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"
+#include "content/renderer/media/webrtc_audio_capturer.h"
+
+using media::AudioBus;
+using media::AudioFifo;
+using media::AudioParameters;
+using media::ChannelLayout;
+using media::CHANNEL_LAYOUT_MONO;
+using media::CHANNEL_LAYOUT_STEREO;
+
+static const int kFifoSize = 2048;
+
+namespace content {
+
+WebAudioCapturerSource::WebAudioCapturerSource(WebRtcAudioCapturer* capturer)
+    : capturer_(capturer),
+      callback_(0),
+      started_(false) {
+  DVLOG(1) << "WebAudioCapturerSource::WebAudioCapturerSource()";
+}
+
+WebAudioCapturerSource::~WebAudioCapturerSource() {
+  DVLOG(1) << "WebAudioCapturerSource::~WebAudioCapturerSource()";
+}
+
+void WebAudioCapturerSource::setFormat(
+    size_t number_of_channels, float sample_rate) {
+  // See TODO in channel mixing code.
+  // We'll eventually want to handle more than the mono and stereo cases.
+  ChannelLayout channel_layout =
+      number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
+  capturer_->SetCapturerSource(this, channel_layout, sample_rate);
+  capturer_->Start();
+}
+
+void WebAudioCapturerSource::Initialize(
+    const media::AudioParameters& params,
+    media::AudioCapturerSource::CaptureCallback* callback,
+    media::AudioCapturerSource::CaptureEventHandler* event_handler) {
+  DVLOG(1) << "WebAudioCapturerSource::Initialize()";
+  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() {
+  DVLOG(1) << "WebAudioCapturerSource::Start()";
+  // TODO(crogers): check that our client actually is calling Start()
+  // and do something with this information.
+  started_ = true;
+}
+
+void WebAudioCapturerSource::Stop() {
+  DVLOG(1) << "WebAudioCapturerSource::Stop()";
+  started_ = false;
+}
+
+void WebAudioCapturerSource::consumeAudio(
+    const WebKit::WebVector<const float*>& audio_data,
+    size_t number_of_frames) {
+  base::AutoLock auto_lock(lock_);
+
+  if (!callback_)
+    return;
+
+  // TODO(crogers): Handle possible sample-rate mismatch.
+  // Currently we don't even receive this information from WebKit.
+
+  wrapper_bus_->set_frames(number_of_frames);
+
+  // TODO(henrika,crogers): improve how we deal with channels depending on
+  // the source. Today, the native capture channel config is utilized even
+  // for cases where the source is not a microphone.
+  if (wrapper_bus_->channels() == static_cast<int>(audio_data.size())) {
+    for (size_t i = 0; i < audio_data.size(); ++i)
+      wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[0]));
+  } else if (wrapper_bus_->channels() == 2 &&
+             static_cast<int>(audio_data.size()) == 1) {
+    // Mono -> Stereo channel up-mixing.
+    wrapper_bus_->SetChannelData(0, const_cast<float*>(audio_data[0]));
+    wrapper_bus_->SetChannelData(1, const_cast<float*>(audio_data[0]));
+  } else {
+    LOG(ERROR) << "WebAudioCapturerSource::Consume() : Channel mismatch.";
+      return;
+  }
+
+  // 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