Refactor the Get*Hardware* routines a bit.

These methods currently all have a static variable that caches the queried value.
This causes a problem in unit tests where values can change between tests.
I moved them outside of the AudioDevice class and grouped in their own namespace
which owns the globally cached state and allows the cache to be invalidated.

R=henrika
TEST=Run *WebRTC* tests in content.
Review URL: http://codereview.chromium.org/8588030

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@110695 0039d316-1c4b-4281-b951-d872f2087c98

content/content_renderer.gypi

@@ -81,6 +81,8 @@
     'renderer/load_progress_tracker.h',
     'renderer/media/audio_device.cc',
     'renderer/media/audio_device.h',
+    'renderer/media/audio_hardware.cc',
+    'renderer/media/audio_hardware.h',
     'renderer/media/audio_input_device.cc',
     'renderer/media/audio_input_device.h',
     'renderer/media/audio_input_message_filter.cc',

content/renderer/media/audio_device.cc

@@ -233,25 +233,3 @@ void AudioDevice::FireRenderCallback() {
                                   buffer_size_);
   }
 }
-
-double AudioDevice::GetAudioHardwareSampleRate() {
-  // Uses cached value if possible.
-  static double hardware_sample_rate = 0;
-  if (!hardware_sample_rate) {
-    RenderThreadImpl::current()->Send(
-        new ViewHostMsg_GetHardwareSampleRate(&hardware_sample_rate));
-  }
-  return hardware_sample_rate;
-}
-
-size_t AudioDevice::GetAudioHardwareBufferSize() {
-  // Uses cached value if possible.
-  static uint32 buffer_size = 0;
-
-  if (!buffer_size) {
-    RenderThreadImpl::current()->Send(
-        new ViewHostMsg_GetHardwareBufferSize(&buffer_size));
-  }
-
-  return static_cast<size_t>(buffer_size);
-}

content/renderer/media/audio_device.h

@@ -101,9 +101,6 @@ class CONTENT_EXPORT AudioDevice
   double sample_rate() const { return sample_rate_; }
   size_t buffer_size() const { return buffer_size_; }
 
-  static double GetAudioHardwareSampleRate();
-  static size_t GetAudioHardwareBufferSize();
-
   // Methods called on IO thread ----------------------------------------------
   // AudioMessageFilter::Delegate methods, called by AudioMessageFilter.
   virtual void OnRequestPacket(AudioBuffersState buffers_state) OVERRIDE;

content/renderer/media/audio_hardware.cc

+// 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/renderer/media/audio_hardware.h"
+
+#include "base/logging.h"
+#include "content/common/view_messages.h"
+#include "content/renderer/render_thread_impl.h"
+
+static double output_sample_rate = 0.0;
+static double input_sample_rate = 0.0;
+static size_t output_buffer_size = 0;
+
+namespace audio_hardware {
+
+double GetOutputSampleRate() {
+  DCHECK(RenderThreadImpl::current() != NULL);
+
+  if (!output_sample_rate) {
+    RenderThreadImpl::current()->Send(
+        new ViewHostMsg_GetHardwareSampleRate(&output_sample_rate));
+  }
+  return output_sample_rate;
+}
+
+double GetInputSampleRate() {
+  DCHECK(RenderThreadImpl::current() != NULL);
+
+  if (!input_sample_rate) {
+    RenderThreadImpl::current()->Send(
+        new ViewHostMsg_GetHardwareInputSampleRate(&input_sample_rate));
+  }
+  return input_sample_rate;
+}
+
+size_t GetOutputBufferSize() {
+  DCHECK(RenderThreadImpl::current() != NULL);
+
+  if (!output_buffer_size) {
+    uint32 buffer_size = 0;
+    RenderThreadImpl::current()->Send(
+        new ViewHostMsg_GetHardwareBufferSize(&buffer_size));
+    output_buffer_size = buffer_size;
+  }
+
+  return output_buffer_size;
+}
+
+void ResetCache() {
+  DCHECK(RenderThreadImpl::current() != NULL);
+
+  output_sample_rate = 0.0;
+  input_sample_rate = 0.0;
+  output_buffer_size = 0;
+}
+
+}  // namespace audio_hardware

content/renderer/media/audio_hardware.h

+// 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.
+//
+#ifndef CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_
+#define CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_
+#pragma once
+
+#include "base/basictypes.h"
+#include "content/common/content_export.h"
+
+// Contains static methods to query hardware properties from the browser
+// process. Values are cached to avoid unnecessary round trips, but the cache
+// can be cleared if needed (currently only used by tests).
+namespace audio_hardware {
+
+// Fetch the sample rate of the default audio output end point device.
+// Must be called from RenderThreadImpl::current().
+CONTENT_EXPORT double GetOutputSampleRate();
+
+// Fetch the sample rate of the default audio input end point device.
+// Must be called from RenderThreadImpl::current().
+CONTENT_EXPORT double GetInputSampleRate();
+
+// Fetch the buffer size we use for the default output device.
+// Must be called from RenderThreadImpl::current().
+CONTENT_EXPORT size_t GetOutputBufferSize();
+
+// Forces the next call to any of the Get functions to query the hardware
+// and repopulate the cache.
+CONTENT_EXPORT void ResetCache();
+}  // namespace audio_hardware
+
+#endif  // CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_

content/renderer/media/webrtc_audio_device_impl.cc

@@ -7,22 +7,13 @@
 #include "base/bind.h"
 #include "base/string_util.h"
 #include "base/win/windows_version.h"
-#include "content/common/view_messages.h"
+#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_util.h"
 
 static const int64 kMillisecondsBetweenProcessCalls = 5000;
 static const char kVersion[] = "WebRTC AudioDevice 1.0.0.Chrome";
 
-static int GetAudioInputHardwareSampleRate() {
-  static double input_sample_rate = 0;
-  if (!input_sample_rate) {
-    RenderThreadImpl::current()->Send(
-        new ViewHostMsg_GetHardwareInputSampleRate(&input_sample_rate));
-  }
-  return static_cast<int>(input_sample_rate);
-}
-
 WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
     : ref_count_(0),
       render_loop_(base::MessageLoopProxy::current()),
@@ -290,12 +281,13 @@ int32_t WebRtcAudioDeviceImpl::Init() {
   // Ask the browser for the default audio output hardware sample-rate.
   // This request is based on a synchronous IPC message.
   int output_sample_rate =
-      static_cast<int>(AudioDevice::GetAudioHardwareSampleRate());
+      static_cast<int>(audio_hardware::GetOutputSampleRate());
   DVLOG(1) << "Audio output hardware sample rate: " << output_sample_rate;
 
   // Ask the browser for the default audio input hardware sample-rate.
   // This request is based on a synchronous IPC message.
-  int input_sample_rate = GetAudioInputHardwareSampleRate();
+  int input_sample_rate =
+      static_cast<int>(audio_hardware::GetInputSampleRate());
   DVLOG(1) << "Audio input hardware sample rate: " << input_sample_rate;
 
   int input_channels = 0;

content/renderer/media/webrtc_audio_device_unittest.cc

@@ -15,6 +15,7 @@
 #include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
 
 using testing::_;
+using testing::AnyNumber;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
@@ -117,7 +118,7 @@ class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
 // WebRtcAudioDeviceImpl.
 TEST_F(WebRTCAudioDeviceTest, Construct) {
   AudioUtilNoHardware audio_util(48000.0, 48000.0);
-  set_audio_util_callback(&audio_util);
+  SetAudioUtilCallback(&audio_util);
   scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
       new WebRtcAudioDeviceImpl());
   audio_device->SetSessionId(1);
@@ -142,7 +143,7 @@ TEST_F(WebRTCAudioDeviceTest, StartPlayout) {
     return;
 
   AudioUtil audio_util;
-  set_audio_util_callback(&audio_util);
+  SetAudioUtilCallback(&audio_util);
 
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
@@ -151,7 +152,7 @@ TEST_F(WebRTCAudioDeviceTest, StartPlayout) {
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
-      OnDeleteAudioStream(_, 1)).Times(1);
+      OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
       new WebRtcAudioDeviceImpl());
@@ -211,7 +212,7 @@ TEST_F(WebRTCAudioDeviceTest, StartRecording) {
     return;
 
   AudioUtil audio_util;
-  set_audio_util_callback(&audio_util);
+  SetAudioUtilCallback(&audio_util);
 
   // TODO(tommi): extend MediaObserver and MockMediaObserver with support
   // for new interfaces, like OnSetAudioStreamRecording(). When done, add
@@ -278,7 +279,7 @@ TEST_F(WebRTCAudioDeviceTest, PlayLocalFile) {
       GetTestDataPath(FILE_PATH_LITERAL("speechmusic_mono_16kHz.pcm")));
 
   AudioUtil audio_util;
-  set_audio_util_callback(&audio_util);
+  SetAudioUtilCallback(&audio_util);
 
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
@@ -287,7 +288,7 @@ TEST_F(WebRTCAudioDeviceTest, PlayLocalFile) {
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
-      OnDeleteAudioStream(_, 1)).Times(1);
+      OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
       new WebRtcAudioDeviceImpl());
@@ -334,8 +335,15 @@ TEST_F(WebRTCAudioDeviceTest, FullDuplexAudio) {
   if (IsRunningHeadless())
     return;
 
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamStatus(_, 1, StrEq("created")));
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamPlaying(_, 1, true));
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamStatus(_, 1, StrEq("closed")));
+
   AudioUtil audio_util;
-  set_audio_util_callback(&audio_util);
+  SetAudioUtilCallback(&audio_util);
 
   scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
       new WebRtcAudioDeviceImpl());

content/renderer/renderer_webkitplatformsupport_impl.cc

@@ -22,6 +22,7 @@
 #include "content/public/renderer/content_renderer_client.h"
 #include "content/renderer/gpu/webgraphicscontext3d_command_buffer_impl.h"
 #include "content/renderer/media/audio_device.h"
+#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/render_thread_impl.h"
 #include "content/renderer/renderer_webaudiodevice_impl.h"
 #include "content/renderer/renderer_webidbfactory_impl.h"
@@ -564,11 +565,11 @@ RendererWebKitPlatformSupportImpl::createGraphicsContext3D() {
 }
 
 double RendererWebKitPlatformSupportImpl::audioHardwareSampleRate() {
-  return AudioDevice::GetAudioHardwareSampleRate();
+  return audio_hardware::GetOutputSampleRate();
 }
 
 size_t RendererWebKitPlatformSupportImpl::audioHardwareBufferSize() {
-  return AudioDevice::GetAudioHardwareBufferSize();
+  return audio_hardware::GetOutputBufferSize();
 }
 
 WebAudioDevice*

content/test/webrtc_audio_device_test.cc

@@ -19,6 +19,7 @@
 #include "content/common/view_messages.h"
 #include "content/public/browser/browser_thread.h"
 #include "content/public/common/content_paths.h"
+#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_process.h"
 #include "content/renderer/render_thread_impl.h"
@@ -125,10 +126,13 @@ void WebRTCAudioDeviceTest::SetUp() {
 }
 
 void WebRTCAudioDeviceTest::TearDown() {
+  SetAudioUtilCallback(NULL);
+
   ChildProcess::current()->io_message_loop()->PostTask(
       FROM_HERE,
       base::Bind(&SetupTask::UninitializeIOThread, new SetupTask(this)));
-  WaitForIOThreadCompletion();
+  EXPECT_TRUE(event_.TimedWait(
+      base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
   mock_process_.reset();
 }
 
@@ -136,6 +140,13 @@ bool WebRTCAudioDeviceTest::Send(IPC::Message* message) {
   return channel_->Send(message);
 }
 
+void WebRTCAudioDeviceTest::SetAudioUtilCallback(AudioUtilInterface* callback) {
+  // Invalidate any potentially cached values since the new callback should
+  // be used for those queries.
+  audio_hardware::ResetCache();
+  audio_util_callback_ = callback;
+}
+
 void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) {
   // We initialize COM (STA) on our IO thread as is done in Chrome.
   // See BrowserProcessSubThread::Init.
@@ -160,6 +171,7 @@ void WebRTCAudioDeviceTest::UninitializeIOThread() {
   media_stream_manager_.reset();
   test_request_context_ = NULL;
   initialize_com_.reset();
+  event_.Signal();
 }
 
 void WebRTCAudioDeviceTest::CreateChannel(
@@ -182,7 +194,9 @@ void WebRTCAudioDeviceTest::CreateChannel(
 void WebRTCAudioDeviceTest::DestroyChannel() {
   DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO));
   audio_render_host_->OnChannelClosing();
+  audio_render_host_->OnFilterRemoved();
   audio_input_renderer_host_->OnChannelClosing();
+  audio_input_renderer_host_->OnFilterRemoved();
   channel_.reset();
   audio_render_host_ = NULL;
   audio_input_renderer_host_ = NULL;
@@ -237,10 +251,6 @@ bool WebRTCAudioDeviceTest::OnMessageReceived(const IPC::Message& message) {
 
   EXPECT_TRUE(message_is_ok);
 
-  // We leave a DLOG as a hint to the developer in case important IPC messages
-  // are being dropped.
-  DLOG_IF(WARNING, !handled) << "Unhandled IPC message";
-
   return true;
 }
 

content/test/webrtc_audio_device_test.h

@@ -140,9 +140,7 @@ class WebRTCAudioDeviceTest
   // Sends an IPC message to the IO thread channel.
   bool Send(IPC::Message* message);
 
-  void set_audio_util_callback(AudioUtilInterface* callback) {
-    audio_util_callback_ = callback;
-  }
+  void SetAudioUtilCallback(AudioUtilInterface* callback);
 
  protected:
   void InitializeIOThread(const char* thread_name);