Deprecate rtc::TaskQueue variant of AudioProcessing::CreateAndAttachAecDump

Bug: webrtc:14169
Change-Id: I63f40ec18b72cba89eb0b9b298f448ce7f7c4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334201
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41528}

media/BUILD.gn

@@ -516,7 +516,6 @@ rtc_library("rtc_audio_video") {
     "../rtc_base:macromagic",
     "../rtc_base:network_route",
     "../rtc_base:race_checker",
-    "../rtc_base:rtc_task_queue",
     "../rtc_base:safe_conversions",
     "../rtc_base:socket",
     "../rtc_base:ssl",

media/engine/webrtc_voice_engine.cc

@@ -377,9 +377,8 @@ void WebRtcVoiceEngine::Init() {
 
   // TaskQueue expects to be created/destroyed on the same thread.
   RTC_DCHECK(!low_priority_worker_queue_);
-  low_priority_worker_queue_.reset(
+  low_priority_worker_queue_ = task_queue_factory_->CreateTaskQueue(
-      new rtc::TaskQueue(task_queue_factory_->CreateTaskQueue(
+      "rtc-low-prio", webrtc::TaskQueueFactory::Priority::LOW);
-          "rtc-low-prio", webrtc::TaskQueueFactory::Priority::LOW)));
 
   // Load our audio codec lists.
   RTC_LOG(LS_VERBOSE) << "Supported send codecs in order of preference:";

media/engine/webrtc_voice_engine.h

@@ -66,7 +66,6 @@
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/system/file_wrapper.h"
-#include "rtc_base/task_queue.h"
 
 namespace webrtc {
 class AudioFrameProcessor;
@@ -141,7 +140,8 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
   void ApplyOptions(const AudioOptions& options);
 
   webrtc::TaskQueueFactory* const task_queue_factory_;
-  std::unique_ptr<rtc::TaskQueue> low_priority_worker_queue_;
+  std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>
+      low_priority_worker_queue_;
 
   webrtc::AudioDeviceModule* adm();
   webrtc::AudioProcessing* apm() const;

modules/audio_processing/include/audio_processing.cc

@@ -222,22 +222,4 @@ bool AudioProcessing::CreateAndAttachAecDump(FILE* handle,
                                 worker_queue->Get());
 }
 
-bool AudioProcessing::CreateAndAttachAecDump(
-    absl::string_view file_name,
-    int64_t max_log_size_bytes,
-    absl::Nonnull<TaskQueueBase*> worker_queue) {
-  // Newer code should implement this variant,
-  // Older code shouldn't call this variant.
-  RTC_CHECK_NOTREACHED();
-}
-
-bool AudioProcessing::CreateAndAttachAecDump(
-    FILE* handle,
-    int64_t max_log_size_bytes,
-    absl::Nonnull<TaskQueueBase*> worker_queue) {
-  // New ercode should implement this variant,
-  // Older code shouldn't call this variant.
-  RTC_CHECK_NOTREACHED();
-}
-
 }  // namespace webrtc

modules/audio_processing/include/audio_processing.h

@@ -635,22 +635,21 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
   // sucessfully opened, while a value of false indicates that
   // opening the file failed.
   // TODO: bugs.webrtc.org/14169 - Delete rtc::TaskQueue variants
-  virtual bool CreateAndAttachAecDump(absl::string_view file_name,
+  [[deprecated]] bool CreateAndAttachAecDump(absl::string_view file_name,
-                                      int64_t max_log_size_bytes,
+                                             int64_t max_log_size_bytes,
-                                      rtc::TaskQueue* worker_queue);
+                                             rtc::TaskQueue* worker_queue);
-  virtual bool CreateAndAttachAecDump(FILE* handle,
+  [[deprecated]] bool CreateAndAttachAecDump(FILE* handle,
-                                      int64_t max_log_size_bytes,
+                                             int64_t max_log_size_bytes,
-                                      rtc::TaskQueue* worker_queue);
+                                             rtc::TaskQueue* worker_queue);
-  // TODO: bugs.webrtc.org/14169 - Make TaskQueueBase variants pure virtual when
+
-  // implemented by derived classes.
   virtual bool CreateAndAttachAecDump(
       absl::string_view file_name,
       int64_t max_log_size_bytes,
-      absl::Nonnull<TaskQueueBase*> worker_queue);
+      absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
   virtual bool CreateAndAttachAecDump(
       absl::Nonnull<FILE*> handle,
       int64_t max_log_size_bytes,
-      absl::Nonnull<TaskQueueBase*> worker_queue);
+      absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
 
   // TODO(webrtc:5298) Deprecated variant.
   // Attaches provided webrtc::AecDump for recording debugging

test/pc/e2e/BUILD.gn

@@ -99,6 +99,7 @@ if (!build_with_chromium) {
         "../../../api:enable_media_with_defaults",
         "../../../api:time_controller",
         "../../../api/rtc_event_log:rtc_event_log_factory",
+        "../../../api/task_queue",
         "../../../api/task_queue:default_task_queue_factory",
         "../../../api/test/pclf:media_configuration",
         "../../../api/test/pclf:media_quality_test_params",

test/pc/e2e/peer_connection_quality_test.cc

@@ -277,7 +277,7 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
 
   TestPeerFactory test_peer_factory(
       signaling_thread.get(), time_controller_,
-      video_quality_analyzer_injection_helper_.get(), task_queue_.get());
+      video_quality_analyzer_injection_helper_.get(), task_queue_->Get());
   alice_ = test_peer_factory.CreateTestPeer(
       std::move(alice_configurer),
       std::make_unique<FixturePeerConnectionObserver>(

test/pc/e2e/test_peer_factory.h

@@ -18,6 +18,7 @@
 
 #include "absl/strings/string_view.h"
 #include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/task_queue/task_queue_base.h"
 #include "api/test/pclf/media_configuration.h"
 #include "api/test/pclf/media_quality_test_params.h"
 #include "api/test/pclf/peer_configurer.h"
@@ -55,12 +56,22 @@ class TestPeerFactory {
   TestPeerFactory(rtc::Thread* signaling_thread,
                   TimeController& time_controller,
                   VideoQualityAnalyzerInjectionHelper* video_analyzer_helper,
-                  rtc::TaskQueue* task_queue)
+                  TaskQueueBase* task_queue)
       : signaling_thread_(signaling_thread),
         time_controller_(time_controller),
         video_analyzer_helper_(video_analyzer_helper),
         task_queue_(task_queue) {}
 
+  [[deprecated]] TestPeerFactory(
+      rtc::Thread* signaling_thread,
+      TimeController& time_controller,
+      VideoQualityAnalyzerInjectionHelper* video_analyzer_helper,
+      rtc::TaskQueue* task_queue)
+      : TestPeerFactory(signaling_thread,
+                        time_controller,
+                        video_analyzer_helper,
+                        task_queue->Get()) {}
+
   // Setups all components, that should be provided to WebRTC
   // PeerConnectionFactory and PeerConnection creation methods,
   // also will setup dependencies, that are required for media analyzers
@@ -75,7 +86,7 @@ class TestPeerFactory {
   rtc::Thread* signaling_thread_;
   TimeController& time_controller_;
   VideoQualityAnalyzerInjectionHelper* video_analyzer_helper_;
-  rtc::TaskQueue* task_queue_;
+  TaskQueueBase* const task_queue_;
 };
 
 }  // namespace webrtc_pc_e2e