Remove AudioBuffersState usage in Chromium

media/audio/win/audio_low_latency_output_win.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 "media/audio/win/audio_low_latency_output_win.h"
 
 #include <Functiondiscoverykeys_devpkey.h>
 
 #include "base/command_line.h"
 #include "base/debug/trace_event.h"
@@ skipping 458 matching lines @@
                  << std::hex << hr;
       return false;
     }
 
     // Derive the audio delay which corresponds to the delay between
     // a render event and the time when the first audio sample in a
     // packet is played out through the speaker. This delay value
     // can typically be utilized by an acoustic echo-control (AEC)
     // unit at the render side.
     UINT64 position = 0;
-    int audio_delay_bytes = 0;
+    uint32 audio_delay_bytes = 0;
     hr = audio_clock_->GetPosition(&position, NULL);
     if (SUCCEEDED(hr)) {
       // Stream position of the sample that is currently playing
       // through the speaker.
       double pos_sample_playing_frames = format_.Format.nSamplesPerSec *
           (static_cast<double>(position) / device_frequency);
 
       // Stream position of the last sample written to the endpoint
       // buffer. Note that, the packet we are about to receive in
       // the upcoming callback is also included.
       size_t pos_last_sample_written_frames =
           num_written_frames_ + packet_size_frames_;
 
       // Derive the actual delay value which will be fed to the
       // render client using the OnMoreData() callback.
       audio_delay_bytes = (pos_last_sample_written_frames -
           pos_sample_playing_frames) *  format_.Format.nBlockAlign;
     }
 
     // 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(), AudioBuffersState(0, audio_delay_bytes));
+        audio_bus_.get(), audio_delay_bytes);
     uint32 num_filled_bytes = frames_filled * format_.Format.nBlockAlign;
     DCHECK_LE(num_filled_bytes, packet_size_bytes_);
 
     // 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.
     const int bytes_per_sample = format_.Format.wBitsPerSample >> 3;
     audio_bus_->Scale(volume_);
     audio_bus_->ToInterleaved(
         frames_filled, bytes_per_sample, audio_data);
@@ skipping 102 matching lines @@
 
     // Ensure that we don't quit the main thread loop immediately next
     // time Start() is called.
     ResetEvent(stop_render_event_.Get());
   }
 
   source_ = NULL;
 }
 
 }  // namespace media