+        [videoTrack, self]() { return [self removeVideoTrack:videoTrack]; });
   }
   NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:videoTrack];
   if (index == NSNotFound) {

sdk/objc/api/peerconnection/RTCPeerConnection.mm

@@ -392,26 +392,22 @@ void PeerConnectionDelegateAdapter::OnRemoveTrack(
 
 - (RTC_OBJC_TYPE(RTCSessionDescription) *)localDescription {
   // It's only safe to operate on SessionDescriptionInterface on the signaling thread.
-  return _peerConnection->signaling_thread()->Invoke<RTC_OBJC_TYPE(RTCSessionDescription) *>(
+  return _peerConnection->signaling_thread()->BlockingCall([self] {
-      RTC_FROM_HERE, [self] {
+    const webrtc::SessionDescriptionInterface *description = _peerConnection->local_description();
-        const webrtc::SessionDescriptionInterface *description =
+    return description ?
-            _peerConnection->local_description();
+        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
-        return description ?
+        nil;
-            [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
+  });
-            nil;
-      });
 }
 
 - (RTC_OBJC_TYPE(RTCSessionDescription) *)remoteDescription {
   // It's only safe to operate on SessionDescriptionInterface on the signaling thread.
-  return _peerConnection->signaling_thread()->Invoke<RTC_OBJC_TYPE(RTCSessionDescription) *>(
+  return _peerConnection->signaling_thread()->BlockingCall([self] {
-      RTC_FROM_HERE, [self] {
+    const webrtc::SessionDescriptionInterface *description = _peerConnection->remote_description();
-        const webrtc::SessionDescriptionInterface *description =
+    return description ?
-            _peerConnection->remote_description();
+        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
-        return description ?
+        nil;
-            [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
+  });
-            nil;
-      });
 }
 
 - (RTCSignalingState)signalingState {

sdk/objc/api/peerconnection/RTCVideoTrack.mm

@@ -72,7 +72,7 @@
 
 - (void)addRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer {
   if (!_workerThread->IsCurrent()) {
-    _workerThread->Invoke<void>(RTC_FROM_HERE, [renderer, self] { [self addRenderer:renderer]; });
+    _workerThread->BlockingCall([renderer, self] { [self addRenderer:renderer]; });
     return;
   }
 
@@ -93,8 +93,7 @@
 
 - (void)removeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer {
   if (!_workerThread->IsCurrent()) {
-    _workerThread->Invoke<void>(RTC_FROM_HERE,
+    _workerThread->BlockingCall([renderer, self] { [self removeRenderer:renderer]; });
-                                [renderer, self] { [self removeRenderer:renderer]; });
     return;
   }
   __block NSUInteger indexToRemove = NSNotFound;

sdk/objc/native/src/objc_audio_device_delegate.mm

@@ -178,7 +178,7 @@ class AudioDeviceDelegateImpl final : public rtc::RefCountedNonVirtual<AudioDevi
       block();
     }
   } else {
-    thread->Invoke<void>(RTC_FROM_HERE, [block] {
+    thread->BlockingCall([block] {
       @autoreleasepool {
         block();
       }

test/time_controller/BUILD.gn

@@ -60,7 +60,6 @@ if (rtc_include_tests) {
       "../../api:time_controller",
       "../../api/units:time_delta",
       "../../rtc_base",
-      "../../rtc_base:location",
       "../../rtc_base:macromagic",
       "../../rtc_base:rtc_event",
       "../../rtc_base:rtc_task_queue",

test/time_controller/time_controller_conformance_test.cc

@@ -14,7 +14,6 @@
 #include "api/test/time_controller.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/event.h"
-#include "rtc_base/location.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
@@ -121,7 +120,7 @@ TEST_P(SimulatedRealTimeControllerConformanceTest, ThreadPostInvokeOrderTest) {
   // posted/invoked.
   ExecutionOrderKeeper execution_order;
   thread->PostTask([&]() { execution_order.Executed(1); });
-  thread->Invoke<void>(RTC_FROM_HERE, [&]() { execution_order.Executed(2); });
+  thread->BlockingCall([&]() { execution_order.Executed(2); });
   time_controller->AdvanceTime(TimeDelta::Millis(100));
   EXPECT_THAT(execution_order.order(), ElementsAreArray({1, 2}));
   // Destroy `thread` before `execution_order` to be sure `execution_order`
@@ -140,7 +139,7 @@ TEST_P(SimulatedRealTimeControllerConformanceTest,
   ExecutionOrderKeeper execution_order;
   thread->PostTask([&]() {
     thread->PostTask([&]() { execution_order.Executed(2); });
-    thread->Invoke<void>(RTC_FROM_HERE, [&]() { execution_order.Executed(1); });
+    thread->BlockingCall([&]() { execution_order.Executed(1); });
   });
   time_controller->AdvanceTime(TimeDelta::Millis(100));
   EXPECT_THAT(execution_order.order(), ElementsAreArray({1, 2}));

video/BUILD.gn

@@ -106,7 +106,6 @@ rtc_library("video") {
     "../rtc_base:checks",
     "../rtc_base:event_tracer",
     "../rtc_base:histogram_percentile_counter",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:mod_ops",
@@ -405,7 +404,6 @@ rtc_library("video_stream_encoder_impl") {
     "../rtc_base:checks",
     "../rtc_base:criticalsection",
     "../rtc_base:event_tracer",
-    "../rtc_base:location",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:race_checker",

video/call_stats2.cc

@@ -16,7 +16,6 @@
 
 #include "absl/algorithm/container.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "system_wrappers/include/metrics.h"
 
 namespace webrtc {

video/receive_statistics_proxy2.cc

@@ -594,7 +594,7 @@ VideoReceiveStreamInterface::Stats ReceiveStatisticsProxy::GetStats() const {
   RTC_DCHECK_RUN_ON(&main_thread_);
 
   // Like VideoReceiveStreamInterface::GetStats, called on the worker thread
-  // from StatsCollector::ExtractMediaInfo via worker_thread()->Invoke().
+  // from StatsCollector::ExtractMediaInfo via worker_thread()->BlockingCall().
   // WebRtcVideoChannel::GetStats(), GetVideoReceiverInfo.
 
   // Get current frame rates here, as only updating them on new frames prevents

video/rtp_video_stream_receiver2.cc

@@ -42,7 +42,6 @@
 #include "modules/video_coding/nack_requester.h"
 #include "modules/video_coding/packet_buffer.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 #include "system_wrappers/include/metrics.h"

video/video_stream_encoder.cc

@@ -42,7 +42,6 @@
 #include "rtc_base/experiments/alr_experiment.h"
 #include "rtc_base/experiments/encoder_info_settings.h"
 #include "rtc_base/experiments/rate_control_settings.h"
-#include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/system/no_unique_address.h"