Pass delay and timestamp to AudioSourceCallback::OnMoreData.

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 9fa035d80ba6aec06859dfcea5c2f04202e0e1cd..9206e685ba3636e9ba7ca63dc4af1806c75bf55f 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -11,6 +11,7 @@
 #include "base/macros.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/utf_string_conversions.h"
+#include "base/time/time.h"
 #include "base/trace_event/trace_event.h"
 #include "base/win/scoped_propvariant.h"
 #include "media/audio/audio_device_description.h"
@@ -511,7 +512,7 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
     // can typically be utilized by an acoustic echo-control (AEC)
     // unit at the render side.
     UINT64 position = 0;
-    uint32_t audio_delay_bytes = 0;
+    base::TimeTicks target_playout_time = base::TimeTicks::Now();
     hr = audio_clock_->GetPosition(&position, NULL);
     if (SUCCEEDED(hr)) {
       // Stream position of the sample that is currently playing
@@ -525,17 +526,18 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
       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;
+      double audio_delay_frames =
+          pos_last_sample_written_frames - pos_sample_playing_frames;
+
+      target_playout_time += base::TimeDelta::FromSecondsD(
+          audio_delay_frames / format_.Format.nSamplesPerSec);
     }
 
     // Read a data packet from the registered client source and
     // deliver a delay estimate in the same callback to the client.
 
     int frames_filled =
-        source_->OnMoreData(audio_bus_.get(), audio_delay_bytes, 0);
+        source_->OnMoreData(target_playout_time, 0, audio_bus_.get());
     uint32_t num_filled_bytes = frames_filled * format_.Format.nBlockAlign;
     DCHECK_LE(num_filled_bytes, packet_size_bytes_);
 
@@ -641,7 +643,7 @@ HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization(
 }
 
 void WASAPIAudioOutputStream::StopThread() {
-  if (render_thread_ ) {
+  if (render_thread_) {
     if (render_thread_->HasBeenStarted()) {
       // Wait until the thread completes and perform cleanup.
       SetEvent(stop_render_event_.Get());