Change the way that AecDumps are created in APM

This CL changes the way that AecDumps are created in APM. Instead
of being injected, they are now created via the API.

This removes the AecDumpFactory from the API surface of APM and
makes the API more explicit.

The CL will be followed by one more CL that deprecates the usage
of the AttachAecDump API also within the audio_processing
and the fuzzer folders.

The CL also moves the aec_dump.* files from the include folder
to the aec_dump folder and changes the build files. The reasons
for this are that
1) The content of aec_dump.h is not really part of the API
   surface of APM.
2) Those files anyway needed to be moved to a separate build-
   target to avoid a circular build-file dependency caused by
   the other changes in this CL

Bug: webrtc:5298
Change-Id: I7dd6b49de76eb44158472874e1d4ae17dca9be54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174750
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31207}

media/engine/webrtc_voice_engine.cc

@@ -587,14 +587,8 @@ bool WebRtcVoiceEngine::StartAecDump(webrtc::FileWrapper file,
     return false;
   }
 
-  auto aec_dump = webrtc::AecDumpFactory::Create(
+  return ap->CreateAndAttachAecDump(file.Release(), max_size_bytes,
-      std::move(file), max_size_bytes, low_priority_worker_queue_.get());
+                                    low_priority_worker_queue_.get());
-  if (!aec_dump) {
-    return false;
-  }
-
-  ap->AttachAecDump(std::move(aec_dump));
-  return true;
 }
 
 void WebRtcVoiceEngine::StopAecDump() {

modules/audio_processing/BUILD.gn

@@ -50,6 +50,7 @@ rtc_library("api") {
     "../../rtc_base:macromagic",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:arch",
+    "../../rtc_base/system:file_wrapper",
     "../../rtc_base/system:rtc_export",
     "agc:gain_control_interface",
     "//third_party/abseil-cpp/absl/types:optional",
@@ -112,6 +113,20 @@ rtc_library("high_pass_filter") {
   ]
 }
 
+rtc_source_set("aec_dump_interface") {
+  visibility = [ "*" ]
+  sources = [
+    "include/aec_dump.cc",
+    "include/aec_dump.h",
+  ]
+
+  deps = [
+    ":api",
+    ":audio_frame_view",
+    "../../rtc_base:deprecation",
+  ]
+}
+
 rtc_library("audio_processing") {
   visibility = [ "*" ]
   configs += [ ":apm_debug_dump" ]
@@ -134,8 +149,6 @@ rtc_library("audio_processing") {
     "gain_control_impl.h",
     "gain_controller2.cc",
     "gain_controller2.h",
-    "include/aec_dump.cc",
-    "include/aec_dump.h",
     "level_estimator.cc",
     "level_estimator.h",
     "render_queue_item_verifier.h",
@@ -147,6 +160,7 @@ rtc_library("audio_processing") {
 
   defines = []
   deps = [
+    ":aec_dump_interface",
     ":api",
     ":apm_logging",
     ":audio_buffer",
@@ -178,6 +192,7 @@ rtc_library("audio_processing") {
     "../../system_wrappers:field_trial",
     "../../system_wrappers:metrics",
     "aec3",
+    "aec_dump:aec_dump",
     "aecm:aecm_core",
     "agc",
     "agc:gain_control_interface",
@@ -198,6 +213,12 @@ rtc_library("audio_processing") {
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers",
   ]
+
+  if (rtc_enable_protobuf) {
+    deps += [ "aec_dump:aec_dump_impl" ]
+  } else {
+    deps += [ "aec_dump:null_aec_dump_factory" ]
+  }
 }
 
 rtc_library("voice_detection") {
@@ -283,6 +304,7 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "include/mock_audio_processing.h" ]
     deps = [
+      ":aec_dump_interface",
       ":api",
       ":audio_buffer",
       ":audio_processing",

modules/audio_processing/aec_dump/BUILD.gn

@@ -13,7 +13,7 @@ rtc_source_set("aec_dump") {
   sources = [ "aec_dump_factory.h" ]
 
   deps = [
-    "../",
+    "..:aec_dump_interface",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:file_wrapper",
     "../../../rtc_base/system:rtc_export",
@@ -29,6 +29,7 @@ if (rtc_include_tests) {
     ]
 
     deps = [
+      "..:aec_dump_interface",
       "..:audioproc_test_utils",
       "../",
       "../../../test:test_support",
@@ -64,7 +65,7 @@ if (rtc_enable_protobuf) {
 
     deps = [
       ":aec_dump",
-      "../",
+      "..:aec_dump_interface",
       "../../../api/audio:audio_frame_api",
       "../../../api/task_queue",
       "../../../rtc_base:checks",
@@ -104,6 +105,6 @@ rtc_library("null_aec_dump_factory") {
 
   deps = [
     ":aec_dump",
-    "../",
+    "..:aec_dump_interface",
   ]
 }

modules/audio_processing/audio_processing_impl.cc

@@ -22,6 +22,7 @@
 #include "api/audio/audio_frame.h"
 #include "common_audio/audio_converter.h"
 #include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "modules/audio_processing/agc2/gain_applier.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/common.h"
@@ -1532,6 +1533,32 @@ int AudioProcessingImpl::recommended_stream_analog_level() const {
   }
 }
 
+bool AudioProcessingImpl::CreateAndAttachAecDump(const std::string& file_name,
+                                                 int64_t max_log_size_bytes,
+                                                 rtc::TaskQueue* worker_queue) {
+  std::unique_ptr<AecDump> aec_dump =
+      AecDumpFactory::Create(file_name, max_log_size_bytes, worker_queue);
+  if (!aec_dump) {
+    return false;
+  }
+
+  AttachAecDump(std::move(aec_dump));
+  return true;
+}
+
+bool AudioProcessingImpl::CreateAndAttachAecDump(FILE* handle,
+                                                 int64_t max_log_size_bytes,
+                                                 rtc::TaskQueue* worker_queue) {
+  std::unique_ptr<AecDump> aec_dump =
+      AecDumpFactory::Create(handle, max_log_size_bytes, worker_queue);
+  if (!aec_dump) {
+    return false;
+  }
+
+  AttachAecDump(std::move(aec_dump));
+  return true;
+}
+
 void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
   RTC_DCHECK(aec_dump);
   rtc::CritScope cs_render(&crit_render_);

modules/audio_processing/audio_processing_impl.h

@@ -11,8 +11,11 @@
 #ifndef MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 #define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
+#include <stdio.h>
+
 #include <list>
 #include <memory>
+#include <string>
 #include <vector>
 
 #include "api/function_view.h"
@@ -70,6 +73,13 @@ class AudioProcessingImpl : public AudioProcessing {
   int Initialize(const ProcessingConfig& processing_config) override;
   void ApplyConfig(const AudioProcessing::Config& config) override;
   void SetExtraOptions(const webrtc::Config& config) override;
+  bool CreateAndAttachAecDump(const std::string& file_name,
+                              int64_t max_log_size_bytes,
+                              rtc::TaskQueue* worker_queue) override;
+  bool CreateAndAttachAecDump(FILE* handle,
+                              int64_t max_log_size_bytes,
+                              rtc::TaskQueue* worker_queue) override;
+  // TODO(webrtc:5298) Deprecated variant.
   void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
   void DetachAecDump() override;
   void SetRuntimeSetting(RuntimeSetting setting) override;

modules/audio_processing/include/audio_processing.h

@@ -33,8 +33,13 @@
 #include "rtc_base/arraysize.h"
 #include "rtc_base/deprecation.h"
 #include "rtc_base/ref_count.h"
+#include "rtc_base/system/file_wrapper.h"
 #include "rtc_base/system/rtc_export.h"
 
+namespace rtc {
+class TaskQueue;
+}  // namespace rtc
+
 namespace webrtc {
 
 class AecDump;
@@ -600,6 +605,23 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
   // with this chunk of audio.
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
 
+  // Creates and attaches an webrtc::AecDump for recording debugging
+  // information.
+  // The |worker_queue| may not be null and must outlive the created
+  // AecDump instance. |max_log_size_bytes == -1| means the log size
+  // will be unlimited. |handle| may not be null. The AecDump takes
+  // responsibility for |handle| and closes it in the destructor. A
+  // return value of true indicates that the file has been
+  // sucessfully opened, while a value of false indicates that
+  // opening the file failed.
+  virtual bool CreateAndAttachAecDump(const std::string& file_name,
+                                      int64_t max_log_size_bytes,
+                                      rtc::TaskQueue* worker_queue) = 0;
+  virtual bool CreateAndAttachAecDump(FILE* handle,
+                                      int64_t max_log_size_bytes,
+                                      rtc::TaskQueue* worker_queue) = 0;
+
+  // TODO(webrtc:5298) Deprecated variant.
   // Attaches provided webrtc::AecDump for recording debugging
   // information. Log file and maximum file size logic is supposed to
   // be handled by implementing instance of AecDump. Calling this

modules/audio_processing/include/mock_audio_processing.h

@@ -13,7 +13,6 @@
 
 #include <memory>
 
-#include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/include/aec_dump.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
@@ -128,8 +127,15 @@ class MockAudioProcessing : public ::testing::NiceMock<AudioProcessing> {
   MOCK_CONST_METHOD0(delay_offset_ms, int());
   MOCK_METHOD1(set_stream_analog_level, void(int));
   MOCK_CONST_METHOD0(recommended_stream_analog_level, int());
-
+  MOCK_METHOD3(CreateAndAttachAecDump,
-  virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) {}
+               bool(const std::string& file_name,
+                    int64_t max_log_size_bytes,
+                    rtc::TaskQueue* worker_queue));
+  MOCK_METHOD3(CreateAndAttachAecDump,
+               bool(FILE* handle,
+                    int64_t max_log_size_bytes,
+                    rtc::TaskQueue* worker_queue));
+  MOCK_METHOD1(AttachAecDump, void(std::unique_ptr<AecDump>));
   MOCK_METHOD0(DetachAecDump, void());
 
   MOCK_METHOD0(GetStatistics, AudioProcessingStats());

test/pc/e2e/test_peer_factory.cc

@@ -292,8 +292,8 @@ std::unique_ptr<TestPeer> TestPeerFactory::CreateTestPeer(
   rtc::scoped_refptr<AudioProcessing> audio_processing =
       webrtc::AudioProcessingBuilder().Create();
   if (params->aec_dump_path && audio_processing) {
-    audio_processing->AttachAecDump(
+    audio_processing->CreateAndAttachAecDump(*params->aec_dump_path, -1,
-        AecDumpFactory::Create(*params->aec_dump_path, -1, task_queue));
+                                             task_queue);
   }
   rtc::scoped_refptr<AudioDeviceModule> audio_device_module =
       CreateAudioDeviceModule(