Simplified AudioOutputStream interface.

media/audio/simple_sources_unittest.cc

 // Copyright (c) 2010 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/logging.h"
 #include "base/basictypes.h"
 #include "base/scoped_ptr.h"
 #include "base/time.h"
 #include "media/audio/audio_manager.h"
 #include "media/audio/fake_audio_output_stream.h"
@@ skipping 40 matching lines @@
   EXPECT_EQ(kDataSize, push_source.UnProcessedBytes());
 
   // Read everything from the push source.
   for (uint32 i = 0; i < kDataSize; i += kReadSize) {
     uint32 size = std::min(kDataSize - i , kReadSize);
     EXPECT_EQ(size, push_source.OnMoreData(NULL, read_data.get(), size,
                                            AudioBuffersState()));
     EXPECT_EQ(0, memcmp(data.get() + i, read_data.get(), size));
   }
   EXPECT_EQ(0u, push_source.UnProcessedBytes());
-
-  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 uint32 samples = 1024;
   const uint32 bytes_per_sample = 2;
   const int freq = 200;
 
   SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
                              freq, AudioParameters::kTelephoneSampleRate);
 
   AudioManager* audio_man = AudioManager::GetAudioManager();
   ASSERT_TRUE(NULL != audio_man);
   AudioParameters params(
       AudioParameters::AUDIO_MOCK, 1, AudioParameters::kTelephoneSampleRate,
-      bytes_per_sample * 2);
+      bytes_per_sample * 2, samples);
   AudioOutputStream* oas = audio_man->MakeAudioOutputStream(params);
   ASSERT_TRUE(NULL != oas);
-  EXPECT_TRUE(oas->Open(samples * bytes_per_sample));
+  EXPECT_TRUE(oas->Open());
 
   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);
 
   uint32 half_period = AudioParameters::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]);
 }