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33 #include "AudioDestinationChromium.h"
35 #include "AudioSourceProvider.h"
37 #include "WebKitPlatformSupport.h"
39 using namespace WebKit;
43 // Buffer size that the Chromium audio system will call us back with.
45 // For Mac OS X the chromium audio backend uses a low-latency CoreAudio API, so a low buffer size is possible.
46 const unsigned callbackBufferSize = 128;
48 // This value may need to be tuned based on the OS.
49 // FIXME: It may be possible to reduce this value once real-time threads
50 // and other Chromium audio improvements are made.
51 const unsigned callbackBufferSize = 2048;
54 // Buffer size at which the web audio engine will render.
55 const unsigned renderBufferSize = 128;
57 const unsigned renderCountPerCallback = callbackBufferSize / renderBufferSize;
59 // FIXME: add support for multi-channel.
60 const unsigned numberOfChannels = 2;
62 // Factory method: Chromium-implementation
63 PassOwnPtr<AudioDestination> AudioDestination::create(AudioSourceProvider& provider, double sampleRate)
65 return adoptPtr(new AudioDestinationChromium(provider, sampleRate));
68 AudioDestinationChromium::AudioDestinationChromium(AudioSourceProvider& provider, double sampleRate)
69 : m_provider(provider)
70 , m_renderBus(numberOfChannels, renderBufferSize, false)
71 , m_sampleRate(sampleRate)
74 m_audioDevice = adoptPtr(webKitPlatformSupport()->createAudioDevice(callbackBufferSize, numberOfChannels, sampleRate, this));
75 ASSERT(m_audioDevice.get());
78 AudioDestinationChromium::~AudioDestinationChromium()
83 void AudioDestinationChromium::start()
85 if (!m_isPlaying && m_audioDevice.get()) {
86 m_audioDevice->start();
91 void AudioDestinationChromium::stop()
93 if (m_isPlaying && m_audioDevice.get()) {
94 m_audioDevice->stop();
99 double AudioDestination::hardwareSampleRate()
101 return webKitPlatformSupport()->audioHardwareSampleRate();
104 // Pulls on our provider to get the rendered audio stream.
105 void AudioDestinationChromium::render(const WebVector<float*>& audioData, size_t numberOfFrames)
107 bool isNumberOfChannelsGood = audioData.size() == numberOfChannels;
108 if (!isNumberOfChannelsGood) {
109 ASSERT_NOT_REACHED();
113 bool isBufferSizeGood = numberOfFrames == callbackBufferSize;
114 if (!isBufferSizeGood) {
115 ASSERT_NOT_REACHED();
119 // Split up the callback buffer into smaller chunks which we'll render one after the other.
120 for (unsigned i = 0; i < renderCountPerCallback; ++i) {
121 m_renderBus.setChannelMemory(0, audioData[0] + i * renderBufferSize, renderBufferSize);
122 m_renderBus.setChannelMemory(1, audioData[1] + i * renderBufferSize, renderBufferSize);
123 m_provider.provideInput(&m_renderBus, renderBufferSize);
127 } // namespace WebCore
129 #endif // ENABLE(WEB_AUDIO)