Revert 290359 "Remove AudioBuffersState class."

trunk/src/media/audio/win/audio_low_latency_output_win_unittest.cc

Index: trunk/src/media/audio/win/audio_low_latency_output_win_unittest.cc
===================================================================
--- trunk/src/media/audio/win/audio_low_latency_output_win_unittest.cc	(revision 290374)
+++ trunk/src/media/audio/win/audio_low_latency_output_win_unittest.cc	(working copy)
@@ -52,7 +52,7 @@
   // 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 >= value;
+  return arg.hardware_delay_bytes >= value.hardware_delay_bytes;
 }
 
 // Used to terminate a loop from a different thread than the loop belongs to.
@@ -103,7 +103,7 @@
 
   // AudioOutputStream::AudioSourceCallback implementation.
   virtual int OnMoreData(AudioBus* audio_bus,
-                         int total_bytes_delay) {
+                         AudioBuffersState buffers_state) {
     // 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();
@@ -396,11 +396,14 @@
   EXPECT_TRUE(aos->Open());
 
   // Derive the expected size in bytes of each packet.
-  int bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
-      (aosw.bits_per_sample() / 8);
+  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(bytes_per_packet)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
       .WillOnce(DoAll(
           QuitLoop(loop.message_loop_proxy()),
           Return(aosw.samples_per_packet())));
@@ -597,11 +600,14 @@
   EXPECT_TRUE(aos->Open());
 
   // Derive the expected size in bytes of each packet.
-  int bytes_per_packet = aosw.channels() * aosw.samples_per_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(bytes_per_packet)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
       .WillOnce(DoAll(
           QuitLoop(loop.message_loop_proxy()),
           Return(aosw.samples_per_packet())))
@@ -635,11 +641,14 @@
   EXPECT_TRUE(aos->Open());
 
   // Derive the expected size in bytes of each packet.
-  int bytes_per_packet = aosw.channels() * aosw.samples_per_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(bytes_per_packet)))
+  EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(state)))
     .WillOnce(DoAll(
         QuitLoop(loop.message_loop_proxy()),
         Return(aosw.samples_per_packet())))