Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/webaudio_capturer_source.cc

+// Copyright (c) 2012 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 "content/renderer/media/webaudio_capturer_source.h"
+
+#include "base/logging.h"
+
+using media::AudioParameters;
+using media::AudioBus;
+using media::AudioFifo;
+
+static const int kFifoSize = 2048;
+
+namespace content {
+
+WebAudioCapturerSource::WebAudioCapturerSource()
+    : callback_(0),
+      started_(false) {
+}
+
+WebAudioCapturerSource::~WebAudioCapturerSource() {
+}
+
+void WebAudioCapturerSource::Initialize(
+    const media::AudioParameters& params,
+    media::AudioCapturerSource::CaptureCallback* callback,
+    media::AudioCapturerSource::CaptureEventHandler* event_handler) {
+  base::AutoLock auto_lock(lock_);
+  params_ = params;
+  callback_ = callback;
+  wrapper_bus_ = AudioBus::CreateWrapper(params.channels());
+  capture_bus_ = AudioBus::Create(params);
+  fifo_.reset(new AudioFifo(params.channels(), kFifoSize));
+}
+
+void WebAudioCapturerSource::Start() {
+  // TODO(crogers): check that our client actually is calling Start()
+  // and do something with this information.
+  started_ = true;
+}
+
+void WebAudioCapturerSource::Stop() {
+  started_ = false;
+}
+
+void WebAudioCapturerSource::HandleCapture(
+    const WebKit::WebVector<const float*>& audio_data,
+    size_t number_of_frames) {
+  base::AutoLock auto_lock(lock_);
+  if (!callback_)
+    return;
+
+  if (wrapper_bus_->channels() != static_cast<int>(audio_data.size())) {
+    LOG(ERROR) << "WebAudioCapturerSource::Consume() : Channel mismatch.";
+    return;
+  }
+
+  // TODO(crogers): Handle possibly sample-rate mismatch.
+  // Currently we don't even receive this information from WebKit.
+
+  // Wrap.

[tommi (sloooow) - chröme, 2012/11/12 09:46:18]

comment doesn't add much :)

+  wrapper_bus_->set_frames(number_of_frames);
+  for (unsigned i = 0; i < audio_data.size(); ++i)

[tommi (sloooow) - chröme, 2012/11/12 09:46:18]

s/unsigned/size_t or int?

+    wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
+
+  // Handle mismatch between WebAudio buffer-size and WebRTC.
+  int available = fifo_->max_frames() - fifo_->frames();
+  if (available < static_cast<int>(number_of_frames)) {
+    LOG(ERROR) << "WebAudioCapturerSource::Consume() : FIFO overrun.";
+    return;
+  }
+
+  fifo_->Push(wrapper_bus_.get());
+  int capture_frames = params_.frames_per_buffer();
+  while (fifo_->frames() >= capture_frames) {
+    fifo_->Consume(capture_bus_.get(), 0, capture_frames);
+    callback_->Capture(capture_bus_.get(), 0, 1.0);
+  }
+}
+
+}  // namespace content