move the APM to chrome.

content/renderer/media/webrtc_local_audio_source_provider.cc

Index: content/renderer/media/webrtc_local_audio_source_provider.cc
diff --git a/content/renderer/media/webrtc_local_audio_source_provider.cc b/content/renderer/media/webrtc_local_audio_source_provider.cc
index cc70c785a97fd8b783496a356d3d435c42856ecb..f1f944d06142d0effe8e6d2d721ed894531bf212 100644
--- a/content/renderer/media/webrtc_local_audio_source_provider.cc
+++ b/content/renderer/media/webrtc_local_audio_source_provider.cc
@@ -23,10 +23,7 @@ static const size_t kMaxNumberOfBuffers = 10;
 const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128;
 
 WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider()
-    : audio_delay_ms_(0),
-      volume_(1),
-      key_pressed_(false),
-      is_enabled_(false) {
+    : is_enabled_(false) {
 }
 
 WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
@@ -34,9 +31,10 @@ WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
     audio_converter_->RemoveInput(this);
 }
 
-void WebRtcLocalAudioSourceProvider::Initialize(
-    const media::AudioParameters& source_params) {
+void WebRtcLocalAudioSourceProvider::SetCaptureFormat(
+    const media::AudioParameters& params) {
   DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(params.IsValid());
 
   // Use the native audio output hardware sample-rate for the sink.
   if (RenderThreadImpl::current()) {
@@ -44,8 +42,8 @@ void WebRtcLocalAudioSourceProvider::Initialize(
         RenderThreadImpl::current()->GetAudioHardwareConfig();
     int sample_rate = hardware_config->GetOutputSampleRate();
     sink_params_.Reset(
-        source_params.format(), media::CHANNEL_LAYOUT_STEREO, 2, 0,
-        sample_rate, source_params.bits_per_sample(),
+        params.format(), media::CHANNEL_LAYOUT_STEREO, 2, 0,
+        sample_rate, params.bits_per_sample(),
         kWebAudioRenderBufferSize);
   } else {
     // This happens on unittests which does not have a valid RenderThreadImpl,
@@ -54,56 +52,51 @@ void WebRtcLocalAudioSourceProvider::Initialize(
   }
 
   base::AutoLock auto_lock(lock_);
-  source_params_ = source_params;
+  source_params_ = params;
   // Create the audio converter with |disable_fifo| as false so that the
   // converter will request source_params.frames_per_buffer() each time.
   // This will not increase the complexity as there is only one client to
   // the converter.
   audio_converter_.reset(
-      new media::AudioConverter(source_params, sink_params_, false));
+      new media::AudioConverter(params, sink_params_, false));
   audio_converter_->AddInput(this);
   fifo_.reset(new media::AudioFifo(
-      source_params.channels(),
-      kMaxNumberOfBuffers * source_params.frames_per_buffer()));
+      params.channels(),
+      kMaxNumberOfBuffers * params.frames_per_buffer()));
+  input_wrapper_ = media::AudioBus::Create(source_params_.channels(),
+                                           source_params_.frames_per_buffer());
 }
 
-void WebRtcLocalAudioSourceProvider::DeliverData(
-    media::AudioBus* audio_source,
+int WebRtcLocalAudioSourceProvider::CaptureData(
+    const std::vector<int>& channels,
+    const int16* audio_data,
+    int sample_rate,
+    int number_of_channels,
+    int number_of_frames,
     int audio_delay_milliseconds,
-    int volume,
+    int current_volume,
+    bool need_audio_processing,
     bool key_pressed) {
   base::AutoLock auto_lock(lock_);
   if (!is_enabled_)
-    return;
+    return 0;
 
   DCHECK(fifo_.get());
-
-  if (fifo_->frames() + audio_source->frames() <= fifo_->max_frames()) {
-    fifo_->Push(audio_source);
+  // TODO(xians): A better way to handle the interleaved and deinterleaved
+  // format switching.
+  DCHECK(input_wrapper_->frames() == number_of_frames);
+  CHECK(input_wrapper_->channels() == number_of_channels);
+  input_wrapper_->FromInterleaved(audio_data, number_of_frames, 2);
+
+  if (fifo_->frames() + number_of_frames <= fifo_->max_frames()) {
+    fifo_->Push(input_wrapper_.get());
   } else {
     // This can happen if the data in FIFO is too slowed to be consumed or
     // WebAudio stops consuming data.
     DLOG(WARNING) << "Local source provicer FIFO is full" << fifo_->frames();
   }
 
-  // Cache the values for GetAudioProcessingParams().
-  last_fill_ = base::TimeTicks::Now();
-  audio_delay_ms_ = audio_delay_milliseconds;
-  volume_ = volume;
-  key_pressed_ = key_pressed;
-}
-
-void WebRtcLocalAudioSourceProvider::GetAudioProcessingParams(
-    int* delay_ms, int* volume, bool* key_pressed) {
-  int elapsed_ms = 0;
-  if (!last_fill_.is_null()) {
-    elapsed_ms = static_cast<int>(
-        (base::TimeTicks::Now() - last_fill_).InMilliseconds());
-  }
-  *delay_ms = audio_delay_ms_ + elapsed_ms + static_cast<int>(
-      1000 * fifo_->frames() / source_params_.sample_rate() + 0.5);
-  *volume = volume_;
-  *key_pressed = key_pressed_;
+  return 0;
 }
 
 void WebRtcLocalAudioSourceProvider::setClient(
@@ -114,20 +107,20 @@ void WebRtcLocalAudioSourceProvider::setClient(
 void WebRtcLocalAudioSourceProvider::provideInput(
     const WebVector<float*>& audio_data, size_t number_of_frames) {
   DCHECK_EQ(number_of_frames, kWebAudioRenderBufferSize);
-  if (!bus_wrapper_ ||
-      static_cast<size_t>(bus_wrapper_->channels()) != audio_data.size()) {
-    bus_wrapper_ = media::AudioBus::CreateWrapper(audio_data.size());
+  if (!output_wrapper_ ||
+      static_cast<size_t>(output_wrapper_->channels()) != audio_data.size()) {
+    output_wrapper_ = media::AudioBus::CreateWrapper(audio_data.size());
   }
 
-  bus_wrapper_->set_frames(number_of_frames);
+  output_wrapper_->set_frames(number_of_frames);
   for (size_t i = 0; i < audio_data.size(); ++i)
-    bus_wrapper_->SetChannelData(i, audio_data[i]);
+    output_wrapper_->SetChannelData(i, audio_data[i]);
 
   base::AutoLock auto_lock(lock_);
   DCHECK(audio_converter_.get());
   DCHECK(fifo_.get());
   is_enabled_ = true;
-  audio_converter_->Convert(bus_wrapper_.get());
+  audio_converter_->Convert(output_wrapper_.get());
 }
 
 double WebRtcLocalAudioSourceProvider::ProvideInput(
@@ -136,11 +129,9 @@ double WebRtcLocalAudioSourceProvider::ProvideInput(
     fifo_->Consume(audio_bus, 0, audio_bus->frames());
   } else {
     audio_bus->Zero();
-    if (!last_fill_.is_null()) {
-      DLOG(WARNING) << "Underrun, FIFO has data " << fifo_->frames()
-                    << " samples but " << audio_bus->frames()
-                    << " samples are needed";
-    }
+    DVLOG(1) << "WARNING: Underrun, FIFO has data " << fifo_->frames()
+             << " samples but " << audio_bus->frames()
+             << " samples are needed";
   }
 
   return 1.0;