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..49bb44805e48e2acdcce3e2e564db727b5c56fef
--- /dev/null
+++ b/content/renderer/media/webrtc_local_audio_source_provider.cc
@@ -0,0 +1,147 @@
+// 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) {
+}
+
+WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
+  if (audio_converter_.get())
+    audio_converter_->RemoveInput(this);
+}
+
+void WebRtcLocalAudioSourceProvider::Initialize(
+    const media::AudioParameters& source_params) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+
+  // Use the native audio output hardware sample-rate for the sink.
+  if (RenderThreadImpl::current()) {

[tommi (sloooow) - chröme, 2013/09/12 11:02:16]

what if this is NULL?  Won't the sink_params be invalid?  Seems to me that we
shouldn't be checking this or there is something seriously wrong. (we already
have the thread check above for being called on the correct thread).

[no longer working on chromium, 2013/09/12 19:14:56]

We check this since it will be NULL for most of the unittests without mocking
the IPC message.

We are still exposing SetSinkParamsForTesting() method to allow the unittests to
inject their own sink params for testing.
If the unittests do not do it, it will crash when creating the audio_converter_.

+    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();

[tommi (sloooow) - chröme, 2013/09/12 11:02:16]

Let's talk about this.  It seems like the WebAudio implementation is using this
to tell webkit what sample rate to expect.  Can we do that?  If not, can you add
documentation here that explains why not?

[no longer working on chromium, 2013/09/12 19:14:56]

Telling webkit about the sample rate will force moving the resampler to webkit,
we would like to do in Chrome instead.

+}
+
+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;
+}
+
+}  // namespace content