Remove AudioBuffersState usage in Chromium

media/audio/win/audio_output_win_unittest.cc

 // Copyright (c) 2012 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 <windows.h>
 #include <mmsystem.h>
 
 #include "base/basictypes.h"
 #include "base/base_paths.h"
 #include "base/memory/aligned_memory.h"
@@ skipping 19 matching lines @@
 using ::testing::NotNull;
 using ::testing::Return;
 
 using base::win::ScopedCOMInitializer;
 
 namespace media {
 
 static const wchar_t kAudioFile1_16b_m_16K[]
     = L"media\\test\\data\\sweep02_16b_mono_16KHz.raw";
 
-static int ClearData(AudioBus* audio_bus, AudioBuffersState buffers_state) {
+static int ClearData(AudioBus* audio_bus, uint32 total_bytes_delay) {
   audio_bus->Zero();
   return audio_bus->frames();
 }
 
 // This class allows to find out if the callbacks are occurring as
 // expected and if any error has been reported.
 class TestSourceBasic : public AudioOutputStream::AudioSourceCallback {
  public:
   explicit TestSourceBasic()
       : callback_count_(0),
         had_error_(0) {
   }
   // AudioSourceCallback::OnMoreData implementation:
   virtual int OnMoreData(AudioBus* audio_bus,
-                         AudioBuffersState buffers_state) {
+                         uint32 total_bytes_delay) {
     ++callback_count_;
     // Touch the channel memory value to make sure memory is good.
     audio_bus->Zero();
     return audio_bus->frames();
   }
   // AudioSourceCallback::OnError implementation:
   virtual void OnError(AudioOutputStream* stream) {
     ++had_error_;
   }
   // Returns how many times OnMoreData() has been called.
@@ skipping 16 matching lines @@
 
 const int kMaxNumBuffers = 3;
 // Specializes TestSourceBasic to simulate a source that blocks for some time
 // in the OnMoreData callback.
 class TestSourceLaggy : public TestSourceBasic {
  public:
   TestSourceLaggy(int laggy_after_buffer, int lag_in_ms)
       : laggy_after_buffer_(laggy_after_buffer), lag_in_ms_(lag_in_ms) {
   }
   virtual int OnMoreData(AudioBus* audio_bus,
-                         AudioBuffersState buffers_state) {
+                         uint32 total_bytes_delay) {
     // Call the base, which increments the callback_count_.
-    TestSourceBasic::OnMoreData(audio_bus, buffers_state);
+    TestSourceBasic::OnMoreData(audio_bus, total_bytes_delay);
     if (callback_count() > kMaxNumBuffers) {
       ::Sleep(lag_in_ms_);
     }
     return audio_bus->frames();
   }
  private:
   int laggy_after_buffer_;
   int lag_in_ms_;
 };
 
@@ skipping 408 matching lines @@
   uint32 bytes_100_ms = samples_100_ms * 2;
 
   // Audio output stream has either a double or triple buffer scheme.
   // We expect the amount of pending bytes will reaching up to 2 times of
   // |bytes_100_ms| depending on number of buffers used.
   // From that it would decrease as we are playing the data but not providing
   // new one. And then we will try to provide zero data so the amount of
   // pending bytes will go down and eventually read zero.
   InSequence s;
 
-  EXPECT_CALL(source, OnMoreData(NotNull(),
-                                 Field(&AudioBuffersState::pending_bytes, 0)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), 0))
       .WillOnce(Invoke(ClearData));
 
   // Note: If AudioManagerWin::NumberOfWaveOutBuffers() ever changes, or if this
   // test is run on Vista, these expectations will fail.
-  EXPECT_CALL(source, OnMoreData(NotNull(),
-                                 Field(&AudioBuffersState::pending_bytes,
-                                       bytes_100_ms)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms))
       .WillOnce(Invoke(ClearData));
-  EXPECT_CALL(source, OnMoreData(NotNull(),
-                                 Field(&AudioBuffersState::pending_bytes,
-                                       2 * bytes_100_ms)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), 2 * bytes_100_ms))
       .WillOnce(Invoke(ClearData));
-  EXPECT_CALL(source, OnMoreData(NotNull(),
-                                 Field(&AudioBuffersState::pending_bytes,
-                                       2 * bytes_100_ms)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), 2 * bytes_100_ms))
       .Times(AnyNumber())
       .WillRepeatedly(Return(0));
-  EXPECT_CALL(source, OnMoreData(NotNull(),
-                                 Field(&AudioBuffersState::pending_bytes,
-                                       bytes_100_ms)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms))
       .Times(AnyNumber())
       .WillRepeatedly(Return(0));
-  EXPECT_CALL(source, OnMoreData(NotNull(),
-                                 Field(&AudioBuffersState::pending_bytes, 0)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), 0))
       .Times(AnyNumber())
       .WillRepeatedly(Return(0));
 
   oas->Start(&source);
   ::Sleep(500);
   oas->Stop();
   oas->Close();
 }
 
 // Simple source that uses a SyncSocket to retrieve the audio data
 // from a potentially remote thread.
 class SyncSocketSource : public AudioOutputStream::AudioSourceCallback {
  public:
   SyncSocketSource(base::SyncSocket* socket, const AudioParameters& params)
       : socket_(socket) {
     // Setup AudioBus wrapping data we'll receive over the sync socket.
     data_size_ = AudioBus::CalculateMemorySize(params);
     data_.reset(static_cast<float*>(
         base::AlignedAlloc(data_size_, AudioBus::kChannelAlignment)));
     audio_bus_ = AudioBus::WrapMemory(params, data_.get());
   }
   ~SyncSocketSource() {}
 
   // AudioSourceCallback::OnMoreData implementation:
   virtual int OnMoreData(AudioBus* audio_bus,
-                         AudioBuffersState buffers_state) {
-    socket_->Send(&buffers_state, sizeof(buffers_state));
+                         uint32 total_bytes_delay) {
+    socket_->Send(&total_bytes_delay, sizeof(total_bytes_delay));
     uint32 size = socket_->Receive(data_.get(), data_size_);
     DCHECK_EQ(static_cast<size_t>(size) % sizeof(*audio_bus_->channel(0)), 0U);
     audio_bus_->CopyTo(audio_bus);
     return audio_bus_->frames();
   }
-  virtual int OnMoreIOData(AudioBus* source,
-                           AudioBus* dest,
-                           AudioBuffersState buffers_state) {
-    NOTREACHED();
-    return 0;
-  }
+
   // AudioSourceCallback::OnError implementation:
   virtual void OnError(AudioOutputStream* stream) {
   }
 
  private:
   base::SyncSocket* socket_;
   int data_size_;
   scoped_ptr<float, base::AlignedFreeDeleter> data_;
   scoped_ptr<AudioBus> audio_bus_;
 };
@@ skipping 18 matching lines @@
 
   // Setup AudioBus wrapping data we'll pass over the sync socket.
   scoped_ptr<float, base::AlignedFreeDeleter> data(static_cast<float*>(
       base::AlignedAlloc(ctx.packet_size_bytes, AudioBus::kChannelAlignment)));
   scoped_ptr<AudioBus> audio_bus = AudioBus::WrapMemory(
       ctx.channels, ctx.frames, data.get());
 
   SineWaveAudioSource sine(1, ctx.sine_freq, ctx.sample_rate);
   const int kTwoSecFrames = ctx.sample_rate * 2;
 
-  AudioBuffersState buffers_state;
+  uint32 total_bytes_delay = 0;
   int times = 0;
   for (int ix = 0; ix < kTwoSecFrames; ix += ctx.frames) {
-    if (ctx.socket->Receive(&buffers_state, sizeof(buffers_state)) == 0)
+    if (ctx.socket->Receive(&total_bytes_delay, sizeof(total_bytes_delay)) == 0)
       break;
-    if ((times > 0) && (buffers_state.pending_bytes < 1000)) __debugbreak();
-    sine.OnMoreData(audio_bus.get(), buffers_state);
+    if ((times > 0) && (total_bytes_delay < 1000)) __debugbreak();
+    sine.OnMoreData(audio_bus.get(), total_bytes_delay);
     ctx.socket->Send(data.get(), ctx.packet_size_bytes);
     ++times;
   }
 
   return 0;
 }
 
 // Test the basic operation of AudioOutputStream used with a SyncSocket.
 // The emphasis is to verify that it is possible to feed data to the audio
 // layer using a source based on SyncSocket. In a real situation we would
@@ skipping 40 matching lines @@
   oas->Start(&source);
 
   ::WaitForSingleObject(thread, INFINITE);
   ::CloseHandle(thread);
 
   oas->Stop();
   oas->Close();
 }
 
 }  // namespace media