Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )

Reason for revert:
Breaks internal project.

Original issue's description:
> Activate 'offload debug dump recordings from audio thread to TaskQueue'.
>
> A low priority task queue is added to WebRTCVoiceEngine. The
> start/stop debug calls make file logging happen on the task queue.
>
> In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> so that it can be shared for low priority tasks between different
> subcomponents. It will require some changes to MediaEngine,
> CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> projects.
>
> A task queue must be created and destroyed from the same thread. With
> this CL that will be the worker thread, which creates and destroys
> WebRTCVoiceEngine. With the dependent CL, it will probably change to
> the signaling thread.
>
> NOTRY=True # tests just passed
>
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2896813002
> Cr-Commit-Position: refs/heads/master@{#18252}
> Committed: c61bf947b4

TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2904893002
Cr-Commit-Position: refs/heads/master@{#18255}

webrtc/media/BUILD.gn

@@ -220,11 +220,6 @@ rtc_static_library("rtc_media") {
     public_configs += [ ":rtc_media_defines_config" ]
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
-  if (rtc_enable_protobuf) {
-    deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ]
-  } else {
-    deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
-  }
   deps += [
     ":rtc_media_base",
     "..:webrtc_common",
@@ -242,7 +237,6 @@ rtc_static_library("rtc_media") {
     "../modules/audio_device:audio_device",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/audio_processing:audio_processing",
-    "../modules/audio_processing/aec_dump",
     "../modules/video_capture:video_capture_module",
     "../modules/video_coding",
     "../modules/video_coding:webrtc_h264",

webrtc/media/engine/webrtcvoiceengine.cc

@@ -38,7 +38,6 @@
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
-#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/metrics.h"
@@ -229,8 +228,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
     VoEWrapper* voe_wrapper)
-    : low_priority_worker_queue_("rtc-low-prio", rtc::TaskQueue::Priority::LOW),
-      adm_(adm),
+    : adm_(adm),
       encoder_factory_(encoder_factory),
       decoder_factory_(decoder_factory),
       voe_wrapper_(voe_wrapper) {
@@ -689,28 +687,46 @@ void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
 bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
                                      int64_t max_size_bytes) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  auto aec_dump = webrtc::AecDumpFactory::Create(file, max_size_bytes,
-                                                 &low_priority_worker_queue_);
-  if (!aec_dump) {
+  FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
+  if (!aec_dump_file_stream) {
+    LOG(LS_ERROR) << "Could not open AEC dump file stream.";
+    if (!rtc::ClosePlatformFile(file))
+      LOG(LS_WARNING) << "Could not close file.";
     return false;
   }
-  apm()->AttachAecDump(std::move(aec_dump));
+  StopAecDump();
+  if (apm()->StartDebugRecording(aec_dump_file_stream, max_size_bytes) !=
+      webrtc::AudioProcessing::kNoError) {
+    LOG_RTCERR0(StartDebugRecording);
+    fclose(aec_dump_file_stream);
+    return false;
+  }
+  is_dumping_aec_ = true;
   return true;
 }
 
 void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-
-  auto aec_dump =
-      webrtc::AecDumpFactory::Create(filename, -1, &low_priority_worker_queue_);
-  if (aec_dump) {
-    apm()->AttachAecDump(std::move(aec_dump));
+  if (!is_dumping_aec_) {
+    // Start dumping AEC when we are not dumping.
+    if (apm()->StartDebugRecording(filename.c_str(), -1) !=
+        webrtc::AudioProcessing::kNoError) {
+      LOG_RTCERR1(StartDebugRecording, filename.c_str());
+    } else {
+      is_dumping_aec_ = true;
+    }
   }
 }
 
 void WebRtcVoiceEngine::StopAecDump() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  apm()->DetachAecDump();
+  if (is_dumping_aec_) {
+    // Stop dumping AEC when we are dumping.
+    if (apm()->StopDebugRecording() != webrtc::AudioProcessing::kNoError) {
+      LOG_RTCERR0(StopDebugRecording);
+    }
+    is_dumping_aec_ = false;
+  }
 }
 
 int WebRtcVoiceEngine::CreateVoEChannel() {

webrtc/media/engine/webrtcvoiceengine.h

@@ -22,7 +22,6 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
@@ -111,9 +110,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
-
-  rtc::TaskQueue low_priority_worker_queue_;
-
   webrtc::AudioDeviceModule* adm();
   webrtc::AudioProcessing* apm();
   webrtc::voe::TransmitMixer* transmit_mixer();

webrtc/media/engine/webrtcvoiceengine_unittest.cc

@@ -125,7 +125,6 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
   EXPECT_CALL(apm, ApplyConfig(testing::_));
   EXPECT_CALL(apm, SetExtraOptions(testing::_));
   EXPECT_CALL(apm, Initialize()).WillOnce(Return(0));
-  EXPECT_CALL(apm, DetachAecDump());
   StrictMock<MockTransmitMixer> transmit_mixer;
   EXPECT_CALL(transmit_mixer, EnableStereoChannelSwapping(false));
   cricket::FakeWebRtcVoiceEngine voe(&apm, &transmit_mixer);
@@ -164,7 +163,6 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     EXPECT_CALL(apm_, ApplyConfig(testing::_));
     EXPECT_CALL(apm_, SetExtraOptions(testing::_));
     EXPECT_CALL(apm_, Initialize()).WillOnce(Return(0));
-    EXPECT_CALL(apm_, DetachAecDump());
     // Default Options.
     EXPECT_CALL(apm_ec_, Enable(true)).WillOnce(Return(0));
     EXPECT_CALL(apm_ec_, enable_metrics(true)).WillOnce(Return(0));