Index: Source/platform/audio/AudioDestination.cpp |
diff --git a/Source/platform/audio/AudioDestination.cpp b/Source/platform/audio/AudioDestination.cpp |
index 6a9e1dea2408bf45ef50b7d8fd1403a084c603d1..946a05440a5050b31fd92bf695b77017d3b72343 100644 |
--- a/Source/platform/audio/AudioDestination.cpp |
+++ b/Source/platform/audio/AudioDestination.cpp |
@@ -59,7 +59,7 @@ AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inpu |
, 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 |
@@ -82,7 +82,7 @@ AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inpu |
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 |
@@ -127,15 +127,15 @@ void AudioDestination::stop() |
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) { |