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

content/renderer/media/webaudio_capturer_source.cc

Index: content/renderer/media/webaudio_capturer_source.cc
diff --git a/content/renderer/media/webaudio_capturer_source.cc b/content/renderer/media/webaudio_capturer_source.cc
index bcebcbb3e24e523288827db01d6b740a019a29fd..6a4966070870b38b1e0627cb6aeca2e93da426c2 100644
--- a/content/renderer/media/webaudio_capturer_source.cc
+++ b/content/renderer/media/webaudio_capturer_source.cc
@@ -4,27 +4,28 @@
 
 #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) {
-}
+      rechunker_(base::TimeDelta::FromMilliseconds(10),
+                 base::Bind(&WebAudioCapturerSource::DeliverRechunkedAudio,
+                            base::Unretained(this))),
+      blink_source_(blink_source) {}
 
 WebAudioCapturerSource::~WebAudioCapturerSource() {
   DCHECK(thread_checker_.CalledOnValidThread());
@@ -48,20 +49,19 @@ void WebAudioCapturerSource::setFormat(
   // 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.
+  rechunker_.SetSampleRate(sample_rate);
   params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
-                sample_rate, 16, sample_rate / 100);
+                sample_rate, 16, rechunker_.output_frames());
 
   // 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) {
@@ -85,6 +85,11 @@ void WebAudioCapturerSource::Stop() {
 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
+  base::TimeTicks reference_time = base::TimeTicks::Now();
+
   base::AutoLock auto_lock(lock_);
   if (!track_)
     return;
@@ -96,36 +101,20 @@ void WebAudioCapturerSource::consumeAudio(
   }
 
   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;
-  }
-
-  // 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);
+  // The following will result in zero, one, or multiple synchronous calls to
+  // DeliverRechunkedAudio().
+  rechunker_.Push(*wrapper_bus_, reference_time - base::TimeTicks());

[o1ka, 2016/02/22 13:04:47]

I like how neat it becomes!

[miu, 2016/02/23 04:27:41]

Acknowledged.

+}
 
-    // 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::DeliverRechunkedAudio(
+    const media::AudioBus& audio_bus,
+    base::TimeDelta reference_timestamp) {
+  lock_.AssertAcquired();
+  track_->Capture(audio_bus, base::TimeTicks() + reference_timestamp, false);
 }
 
 // If registered as audio consumer in |blink_source_|, deregister from