Refactoring to share MockAudioOutputStream implementations across 3 platforms

Re-trying the previous CL. The last change got reverted because of valgrind
failure.

TBR=scherkus

Original CL: http://codereview.chromium.org/155471

Review URL: http://codereview.chromium.org/159167

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@21242 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 46 insertions, 0 deletions

diff --git a/media/audio/simple_sources_unittest.cc b/media/audio/simple_sources_unittest.cc
index 32176b7..ef6a5a7 100644
--- a/media/audio/simple_sources_unittest.cc
+++ b/media/audio/simple_sources_unittest.cc
@@ -6,6 +6,7 @@
 #include "base/basictypes.h"
 #include "base/scoped_ptr.h"
 #include "base/time.h"
+#include "media/audio/fake_audio_output_stream.h"
 #include "media/audio/simple_sources.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -60,3 +61,48 @@ TEST(SimpleSourcesTest, PushSourceSmallerWrite) {
 
   push_source.OnClose(NULL);
 }
+
+// Validate that the SineWaveAudioSource writes the expected values for
+// the FORMAT_16BIT_MONO. The values are carefully selected so rounding issues
+// do not affect the result. We also test that AudioManager::GetLastMockBuffer
+// works.
+TEST(SimpleSources, SineWaveAudio16MonoTest) {
+  const size_t samples = 1024;
+  const size_t bytes_per_sample = 2;
+  const int freq = 200;
+
+  SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
+                             freq, AudioManager::kTelephoneSampleRate);
+
+  AudioManager* audio_man = AudioManager::GetAudioManager();
+  ASSERT_TRUE(NULL != audio_man);
+  AudioOutputStream* oas =
+      audio_man->MakeAudioStream(AudioManager::AUDIO_MOCK, 1,
+                                 AudioManager::kTelephoneSampleRate,
+                                 bytes_per_sample * 2);
+  ASSERT_TRUE(NULL != oas);
+  EXPECT_TRUE(oas->Open(samples * bytes_per_sample));
+
+  oas->Start(&source);
+  oas->Stop();
+  oas->Close();
+
+  ASSERT_TRUE(FakeAudioOutputStream::GetLastFakeStream());
+  const int16* last_buffer =
+      reinterpret_cast<int16*>(
+          FakeAudioOutputStream::GetLastFakeStream()->buffer());
+  ASSERT_TRUE(NULL != last_buffer);
+
+  size_t half_period = AudioManager::kTelephoneSampleRate / (freq * 2);
+
+  // Spot test positive incursion of sine wave.
+  EXPECT_EQ(0, last_buffer[0]);
+  EXPECT_EQ(5126, last_buffer[1]);
+  EXPECT_TRUE(last_buffer[1] < last_buffer[2]);
+  EXPECT_TRUE(last_buffer[2] < last_buffer[3]);
+  // Spot test negative incursion of sine wave.
+  EXPECT_EQ(0, last_buffer[half_period]);
+  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]);
+}