Use AudioDeviceModule instead of TestAudioDeviceModule.

This is step to allow migration of Test ADM to the AudioDeviceModuleImpl
as a base class to include AudioDeviceBuffer into SUT.

Also it will allow to remove WaitForRecordingEnd() method from Test
ADM

Bug: b/272350185, webrtc:15081
Change-Id: If2aa43ec0c31f6ad9aab8aa3e36cabc4a7a73c22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300862
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39849}

audio/BUILD.gn

@@ -134,6 +134,8 @@ if (rtc_include_tests) {
       "../api/task_queue",
       "../call:fake_network",
       "../call:simulated_network",
+      "../modules/audio_device:audio_device_api",
+      "../modules/audio_device:audio_device_impl",
       "../system_wrappers",
       "../test:test_common",
       "../test:test_support",

audio/test/audio_end_to_end_test.cc

@@ -16,16 +16,16 @@
 #include "api/task_queue/task_queue_base.h"
 #include "call/fake_network_pipe.h"
 #include "call/simulated_network.h"
+#include "modules/audio_device/include/test_audio_device.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/gtest.h"
 
 namespace webrtc {
 namespace test {
 namespace {
-// Wait half a second between stopping sending and stopping receiving audio.
-constexpr int kExtraRecordTimeMs = 500;
 
 constexpr int kSampleRate = 48000;
+
 }  // namespace
 
 AudioEndToEndTest::AudioEndToEndTest()
@@ -54,8 +54,8 @@ AudioEndToEndTest::CreateRenderer() {
 }
 
 void AudioEndToEndTest::OnFakeAudioDevicesCreated(
-    TestAudioDeviceModule* send_audio_device,
+    AudioDeviceModule* send_audio_device,
-    TestAudioDeviceModule* recv_audio_device) {
+    AudioDeviceModule* recv_audio_device) {
   send_audio_device_ = send_audio_device;
 }
 
@@ -81,11 +81,5 @@ void AudioEndToEndTest::OnAudioStreamsCreated(
   receive_stream_ = receive_streams[0];
 }
 
-void AudioEndToEndTest::PerformTest() {
-  // Wait until the input audio file is done...
-  send_audio_device_->WaitForRecordingEnd();
-  // and some extra time to account for network delay.
-  SleepMs(GetSendTransportConfig().queue_delay_ms + kExtraRecordTimeMs);
-}
 }  // namespace test
 }  // namespace webrtc

audio/test/audio_end_to_end_test.h

@@ -16,6 +16,8 @@
 
 #include "api/task_queue/task_queue_base.h"
 #include "api/test/simulated_network.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/test_audio_device.h"
 #include "test/call_test.h"
 
 namespace webrtc {
@@ -26,7 +28,7 @@ class AudioEndToEndTest : public test::EndToEndTest {
   AudioEndToEndTest();
 
  protected:
-  TestAudioDeviceModule* send_audio_device() { return send_audio_device_; }
+  AudioDeviceModule* send_audio_device() { return send_audio_device_; }
   const AudioSendStream* send_stream() const { return send_stream_; }
   const AudioReceiveStreamInterface* receive_stream() const {
     return receive_stream_;
@@ -39,9 +41,8 @@ class AudioEndToEndTest : public test::EndToEndTest {
   std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
   std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override;
 
-  void OnFakeAudioDevicesCreated(
+  void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
-      TestAudioDeviceModule* send_audio_device,
+                                 AudioDeviceModule* recv_audio_device) override;
-      TestAudioDeviceModule* recv_audio_device) override;
 
   void ModifyAudioConfigs(AudioSendStream::Config* send_config,
                           std::vector<AudioReceiveStreamInterface::Config>*
@@ -50,10 +51,8 @@ class AudioEndToEndTest : public test::EndToEndTest {
                              const std::vector<AudioReceiveStreamInterface*>&
                                  receive_streams) override;
 
-  void PerformTest() override;
-
  private:
-  TestAudioDeviceModule* send_audio_device_ = nullptr;
+  AudioDeviceModule* send_audio_device_ = nullptr;
   AudioSendStream* send_stream_ = nullptr;
   AudioReceiveStreamInterface* receive_stream_ = nullptr;
 };

audio/test/audio_stats_test.cc

@@ -17,6 +17,9 @@ namespace webrtc {
 namespace test {
 namespace {
 
+// Wait half a second between stopping sending and stopping receiving audio.
+constexpr int kExtraRecordTimeMs = 500;
+
 bool IsNear(int reference, int v) {
   // Margin is 10%.
   const int error = reference / 10 + 1;
@@ -41,7 +44,8 @@ class NoLossTest : public AudioEndToEndTest {
   void PerformTest() override {
     SleepMs(kTestDurationMs);
     send_audio_device()->StopRecording();
-    AudioEndToEndTest::PerformTest();
+    // and some extra time to account for network delay.
+    SleepMs(GetSendTransportConfig().queue_delay_ms + kExtraRecordTimeMs);
   }
 
   void OnStreamsStopped() override {

audio/test/low_bandwidth_audio_test.cc

@@ -57,7 +57,8 @@ class AudioQualityTest : public AudioEndToEndTest {
       // Let the recording run for a small amount of time to check if it works.
       SleepMs(1000);
     } else {
-      AudioEndToEndTest::PerformTest();
+      // Sleep for whole audio duration which is 5.4 seconds.
+      SleepMs(5400);
     }
   }
 

modules/audio_device/include/test_audio_device.cc

@@ -439,7 +439,7 @@ size_t TestAudioDeviceModule::SamplesPerFrame(int sampling_frequency_in_hz) {
   return rtc::CheckedDivExact(sampling_frequency_in_hz, kFramesPerSecond);
 }
 
-rtc::scoped_refptr<TestAudioDeviceModule> TestAudioDeviceModule::Create(
+rtc::scoped_refptr<AudioDeviceModule> TestAudioDeviceModule::Create(
     TaskQueueFactory* task_queue_factory,
     std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
     std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,

modules/audio_device/include/test_audio_device.h

@@ -82,7 +82,7 @@ class TestAudioDeviceModule : public AudioDeviceModule {
   // `renderer` is an object that receives audio data that would have been
   // played out. Can be nullptr if this device is never used for playing.
   // Use one of the Create... functions to get these instances.
-  static rtc::scoped_refptr<TestAudioDeviceModule> Create(
+  static rtc::scoped_refptr<AudioDeviceModule> Create(
       TaskQueueFactory* task_queue_factory,
       std::unique_ptr<Capturer> capturer,
       std::unique_ptr<Renderer> renderer,

test/BUILD.gn

@@ -1099,6 +1099,7 @@ rtc_library("test_common") {
     "../call:simulated_network",
     "../call:simulated_packet_receiver",
     "../call:video_stream_api",
+    "../modules/audio_device:audio_device_api",
     "../modules/audio_device:audio_device_impl",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/rtp_rtcp",

test/call_test.cc

@@ -22,6 +22,8 @@
 #include "call/fake_network_pipe.h"
 #include "call/packet_receiver.h"
 #include "call/simulated_network.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/test_audio_device.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
@@ -770,9 +772,9 @@ std::unique_ptr<TestAudioDeviceModule::Renderer> BaseTest::CreateRenderer() {
   return TestAudioDeviceModule::CreateDiscardRenderer(48000);
 }
 
-void BaseTest::OnFakeAudioDevicesCreated(
+void BaseTest::OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
-    TestAudioDeviceModule* send_audio_device,
+                                         AudioDeviceModule* recv_audio_device) {
-    TestAudioDeviceModule* recv_audio_device) {}
+}
 
 void BaseTest::ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) {}
 

test/call_test.h

@@ -26,6 +26,7 @@
 #include "api/units/time_delta.h"
 #include "api/video/video_bitrate_allocator_factory.h"
 #include "call/call.h"
+#include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_device/include/test_audio_device.h"
 #include "test/encoder_settings.h"
 #include "test/fake_decoder.h"
@@ -271,8 +272,8 @@ class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
   std::vector<RtpExtension> rtp_extensions_;
   rtc::scoped_refptr<AudioProcessing> apm_send_;
   rtc::scoped_refptr<AudioProcessing> apm_recv_;
-  rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
+  rtc::scoped_refptr<AudioDeviceModule> fake_send_audio_device_;
-  rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
+  rtc::scoped_refptr<AudioDeviceModule> fake_recv_audio_device_;
 };
 
 class BaseTest : public RtpRtcpObserver {
@@ -290,9 +291,8 @@ class BaseTest : public RtpRtcpObserver {
 
   virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
   virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
-  virtual void OnFakeAudioDevicesCreated(
+  virtual void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
-      TestAudioDeviceModule* send_audio_device,
+                                         AudioDeviceModule* recv_audio_device);
-      TestAudioDeviceModule* recv_audio_device);
 
   virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
   virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);