Low-bandwidth audio testing

webrtc/test/call_test.cc

Index: webrtc/test/call_test.cc
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 5bfcb4d1a15f0fa358f72afc803b062acd0fe31c..1b1ed51d93ccfa0f029cca76b36bc46771943150 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -50,6 +50,9 @@ void CallTest::RunBaseTest(BaseTest* test) {
   RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
   Call::Config send_config(test->GetSenderCallConfig());
   if (num_audio_streams_ > 0) {
+    CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
+    test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
+                                    fake_recv_audio_device_.get());

[stefan-webrtc, 2017/03/20 14:40:10]

It would be nice if this was done in roughly the same place as fake video
capturers are created, see lines 119-126. Maybe those can be moved up here, or
the other way around?

[oprypin_webrtc, 2017/03/20 15:09:16]

I have not changed the logic of CallTest here, only moved CreateFakeAudioDevices
outside of CreateVoiceEngines and made it pluggable.
This cannot be moved down because CreateVoiceEngines relies on the audio
devices. Lines 119-126 cannot be moved here because CreateFrameGeneratorCapturer
relies on the result of CreateVideoStreams.

[stefan-webrtc, 2017/03/20 15:42:32]

Ack

     CreateVoiceEngines();
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe_send_.voice_engine;
@@ -132,6 +135,8 @@ void CallTest::RunBaseTest(BaseTest* test) {
   DestroyCalls();
   if (num_audio_streams_ > 0)
     DestroyVoiceEngines();
+
+  test->OnTestFinished();
 }
 
 void CallTest::Start() {
@@ -298,11 +303,13 @@ void CallTest::CreateFrameGeneratorCapturer(int framerate,
       VideoSendStream::DegradationPreference::kBalanced);
 }
 
-void CallTest::CreateFakeAudioDevices() {
+void CallTest::CreateFakeAudioDevices(
+    std::unique_ptr<FakeAudioDevice::Capturer> capturer,
+    std::unique_ptr<FakeAudioDevice::Renderer> renderer) {
   fake_send_audio_device_.reset(new FakeAudioDevice(
-      FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000), nullptr, 1.f));
+      std::move(capturer), nullptr, 1.f));
   fake_recv_audio_device_.reset(new FakeAudioDevice(
-      nullptr, FakeAudioDevice::CreateDiscardRenderer(48000), 1.f));
+      nullptr, std::move(renderer), 1.f));
 }
 
 void CallTest::CreateVideoStreams() {
@@ -361,7 +368,6 @@ void CallTest::DestroyStreams() {
 }
 
 void CallTest::CreateVoiceEngines() {
-  CreateFakeAudioDevices();
   voe_send_.voice_engine = VoiceEngine::Create();
   voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
   EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
@@ -425,6 +431,18 @@ BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
 BaseTest::~BaseTest() {
 }
 
+std::unique_ptr<FakeAudioDevice::Capturer> BaseTest::CreateCapturer() {
+  return FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000);
+}
+
+std::unique_ptr<FakeAudioDevice::Renderer> BaseTest::CreateRenderer() {
+  return FakeAudioDevice::CreateDiscardRenderer(48000);
+}
+
+void BaseTest::OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
+                                         FakeAudioDevice* recv_audio_device) {
+}
+
 Call::Config BaseTest::GetSenderCallConfig() {
   return Call::Config(&event_log_);
 }
@@ -489,6 +507,9 @@ void BaseTest::OnFrameGeneratorCapturerCreated(
     FrameGeneratorCapturer* frame_generator_capturer) {
 }
 
+void BaseTest::OnTestFinished() {
+}
+
 SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }