Remove AudioBuffersState usage in Chromium

media/audio/win/audio_low_latency_output_win.cc

Index: media/audio/win/audio_low_latency_output_win.cc
diff --git a/media/audio/win/audio_low_latency_output_win.cc b/media/audio/win/audio_low_latency_output_win.cc
index 53dcbfe81927e9dc9023216b417539a406d8e962..8c9ff2f64e5c6e537a8e6b88ae9307f6c07803ed 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -476,7 +476,7 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
     // 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
@@ -498,13 +498,9 @@ 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(), 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_);