Migrate CreateVoipEngine to take audio_processing_factory instead of audio_processing

This would allow users of the voip engine to migrate away from the AudioProcessingBuilder

Bug: webrtc:369904700
Change-Id: Ie4f6f4579e185ff6366333a3f37e6aaa23b892b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365920
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43255}

api/voip/BUILD.gn

@@ -43,6 +43,7 @@ rtc_library("voip_engine_factory") {
     "../audio:audio_device",
     "../audio:audio_processing",
     "../audio_codecs:audio_codecs_api",
+    "../environment",
     "../environment:environment_factory",
     "../task_queue",
   ]

api/voip/test/voip_engine_factory_unittest.cc

@@ -10,6 +10,7 @@
 
 #include "api/voip/voip_engine_factory.h"
 
+#include <memory>
 #include <utility>
 
 #include "api/make_ref_counted.h"
@@ -24,14 +25,16 @@
 namespace webrtc {
 namespace {
 
+using ::testing::NiceMock;
+
 // Create voip engine with mock modules as normal use case.
 TEST(VoipEngineFactoryTest, CreateEngineWithMockModules) {
   VoipEngineConfig config;
   config.encoder_factory = rtc::make_ref_counted<MockAudioEncoderFactory>();
   config.decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
   config.task_queue_factory = CreateDefaultTaskQueueFactory();
-  config.audio_processing =
-      rtc::make_ref_counted<testing::NiceMock<test::MockAudioProcessing>>();
+  config.audio_processing_factory =
+      std::make_unique<NiceMock<test::MockAudioProcessingFactory>>();
   config.audio_device_module = test::MockAudioDeviceModule::CreateNice();
 
   auto voip_engine = CreateVoipEngine(std::move(config));

api/voip/voip_engine_factory.cc

@@ -13,7 +13,10 @@
 #include <memory>
 #include <utility>
 
+#include "api/audio/audio_processing.h"
+#include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
+#include "api/scoped_refptr.h"
 #include "api/voip/voip_engine.h"
 #include "audio/voip/voip_core.h"
 #include "rtc_base/checks.h"
@@ -27,15 +30,26 @@ std::unique_ptr<VoipEngine> CreateVoipEngine(VoipEngineConfig config) {
   RTC_CHECK(config.task_queue_factory);
   RTC_CHECK(config.audio_device_module);
 
-  if (!config.audio_processing) {
+  Environment env = CreateEnvironment(std::move(config.task_queue_factory));
+
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+  RTC_CHECK(config.audio_processing == nullptr ||
+            config.audio_processing_factory == nullptr);
+  scoped_refptr<AudioProcessing> audio_processing =
+      std::move(config.audio_processing);
+#pragma clang diagnostic pop
+  if (config.audio_processing_factory != nullptr) {
+    audio_processing = config.audio_processing_factory->Create(env);
+  }
+
+  if (audio_processing == nullptr) {
     RTC_DLOG(LS_INFO) << "No audio processing functionality provided.";
   }
 
   return std::make_unique<VoipCore>(
-      CreateEnvironment(std::move(config.task_queue_factory)),
-      std::move(config.encoder_factory), std::move(config.decoder_factory),
-      std::move(config.audio_device_module),
-      std::move(config.audio_processing));
+      env, std::move(config.encoder_factory), std::move(config.decoder_factory),
+      std::move(config.audio_device_module), std::move(audio_processing));
 }
 
 }  // namespace webrtc

api/voip/voip_engine_factory.h

@@ -28,17 +28,27 @@ namespace webrtc {
 // marked with comments as either mandatory or optional and default
 // implementations that applications can use.
 struct VoipEngineConfig {
+  // TODO: bugs.webrtc.org/369904700 - Remove explicit default constructors
+  // when deprecated `audio_processing` is removed and thus implicit
+  // constructors won't be considered deprecated.
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+  VoipEngineConfig() = default;
+  VoipEngineConfig(VoipEngineConfig&&) = default;
+  VoipEngineConfig& operator=(VoipEngineConfig&&) = default;
+#pragma clang diagnostic pop
+
   // Mandatory (e.g. api/audio_codec/builtin_audio_encoder_factory).
   // AudioEncoderFactory provides a set of audio codecs for VoipEngine to encode
   // the audio input sample. Application can choose to limit the set to reduce
   // application footprint.
-  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
+  scoped_refptr<AudioEncoderFactory> encoder_factory;
 
   // Mandatory (e.g. api/audio_codec/builtin_audio_decoder_factory).
   // AudioDecoderFactory provides a set of audio codecs for VoipEngine to decode
   // the received RTP packets from remote media endpoint. Application can choose
   // to limit the set to reduce application footprint.
-  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
+  scoped_refptr<AudioDecoderFactory> decoder_factory;
 
   // Mandatory (e.g. api/task_queue/default_task_queue_factory).
   // TaskQeueuFactory provided for VoipEngine to work asynchronously on its
@@ -49,15 +59,19 @@ struct VoipEngineConfig {
   // AudioDeviceModule that periocally provides audio input samples from
   // recording device (e.g. microphone) and requests audio output samples to
   // play through its output device (e.g. speaker).
-  rtc::scoped_refptr<AudioDeviceModule> audio_device_module;
+  scoped_refptr<AudioDeviceModule> audio_device_module;
 
-  // Optional (e.g. modules/audio_processing/include).
+  // Optional (e.g. api/audio/builtin_audio_processing_factory).
   // AudioProcessing provides audio procesing functionalities (e.g. acoustic
   // echo cancellation, noise suppression, gain control, etc) on audio input
   // samples for VoipEngine. When optionally not set, VoipEngine will not have
   // such functionalities to perform on audio input samples received from
   // AudioDeviceModule.
-  rtc::scoped_refptr<AudioProcessing> audio_processing;
+  std::unique_ptr<AudioProcessingFactory> audio_processing_factory;
+
+  // TODO: bugs.webrtc.org/369904700 - Remove when users are migrated to set
+  // `audio_processing_factory` instead.
+  [[deprecated]] scoped_refptr<AudioProcessing> audio_processing;
 };
 
 // Creates a VoipEngine instance with provided VoipEngineConfig.

examples/androidvoip/BUILD.gn

@@ -56,6 +56,7 @@ if (is_android) {
 
     deps = [
       ":generated_jni",
+      "../../api/audio:builtin_audio_processing_factory",
       "../../rtc_base:async_packet_socket",
       "../../rtc_base:async_udp_socket",
       "../../rtc_base:logging",

examples/androidvoip/jni/android_voip_client.cc

@@ -20,6 +20,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
+#include "api/audio/builtin_audio_processing_factory.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/task_queue/default_task_queue_factory.h"
@@ -130,7 +131,8 @@ void AndroidVoipClient::Init(
   config.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
   config.audio_device_module =
       webrtc::CreateJavaAudioDeviceModule(env, application_context.obj());
-  config.audio_processing = webrtc::AudioProcessingBuilder().Create();
+  config.audio_processing_factory =
+      std::make_unique<webrtc::BuiltinAudioProcessingFactory>();
 
   voip_thread_->Start();