Enable the webrtc audio tests on the build bots that have audio hardware.

BUG=none
TEST=This enables the audio tests in content_unittests on bots with audio hardware.

Committed: http://src.chromium.org/viewvc/chrome?view=rev&revision=130019

Review URL: https://chromiumcodereview.appspot.com/9921035

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@131757 0039d316-1c4b-4281-b951-d872f2087c98

3 files changed, 23 insertions, 14 deletions

diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
index e9323f2..cf3a453 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -75,13 +75,6 @@ class AudioUtilNoHardware : public AudioUtilInterface {
   DISALLOW_COPY_AND_ASSIGN(AudioUtilNoHardware);
 };
 
-bool IsRunningHeadless() {
-  scoped_ptr<base::Environment> env(base::Environment::Create());
-  if (env->HasVar("CHROME_HEADLESS"))
-    return true;
-  return false;
-}
-
 // Return true if at least one element in the array matches |value|.
 bool FindElementInArray(int* array, int size, int value) {
   return (std::find(&array[0], &array[0] + size, value) != &array[size]);
@@ -106,7 +99,7 @@ bool HardwareSampleRatesAreValid() {
 
   if (!FindElementInArray(valid_input_rates, arraysize(valid_input_rates),
                           input_sample_rate)) {
-    DLOG(WARNING) << "Non-supported input sample rate detected.";
+    LOG(WARNING) << "Non-supported input sample rate detected.";
     return false;
   }
 
@@ -115,7 +108,7 @@ bool HardwareSampleRatesAreValid() {
       static_cast<int>(audio_hardware::GetOutputSampleRate());
   if (!FindElementInArray(valid_output_rates, arraysize(valid_output_rates),
                           output_sample_rate)) {
-    DLOG(WARNING) << "Non-supported output sample rate detected.";
+    LOG(WARNING) << "Non-supported output sample rate detected.";
     return false;
   }
 
@@ -260,8 +253,10 @@ TEST_F(WebRTCAudioDeviceTest, Construct) {
 // verify that streaming starts correctly.
 // Disabled when running headless since the bots don't have the required config.
 TEST_F(WebRTCAudioDeviceTest, StartPlayout) {
-  if (IsRunningHeadless())
+  if (!has_output_devices_) {
+    LOG(WARNING) << "No output device detected.";
     return;
+  }
 
   AudioUtil audio_util;
   SetAudioUtilCallback(&audio_util);
@@ -332,8 +327,10 @@ TEST_F(WebRTCAudioDeviceTest, StartPlayout) {
 // that the audio capturing starts as it should.
 // Disabled when running headless since the bots don't have the required config.
 TEST_F(WebRTCAudioDeviceTest, StartRecording) {
-  if (IsRunningHeadless())
+  if (!has_input_devices_ || !has_output_devices_) {
+    LOG(WARNING) << "Missing audio devices.";
     return;
+  }
 
   AudioUtil audio_util;
   SetAudioUtilCallback(&audio_util);
@@ -399,8 +396,10 @@ TEST_F(WebRTCAudioDeviceTest, StartRecording) {
 // Uses WebRtcAudioDeviceImpl to play a local wave file.
 // Disabled when running headless since the bots don't have the required config.
 TEST_F(WebRTCAudioDeviceTest, PlayLocalFile) {
-  if (IsRunningHeadless())
+  if (!has_output_devices_) {
+    LOG(WARNING) << "No output device detected.";
     return;
+  }
 
   std::string file_path(
       GetTestDataPath(FILE_PATH_LITERAL("speechmusic_mono_16kHz.pcm")));
@@ -468,8 +467,10 @@ TEST_F(WebRTCAudioDeviceTest, PlayLocalFile) {
 // TODO(henrika): improve quality by using a wideband codec, enabling noise-
 // suppressions etc.
 TEST_F(WebRTCAudioDeviceTest, FullDuplexAudioWithAGC) {
-  if (IsRunningHeadless())
+  if (!has_output_devices_ || !has_input_devices_) {
+    LOG(WARNING) << "Missing audio devices.";
     return;
+  }
 
   AudioUtil audio_util;
   SetAudioUtilCallback(&audio_util);
diff --git a/content/test/webrtc_audio_device_test.cc b/content/test/webrtc_audio_device_test.cc
index 0f11917..c9e72ec6 100644
--- a/content/test/webrtc_audio_device_test.cc
+++ b/content/test/webrtc_audio_device_test.cc
@@ -88,7 +88,8 @@ ACTION_P(QuitMessageLoop, loop_or_proxy) {
 }  // end namespace
 
 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
-    : render_thread_(NULL), audio_util_callback_(NULL) {
+    : render_thread_(NULL), audio_util_callback_(NULL),
+      has_input_devices_(false), has_output_devices_(false) {
 }
 
 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {}
@@ -171,6 +172,9 @@ void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) {
   resource_context_->set_request_context(test_request_context_.get());
   media_observer_.reset(new MockMediaObserver());
 
+  has_input_devices_ = audio_manager_->HasAudioInputDevices();
+  has_output_devices_ = audio_manager_->HasAudioOutputDevices();
+
   // Create an IPC channel that handles incoming messages on the IO thread.
   CreateChannel(thread_name);
 }
diff --git a/content/test/webrtc_audio_device_test.h b/content/test/webrtc_audio_device_test.h
index dfe9f27..e360f4f 100644
--- a/content/test/webrtc_audio_device_test.h
+++ b/content/test/webrtc_audio_device_test.h
@@ -184,6 +184,10 @@ class WebRTCAudioDeviceTest
   scoped_ptr<content::TestBrowserThread> io_thread_;
   // COM initialization on the IO thread for Windows.
   scoped_ptr<base::win::ScopedCOMInitializer> initialize_com_;
+
+  // These are initialized when we set up our IO thread.
+  bool has_input_devices_;
+  bool has_output_devices_;
 };
 
 // A very basic implementation of webrtc::Transport that acts as a transport