Remove AudioBuffersState usage in Chromium

media/audio/win/audio_low_latency_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/environment.h"
 #include "base/files/file_util.h"
@@ skipping 34 matching lines @@
 static const size_t kFileDurationMs = 20000;
 static const size_t kNumFileSegments = 2;
 static const int kBitsPerSample = 16;
 static const size_t kMaxDeltaSamples = 1000;
 static const char kDeltaTimeMsFileName[] = "delta_times_ms.txt";
 
 MATCHER_P(HasValidDelay, value, "") {
   // It is difficult to come up with a perfect test condition for the delay
   // estimation. For now, verify that the produced output delay is always
   // larger than the selected buffer size.
-  return arg.hardware_delay_bytes >= value.hardware_delay_bytes;
+  return arg >= value;
 }
 
 // Used to terminate a loop from a different thread than the loop belongs to.
 // |loop| should be a MessageLoopProxy.
 ACTION_P(QuitLoop, loop) {
   loop->PostTask(FROM_HERE, base::MessageLoop::QuitClosure());
 }
 
 // This audio source implementation should be used for manual tests only since
 // it takes about 20 seconds to play out a file.
@@ skipping 30 matching lines @@
     while (elements_written < elements_to_write_) {
       fprintf(text_file_, "%d\n", delta_times_[elements_written]);
       ++elements_written;
     }
 
     base::CloseFile(text_file_);
   }
 
   // AudioOutputStream::AudioSourceCallback implementation.
   virtual int OnMoreData(AudioBus* audio_bus,
-                         AudioBuffersState buffers_state) {
+                         uint32 total_bytes_delay) {
     // Store time difference between two successive callbacks in an array.
     // These values will be written to a file in the destructor.
     const base::TimeTicks now_time = base::TimeTicks::Now();
     const int diff = (now_time - previous_call_time_).InMilliseconds();
     previous_call_time_ = now_time;
     if (elements_to_write_ < kMaxDeltaSamples) {
       delta_times_[elements_to_write_] = diff;
       ++elements_to_write_;
     }
 
@@ skipping 273 matching lines @@
   MockAudioSourceCallback source;
 
   // Create default WASAPI output stream which plays out in stereo using
   // the shared mixing rate. The default buffer size is 10ms.
   AudioOutputStreamWrapper aosw(audio_manager.get());
   AudioOutputStream* aos = aosw.Create();
   EXPECT_TRUE(aos->Open());
 
   // Derive the expected size in bytes of each packet.
   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
-                           (aosw.bits_per_sample() / 8);
-
-  // Set up expected minimum delay estimation.
-  AudioBuffersState state(0, bytes_per_packet);
+      (aosw.bits_per_sample() / 8);
 
   // Wait for the first callback and verify its parameters.
-  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
       .WillOnce(DoAll(
           QuitLoop(loop.message_loop_proxy()),
           Return(aosw.samples_per_packet())));
 
   aos->Start(&source);
   loop.PostDelayedTask(FROM_HERE, base::MessageLoop::QuitClosure(),
                        TestTimeouts::action_timeout());
   loop.Run();
   aos->Stop();
   aos->Close();
@@ skipping 179 matching lines @@
   // Create exclusive-mode WASAPI output stream which plays out in stereo
   // using the minimum buffer size at 48kHz sample rate.
   AudioOutputStreamWrapper aosw(audio_manager.get());
   AudioOutputStream* aos = aosw.Create(48000, 160);
   EXPECT_TRUE(aos->Open());
 
   // Derive the expected size in bytes of each packet.
   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
       (aosw.bits_per_sample() / 8);
 
-  // Set up expected minimum delay estimation.
-  AudioBuffersState state(0, bytes_per_packet);
-
  // Wait for the first callback and verify its parameters.
-  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
       .WillOnce(DoAll(
           QuitLoop(loop.message_loop_proxy()),
           Return(aosw.samples_per_packet())))
       .WillRepeatedly(Return(aosw.samples_per_packet()));
 
   aos->Start(&source);
   loop.PostDelayedTask(FROM_HERE, base::MessageLoop::QuitClosure(),
                        TestTimeouts::action_timeout());
   loop.Run();
   aos->Stop();
@@ skipping 16 matching lines @@
   // Create exclusive-mode WASAPI output stream which plays out in stereo
   // using the minimum buffer size at 44.1kHz sample rate.
   AudioOutputStreamWrapper aosw(audio_manager.get());
   AudioOutputStream* aos = aosw.Create(44100, 160);
   EXPECT_TRUE(aos->Open());
 
   // Derive the expected size in bytes of each packet.
   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
       (aosw.bits_per_sample() / 8);
 
-  // Set up expected minimum delay estimation.
-  AudioBuffersState state(0, bytes_per_packet);
-
   // Wait for the first callback and verify its parameters.
-  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
     .WillOnce(DoAll(
         QuitLoop(loop.message_loop_proxy()),
         Return(aosw.samples_per_packet())))
     .WillRepeatedly(Return(aosw.samples_per_packet()));
 
   aos->Start(&source);
   loop.PostDelayedTask(FROM_HERE, base::MessageLoop::QuitClosure(),
                         TestTimeouts::action_timeout());
   loop.Run();
   aos->Stop();
   aos->Close();
 }
 
 }  // namespace media