Switch browser side audio capture path to use base time primitives.

media/audio/win/audio_low_latency_input_win.cc

Index: media/audio/win/audio_low_latency_input_win.cc
diff --git a/media/audio/win/audio_low_latency_input_win.cc b/media/audio/win/audio_low_latency_input_win.cc
index 22355580aac45b2c656ae6066bd25fd1b72f1a3f..94fc69e9ad05c6d90031a8331e653ba816fca7e0 100644
--- a/media/audio/win/audio_low_latency_input_win.cc
+++ b/media/audio/win/audio_low_latency_input_win.cc
@@ -326,7 +326,6 @@ void WASAPIAudioInputStream::Run() {
                2 * packet_size_frames_ * frame_size_);
   std::unique_ptr<uint8_t[]> capture_buffer(new uint8_t[capture_buffer_size]);
 
-  LARGE_INTEGER now_count = {};
   bool recording = true;
   bool error = false;
   double volume = GetVolume();
@@ -401,8 +400,21 @@ void WASAPIAudioInputStream::Run() {
         // The value contains two parts (A+B), where A is the delay of the
         // first audio frame in the packet and B is the extra delay
         // contained in any stored data. Unit is in audio frames.
-        QueryPerformanceCounter(&now_count);
         // first_audio_frame_timestamp will be 0 if we didn't get a timestamp.
+
+        base::TimeTicks delay_timestamp =
+            base::TimeTicks::FromQPCValue(first_audio_frame_timestamp);
+
+        // TODO(dalecurtis): This is all wrong and brain is mush, fix tomorrow.
+        // Needs to compensate for input buffers from past cycles.
+        base::TimeDelta delay =
+            first_audio_frame_timestamp
+                ? base::TimeTicks::Now() - delay_timestamp
+                : base::TimeDelta::FromSecondsD(
+                      num_frames_to_read /
+                      static_cast<double>(format_.nSamplesPerSec));
+
+#if 0
         double audio_delay_frames =
             first_audio_frame_timestamp == 0
                 ? num_frames_to_read
@@ -411,7 +423,7 @@ void WASAPIAudioInputStream::Run() {
                    10000.0) *
                           ms_to_frame_count_ +
                       buffer_frame_index - num_frames_to_read;
-
+#endif
         // Get a cached AGC volume level which is updated once every second
         // on the audio manager thread. Note that, |volume| is also updated
         // each time SetVolume() is called through IPC by the render-side AGC.
@@ -419,7 +431,6 @@ void WASAPIAudioInputStream::Run() {
 
         // Deliver captured data to the registered consumer using a packet
         // size which was specified at construction.
-        uint32_t delay_frames = static_cast<uint32_t>(audio_delay_frames + 0.5);
         while (buffer_frame_index >= packet_size_frames_) {
           // Copy data to audio bus to match the OnData interface.
           uint8_t* audio_data =
@@ -429,8 +440,7 @@ void WASAPIAudioInputStream::Run() {
 
           // Deliver data packet, delay estimation and volume level to
           // the user.
-          sink_->OnData(this, audio_bus_.get(), delay_frames * frame_size_,
-                        volume);
+          sink_->OnData(this, audio_bus_.get(), delay, delay_timestamp, volume);
 
           // Store parts of the recorded data which can't be delivered
           // using the current packet size. The stored section will be used
@@ -445,11 +455,6 @@ void WASAPIAudioInputStream::Run() {
 
           DCHECK_GE(buffer_frame_index, packet_size_frames_);
           buffer_frame_index -= packet_size_frames_;
-          if (delay_frames > packet_size_frames_) {
-            delay_frames -= packet_size_frames_;
-          } else {
-            delay_frames = 0;
-          }
         }
       } break;
       default: