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..c51548a4776f9d57704522236c9fe17b031c134d 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"
@@ -505,37 +506,38 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
       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.
+    // Find the time when the next sample written to the buffer is expected to
+    // be played out through the speaker.
     UINT64 position = 0;
-    uint32_t audio_delay_bytes = 0;
-    hr = audio_clock_->GetPosition(&position, NULL);
+    UINT64 qps_position = 0;
+    base::TimeTicks target_playout_time;
+    hr = audio_clock_->GetPosition(&position, &qps_position);
     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;
+      // Number of frames already played out through the speaker.
+      const uint64_t played_out_frames =
+          format_.Format.nSamplesPerSec * position / device_frequency;
+
+      // Number of frames that have been written to the buffer but not yet
+      // played out. Note that the packet about to be received in the upcoming
+      // callback is included.
+      const uint64_t delay_frames =
+          num_written_frames_ + packet_size_frames_ - played_out_frames;

[miu, 2016/09/16 18:35:58]

Is is correct to include the |packet_size_frames_| in the new calculation?
Unless I'm missing something, it seems that would set |target_playout_delay| to
the *last* sample about to be rendered by OnMoreData(), rather than the *first*
one (which is what the interface intends).

[jameswest, 2016/09/16 21:59:43]

I believe you're correct. I'll change it.

[jameswest, 2016/09/19 23:32:36]

Done.

+
+      // Convert the delay from frames to time.
+      const base::TimeDelta delay = base::TimeDelta::FromMicroseconds(
+          delay_frames * base::Time::kMicrosecondsPerSecond /
+          format_.Format.nSamplesPerSec);
+
+      target_playout_time = base::TimeTicks::FromQPSValue(qps_position) + delay;
+    } else {
+      target_playout_time = base::TimeTicks::Now();
     }
 
     // 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());