stopping the audio thread before destroying the AudioManager<Platform>

media/audio/simple_sources_unittest.cc

Index: media/audio/simple_sources_unittest.cc
diff --git a/media/audio/simple_sources_unittest.cc b/media/audio/simple_sources_unittest.cc
index ef00ed77c20a96f6c869608bb207f7aeaf37739f..95f2241ffc3997c969ca9f880ab63e6c8d82ffc5 100644
--- a/media/audio/simple_sources_unittest.cc
+++ b/media/audio/simple_sources_unittest.cc
@@ -78,12 +78,13 @@ TEST(SimpleSources, SineWaveAudio16MonoTest) {
 
   oas->Start(&source);
   oas->Stop();
-  oas->Close();
 
-  ASSERT_TRUE(FakeAudioOutputStream::GetLastFakeStream());
+  FakeAudioOutputStream* current_fake_stream =
+      reinterpret_cast<FakeAudioOutputStream*>(oas);

[tommi (sloooow) - chröme, 2012/03/13 14:16:52]

use static_cast

[no longer working on chromium, 2012/03/13 15:03:39]

Done.

+  ASSERT_TRUE(current_fake_stream);
   const int16* last_buffer =
       reinterpret_cast<int16*>(
-          FakeAudioOutputStream::GetLastFakeStream()->buffer());
+          current_fake_stream->buffer());
   ASSERT_TRUE(NULL != last_buffer);
 
   uint32 half_period = AudioParameters::kTelephoneSampleRate / (freq * 2);
@@ -98,4 +99,5 @@ TEST(SimpleSources, SineWaveAudio16MonoTest) {
   EXPECT_EQ(-5126, last_buffer[half_period + 1]);
   EXPECT_TRUE(last_buffer[half_period + 1] > last_buffer[half_period + 2]);
   EXPECT_TRUE(last_buffer[half_period + 2] > last_buffer[half_period + 3]);
+  oas->Close();
 }