First unit tests for WebRTCAudioDevice.

content/renderer/media/webrtc_audio_device_unittest.cc

Index: content/renderer/media/webrtc_audio_device_unittest.cc
===================================================================
--- content/renderer/media/webrtc_audio_device_unittest.cc	(revision 0)
+++ content/renderer/media/webrtc_audio_device_unittest.cc	(working copy)
@@ -0,0 +1,482 @@
+// 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 "base/bind.h"
+#include "base/message_loop.h"
+#include "base/path_service.h"
+#include "base/process_util.h"
+#include "base/synchronization/waitable_event.h"
+#include "base/test/test_timeouts.h"
+#include "base/time.h"
+

[Paweł Hajdan Jr., 2011/11/02 09:29:15]

nit: We're not doing gaps like this in the #includes.

[tommi (sloooow) - chröme, 2011/11/02 10:59:23]

Done.

+#include "content/browser/mock_resource_context.h"
+#include "content/browser/renderer_host/media/audio_renderer_host.h"
+#include "content/browser/renderer_host/media/mock_media_observer.h"
+#include "content/common/child_process.h"
+#include "content/common/child_thread.h"
+#include "content/common/media/audio_messages.h"
+#include "content/common/view_messages.h"
+#include "content/public/common/content_paths.h"
+#include "content/renderer/media/audio_renderer_impl.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/mock_content_renderer_client.h"
+#include "content/renderer/render_process.h"
+#include "content/renderer/render_thread_impl.h"
+#include "content/test/test_browser_thread.h"
+
+#include "ipc/ipc_channel.h"
+
+#include "media/audio/audio_util.h"
+#include "media/base/data_buffer.h"
+#include "media/base/mock_callback.h"
+#include "media/base/mock_filter_host.h"
+#include "media/base/mock_filters.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;
+
+namespace {
+// This class is a mock of the child process singleton which is needed
+// to be able to create a RenderThread object.
+class MockRenderProcess : public RenderProcess {
+ public:
+  MockRenderProcess() {}
+  virtual ~MockRenderProcess() {}
+
+  // 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(MockRenderProcess);
+};
+
+class WebRTCMockResourceContext : public content::ResourceContext {
+ public:
+  WebRTCMockResourceContext() {}
+  virtual ~WebRTCMockResourceContext() {}
+  virtual void EnsureInitialized() const OVERRIDE {}
+};
+
+// Convenience class for WebRTC interfaces.  Fetches the wrapped interface
+// in the constructor via WebRTC's GetInterface mechanism and then releases
+// the reference in the destructor.
+template<typename T>
+class ScopedWebRTCPtr {
+ public:
+  template<typename Engine>
+  explicit ScopedWebRTCPtr(Engine* e)
+      : ptr_(T::GetInterface(e)) {}
+  explicit ScopedWebRTCPtr(T* p) : ptr_(p) {}
+  ~ScopedWebRTCPtr() { reset(); }
+  T* operator->() const { return ptr_; }
+  T* get() const { return ptr_; }
+
+  // Releases the current pointer.
+  void reset() {
+    if (ptr_) {
+      ptr_->Release();
+      ptr_ = NULL;
+    }
+  }
+
+  bool valid() const { return ptr_ != NULL; }
+
+ private:
+  T* ptr_;
+};
+
+// Wrapper to automatically calling T::Delete in the destructor.
+// This is useful for some WebRTC objects that have their own Create/Delete
+// methods and we can't use our our scoped_* classes.
+template <typename T>
+class AutoDelete {
+ public:
+  AutoDelete() : ptr_(NULL) {}
+  explicit AutoDelete(T* ptr) : ptr_(ptr) {}
+  ~AutoDelete() { reset(); }
+
+  void reset() {
+    if (ptr_) {
+      T::Delete(ptr_);
+      ptr_ = NULL;
+    }
+  }
+
+  T* operator->() { return ptr_; }
+  T* get() const { return ptr_; }
+
+  bool valid() const { return ptr_ != NULL; }
+
+ protected:
+  T* ptr_;
+};
+
+// This task signals an event owned by the task owner when run and quits.
+class SignalTask : public Task {

[Paweł Hajdan Jr., 2011/11/02 09:29:15]

We already have a class like this (chrome/test/base/signaling_task). Please move
it to base/test or content/test and re-use instead of duplicating.

[tommi (sloooow) - chröme, 2011/11/02 10:59:23]

http://codereview.chromium.org/8438034/

+ public:
+  explicit SignalTask(base::WaitableEvent* event) : event_(event) {}
+  virtual ~SignalTask() {}
+  virtual void Run() { event_->Signal(); }
+ private:
+  base::WaitableEvent* event_;
+  DISALLOW_COPY_AND_ASSIGN(SignalTask);
+};
+
+// Quits the current message loop when run and also sets a supplied bool
+// to true.  If the task never ran, the bool should remain false.
+class QuitAndRaiseTask : public Task {
+ public:
+  explicit QuitAndRaiseTask(bool* was_run) : was_run_(was_run) {
+    *was_run = false;
+  }
+
+  virtual void Run() {
+    *was_run_ = true;
+    MessageLoop::current()->Quit();
+  }
+ protected:
+  bool* was_run_;
+};
+
+ACTION_P(QuitMessageLoop, loop_or_proxy) {
+  LOG(WARNING) << __FUNCTION__;
+  loop_or_proxy->PostTask(FROM_HERE, new MessageLoop::QuitTask());
+}
+
+}  // end namespace
+
+// IPC::Channel::Listener implementation for the WebRTCAudioDeviceTest test
+// fixture.  Provides basic implementations for initialization tasks and
+// forwards callbacks to audio ipc filters.
+template <class T>
+class WebRTCAudioDeviceTestChannelListener : public IPC::Channel::Listener {
+ public:
+  WebRTCAudioDeviceTestChannelListener() {}
+  ~WebRTCAudioDeviceTestChannelListener() {}
+
+  void OnGetHardwareSampleRate(double* sample_rate) {
+    *sample_rate = media::GetAudioHardwareSampleRate();
+  }
+
+  void OnGetHardwareInputSampleRate(double* sample_rate) {
+    *sample_rate = media::GetAudioInputHardwareSampleRate();
+  }
+
+  bool Send(IPC::Message* message) {
+    return channel_->Send(message);
+  }
+
+  // IPC::Channel::Listener implementation.
+  virtual bool OnMessageReceived(const IPC::Message& message) {
+    T* me = static_cast<T*>(this);
+    if (me->render_thread()) {
+      IPC::ChannelProxy::MessageFilter* filter =
+          me->render_thread()->audio_input_message_filter();
+      if (filter->OnMessageReceived(message))
+        return true;
+
+      filter = me->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;
+    IPC_BEGIN_MESSAGE_MAP(WebRTCAudioDeviceTest, message)

[Paweł Hajdan Jr., 2011/11/02 09:29:15]

Please use MESSAGE_MAP_EX to avoid an implicit DCHECK in MESSAGE_MAP.

[tommi (sloooow) - chröme, 2011/11/02 10:59:23]

Done.

+      IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareSampleRate,
+                          OnGetHardwareSampleRate)
+      IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputSampleRate,
+                          OnGetHardwareInputSampleRate)
+      IPC_MESSAGE_UNHANDLED(handled = false)
+    IPC_END_MESSAGE_MAP()
+
+    if (!handled) {
+      DLOG(WARNING) << "Unhandled IPC message";

[Paweł Hajdan Jr., 2011/11/02 09:29:15]

Don't you want to make the test fail in that case? ADD_FAILURE could be what you
really want.

[tommi (sloooow) - chröme, 2011/11/02 10:59:23]

I added this comment:

    if (!handled) {
      // We don't do a NOTREACHED/NOTIMPLEMENTED here since this is FYI
      // 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;
+  }
+
+ protected:
+  void 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 DestroyChannel() {
+    DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
+    channel_.reset();
+    audio_render_host_ = NULL;
+  }
+
+  scoped_ptr<IPC::Channel> channel_;
+  scoped_refptr<AudioRendererHost> audio_render_host_;
+};
+
+class WebRTCAudioDeviceTest
+    : public ::testing::Test,
+      public WebRTCAudioDeviceTestChannelListener<WebRTCAudioDeviceTest> {
+ public:
+  class SetupTask : public base::RefCountedThreadSafe<SetupTask> {
+   public:
+    explicit SetupTask(WebRTCAudioDeviceTest* test) : test_(test) {}
+    void InitializeIOThread(const char* thread_name) {
+      test_->InitializeIOThread(thread_name);
+    }
+    void UninitializeIOThread() { test_->UninitializeIOThread(); }
+   protected:
+    WebRTCAudioDeviceTest* test_;
+  };
+
+  WebRTCAudioDeviceTest() : render_thread_(NULL), event_(false, false) {}
+  virtual ~WebRTCAudioDeviceTest() {}
+
+  static void SetUpTestCase() {

[Paweł Hajdan Jr., 2011/11/02 09:29:15]

Shouldn't you have a corresponding TearDownTestCase? Also, why not just in
SetUp? The operation doesn't seem to be expensive.

[tommi (sloooow) - chröme, 2011/11/02 10:59:23]

This fixture is based in parts on the AudioRendererImplTest fixture (see
audio_renderer_impl_unittest.cc), where this part is done the same way.

I think it's fine to just move this to SetUp though, so I've do that.

+    // Set low latency mode, as it soon would be on by default.
+    if (AudioRendererImpl::latency_type() ==
+        AudioRendererImpl::kUninitializedLatency) {
+      AudioRendererImpl::set_latency_type(AudioRendererImpl::kLowLatency);
+    }
+    DCHECK_EQ(AudioRendererImpl::kLowLatency,
+              AudioRendererImpl::latency_type());
+  }
+
+  virtual void SetUp() {
+    // 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.
+    content::GetContentClient()->set_renderer(&mock_content_renderer_client_);

[Paweł Hajdan Jr., 2011/11/02 09:29:15]

Is this undone in TearDown? Please use a scoped object to guarantee this.

[tommi (sloooow) - chröme, 2011/11/02 10:59:23]

This is another thing that is borrowed from AudioRendererImplTest.  Do you mind
taking a look at it? Perhaps there are some bugs we could fix there as well.

+    mock_process_.reset(new MockRenderProcess);
+    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_);
+  }
+
+  virtual void TearDown() {
+    ChildProcess::current()->io_message_loop()->PostTask(
+        FROM_HERE,
+        base::Bind(&SetupTask::UninitializeIOThread, new SetupTask(this)));
+    WaitForIOThreadCompletion();
+    mock_process_.reset();
+  }
+
+  RenderThreadImpl* render_thread() const {
+    return render_thread_;
+  }
+
+ protected:
+  void 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 UninitializeIOThread() {
+    DestroyChannel();
+    resource_context_.reset();
+    test_request_context_ = NULL;
+  }
+
+  // Posts a final task to the IO message loop and waits for completion.
+  void WaitForIOThreadCompletion() {
+    ChildProcess::current()->io_message_loop()->PostTask(
+        FROM_HERE, new SignalTask(&event_));
+    EXPECT_TRUE(event_.TimedWait(
+        base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
+  }
+
+  // Convenience getter for gmock.
+  MockMediaObserver& media_observer() const {
+    return *media_observer_.get();
+  }
+
+  std::string GetTestDataPath(const FilePath::StringType& file_name) {
+    FilePath path;
+    PathService::Get(content::DIR_TEST_DATA, &path);

[Paweł Hajdan Jr., 2011/11/02 09:29:15]

Check the return value.

[tommi (sloooow) - chröme, 2011/11/02 10:59:23]

Done.

+    path = path.Append(file_name);
+#ifdef OS_WIN
+    return WideToUTF8(path.value());
+#else
+    return path.value();
+#endif
+  }
+
+  // Actual implementation of the short and long PlayFile tests.
+  // If the |duration| parameter is 0, then the function determines the length
+  // of the audio file and plays the entire file.  Otherwise, |duration|
+  // specifies the number of milliseconds to allow the file to play.
+  void PlayLocalFile(int duration);
+
+  MessageLoopForUI message_loop_;
+  content::MockContentRendererClient mock_content_renderer_client_;
+  RenderThreadImpl* render_thread_;  // owned by mock_process_
+  scoped_ptr<MockRenderProcess> mock_process_;
+  media::MockFilterHost host_;
+  base::WaitableEvent event_;
+  scoped_ptr<MockMediaObserver> media_observer_;
+  scoped_ptr<content::ResourceContext> resource_context_;
+  scoped_refptr<net::URLRequestContext> test_request_context_;
+
+  // Initialized on the main test thread that we mark as the UI thread.
+  scoped_ptr<content::TestBrowserThread> ui_thread_;
+  // Initialized on our IO thread to satisfy BrowserThread::IO checks.
+  scoped_ptr<content::TestBrowserThread> io_thread_;
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(WebRTCAudioDeviceTest);

[Paweł Hajdan Jr., 2011/11/02 09:29:15]

nit: No need for that in a test fixture.

[tommi (sloooow) - chröme, 2011/11/02 10:59:23]

Done.

+};
+
+// A very basic implementation of webrtc::Transport that acts as a transport
+// but just forwards all calls to a local webrtc::VoENetwork implementation.
+// Ownership of the VoENetwork object lies outside the class.
+class TransportImpl : public webrtc::Transport {
+ public:
+  explicit TransportImpl(webrtc::VoENetwork* network) : network_(network) {}
+  ~TransportImpl() {}
+
+  virtual int SendPacket(int channel, const void* data, int len) {
+    DVLOG(1) << __FUNCTION__;
+    return network_->ReceivedRTPPacket(channel, data, len);
+  }
+
+  virtual int SendRTCPPacket(int channel, const void* data, int len) {
+    DVLOG(1) << __FUNCTION__;
+    return network_->ReceivedRTCPPacket(channel, data, len);
+  }
+
+ private:
+  webrtc::VoENetwork* network_;
+};
+
+// Basic test that instantiates and initializes an instance of
+// WebRtcAudioDeviceImpl.
+TEST_F(WebRTCAudioDeviceTest, Construct) {
+  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+      new WebRtcAudioDeviceImpl());
+  audio_device->SetSessionId(1);
+
+  AutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
+
+  ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
+  int err = base->Init(audio_device);
+  EXPECT_EQ(0, err);
+}
+
+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);
+
+  AutoDelete<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<TransportImpl> transport(new TransportImpl(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));
+  }
+}
+
+// Plays a local file.  This test usually takes just under a minute to run
+// and requires an audio card to work, so disabled by default.
+TEST_F(WebRTCAudioDeviceTest, DISABLED_PlayLocalFileLong) {
+  PlayLocalFile(0);
+}
+
+TEST_F(WebRTCAudioDeviceTest, PlayLocalFile) {
+  PlayLocalFile(TestTimeouts::action_timeout_ms());
+}
Property changes on: content/renderer/media/webrtc_audio_device_unittest.cc
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