Revert 290359 "Remove AudioBuffersState class."

Broke ChromeOS bots
http://build.chromium.org/p/chromium.chrome/builders/Google%20Chrome%20ChromeOS/builds/70835/steps/compile/logs/stdio#error1

> Remove AudioBuffersState class.
> 
> The AudioBuffersState object doesn't appear to be necessary anymore.
> Most code either completely ignores this information or doesn't actually
> care about the difference between pending_bytes and hardware_delay_bytes.
> Also usually only one of the 2 fields was actually being used at a time.
> This change removes the class and simply uses an int that represent the
> total number of delay bytes.
> 
> BUG=125685
> 
> Review URL: https://codereview.chromium.org/467833002

TBR=acolwell@chromium.org

Review URL: https://codereview.chromium.org/487543002

Cr-Commit-Position: refs/heads/master@{#290376}
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@290376 0039d316-1c4b-4281-b951-d872f2087c98

35 files changed, 177 insertions, 82 deletions

diff --git a/media/audio/BUILD.gn b/media/audio/BUILD.gn
index 0f48af0..e6ac957 100644
--- a/media/audio/BUILD.gn
+++ b/media/audio/BUILD.gn
@@ -36,6 +36,8 @@ if (!linux_link_pulseaudio) {
 source_set("audio") {
   sources = [
     "agc_audio_stream.h",
+    "audio_buffers_state.cc",
+    "audio_buffers_state.h",
     "audio_device_name.cc",
     "audio_device_name.h",
     "audio_device_thread.cc",
diff --git a/media/audio/alsa/alsa_output.cc b/media/audio/alsa/alsa_output.cc
index bb03825..690d738 100644
--- a/media/audio/alsa/alsa_output.cc
+++ b/media/audio/alsa/alsa_output.cc
@@ -360,7 +360,7 @@ void AlsaPcmOutputStream::BufferPacket(bool* source_exhausted) {
     scoped_refptr<media::DataBuffer> packet =
         new media::DataBuffer(packet_size_);
     int frames_filled = RunDataCallback(
-        audio_bus_.get(), hardware_delay);
+        audio_bus_.get(), AudioBuffersState(0, hardware_delay));
 
     size_t packet_size = frames_filled * bytes_per_frame_;
     DCHECK_LE(packet_size, packet_size_);
@@ -740,11 +740,11 @@ bool AlsaPcmOutputStream::IsOnAudioThread() const {
 }
 
 int AlsaPcmOutputStream::RunDataCallback(AudioBus* audio_bus,
-                                         int total_bytes_delay) {
+                                         AudioBuffersState buffers_state) {
   TRACE_EVENT0("audio", "AlsaPcmOutputStream::RunDataCallback");
 
   if (source_callback_)
-    return source_callback_->OnMoreData(audio_bus, total_bytes_delay);
+    return source_callback_->OnMoreData(audio_bus, buffers_state);
 
   return 0;
 }
diff --git a/media/audio/alsa/alsa_output.h b/media/audio/alsa/alsa_output.h
index 6a2817c..1805645 100644
--- a/media/audio/alsa/alsa_output.h
+++ b/media/audio/alsa/alsa_output.h
@@ -154,7 +154,7 @@ class MEDIA_EXPORT AlsaPcmOutputStream : public AudioOutputStream {
   // is passed into the output stream, but ownership is not transfered which
   // requires a synchronization on access of the |source_callback_| to avoid
   // using a deleted callback.
-  int RunDataCallback(AudioBus* audio_bus, int total_bytes_delay);
+  int RunDataCallback(AudioBus* audio_bus, AudioBuffersState buffers_state);
   void RunErrorCallback(int code);
 
   // Changes the AudioSourceCallback to proxy calls to.  Pass in NULL to
diff --git a/media/audio/alsa/alsa_output_unittest.cc b/media/audio/alsa/alsa_output_unittest.cc
index 3b6592b..8b0aeae 100644
--- a/media/audio/alsa/alsa_output_unittest.cc
+++ b/media/audio/alsa/alsa_output_unittest.cc
@@ -644,7 +644,10 @@ TEST_F(AlsaPcmOutputStreamTest, BufferPacket_Underrun) {
       .WillOnce(Return(SND_PCM_STATE_XRUN));
   EXPECT_CALL(mock_alsa_wrapper_, PcmAvailUpdate(_))
       .WillRepeatedly(Return(0));  // Buffer is full.
-  EXPECT_CALL(mock_callback, OnMoreData(_, 0))
+  EXPECT_CALL(mock_callback,
+              OnMoreData(_, AllOf(
+                  Field(&AudioBuffersState::pending_bytes, 0),
+                  Field(&AudioBuffersState::hardware_delay_bytes, 0))))
       .WillOnce(DoAll(ClearBuffer(), Return(kTestFramesPerPacket / 2)));
 
   bool source_exhausted;
diff --git a/media/audio/android/audio_android_unittest.cc b/media/audio/android/audio_android_unittest.cc
index d333237..177953d 100644
--- a/media/audio/android/audio_android_unittest.cc
+++ b/media/audio/android/audio_android_unittest.cc
@@ -123,7 +123,7 @@ static void CheckDeviceNames(const AudioDeviceNames& device_names) {
 }
 
 // We clear the data bus to ensure that the test does not cause noise.
-static int RealOnMoreData(AudioBus* dest, int total_bytes_delay) {
+static int RealOnMoreData(AudioBus* dest, AudioBuffersState buffers_state) {
   dest->Zero();
   return dest->frames();
 }
@@ -178,7 +178,7 @@ class FileAudioSource : public AudioOutputStream::AudioSourceCallback {
   // Use samples read from a data file and fill up the audio buffer
   // provided to us in the callback.
   virtual int OnMoreData(AudioBus* audio_bus,
-                         int total_bytes_delay) OVERRIDE {
+                         AudioBuffersState buffers_state) OVERRIDE {
     bool stop_playing = false;
     int max_size =
         audio_bus->frames() * audio_bus->channels() * kBytesPerSample;
@@ -354,7 +354,7 @@ class FullDuplexAudioSinkSource
 
   // AudioOutputStream::AudioSourceCallback implementation
   virtual int OnMoreData(AudioBus* dest,
-                         int total_bytes_delay) OVERRIDE {
+                         AudioBuffersState buffers_state) OVERRIDE {
     const int size_in_bytes =
         (params_.bits_per_sample() / 8) * dest->frames() * dest->channels();
     EXPECT_EQ(size_in_bytes, params_.GetBytesPerBuffer());
diff --git a/media/audio/android/opensles_output.cc b/media/audio/android/opensles_output.cc
index 4ac0af2..41c03c7 100644
--- a/media/audio/android/opensles_output.cc
+++ b/media/audio/android/opensles_output.cc
@@ -326,7 +326,7 @@ void OpenSLESOutputStream::FillBufferQueueNoLock() {
   // delay estimation.
   const uint32 hardware_delay = buffer_size_bytes_;
   int frames_filled = callback_->OnMoreData(
-      audio_bus_.get(), hardware_delay);
+      audio_bus_.get(), AudioBuffersState(0, hardware_delay));
   if (frames_filled <= 0) {
     // Audio source is shutting down, or halted on error.
     return;
diff --git a/media/audio/audio_buffers_state.cc b/media/audio/audio_buffers_state.cc
new file mode 100644
index 0000000..6c4f950
--- /dev/null
+++ b/media/audio/audio_buffers_state.cc
@@ -0,0 +1,20 @@
+// 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 "media/audio/audio_buffers_state.h"
+
+namespace media {
+
+AudioBuffersState::AudioBuffersState()
+    : pending_bytes(0),
+      hardware_delay_bytes(0) {
+}
+
+AudioBuffersState::AudioBuffersState(int pending_bytes,
+                                     int hardware_delay_bytes)
+    : pending_bytes(pending_bytes),
+      hardware_delay_bytes(hardware_delay_bytes) {
+}
+
+}  // namespace media
diff --git a/media/audio/audio_buffers_state.h b/media/audio/audio_buffers_state.h
new file mode 100644
index 0000000..79244ae
--- /dev/null
+++ b/media/audio/audio_buffers_state.h
@@ -0,0 +1,32 @@
+// 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.
+
+#ifndef MEDIA_AUDIO_AUDIO_BUFFERS_STATE_H_
+#define MEDIA_AUDIO_AUDIO_BUFFERS_STATE_H_
+
+#include "media/base/media_export.h"
+
+namespace media {
+
+// AudioBuffersState struct stores current state of audio buffers.
+// It is used for audio synchronization.
+struct MEDIA_EXPORT AudioBuffersState {
+  AudioBuffersState();
+  AudioBuffersState(int pending_bytes, int hardware_delay_bytes);
+
+  int total_bytes() {
+    return pending_bytes + hardware_delay_bytes;
+  }
+
+  // Number of bytes we currently have in our software buffer.
+  int pending_bytes;
+
+  // Number of bytes that have been written to the device, but haven't
+  // been played yet.
+  int hardware_delay_bytes;
+};
+
+}  // namespace media
+
+#endif  // MEDIA_AUDIO_AUDIO_BUFFERS_STATE_H_
diff --git a/media/audio/audio_io.h b/media/audio/audio_io.h
index cb0ca59..1e1eba4 100644
--- a/media/audio/audio_io.h
+++ b/media/audio/audio_io.h
@@ -6,6 +6,7 @@
 #define MEDIA_AUDIO_AUDIO_IO_H_
 
 #include "base/basictypes.h"
+#include "media/audio/audio_buffers_state.h"
 #include "media/base/audio_bus.h"
 
 // Low-level audio output support. To make sound there are 3 objects involved:
@@ -57,10 +58,10 @@ class MEDIA_EXPORT AudioOutputStream {
   class MEDIA_EXPORT AudioSourceCallback {
    public:
     // Provide more data by fully filling |dest|.  The source will return
-    // the number of frames it filled.  |total_bytes_delay| contains current
-    // number of bytes of delay buffered by the AudioOutputStream.
+    // the number of frames it filled.  |buffers_state| contains current state
+    // of the buffers, and can be used by the source to calculate delay.
     virtual int OnMoreData(AudioBus* dest,
-                           int total_bytes_delay) = 0;
+                           AudioBuffersState buffers_state) = 0;
 
     // There was an error while playing a buffer. Audio source cannot be
     // destroyed yet. No direct action needed by the AudioStream, but it is
diff --git a/media/audio/audio_low_latency_input_output_unittest.cc b/media/audio/audio_low_latency_input_output_unittest.cc
index 0a572c9..5632896 100644
--- a/media/audio/audio_low_latency_input_output_unittest.cc
+++ b/media/audio/audio_low_latency_input_output_unittest.cc
@@ -226,14 +226,23 @@ class FullDuplexAudioSinkSource
 
   // AudioOutputStream::AudioSourceCallback.
   virtual int OnMoreData(AudioBus* audio_bus,
-                         int total_bytes_delay) OVERRIDE {
+                         AudioBuffersState buffers_state) OVERRIDE {
     base::AutoLock lock(lock_);
 
     // Update one component in the AudioDelayState for the packet
     // which is about to be played out.
     if (output_elements_to_write_ < kMaxDelayMeasurements) {
+      int output_delay_bytes = buffers_state.hardware_delay_bytes;
+#if defined(OS_WIN)
+      // Special fix for Windows in combination with Wave where the
+      // pending bytes field of the audio buffer state is used to
+      // report the delay.
+      if (!CoreAudioUtil::IsSupported()) {
+        output_delay_bytes = buffers_state.pending_bytes;
+      }
+#endif
       delay_states_[output_elements_to_write_].output_delay_ms =
-          BytesToMilliseconds(total_bytes_delay);
+          BytesToMilliseconds(output_delay_bytes);
       ++output_elements_to_write_;
     }
 
diff --git a/media/audio/audio_output_controller.cc b/media/audio/audio_output_controller.cc
index 845af9a..fb01254 100644
--- a/media/audio/audio_output_controller.cc
+++ b/media/audio/audio_output_controller.cc
@@ -290,7 +290,7 @@ void AudioOutputController::DoReportError() {
 }
 
 int AudioOutputController::OnMoreData(AudioBus* dest,
-                                      int total_bytes_delay) {
+                                      AudioBuffersState buffers_state) {
   TRACE_EVENT0("audio", "AudioOutputController::OnMoreData");
 
   // Indicate that we haven't wedged (at least not indefinitely, WedgeCheck()
@@ -304,7 +304,7 @@ int AudioOutputController::OnMoreData(AudioBus* dest,
 
   const int frames = dest->frames();
   sync_reader_->UpdatePendingBytes(
-      total_bytes_delay + frames * params_.GetBytesPerFrame());
+      buffers_state.total_bytes() + frames * params_.GetBytesPerFrame());
 
 #if defined(AUDIO_POWER_MONITORING)
   power_monitor_.Scan(*dest, frames);
diff --git a/media/audio/audio_output_controller.h b/media/audio/audio_output_controller.h
index 795b4cf..0b02ee2 100644
--- a/media/audio/audio_output_controller.h
+++ b/media/audio/audio_output_controller.h
@@ -149,7 +149,7 @@ class MEDIA_EXPORT AudioOutputController
 
   // AudioSourceCallback implementation.
   virtual int OnMoreData(AudioBus* dest,
-                         int total_bytes_delay) OVERRIDE;
+                         AudioBuffersState buffers_state) OVERRIDE;
   virtual void OnError(AudioOutputStream* stream) OVERRIDE;
 
   // AudioDeviceListener implementation.  When called AudioOutputController will
diff --git a/media/audio/audio_output_controller_unittest.cc b/media/audio/audio_output_controller_unittest.cc
index a4ecdb8..125763c 100644
--- a/media/audio/audio_output_controller_unittest.cc
+++ b/media/audio/audio_output_controller_unittest.cc
@@ -192,7 +192,7 @@ class AudioOutputControllerTest : public testing::Test {
     scoped_ptr<AudioBus> dest = AudioBus::Create(params_);
     ASSERT_TRUE(!!mock_stream_.callback());
     const int frames_read =
-        mock_stream_.callback()->OnMoreData(dest.get(), 0);
+        mock_stream_.callback()->OnMoreData(dest.get(), AudioBuffersState());
     EXPECT_LT(0, frames_read);
     EXPECT_EQ(kBufferNonZeroData, dest->channel(0)[0]);
   }
diff --git a/media/audio/audio_output_proxy_unittest.cc b/media/audio/audio_output_proxy_unittest.cc
index 5f0ab06..b8f23ac 100644
--- a/media/audio/audio_output_proxy_unittest.cc
+++ b/media/audio/audio_output_proxy_unittest.cc
@@ -25,6 +25,7 @@ using ::testing::NotNull;
 using ::testing::Return;
 using ::testing::SetArrayArgument;
 using media::AudioBus;
+using media::AudioBuffersState;
 using media::AudioInputStream;
 using media::AudioManager;
 using media::AudioManagerBase;
@@ -124,7 +125,7 @@ class MockAudioManager : public AudioManagerBase {
 
 class MockAudioSourceCallback : public AudioOutputStream::AudioSourceCallback {
  public:
-  int OnMoreData(AudioBus* audio_bus, int total_bytes_delay) {
+  int OnMoreData(AudioBus* audio_bus, AudioBuffersState buffers_state) {
     audio_bus->Zero();
     return audio_bus->frames();
   }
diff --git a/media/audio/audio_output_resampler.cc b/media/audio/audio_output_resampler.cc
index 6375096..15633bb 100644
--- a/media/audio/audio_output_resampler.cc
+++ b/media/audio/audio_output_resampler.cc
@@ -30,7 +30,7 @@ class OnMoreDataConverter
 
   // AudioSourceCallback interface.
   virtual int OnMoreData(AudioBus* dest,
-                         int total_bytes_delay) OVERRIDE;
+                         AudioBuffersState buffers_state) OVERRIDE;
   virtual void OnError(AudioOutputStream* stream) OVERRIDE;
 
   // Sets |source_callback_|.  If this is not a new object, then Stop() must be
@@ -54,9 +54,9 @@ class OnMoreDataConverter
   // Source callback.
   AudioOutputStream::AudioSourceCallback* source_callback_;
 
-  // Last |total_bytes_delay| received via OnMoreData(), used to correct
+  // Last AudioBuffersState object received via OnMoreData(), used to correct
   // playback delay by ProvideInput() and passed on to |source_callback_|.
-  int current_total_bytes_delay_;
+  AudioBuffersState current_buffers_state_;
 
   const int input_bytes_per_second_;
 
@@ -327,8 +327,8 @@ void OnMoreDataConverter::Stop() {
 }
 
 int OnMoreDataConverter::OnMoreData(AudioBus* dest,
-                                    int total_bytes_delay) {
-  current_total_bytes_delay_ = total_bytes_delay;
+                                    AudioBuffersState buffers_state) {
+  current_buffers_state_ = buffers_state;
   audio_converter_.Convert(dest);
 
   // Always return the full number of frames requested, ProvideInput()
@@ -341,12 +341,13 @@ double OnMoreDataConverter::ProvideInput(AudioBus* dest,
   // Adjust playback delay to include |buffer_delay|.
   // TODO(dalecurtis): Stop passing bytes around, it doesn't make sense since
   // AudioBus is just float data.  Use TimeDelta instead.
-  int new_total_bytes_delay =
-      io_ratio_ * (current_total_bytes_delay_ +
+  AudioBuffersState new_buffers_state;
+  new_buffers_state.pending_bytes =
+      io_ratio_ * (current_buffers_state_.total_bytes() +
                    buffer_delay.InSecondsF() * input_bytes_per_second_);
 
   // Retrieve data from the original callback.
-  const int frames = source_callback_->OnMoreData(dest, new_total_bytes_delay);
+  const int frames = source_callback_->OnMoreData(dest, new_buffers_state);
 
   // Zero any unfilled frames if anything was filled, otherwise we'll just
   // return a volume of zero and let AudioConverter drop the output.
diff --git a/media/audio/audio_output_resampler.h b/media/audio/audio_output_resampler.h
index af7ebf2..fa488aa 100644
--- a/media/audio/audio_output_resampler.h
+++ b/media/audio/audio_output_resampler.h
@@ -24,8 +24,8 @@ class OnMoreDataConverter;
 // AudioConverter class for details on the conversion process.
 //
 // AOR works by intercepting the AudioSourceCallback provided to StartStream()
-// and redirecting it through an AudioConverter instance.  |total_bytes_delay|
-// is adjusted for buffer delay caused by the conversion process.
+// and redirecting it through an AudioConverter instance.  AudioBuffersState is
+// adjusted for buffer delay caused by the conversion process.
 //
 // AOR will automatically fall back from AUDIO_PCM_LOW_LATENCY to
 // AUDIO_PCM_LINEAR if the output device fails to open at the requested output
diff --git a/media/audio/cras/cras_unified.cc b/media/audio/cras/cras_unified.cc
index 232fee1..25af383 100644
--- a/media/audio/cras/cras_unified.cc
+++ b/media/audio/cras/cras_unified.cc
@@ -358,7 +358,8 @@ uint32 CrasUnifiedStream::ReadWriteAudio(size_t frames,
   total_delay_bytes += GetBytesLatency(latency_ts);
 
   int frames_filled = source_callback_->OnMoreData(
-      output_bus_.get(), total_delay_bytes);
+      output_bus_.get(),
+      AudioBuffersState(0, total_delay_bytes));
 
   output_bus_->ToInterleaved(frames_filled, bytes_per_sample, output_samples);
 
@@ -375,7 +376,7 @@ uint32 CrasUnifiedStream::WriteAudio(size_t frames,
   cras_client_calc_playback_latency(sample_ts, &latency_ts);
 
   int frames_filled = source_callback_->OnMoreData(
-      output_bus_.get(), GetBytesLatency(latency_ts));
+      output_bus_.get(), AudioBuffersState(0, GetBytesLatency(latency_ts)));
 
   // Note: If this ever changes to output raw float the data must be clipped and
   // sanitized since it may come from an untrusted source such as NaCl.
diff --git a/media/audio/fake_audio_consumer_unittest.cc b/media/audio/fake_audio_consumer_unittest.cc
index ad04cb6..cb2f75c 100644
--- a/media/audio/fake_audio_consumer_unittest.cc
+++ b/media/audio/fake_audio_consumer_unittest.cc
@@ -5,6 +5,7 @@
 #include "base/bind.h"
 #include "base/message_loop/message_loop.h"
 #include "base/time/time.h"
+#include "media/audio/audio_buffers_state.h"
 #include "media/audio/audio_parameters.h"
 #include "media/audio/fake_audio_consumer.h"
 #include "media/audio/simple_sources.h"
@@ -29,7 +30,7 @@ class FakeAudioConsumerTest : public testing::Test {
   virtual ~FakeAudioConsumerTest() {}
 
   void ConsumeData(AudioBus* audio_bus) {
-    source_.OnMoreData(audio_bus, 0);
+    source_.OnMoreData(audio_bus, AudioBuffersState());
   }
 
   void RunOnAudioThread() {
diff --git a/media/audio/fake_audio_output_stream.cc b/media/audio/fake_audio_output_stream.cc
index d5e0d5f..0448c23 100644
--- a/media/audio/fake_audio_output_stream.cc
+++ b/media/audio/fake_audio_output_stream.cc
@@ -61,7 +61,7 @@ void FakeAudioOutputStream::GetVolume(double* volume) {
 
 void FakeAudioOutputStream::CallOnMoreData(AudioBus* audio_bus) {
   DCHECK(audio_manager_->GetWorkerTaskRunner()->BelongsToCurrentThread());
-  callback_->OnMoreData(audio_bus, 0);
+  callback_->OnMoreData(audio_bus, AudioBuffersState());
 }
 
 }  // namespace media
diff --git a/media/audio/mac/audio_auhal_mac.cc b/media/audio/mac/audio_auhal_mac.cc
index 92a1b18..41fc57c 100644
--- a/media/audio/mac/audio_auhal_mac.cc
+++ b/media/audio/mac/audio_auhal_mac.cc
@@ -228,8 +228,9 @@ void AUHALStream::ProvideInput(int frame_delay, AudioBus* dest) {
   // Supply the input data and render the output data.
   source_->OnMoreData(
       dest,
-      current_hardware_pending_bytes_ +
-      frame_delay * params_.GetBytesPerFrame());
+      AudioBuffersState(0,
+                        current_hardware_pending_bytes_ +
+                            frame_delay * params_.GetBytesPerFrame()));
   dest->Scale(volume_);
 }
 
diff --git a/media/audio/mock_audio_source_callback.h b/media/audio/mock_audio_source_callback.h
index 89814cf..d24ce44 100644
--- a/media/audio/mock_audio_source_callback.h
+++ b/media/audio/mock_audio_source_callback.h
@@ -16,7 +16,7 @@ class MockAudioSourceCallback : public AudioOutputStream::AudioSourceCallback {
   virtual ~MockAudioSourceCallback();
 
   MOCK_METHOD2(OnMoreData, int(AudioBus* audio_bus,
-                               int total_bytes_delay));
+                               AudioBuffersState buffers_state));
   MOCK_METHOD1(OnError, void(AudioOutputStream* stream));
 
  private:
diff --git a/media/audio/pulse/pulse_output.cc b/media/audio/pulse/pulse_output.cc
index 1048113..e0a0b42 100644
--- a/media/audio/pulse/pulse_output.cc
+++ b/media/audio/pulse/pulse_output.cc
@@ -133,7 +133,7 @@ void PulseAudioOutputStream::FulfillWriteRequest(size_t requested_bytes) {
       const uint32 hardware_delay = pulse::GetHardwareLatencyInBytes(
           pa_stream_, params_.sample_rate(), params_.GetBytesPerFrame());
       frames_filled = source_callback_->OnMoreData(
-          audio_bus_.get(), hardware_delay);
+          audio_bus_.get(), AudioBuffersState(0, hardware_delay));
 
       // Zero any unfilled data so it plays back as silence.
       if (frames_filled < audio_bus_->frames()) {
diff --git a/media/audio/simple_sources.cc b/media/audio/simple_sources.cc
index 2c22e16..039029e 100644
--- a/media/audio/simple_sources.cc
+++ b/media/audio/simple_sources.cc
@@ -29,7 +29,7 @@ SineWaveAudioSource::SineWaveAudioSource(int channels,
 // The implementation could be more efficient if a lookup table is constructed
 // but it is efficient enough for our simple needs.
 int SineWaveAudioSource::OnMoreData(AudioBus* audio_bus,
-                                    int total_bytes_delay) {
+                                    AudioBuffersState audio_buffers) {
   base::AutoLock auto_lock(time_lock_);
   callbacks_++;
 
diff --git a/media/audio/simple_sources.h b/media/audio/simple_sources.h
index feda33b..6303386 100644
--- a/media/audio/simple_sources.h
+++ b/media/audio/simple_sources.h
@@ -28,7 +28,7 @@ class MEDIA_EXPORT SineWaveAudioSource
 
   // Implementation of AudioSourceCallback.
   virtual int OnMoreData(AudioBus* audio_bus,
-                         int total_bytes_delay) OVERRIDE;
+                         AudioBuffersState audio_buffers) OVERRIDE;
   virtual void OnError(AudioOutputStream* stream) OVERRIDE;
 
   // The number of OnMoreData()+OnMoreIOData() and OnError() calls respectively.
diff --git a/media/audio/simple_sources_unittest.cc b/media/audio/simple_sources_unittest.cc
index 51cb740..5ee8686 100644
--- a/media/audio/simple_sources_unittest.cc
+++ b/media/audio/simple_sources_unittest.cc
@@ -26,7 +26,7 @@ TEST(SimpleSources, SineWaveAudioSource) {
 
   SineWaveAudioSource source(1, freq, params.sample_rate());
   scoped_ptr<AudioBus> audio_bus = AudioBus::Create(params);
-  source.OnMoreData(audio_bus.get(), 0);
+  source.OnMoreData(audio_bus.get(), AudioBuffersState());
   EXPECT_EQ(1, source.callbacks());
   EXPECT_EQ(0, source.errors());
 
@@ -56,13 +56,13 @@ TEST(SimpleSources, SineWaveAudioCapped) {
 
   scoped_ptr<AudioBus> audio_bus = AudioBus::Create(1, 2 * kSampleCap);
   EXPECT_EQ(source.OnMoreData(
-      audio_bus.get(), 0), kSampleCap);
+      audio_bus.get(), AudioBuffersState()), kSampleCap);
   EXPECT_EQ(1, source.callbacks());
-  EXPECT_EQ(source.OnMoreData(audio_bus.get(), 0), 0);
+  EXPECT_EQ(source.OnMoreData(audio_bus.get(), AudioBuffersState()), 0);
   EXPECT_EQ(2, source.callbacks());
   source.Reset();
   EXPECT_EQ(source.OnMoreData(
-      audio_bus.get(), 0), kSampleCap);
+      audio_bus.get(), AudioBuffersState()), kSampleCap);
   EXPECT_EQ(3, source.callbacks());
   EXPECT_EQ(0, source.errors());
 }
diff --git a/media/audio/sounds/audio_stream_handler.cc b/media/audio/sounds/audio_stream_handler.cc
index 7e48b2d..645fcb3 100644
--- a/media/audio/sounds/audio_stream_handler.cc
+++ b/media/audio/sounds/audio_stream_handler.cc
@@ -105,7 +105,7 @@ class AudioStreamHandler::AudioStreamContainer
   // AudioOutputStream::AudioSourceCallback overrides:
   // Following methods could be called from *ANY* thread.
   virtual int OnMoreData(AudioBus* dest,
-                         int /* total_bytes_delay */) OVERRIDE {
+                         AudioBuffersState /* state */) OVERRIDE {
     base::AutoLock al(state_lock_);
     size_t bytes_written = 0;
 
diff --git a/media/audio/virtual_audio_input_stream_unittest.cc b/media/audio/virtual_audio_input_stream_unittest.cc
index c82a9ff..3aa87b0 100644
--- a/media/audio/virtual_audio_input_stream_unittest.cc
+++ b/media/audio/virtual_audio_input_stream_unittest.cc
@@ -67,9 +67,8 @@ class TestAudioSource : public SineWaveAudioSource {
   virtual ~TestAudioSource() {}
 
   virtual int OnMoreData(AudioBus* audio_bus,
-                         int total_bytes_delay) OVERRIDE {
-    const int ret = SineWaveAudioSource::OnMoreData(audio_bus,
-                                                    total_bytes_delay);
+                         AudioBuffersState audio_buffers) OVERRIDE {
+    const int ret = SineWaveAudioSource::OnMoreData(audio_bus, audio_buffers);
     data_pulled_.Signal();
     return ret;
   }
diff --git a/media/audio/virtual_audio_output_stream.cc b/media/audio/virtual_audio_output_stream.cc
index 016c2f3..43b83cf 100644
--- a/media/audio/virtual_audio_output_stream.cc
+++ b/media/audio/virtual_audio_output_stream.cc
@@ -77,7 +77,7 @@ double VirtualAudioOutputStream::ProvideInput(AudioBus* audio_bus,
   // platform.
   DCHECK(callback_);
 
-  const int frames = callback_->OnMoreData(audio_bus, 0);
+  const int frames = callback_->OnMoreData(audio_bus, AudioBuffersState());
   if (frames < audio_bus->frames())
     audio_bus->ZeroFramesPartial(frames, audio_bus->frames() - frames);
 
diff --git a/media/audio/win/audio_low_latency_output_win.cc b/media/audio/win/audio_low_latency_output_win.cc
index 99576a6..15cabbaf 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -498,9 +498,13 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
 
     // Read a data packet from the registered client source and
     // deliver a delay estimate in the same callback to the client.
+    // A time stamp is also stored in the AudioBuffersState. This
+    // time stamp can be used at the client side to compensate for
+    // the delay between the usage of the delay value and the time
+    // of generation.
 
     int frames_filled = source_->OnMoreData(
-        audio_bus_.get(), audio_delay_bytes);
+        audio_bus_.get(), AudioBuffersState(0, audio_delay_bytes));
     uint32 num_filled_bytes = frames_filled * format_.Format.nBlockAlign;
     DCHECK_LE(num_filled_bytes, packet_size_bytes_);
 
diff --git a/media/audio/win/audio_low_latency_output_win_unittest.cc b/media/audio/win/audio_low_latency_output_win_unittest.cc
index 30fa9ca..ed03d2b 100644
--- a/media/audio/win/audio_low_latency_output_win_unittest.cc
+++ b/media/audio/win/audio_low_latency_output_win_unittest.cc
@@ -52,7 +52,7 @@ 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 >= 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 @@ class ReadFromFileAudioSource : public AudioOutputStream::AudioSourceCallback {
 
   // 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 @@ TEST(WASAPIAudioOutputStreamTest, ValidPacketSize) {
   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 @@ TEST(WASAPIAudioOutputStreamTest, ExclusiveModeMinBufferSizeAt48kHz) {
   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 @@ TEST(WASAPIAudioOutputStreamTest, ExclusiveModeMinBufferSizeAt44kHz) {
   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())))
diff --git a/media/audio/win/audio_output_win_unittest.cc b/media/audio/win/audio_output_win_unittest.cc
index 6257a60..2902657 100644
--- a/media/audio/win/audio_output_win_unittest.cc
+++ b/media/audio/win/audio_output_win_unittest.cc
@@ -37,7 +37,7 @@ 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, int total_bytes_delay) {
+static int ClearData(AudioBus* audio_bus, AudioBuffersState buffers_state) {
   audio_bus->Zero();
   return audio_bus->frames();
 }
@@ -52,7 +52,7 @@ class TestSourceBasic : public AudioOutputStream::AudioSourceCallback {
   }
   // AudioSourceCallback::OnMoreData implementation:
   virtual int OnMoreData(AudioBus* audio_bus,
-                         int total_bytes_delay) {
+                         AudioBuffersState buffers_state) {
     ++callback_count_;
     // Touch the channel memory value to make sure memory is good.
     audio_bus->Zero();
@@ -89,9 +89,9 @@ class TestSourceLaggy : public TestSourceBasic {
       : laggy_after_buffer_(laggy_after_buffer), lag_in_ms_(lag_in_ms) {
   }
   virtual int OnMoreData(AudioBus* audio_bus,
-                         int total_bytes_delay) {
+                         AudioBuffersState buffers_state) {
     // Call the base, which increments the callback_count_.
-    TestSourceBasic::OnMoreData(audio_bus, total_bytes_delay);
+    TestSourceBasic::OnMoreData(audio_bus, buffers_state);
     if (callback_count() > kMaxNumBuffers) {
       ::Sleep(lag_in_ms_);
     }
@@ -520,22 +520,32 @@ TEST(WinAudioTest, PCMWaveStreamPendingBytes) {
   // pending bytes will go down and eventually read zero.
   InSequence s;
 
-  EXPECT_CALL(source, OnMoreData(NotNull(), 0))
+  EXPECT_CALL(source, OnMoreData(NotNull(),
+                                 Field(&AudioBuffersState::pending_bytes, 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(), bytes_100_ms))
+  EXPECT_CALL(source, OnMoreData(NotNull(),
+                                 Field(&AudioBuffersState::pending_bytes,
+                                       bytes_100_ms)))
       .WillOnce(Invoke(ClearData));
-  EXPECT_CALL(source, OnMoreData(NotNull(), 2 * bytes_100_ms))
+  EXPECT_CALL(source, OnMoreData(NotNull(),
+                                 Field(&AudioBuffersState::pending_bytes,
+                                       2 * bytes_100_ms)))
       .WillOnce(Invoke(ClearData));
-  EXPECT_CALL(source, OnMoreData(NotNull(), 2 * bytes_100_ms))
+  EXPECT_CALL(source, OnMoreData(NotNull(),
+                                 Field(&AudioBuffersState::pending_bytes,
+                                       2 * bytes_100_ms)))
       .Times(AnyNumber())
       .WillRepeatedly(Return(0));
-  EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms))
+  EXPECT_CALL(source, OnMoreData(NotNull(),
+                                 Field(&AudioBuffersState::pending_bytes,
+                                       bytes_100_ms)))
       .Times(AnyNumber())
       .WillRepeatedly(Return(0));
-  EXPECT_CALL(source, OnMoreData(NotNull(), 0))
+  EXPECT_CALL(source, OnMoreData(NotNull(),
+                                 Field(&AudioBuffersState::pending_bytes, 0)))
       .Times(AnyNumber())
       .WillRepeatedly(Return(0));
 
@@ -561,8 +571,8 @@ class SyncSocketSource : public AudioOutputStream::AudioSourceCallback {
 
   // AudioSourceCallback::OnMoreData implementation:
   virtual int OnMoreData(AudioBus* audio_bus,
-                         int total_bytes_delay) {
-    socket_->Send(&total_bytes_delay, sizeof(total_bytes_delay));
+                         AudioBuffersState buffers_state) {
+    socket_->Send(&buffers_state, sizeof(buffers_state));
     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);
@@ -570,7 +580,7 @@ class SyncSocketSource : public AudioOutputStream::AudioSourceCallback {
   }
   virtual int OnMoreIOData(AudioBus* source,
                            AudioBus* dest,
-                           int total_bytes_delay) {
+                           AudioBuffersState buffers_state) {
     NOTREACHED();
     return 0;
   }
@@ -612,13 +622,13 @@ DWORD __stdcall SyncSocketThread(void* context) {
   SineWaveAudioSource sine(1, ctx.sine_freq, ctx.sample_rate);
   const int kTwoSecFrames = ctx.sample_rate * 2;
 
-  int total_bytes_delay = 0;
+  AudioBuffersState buffers_state;
   int times = 0;
   for (int ix = 0; ix < kTwoSecFrames; ix += ctx.frames) {
-    if (ctx.socket->Receive(&total_bytes_delay, sizeof(total_bytes_delay)) == 0)
+    if (ctx.socket->Receive(&buffers_state, sizeof(buffers_state)) == 0)
       break;
-    if ((times > 0) && (total_bytes_delay < 1000)) __debugbreak();
-    sine.OnMoreData(audio_bus.get(), total_bytes_delay);
+    if ((times > 0) && (buffers_state.pending_bytes < 1000)) __debugbreak();
+    sine.OnMoreData(audio_bus.get(), buffers_state);
     ctx.socket->Send(data.get(), ctx.packet_size_bytes);
     ++times;
   }
diff --git a/media/audio/win/waveout_output_win.cc b/media/audio/win/waveout_output_win.cc
index 4d11e08..0f54817 100644
--- a/media/audio/win/waveout_output_win.cc
+++ b/media/audio/win/waveout_output_win.cc
@@ -325,10 +325,9 @@ void PCMWaveOutAudioOutputStream::QueueNextPacket(WAVEHDR *buffer) {
   // return to us how many bytes were used.
   // TODO(fbarchard): Handle used 0 by queueing more.
 
-  // TODO(sergeyu): Specify correct hardware delay for |total_delay_bytes|.
-  int total_delay_bytes = pending_bytes_;
+  // TODO(sergeyu): Specify correct hardware delay for AudioBuffersState.
   int frames_filled = callback_->OnMoreData(
-      audio_bus_.get(), total_delay_bytes);
+      audio_bus_.get(), AudioBuffersState(pending_bytes_, 0));
   uint32 used = frames_filled * audio_bus_->channels() *
       format_.Format.wBitsPerSample / 8;
 
diff --git a/media/cast/test/receiver.cc b/media/cast/test/receiver.cc
index 7ded86a..65b0e60 100644
--- a/media/cast/test/receiver.cc
+++ b/media/cast/test/receiver.cc
@@ -316,7 +316,7 @@ class NaivePlayer : public InProcessReceiver,
   ////////////////////////////////////////////////////////////////////
   // AudioSourceCallback implementation.
 
-  virtual int OnMoreData(AudioBus* dest, int total_bytes_delay)
+  virtual int OnMoreData(AudioBus* dest, AudioBuffersState buffers_state)
       OVERRIDE {
     // Note: This method is being invoked by a separate thread unknown to us
     // (i.e., outside of CastEnvironment).
@@ -329,8 +329,8 @@ class NaivePlayer : public InProcessReceiver,
         base::AutoLock auto_lock(audio_lock_);
 
         // Prune the queue, skipping entries that are too old.
-        // TODO(miu): Use |total_bytes_delay| to account for audio buffering
-        // delays upstream.
+        // TODO(miu): Use |buffers_state| to account for audio buffering delays
+        // upstream.
         const base::TimeTicks earliest_time_to_play =
             cast_env()->Clock()->NowTicks() - max_frame_age_;
         while (!audio_playout_queue_.empty() &&
diff --git a/media/cast/test/utility/audio_utility.cc b/media/cast/test/utility/audio_utility.cc
index 094f141..8dde4dd 100644
--- a/media/cast/test/utility/audio_utility.cc
+++ b/media/cast/test/utility/audio_utility.cc
@@ -36,7 +36,7 @@ scoped_ptr<AudioBus> TestAudioBusFactory::NextAudioBus(
   const int num_samples = static_cast<int>((sample_rate_ * duration) /
                                            base::TimeDelta::FromSeconds(1));
   scoped_ptr<AudioBus> bus(AudioBus::Create(num_channels_, num_samples));
-  source_.OnMoreData(bus.get(), 0);
+  source_.OnMoreData(bus.get(), AudioBuffersState());
   bus->Scale(volume_);
   return bus.Pass();
 }
diff --git a/media/media.gyp b/media/media.gyp
index 5adb828..027cb55 100644
--- a/media/media.gyp
+++ b/media/media.gyp
@@ -91,6 +91,8 @@
         'audio/android/opensles_output.cc',
         'audio/android/opensles_output.h',
         'audio/android/opensles_wrapper.cc',
+        'audio/audio_buffers_state.cc',
+        'audio/audio_buffers_state.h',
         'audio/audio_device_name.cc',
         'audio/audio_device_name.h',
         'audio/audio_device_thread.cc',