Cleanup namespace usage from animation to exported in platform/

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 41 matching lines @@
 
 AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
     : m_callback(callback)
     , m_numberOfOutputChannels(numberOfOutputChannels)
     , m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize))
     , m_renderBus(AudioBus::create(numberOfOutputChannels, renderBufferSize, false))
     , m_sampleRate(sampleRate)
     , m_isPlaying(false)
 {
     // Use the optimal buffer size recommended by the audio backend.
-    m_callbackBufferSize = blink::Platform::current()->audioHardwareBufferSize();
+    m_callbackBufferSize = Platform::current()->audioHardwareBufferSize();
 
 #if OS(ANDROID)
     // The optimum low-latency hardware buffer size is usually too small on Android for WebAudio to
     // render without glitching. So, if it is small, use a larger size. If it was already large, use
     // the requested size.
     //
     // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 for a Galaxy Nexus),
     // cause significant processing jitter. Sometimes multiple blocks will processed, but other
     // times will not be since the FIFO can satisfy the request. By using a larger
     // callbackBufferSize, we smooth out the jitter.
     const size_t kSmallBufferSize = 1024;
     const size_t kDefaultCallbackBufferSize = 2048;
 
     if (m_callbackBufferSize <= kSmallBufferSize)
         m_callbackBufferSize = kDefaultCallbackBufferSize;
 #endif
 
     // Quick exit if the requested size is too large.
     ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize);
     if (m_callbackBufferSize + renderBufferSize > fifoSize)
         return;
 
-    m_audioDevice = adoptPtr(blink::Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId));
+    m_audioDevice = adoptPtr(Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId));
     ASSERT(m_audioDevice);
 
     // Create a FIFO to handle the possibility of the callback size
     // not being a multiple of the render size. If the FIFO already
     // contains enough data, the data will be provided directly.
     // Otherwise, the FIFO will call the provider enough times to
     // satisfy the request for data.
     m_fifo = adoptPtr(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize, renderBufferSize));
 
     // Input buffering.
@@ skipping 24 matching lines @@
 void AudioDestination::stop()
 {
     if (m_isPlaying && m_audioDevice) {
         m_audioDevice->stop();
         m_isPlaying = false;
     }
 }
 
 float AudioDestination::hardwareSampleRate()
 {
-    return static_cast<float>(blink::Platform::current()->audioHardwareSampleRate());
+    return static_cast<float>(Platform::current()->audioHardwareSampleRate());
 }
 
 unsigned long AudioDestination::maxChannelCount()
 {
-    return static_cast<float>(blink::Platform::current()->audioHardwareOutputChannels());
+    return static_cast<float>(Platform::current()->audioHardwareOutputChannels());
 }
 
-void AudioDestination::render(const blink::WebVector<float*>& sourceData, const blink::WebVector<float*>& audioData, size_t numberOfFrames)
+void AudioDestination::render(const WebVector<float*>& sourceData, const WebVector<float*>& audioData, size_t numberOfFrames)
 {
     bool isNumberOfChannelsGood = audioData.size() == m_numberOfOutputChannels;
     if (!isNumberOfChannelsGood) {
         ASSERT_NOT_REACHED();
         return;
     }
 
     bool isBufferSizeGood = numberOfFrames == m_callbackBufferSize;
     if (!isBufferSizeGood) {
         ASSERT_NOT_REACHED();
@@ skipping 22 matching lines @@
         m_inputFifo->consume(m_inputBus.get(), framesToProcess);
         sourceBus = m_inputBus.get();
     }
 
     m_callback.render(sourceBus, bus, framesToProcess);
 }
 
 } // namespace blink
 
 #endif // ENABLE(WEB_AUDIO)