Introduce media::AudioPushFifo and a couple of use cases (and clean-ups).

content/renderer/media/webaudio_capturer_source.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/renderer/media/webaudio_capturer_source.h"
 
+#include "base/bind.h"
+#include "base/bind_helpers.h"
 #include "base/logging.h"
-#include "base/time/time.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 
 using media::AudioBus;
-using media::AudioFifo;
 using media::AudioParameters;
 using media::ChannelLayout;
 using media::CHANNEL_LAYOUT_MONO;
 using media::CHANNEL_LAYOUT_STEREO;
 
-static const int kMaxNumberOfBuffersInFifo = 5;
-
 namespace content {
 
 WebAudioCapturerSource::WebAudioCapturerSource(
     const blink::WebMediaStreamSource& blink_source)
     : track_(NULL),
       audio_format_changed_(false),
-      blink_source_(blink_source) {
-}
+      fifo_(base::Bind(&WebAudioCapturerSource::DeliverRebufferedAudio,
+                       base::Unretained(this))),
+      blink_source_(blink_source) {}
 
 WebAudioCapturerSource::~WebAudioCapturerSource() {
   DCHECK(thread_checker_.CalledOnValidThread());
   removeFromBlinkSource();
 }
 
 void WebAudioCapturerSource::setFormat(
     size_t number_of_channels, float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
            << sample_rate << ")";
 
   // If the channel count is greater than 8, use discrete layout. However,
   // anything beyond 8 is ignored by the subsequent (WebRTC) audio pipeline.
   ChannelLayout channel_layout =
       number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE
                              : media::GuessChannelLayout(number_of_channels);
 
   base::AutoLock auto_lock(lock_);
 
   // Set the format used by this WebAudioCapturerSource. We are using 10ms data
   // as buffer size since that is the native buffer size of WebRtc packet
   // running on.
+  fifo_.Reset(sample_rate / 100);
   params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
-                sample_rate, 16, sample_rate / 100);
+                sample_rate, 16, fifo_.frames_per_buffer());
 
   // Take care of the discrete channel layout case.
   params_.set_channels_for_discrete(number_of_channels);
 
   audio_format_changed_ = true;
 
-  wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
-  capture_bus_ = AudioBus::Create(params_);
-
-  fifo_.reset(new AudioFifo(
-      params_.channels(),
-      kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
+  if (!wrapper_bus_ ||
+      wrapper_bus_->channels() != static_cast<int>(number_of_channels)) {
+    wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
+  }
 }
 
 void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(track);
   base::AutoLock auto_lock(lock_);
   track_ = track;
 }
 
 void WebAudioCapturerSource::Stop() {
   DCHECK(thread_checker_.CalledOnValidThread());
   {
     base::AutoLock auto_lock(lock_);
     track_ = NULL;
   }
   // removeFromBlinkSource() should not be called while |lock_| is acquired,
   // as it could result in a deadlock.
   removeFromBlinkSource();
 }
 
 void WebAudioCapturerSource::consumeAudio(
     const blink::WebVector<const float*>& audio_data,
     size_t number_of_frames) {
+  // TODO(miu): Plumbing is needed to determine the actual capture timestamp
+  // of the audio, instead of just snapshotting TimeTicks::Now(), for proper
+  // audio/video sync.  http://crbug.com/335335
+  current_reference_time_ = base::TimeTicks::Now();
+
   base::AutoLock auto_lock(lock_);
   if (!track_)
     return;
 
   // Update the downstream client if the audio format has been changed.
   if (audio_format_changed_) {
     track_->OnSetFormat(params_);
     audio_format_changed_ = false;
   }
 
   wrapper_bus_->set_frames(number_of_frames);
-
-  // Make sure WebKit is honoring what it told us up front
-  // about the channels.
   DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
-
   for (size_t i = 0; i < audio_data.size(); ++i)
     wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
 
-  // Handle mismatch between WebAudio buffer-size and WebRTC.
-  int available = fifo_->max_frames() - fifo_->frames();
-  if (available < static_cast<int>(number_of_frames)) {
-    NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
-    return;
-  }
+  // The following will result in zero, one, or multiple synchronous calls to
+  // DeliverRebufferedAudio().
+  fifo_.Push(*wrapper_bus_);
+}
 
-  // Compute the estimated capture time of the first sample frame of audio that
-  // will be consumed from the FIFO in the loop below.
-  base::TimeTicks estimated_capture_time = base::TimeTicks::Now() -
-      fifo_->frames() * base::TimeDelta::FromSeconds(1) / params_.sample_rate();
-
-  fifo_->Push(wrapper_bus_.get());
-  while (fifo_->frames() >= capture_bus_->frames()) {
-    fifo_->Consume(capture_bus_.get(), 0, capture_bus_->frames());
-    track_->Capture(*capture_bus_, estimated_capture_time, false);
-
-    // Advance the estimated capture time for the next FIFO consume operation.
-    estimated_capture_time +=
-        capture_bus_->frames() * base::TimeDelta::FromSeconds(1) /
-            params_.sample_rate();
-  }
+void WebAudioCapturerSource::DeliverRebufferedAudio(
+    const media::AudioBus& audio_bus,
+    int frame_delay) {
+  lock_.AssertAcquired();
+  const base::TimeTicks reference_time =
+      current_reference_time_ +
+      base::TimeDelta::FromMicroseconds(frame_delay *
+                                        base::Time::kMicrosecondsPerSecond /
+                                        params_.sample_rate());
+  track_->Capture(audio_bus, reference_time, false);
 }
 
 // If registered as audio consumer in |blink_source_|, deregister from
 // |blink_source_| and stop keeping a reference to |blink_source_|.
 // Failure to call this method when stopping the track might leave an invalid
 // WebAudioCapturerSource reference still registered as an audio consumer on
 // the blink side.
 void WebAudioCapturerSource::removeFromBlinkSource() {
   if (!blink_source_.isNull()) {
     blink_source_.removeAudioConsumer(this);
     blink_source_.reset();
   }
 }
 
 }  // namespace content