[DO NOT SUBMIT] AudioWorklet FS2: audio thread

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

+// Copyright 2016 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "platform/audio/AudioRenderSink.h"
+
+#include "platform/audio/AudioPullFIFO.h"
+#include "public/platform/Platform.h"
+#include "public/platform/WebSecurityOrigin.h"
+
+namespace blink {
+
+// WebAudio engine's render quantum size.
+const unsigned kRenderQuantumSize = 128;
+
+// Default pulling FIFO size.
+const size_t kPullFifoSize = 8192;
+
+AudioRenderSink::AudioRenderSink(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate, const PassRefPtr<SecurityOrigin>& securityOrigin)
+    : m_isRunning(false)
+    , m_sampleRate(sampleRate)
+    , m_numberOfOutputChannels(numberOfOutputChannels)
+    , m_callback(callback)
+    , m_inputBus(AudioBus::create(numberOfInputChannels, kRenderQuantumSize))
+    , m_outputBus(AudioBus::create(numberOfOutputChannels, kRenderQuantumSize, false))
+{
+    // Use the optimal buffer size recommended by the audio stack. It differs
+    // from the render quantum size and it varies across the platform.
+    size_t recommendedBufferSize = Platform::current()->audioHardwareBufferSize();
+    m_callbackBufferSize = recommendedBufferSize;
+
+    m_webAudioDevice = adoptPtr(Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId, securityOrigin));
+    DCHECK(m_webAudioDevice);
+
+    // TODO(hongchan): the ownership and the structure are weird. This
+    // AudioRenderSink is also AudioSourceProvider, this can provide input for
+    // its own FIFO.
+    m_pullFifo = adoptPtr(new AudioPullFIFO(numberOfOutputChannels, kPullFifoSize, *this, kRenderQuantumSize));
+
+    LOG(INFO) << "WEBAUDIO: hardware buffer size = " << recommendedBufferSize;
+}
+
+AudioRenderSink::~AudioRenderSink()
+{
+    stop();
+}
+
+PassOwnPtr<AudioRenderSink> AudioRenderSink::create(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate, const PassRefPtr<SecurityOrigin>& securityOrigin)
+{
+    return adoptPtr(new AudioRenderSink(callback, inputDeviceId, numberOfInputChannels, numberOfOutputChannels, sampleRate, securityOrigin));
+}
+
+size_t AudioRenderSink::renderQuantumSize()
+{
+    return kRenderQuantumSize;
+}
+
+unsigned AudioRenderSink::audioHardwareBufferSize()
+{
+    return static_cast<unsigned>(Platform::current()->audioHardwareBufferSize());
+}
+
+float AudioRenderSink::audioHardwareSampleRate()
+{
+    return static_cast<float>(Platform::current()->audioHardwareSampleRate());
+}
+
+unsigned AudioRenderSink::audioHardwareChannelCount()
+{
+    return static_cast<unsigned>(Platform::current()->audioHardwareOutputChannels());

[o1ka, 2016/05/27 14:28:23]

If WebAudioDevice provides all these information, can it be used to return all
those hardware values instead of accessing Platform::current()?

+}
+
+void AudioRenderSink::start()
+{
+    if (!m_isRunning && m_webAudioDevice) {
+        m_webAudioDevice->start();
+        m_isRunning = true;
+    }
+}
+
+void AudioRenderSink::stop()
+{
+    if (m_isRunning && m_webAudioDevice) {
+        m_webAudioDevice->stop();
+        m_isRunning = false;
+    }
+}
+
+void AudioRenderSink::render(const WebVector<float*>& source, const WebVector<float*>& destination, size_t framesToProcess)
+{
+    // If the number of channels or the number of frames to process do not match
+    // stop the render and return.
+    if (!(destination.size() == m_numberOfOutputChannels)
+        || !(framesToProcess == m_callbackBufferSize)) {
+        NOTREACHED();
+        return;
+    }
+
+    // Fetch the input stream and make the data available in the input FIFO.
+    if (source.size() >= 2) {
+        // TODO: implement this.
+    }
+
+    for (unsigned i = 0; i < m_numberOfOutputChannels; ++i)
+        m_outputBus->setChannelMemory(i, destination[i], framesToProcess);
+
+    // Fill the destination buffer by consuming data in the pull FIFO.
+    m_pullFifo->consume(m_outputBus.get(), framesToProcess);
+
+    // LOG(INFO) << "AudioRenderSink::render" << framesToProcess;
+}
+
+void AudioRenderSink::provideInput(AudioBus* outputBus, size_t numberOfFrames)
+{
+    AudioBus* inputBus = m_inputBus.get();
+    m_callback.render(inputBus, outputBus, numberOfFrames);
+}
+
+} // namespace blink