First unit tests for WebRTCAudioDevice.

content/test/webrtc_audio_device_test.cc

Index: content/test/webrtc_audio_device_test.cc
===================================================================
--- content/test/webrtc_audio_device_test.cc	(revision 0)
+++ content/test/webrtc_audio_device_test.cc	(working copy)
@@ -0,0 +1,302 @@
+// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "content/test/webrtc_audio_device_test.h"

[Paweł Hajdan Jr., 2011/11/02 18:00:24]

Ooops, one thing I didn't initially notice is that this is placed in
content/test . Please don't do that. It should only be used for testing
infrastructure, and tests should live as close to code being tested as possible.

[tommi (sloooow) - chröme, 2011/11/03 07:56:52]

You didn't notice before because this file wasn't here the first time around :)

Due to deps problems I split up the test file to three files.  A .cc/.h pair in
content/test that can include from content/browser and a _unittest.cc file that
has (or will have) the unit tests themselves in the same directory as the code
being tested.

When I ran into the deps problem, I figured I had two options:
a) Modify the DEPS files (or create a new one) to allow me to include from
content/browser.
b) Split the files up like I did.

I discussed these with other engs and we found other examples of doing it this
way.  Besides, I'm not familiar enough with the background behind the content/
include rules.

Some of the files I depend on might belong in content/test actually, like
mock_media_observer.*, but not others like resource_context.h

How do you propose we solve this?

[Paweł Hajdan Jr., 2011/11/04 07:51:32]

Ah, I see. Do you expect more tests to use webrtc_audio_device_test.h though? As
an alternative you could create content/renderer/test and put the test there and
add a DEPS file. I think that's how some renderer_host test deps were solved.

[tommi (sloooow) - chröme, 2011/11/04 08:59:00]

More tests are going to be added but I don't know if there will be more files
than the _unittest file in this cl that will use it.  Hmm.. actually, some of
this code could be shared with the audio_renderer tests.

+
+#include "base/bind.h"
+#include "base/message_loop.h"
+#include "base/synchronization/waitable_event.h"
+#include "base/test/signaling_task.h"
+#include "base/test/test_timeouts.h"
+#include "content/browser/renderer_host/media/audio_renderer_host.h"
+#include "content/browser/renderer_host/media/mock_media_observer.h"
+#include "content/browser/resource_context.h"
+#include "content/common/view_messages.h"
+#include "content/public/common/content_paths.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/render_process.h"
+#include "content/renderer/render_thread_impl.h"
+#include "content/test/test_browser_thread.h"
+#include "net/url_request/url_request_test_util.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h"
+#include "third_party/webrtc/voice_engine/main/interface/voe_base.h"
+#include "third_party/webrtc/voice_engine/main/interface/voe_file.h"
+#include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
+
+using testing::_;
+using testing::InvokeWithoutArgs;
+using testing::Return;
+using testing::StrEq;
+
+// This class is a mock of the child process singleton which is needed
+// to be able to create a RenderThread object.
+class WebRTCMockRenderProcess : public RenderProcess {
+ public:
+  WebRTCMockRenderProcess() {}
+  virtual ~WebRTCMockRenderProcess() {}
+
+  // RenderProcess implementation.
+  virtual skia::PlatformCanvas* GetDrawingCanvas(TransportDIB** memory,
+      const gfx::Rect& rect) { return NULL; }
+  virtual void ReleaseTransportDIB(TransportDIB* memory) {}
+  virtual bool UseInProcessPlugins() const { return false; }
+  virtual bool HasInitializedMediaLibrary() const { return false; }
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(WebRTCMockRenderProcess);
+};
+
+namespace {
+
+class WebRTCMockResourceContext : public content::ResourceContext {
+ public:
+  WebRTCMockResourceContext() {}
+  virtual ~WebRTCMockResourceContext() {}
+  virtual void EnsureInitialized() const OVERRIDE {}
+};
+
+ACTION_P(QuitMessageLoop, loop_or_proxy) {
+  LOG(WARNING) << __FUNCTION__;

[Paweł Hajdan Jr., 2011/11/02 18:00:24]

nit: Isn't this cluttering the logs too much? Ideally a passing test should
generate no logs.

[tommi (sloooow) - chröme, 2011/11/03 07:56:52]

This has been removed. It was there just for my benefit while debugging.

+  loop_or_proxy->PostTask(FROM_HERE, new MessageLoop::QuitTask());
+}
+
+}  // end namespace
+
+WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
+    : render_thread_(NULL), event_(false, false), audio_util_callback_(NULL) {}
+WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {}
+
+void WebRTCAudioDeviceTest::SetUp() {
+  // Set low latency mode, as it soon would be on by default.
+  if (AudioRendererImpl::latency_type() != AudioRendererImpl::kLowLatency)
+    AudioRendererImpl::set_latency_type(AudioRendererImpl::kLowLatency);
+
+  DCHECK_EQ(AudioRendererImpl::kLowLatency,

[Paweł Hajdan Jr., 2011/11/02 18:00:24]

nit: I think ASSERT_EQ would fit better (not crashing entire test binary).

[tommi (sloooow) - chröme, 2011/11/03 07:56:52]

agreed.

+            AudioRendererImpl::latency_type());
+
+  // This part sets up a RenderThread environment to ensure that
+  // RenderThread::current() (<=> TLS pointer) is valid.
+  // Main parts are inspired by the RenderViewFakeResourcesTest.
+  // Note that, the IPC part is not utilized in this test.
+  saved_content_renderer_ = content::GetContentClient()->renderer();

[Paweł Hajdan Jr., 2011/11/02 18:00:24]

Could you create a scoped object for this and put it in content/test? Restoring
is better than not doing it, but a scoper is even harder to misuse.

[tommi (sloooow) - chröme, 2011/11/03 07:56:52]

Will do. (I'm at home right now, will post an update later)

+  content::GetContentClient()->set_renderer(&mock_content_renderer_client_);
+  mock_process_.reset(new WebRTCMockRenderProcess());
+  ui_thread_.reset(new content::TestBrowserThread(BrowserThread::UI,
+                                                  MessageLoop::current()));
+
+  // Construct the resource context on the UI thread.
+  resource_context_.reset(new WebRTCMockResourceContext());
+
+  static const char kThreadName[] = "RenderThread";
+  ChildProcess::current()->io_message_loop()->PostTask(
+      FROM_HERE,
+      base::Bind(&SetupTask::InitializeIOThread, new SetupTask(this),
+                 kThreadName));
+  WaitForIOThreadCompletion();
+
+  render_thread_ = new RenderThreadImpl(kThreadName);
+  mock_process_->set_main_thread(render_thread_);
+}
+
+void WebRTCAudioDeviceTest::TearDown() {
+  ChildProcess::current()->io_message_loop()->PostTask(
+      FROM_HERE,
+      base::Bind(&SetupTask::UninitializeIOThread, new SetupTask(this)));
+  WaitForIOThreadCompletion();
+  mock_process_.reset();
+
+  content::GetContentClient()->set_renderer(saved_content_renderer_);
+}
+
+bool WebRTCAudioDeviceTest::Send(IPC::Message* message) {
+  return channel_->Send(message);
+}
+
+void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) {
+  // Set the current thread as the IO thread.
+  io_thread_.reset(new content::TestBrowserThread(BrowserThread::IO,
+                                                  MessageLoop::current()));
+  test_request_context_ = new TestURLRequestContext();
+  resource_context_->set_request_context(test_request_context_.get());
+  media_observer_.reset(new MockMediaObserver());
+  resource_context_->set_media_observer(media_observer_.get());
+
+  CreateChannel(thread_name, resource_context_.get());
+}
+
+void WebRTCAudioDeviceTest::UninitializeIOThread() {
+  DestroyChannel();
+  resource_context_.reset();
+  test_request_context_ = NULL;
+}
+
+void WebRTCAudioDeviceTest::CreateChannel(const char* name,
+    content::ResourceContext* resource_context) {
+  DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
+  audio_render_host_ = new AudioRendererHost(resource_context);
+  audio_render_host_->OnChannelConnected(base::GetCurrentProcId());
+
+  channel_.reset(new IPC::Channel(name, IPC::Channel::MODE_SERVER, this));
+  ASSERT_TRUE(channel_->Connect());
+
+  audio_render_host_->OnFilterAdded(channel_.get());
+}
+
+void WebRTCAudioDeviceTest::DestroyChannel() {
+  DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
+  channel_.reset();
+  audio_render_host_ = NULL;
+}
+
+void WebRTCAudioDeviceTest::OnGetHardwareSampleRate(double* sample_rate) {
+  DLOG_IF(WARNING, !audio_util_callback_) << "Can't get output sample rate";
+  *sample_rate = audio_util_callback_ ?
+      audio_util_callback_->GetAudioHardwareSampleRate() : 0.0;
+}
+
+void WebRTCAudioDeviceTest::OnGetHardwareInputSampleRate(double* sample_rate) {
+  DLOG_IF(WARNING, !audio_util_callback_) << "Can't get input sample rate";

[Paweł Hajdan Jr., 2011/11/02 18:00:24]

Should that possibly be EXCEPT_TRUE? I generally don't like "warnings" that
don't make the test fails. Either this condition means things are broken, or
it's possibly not really a warning.

[tommi (sloooow) - chröme, 2011/11/03 07:56:52]

Yes, that makes sense as things are right now.

+  *sample_rate = audio_util_callback_ ?
+      audio_util_callback_->GetAudioInputHardwareSampleRate() : 0.0;
+}
+
+// IPC::Channel::Listener implementation.
+bool WebRTCAudioDeviceTest::OnMessageReceived(const IPC::Message& message) {
+  if (render_thread_) {
+    IPC::ChannelProxy::MessageFilter* filter =
+        render_thread_->audio_input_message_filter();
+    if (filter->OnMessageReceived(message))
+      return true;
+
+    filter = render_thread_->audio_message_filter();
+    if (filter->OnMessageReceived(message))
+      return true;
+  }
+
+  if (audio_render_host_.get()) {
+    bool message_was_ok = false;
+    if (audio_render_host_->OnMessageReceived(message, &message_was_ok))
+      return true;
+  }
+
+  bool handled = true;
+  bool message_is_ok = true;

[Paweł Hajdan Jr., 2011/11/02 18:00:24]

Now could you EXPECT_TRUE(message_is_ok) after END_MESSAGE_MAP_EX?

[tommi (sloooow) - chröme, 2011/11/03 07:56:52]

Sure

+  IPC_BEGIN_MESSAGE_MAP_EX(WebRTCAudioDeviceTest, message, message_is_ok)
+    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareSampleRate,
+                        OnGetHardwareSampleRate)
+    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputSampleRate,
+                        OnGetHardwareInputSampleRate)
+    IPC_MESSAGE_UNHANDLED(handled = false)
+  IPC_END_MESSAGE_MAP_EX()
+
+  if (!handled) {
+    // We don't do a NOTREACHED/NOTIMPLEMENTED here since this is FYI

[Paweł Hajdan Jr., 2011/11/02 18:00:24]

I think it'd be much simpler and cleaner to just say EXPECT_TRUE(handled) here.

A log that is only done in Debug mode might be missed by someone working in
Release mode, and if the test produces lots of other output it's quite easy to
miss anyway.

If you still want to make it non-fatal, consider converting DLOG -> LOG, and
please make the comment shorter (no more than 1-2 lines), like "Unhandled
messages are not fatal, but may be indication of problems.".

Also, maybe (D)LOG_IF would be slightly shorter.

[tommi (sloooow) - chröme, 2011/11/03 07:56:52]

Right now there are 1-2 such cases for each unit test but that's fine.
I would like to avoid entries in the release output log that could indicate an
error while the test is succeeding.

I'll change it to DLOG_IF

+    // for the developer in case the test stops working.  If new IPC messages
+    // are added but not handled in the test for some reason, the test might
+    // break. This DLOG can save time in tracking down why.
+    DLOG(WARNING) << "Unhandled IPC message";
+  }
+
+  return true;
+}
+
+// Posts a final task to the IO message loop and waits for completion.
+void WebRTCAudioDeviceTest::WaitForIOThreadCompletion() {
+  ChildProcess::current()->io_message_loop()->PostTask(
+      FROM_HERE, new base::SignalingTask(&event_));
+  EXPECT_TRUE(event_.TimedWait(
+      base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
+}
+
+std::string WebRTCAudioDeviceTest::GetTestDataPath(
+    const FilePath::StringType& file_name) {
+  FilePath path;
+  EXPECT_TRUE(PathService::Get(content::DIR_TEST_DATA, &path));
+  path = path.Append(file_name);
+#ifdef OS_WIN
+  return WideToUTF8(path.value());
+#else
+  return path.value();
+#endif
+}
+
+void WebRTCAudioDeviceTest::PlayLocalFile(int duration) {
+  EXPECT_GE(duration, 0);
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
+
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamPlaying(_, 1, true)).Times(1);
+
+  // When the "closed" event is triggered, we end the test.
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamStatus(_, 1, StrEq("closed")))
+          .WillOnce(QuitMessageLoop(message_loop_.message_loop_proxy()));
+
+  EXPECT_CALL(media_observer(),
+      OnDeleteAudioStream(_, 1)).Times(1);
+
+  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+      new WebRtcAudioDeviceImpl());
+  audio_device->SetSessionId(1);
+
+  WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
+
+  ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
+  int err = base->Init(audio_device);
+  EXPECT_EQ(0, err);
+  if (err == 0) {
+    ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());
+    EXPECT_EQ(0, audio_processing->SetAgcStatus(true,
+        webrtc::kAgcAdaptiveDigital));
+
+    int ch = base->CreateChannel();
+    EXPECT_NE(-1, ch);
+
+    ScopedWebRTCPtr<webrtc::VoENetwork> network(engine.get());
+    scoped_ptr<WebRTCTransportImpl> transport(
+        new WebRTCTransportImpl(network.get()));
+    EXPECT_EQ(0, network->RegisterExternalTransport(ch, *transport.get()));
+    EXPECT_EQ(0, base->StartReceive(ch));
+    EXPECT_EQ(0, base->StartPlayout(ch));
+    EXPECT_EQ(0, base->StartSend(ch));
+
+    std::string file_path(
+        GetTestDataPath(FILE_PATH_LITERAL("speechmusic_mono_16kHz.pcm")));
+
+    ScopedWebRTCPtr<webrtc::VoEFile> file(engine.get());
+    if (duration == 0) {
+      EXPECT_EQ(0, file->GetFileDuration(file_path.c_str(), duration,
+                                         webrtc::kFileFormatPcm16kHzFile));
+      EXPECT_NE(0, duration);
+    }
+
+    EXPECT_EQ(0, file->StartPlayingFileLocally(ch, file_path.c_str(), false,
+        webrtc::kFileFormatPcm16kHzFile));
+
+    message_loop_.PostDelayedTask(FROM_HERE,
+                                  new MessageLoop::QuitTask(), duration);
+    message_loop_.Run();
+
+    EXPECT_EQ(0, network->DeRegisterExternalTransport(ch));
+  }
+}
+
+WebRTCTransportImpl::WebRTCTransportImpl(webrtc::VoENetwork* network)
+    : network_(network) {
+}
+
+WebRTCTransportImpl::~WebRTCTransportImpl() {
+}
+
+int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
+  return network_->ReceivedRTPPacket(channel, data, len);
+}
+
+int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
+                                        int len) {
+  return network_->ReceivedRTCPPacket(channel, data, len);
+}
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