Pass delay and timestamp to AudioSourceCallback::OnMoreData.

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/logging.h"
 #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"
 #include "media/audio/win/audio_manager_win.h"
 #include "media/audio/win/avrt_wrapper_win.h"
 #include "media/audio/win/core_audio_util_win.h"
 #include "media/base/limits.h"
 #include "media/base/media_switches.h"
 
 using base::win::ScopedComPtr;
@@ skipping 480 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;
-    uint32_t audio_delay_bytes = 0;
+    base::TimeTicks target_playout_time = base::TimeTicks::Now();

[Mikhail, 2016/09/08 07:44:12]

wouldn't it be more accurate to use the following 'audio_clock_->GetPosition()'
call (providing 2nd argument to it) to retrieve the current playout time than
just calling 'TimeTicks::Now()'? 

TimeTicks can be obtained from the returned QPC value (pls. look at
https://codereview.chromium.org/2060833002/diff/230001/media/audio/win/audio_...)

[miu, 2016/09/08 20:46:48]

Good catch! Yes, that sounds like a great idea. WDYT, James?

However, note that the GetPosition() function return the time at which the
position was measured (i.e., it is not the target_playout_time). So, you'd need
to take that time and also add the delay based on the number of frames that were
queued.

Reference:
https://msdn.microsoft.com/en-us/library/windows/desktop/dd370889(v=vs.85).aspx

[James West, 2016/09/13 07:40:50]

I agree that's the correct way to do it. I implemented it and cleaned up the
logic a bit.

     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;
+      int audio_delay_frames =
+          pos_last_sample_written_frames - pos_sample_playing_frames;
+
+      target_playout_time += base::TimeDelta::FromMicroseconds(
+          audio_delay_frames * base::Time::kMicrosecondsPerSecond /
+          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_);
 
     // 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 85 matching lines @@
     DVLOG(1) << "IAudioClient::GetBufferSize: " << std::hex << hr;
     return hr;
   }
 
   *endpoint_buffer_size = buffer_size_in_frames;
   DVLOG(2) << "endpoint buffer size: " << buffer_size_in_frames;
   return hr;
 }
 
 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());
       render_thread_->Join();
     }
 
     render_thread_.reset();
 
     // 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