Introduce PushPullFIFO class and remove other FIFOs

third_party/WebKit/Source/platform/audio/AudioDestination.cpp

 /*
  * Copyright (C) 2010 Google Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
  * are met:
  *
  * 1.  Redistributions of source code must retain the above copyright
  *     notice, this list of conditions and the following disclaimer.
  * 2.  Redistributions in binary form must reproduce the above copyright
@@ skipping 11 matching lines @@
  * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
  * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
  * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
  * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
  * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "platform/audio/AudioDestination.h"
 
 #include "platform/Histogram.h"
-#include "platform/audio/AudioPullFIFO.h"
 #include "platform/audio/AudioUtilities.h"
+#include "platform/audio/PushPullFIFO.h"
 #include "platform/weborigin/SecurityOrigin.h"
 #include "public/platform/Platform.h"
 #include "public/platform/WebSecurityOrigin.h"
 #include "wtf/PtrUtil.h"
 #include <memory>
 
 namespace blink {
 
 // FIFO Size.
 //
@@ skipping 16 matching lines @@
                                    unsigned numberOfOutputChannels,
                                    float sampleRate,
                                    PassRefPtr<SecurityOrigin> securityOrigin)
     : m_numberOfOutputChannels(numberOfOutputChannels),
       m_sampleRate(sampleRate),
       m_isPlaying(false),
       m_callback(callback),
       m_outputBus(AudioBus::create(numberOfOutputChannels,
                                    AudioUtilities::kRenderQuantumFrames,
                                    false)),
+      m_renderBus(AudioBus::create(numberOfOutputChannels,
+                                   AudioUtilities::kRenderQuantumFrames)),
       m_framesElapsed(0) {
   // Calculate the optimum buffer size first.
   if (calculateBufferSize()) {
     // Create WebAudioDevice. blink::WebAudioDevice is designed to support the
     // local input (e.g. loopback from OS audio system), but Chromium's media
     // renderer does not support it currently. Thus, we use zero for the number
     // of input channels.
     m_webAudioDevice = WTF::wrapUnique(Platform::current()->createAudioDevice(
         m_callbackBufferSize, 0, numberOfOutputChannels, sampleRate, this,
         String(), std::move(securityOrigin)));
     DCHECK(m_webAudioDevice);
 
     // Create a FIFO.
-    m_fifo = WTF::wrapUnique(
-        new AudioPullFIFO(*this, numberOfOutputChannels, kFIFOSize,
-                          AudioUtilities::kRenderQuantumFrames));
+    m_fifo =
+        WTF::wrapUnique(new PushPullFIFO(numberOfOutputChannels, kFIFOSize));
   } else {
     NOTREACHED();
   }
 }
 
 AudioDestination::~AudioDestination() {
   stop();
 }
 
 void AudioDestination::render(const WebVector<float*>& destinationData,
                               size_t numberOfFrames,
                               double delay,
                               double delayTimestamp,
                               size_t priorFramesSkipped) {
   DCHECK_EQ(destinationData.size(), m_numberOfOutputChannels);
-  if (destinationData.size() != m_numberOfOutputChannels)
-    return;
-
   DCHECK_EQ(numberOfFrames, m_callbackBufferSize);
-  if (numberOfFrames != m_callbackBufferSize)
+  if (destinationData.size() != m_numberOfOutputChannels ||
+      numberOfFrames != m_callbackBufferSize)
     return;
 
   m_framesElapsed -= std::min(m_framesElapsed, priorFramesSkipped);
   double outputPosition =
       m_framesElapsed / static_cast<double>(m_sampleRate) - delay;
   m_outputPosition.position = outputPosition;
   m_outputPosition.timestamp = delayTimestamp;
   m_outputPositionReceivedTimestamp = base::TimeTicks::Now();
 
   // Associate the destination data array with the output bus then fill the
   // FIFO.
   for (unsigned i = 0; i < m_numberOfOutputChannels; ++i)
     m_outputBus->setChannelMemory(i, destinationData[i], numberOfFrames);
-  m_fifo->consume(m_outputBus.get(), numberOfFrames);
+
+  size_t framesToPull = numberOfFrames - m_fifo->framesInFIFO();

[Raymond Toy, 2017/01/09 22:46:02]

Although this isn't caused by this CL, what happens if numberOfFrames <
framesInFIFO()?  Should there at least be a DCHECK for this?

[hongchan, 2017/01/10 21:15:58]

Yes, this is actually unsafe. framesToPull can be a huge number. Glad we caught
this here.

+  while (framesToPull > 0) {
+    // If platform buffer is more than two times longer than |framesToProcess|
+    // we do not want output position to get stuck so we promote it
+    // using the elapsed time from the moment it was initially obtained.
+    if (m_callbackBufferSize > AudioUtilities::kRenderQuantumFrames * 2) {
+      double delta =
+          (base::TimeTicks::Now() - m_outputPositionReceivedTimestamp)
+              .InSecondsF();
+      m_outputPosition.position += delta;
+      m_outputPosition.timestamp += delta;
+    }
+
+    // Some implementations give only rough estimation of |delay| so
+    // we might have negative estimation |outputPosition| value.
+    if (m_outputPosition.position < 0.0)
+      m_outputPosition.position = 0.0;
+
+    m_callback.render(nullptr, m_renderBus.get(),
+                      AudioUtilities::kRenderQuantumFrames, m_outputPosition);
+    m_fifo->push(m_renderBus.get());
+
+    framesToPull -= AudioUtilities::kRenderQuantumFrames;
+  }
+
+  m_fifo->pull(m_outputBus.get(), numberOfFrames);
 
   m_framesElapsed += numberOfFrames;
 }
 
-void AudioDestination::provideInput(AudioBus* outputBus,
-                                    size_t framesToProcess) {
-  AudioIOPosition outputPosition = m_outputPosition;
-
-  // If platform buffer is more than two times longer than |framesToProcess|
-  // we do not want output position to get stuck so we promote it
-  // using the elapsed time from the moment it was initially obtained.
-  if (m_callbackBufferSize > framesToProcess * 2) {
-    double delta = (base::TimeTicks::Now() - m_outputPositionReceivedTimestamp)
-                       .InSecondsF();
-    outputPosition.position += delta;
-    outputPosition.timestamp += delta;
-  }
-
-  // Some implementations give only rough estimation of |delay| so
-  // we might have negative estimation |outputPosition| value.
-  if (outputPosition.position < 0.0)
-    outputPosition.position = 0.0;
-
-  // To fill the FIFO, start the render call chain of the destination node.
-  m_callback.render(nullptr, outputBus, framesToProcess, outputPosition);
-}
-
 void AudioDestination::start() {
   if (m_webAudioDevice && !m_isPlaying) {
     m_webAudioDevice->start();
     m_isPlaying = true;
   }
 }
 
 void AudioDestination::stop() {
   if (m_webAudioDevice && m_isPlaying) {
     m_webAudioDevice->stop();
@@ skipping 54 matching lines @@
   callbackBufferSizeHistogram.sample(m_callbackBufferSize);
 
   // Check if the requested buffer size is too large.
   bool isBufferSizeValid =
       m_callbackBufferSize + AudioUtilities::kRenderQuantumFrames <= kFIFOSize;
   DCHECK(isBufferSizeValid);
   return isBufferSizeValid;
 }
 
 }  // namespace blink