Switch LiveAudio to source provider solution.

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
new file mode 100644
index 0000000000000000000000000000000000000000..37cf12da4d417e9da03f50433a5dc62ea4d4d567
--- /dev/null
+++ b/content/renderer/media/webrtc_local_audio_source_provider.cc
@@ -0,0 +1,148 @@
+// Copyright 2013 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/webrtc_local_audio_source_provider.h"
+
+#include "base/logging.h"
+#include "content/renderer/render_thread_impl.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/audio_fifo.h"
+#include "media/base/audio_hardware_config.h"
+#include "third_party/WebKit/public/web/WebAudioSourceProviderClient.h"
+
+using WebKit::WebVector;
+
+namespace content {
+
+static const size_t kMaxNumberOfBuffer = 10;
+static const size_t kWebAudioRenderBufferSize = 128;
+
+WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider()
+    : audio_delay_ms_(0),
+      volume_(1),
+      key_pressed_(false),
+      is_enabled_(false),
+      use_sink_params_for_testing_(false) {

[tommi (sloooow) - chröme, 2013/09/10 16:00:38]

I don't like mixing test code and production code (we definitely should avoid
shipping test code).  Can't the test just pass the sink params via Initialize?

[no longer working on chromium, 2013/09/11 10:22:07]

How about now?

+}
+
+WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
+  if (audio_converter_.get())
+    audio_converter_->RemoveInput(this);
+}
+
+void WebRtcLocalAudioSourceProvider::Initialize(
+    const media::AudioParameters& source_params) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (!use_sink_params_for_testing_) {
+    // Use the native audio output hardware sample-rate for the sink.
+    media::AudioHardwareConfig* hardware_config =
+        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(),
+        kWebAudioRenderBufferSize);
+  }
+
+  base::AutoLock auto_lock(lock_);
+  source_params_ = source_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));
+  audio_converter_->AddInput(this);
+  fifo_.reset(new media::AudioFifo(
+      source_params.channels(),
+      kMaxNumberOfBuffer * source_params.frames_per_buffer()));
+}
+
+void WebRtcLocalAudioSourceProvider::DeliverData(
+    media::AudioBus* audio_source,
+    int audio_delay_milliseconds,
+    int volume,
+    bool key_pressed) {
+  base::AutoLock auto_lock(lock_);
+  if (!is_enabled_)
+    return;
+
+  DCHECK(fifo_.get());
+
+  if (fifo_->frames() + audio_source->frames() <= fifo_->max_frames()) {
+    fifo_->Push(audio_source);
+  } 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_;
+}
+
+void WebRtcLocalAudioSourceProvider::setClient(
+    WebKit::WebAudioSourceProviderClient* client) {
+  NOTIMPLEMENTED();
+}
+
+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());
+  }
+
+  bus_wrapper_->set_frames(number_of_frames);
+  for (size_t i = 0; i < audio_data.size(); ++i)
+    bus_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());
+}
+
+double WebRtcLocalAudioSourceProvider::ProvideInput(
+    media::AudioBus* audio_bus, base::TimeDelta buffer_delay) {
+  if (fifo_->frames() >= audio_bus->frames()) {
+    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";
+    }
+  }
+
+  return 1.0;
+}
+
+void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
+    const media::AudioParameters& sink_params) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  sink_params_ = sink_params;
+  use_sink_params_for_testing_ = true;
+}
+
+}  // namespace content