Make more media APIs aware of |delay| and |delay_timestamp|

content/browser/renderer_host/media/audio_sync_reader.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 "content/browser/renderer_host/media/audio_sync_reader.h"
 
 #include <algorithm>
 #include <string>
 #include <utility>
 
 #include "base/command_line.h"
 #include "base/format_macros.h"
 #include "base/memory/ptr_util.h"
 #include "base/memory/shared_memory.h"
 #include "base/metrics/histogram_macros.h"
 #include "base/strings/stringprintf.h"
 #include "base/trace_event/trace_event.h"
 #include "build/build_config.h"
 #include "content/browser/renderer_host/media/media_stream_manager.h"
 #include "content/public/common/content_switches.h"
+#include "media/audio/audio_device_thread.h"
 #include "media/base/audio_parameters.h"
 
 using media::AudioBus;
 using media::AudioOutputBuffer;
+using Packet = media::AudioDeviceThread::Packet;
 
 namespace {
 
 // Used to log if any audio glitches have been detected during an audio session.
 // Elements in this enum should not be added, deleted or rearranged.
 enum AudioGlitchResult {
   AUDIO_RENDERER_NO_AUDIO_GLITCHES = 0,
   AUDIO_RENDERER_AUDIO_GLITCHES = 1,
   AUDIO_RENDERER_AUDIO_GLITCHES_MAX = AUDIO_RENDERER_AUDIO_GLITCHES
 };
@@ skipping 98 matching lines @@
       !base::CancelableSyncSocket::CreatePair(socket.get(),
                                               foreign_socket.get())) {
     return nullptr;
   }
   return base::WrapUnique(new AudioSyncReader(params, std::move(shared_memory),
                                               std::move(socket),
                                               std::move(foreign_socket)));
 }
 
 // media::AudioOutputController::SyncReader implementations.
-void AudioSyncReader::UpdatePendingBytes(uint32_t bytes,
-                                         uint32_t frames_skipped) {
+void AudioSyncReader::PrepareNextData(base::TimeDelta delay,
+                                      base::TimeTicks delay_timestamp,
+                                      uint32_t frames_skipped) {
   // Increase the number of skipped frames stored in shared memory. We don't
   // send it over the socket since sending more than 4 bytes might lead to being
   // descheduled. The reading side will zero it when consumed.
   AudioOutputBuffer* buffer =
       reinterpret_cast<AudioOutputBuffer*>(shared_memory_->memory());
   buffer->params.frames_skipped += frames_skipped;
 
   // Zero out the entire output buffer to avoid stuttering/repeating-buffers
   // in the anomalous case if the renderer is unable to keep up with real-time.
   output_bus_->Zero();
 
-  socket_->Send(&bytes, sizeof(bytes));
+  Packet packet = {delay.ToInternalValue(), delay_timestamp.ToInternalValue()};

[DaleCurtis, 2016/10/20 18:35:29]

In the past, there were concerns that sending more than 4 bytes of data might
cause descheduling of the real-time audio thread. Probably this needs to be
watched closely to see if we see an increase in AudioRendererMissedDeadline,
AudioRendererAudioGlitches, etc.

[miu, 2016/10/23 01:19:15]

ditto here: Please use InMicroseconds() instead of ToInternalValue().

[Mikhail, 2016/10/24 19:50:23]

Done.

+  socket_->Send(&packet, sizeof(packet));
   ++buffer_index_;
 }
 
 void AudioSyncReader::Read(AudioBus* dest) {
   ++renderer_callback_count_;
   if (!WaitUntilDataIsReady()) {
     ++trailing_renderer_missed_callback_count_;
     ++renderer_missed_callback_count_;
     if (renderer_missed_callback_count_ <= 100) {
       LOG(WARNING) << "AudioSyncReader::Read timed out, audio glitch count="
@@ skipping 65 matching lines @@
                                base::TimeDelta::FromMilliseconds(1),
                                base::TimeDelta::FromMilliseconds(1000),
                                50);
     return false;
   }
 
   return true;
 }
 
 }  // namespace content