Added an "enable-audio-processor" flag and WebRtcAudioProcessor class

content/renderer/media/webrtc_audio_processor.cc

Index: content/renderer/media/webrtc_audio_processor.cc
diff --git a/content/renderer/media/webrtc_audio_processor.cc b/content/renderer/media/webrtc_audio_processor.cc
new file mode 100644
index 0000000000000000000000000000000000000000..ccf6841e03ee0519993a4bc993a4db13a1500027
--- /dev/null
+++ b/content/renderer/media/webrtc_audio_processor.cc
@@ -0,0 +1,359 @@
+// 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_audio_processor.h"
+
+#include "base/command_line.h"
+#include "base/debug/trace_event.h"
+#include "content/public/common/content_switches.h"
+#include "content/renderer/media/webrtc_audio_processor_options.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/audio_converter.h"
+#include "media/base/audio_fifo.h"
+#include "media/base/channel_layout.h"
+
+namespace content {
+
+namespace {
+
+using webrtc::AudioProcessing;
+using webrtc::MediaConstraintsInterface;
+
+#if defined(ANDROID)
+const int kAudioProcessingSampleRate = 16000;
+#else
+const int kAudioProcessingSampleRate = 32000;
+#endif
+const int kAudioProcessingNumberOfChannel = 1;
+
+const int kMaxNumberOfBuffersInFifo = 2;
+
+}  // namespace
+
+class WebRtcAudioProcessor::WebRtcAudioConverter
+    : public media::AudioConverter::InputCallback {
+ public:
+  WebRtcAudioConverter(const media::AudioParameters& source_params,
+                       const media::AudioParameters& sink_params)
+     : source_params_(source_params),
+       sink_params_(sink_params),
+       audio_converter_(source_params, sink_params_, false) {
+    worker_thread_checker_.DetachFromThread();
+
+    audio_converter_.AddInput(this);
+    // Create and initialize audio fifo and audio bus wrapper.
+    // The size of the FIFO should be at least twice of the source buffer size
+    // or twice of the sink buffer size.
+    int buffer_size = std::max(
+        kMaxNumberOfBuffersInFifo * source_params_.frames_per_buffer(),
+        kMaxNumberOfBuffersInFifo * sink_params_.frames_per_buffer());
+    fifo_.reset(new media::AudioFifo(source_params_.channels(), buffer_size));
+    // TODO(xians): Use CreateWrapper to save one memcpy.
+    audio_wrapper_ = media::AudioBus::Create(sink_params_.channels(),
+                                             sink_params_.frames_per_buffer());
+  }
+
+  virtual ~WebRtcAudioConverter() {
+    DCHECK(create_thread_checker_.CalledOnValidThread());
+    audio_converter_.RemoveInput(this);
+  }
+
+  void Push(media::AudioBus* audio_source) {
+    // Called on the audio thread, which is the capture audio thread for
+    // |WebRtcAudioProcessor::capture_converter_|, and render audio thread for
+    // |WebRtcAudioProcessor::render_converter_|.
+    // And it must be the same thread as calling Convert().
+    DCHECK(worker_thread_checker_.CalledOnValidThread());
+    fifo_->Push(audio_source);
+  }
+
+  bool Convert(webrtc::AudioFrame* out) {
+    // Called on the audio thread, which is the capture audio thread for
+    // |WebRtcAudioProcessor::capture_converter_|, and render audio thread for
+    // |WebRtcAudioProcessor::render_converter_|.
+    // Return false if there is no 10ms data in the FIFO.
+    DCHECK(worker_thread_checker_.CalledOnValidThread());
+    if (fifo_->frames() < (source_params_.sample_rate() / 100))
+      return false;
+
+    // Convert 10ms data to the output format, this will trigger ProvideInput().
+    audio_converter_.Convert(audio_wrapper_.get());
+
+    // TODO(xians): Figure out a better way to handle the interleaved and
+    // deinterleaved format switching.
+    audio_wrapper_->ToInterleaved(audio_wrapper_->frames(),
+                                  sink_params_.bits_per_sample() / 8,
+                                  out->data_);
+
+    out->samples_per_channel_ = sink_params_.frames_per_buffer();
+    out->sample_rate_hz_ = sink_params_.sample_rate();
+    out->speech_type_ = webrtc::AudioFrame::kNormalSpeech;
+    out->vad_activity_ = webrtc::AudioFrame::kVadUnknown;
+    out->num_channels_ = sink_params_.channels();
+
+    return true;
+  }
+
+  const media::AudioParameters& source_parameters() const {
+    return source_params_;
+  }
+  const media::AudioParameters& sink_parameters() const {
+    return sink_params_;
+  }
+
+ private:
+  // AudioConverter::InputCallback implementation.
+  virtual double ProvideInput(media::AudioBus* audio_bus,
+                              base::TimeDelta buffer_delay) OVERRIDE {
+    // Called on realtime audio thread.
+    // TODO(xians): Figure out why the first Convert() triggers ProvideInput
+    // two times.
+    if (fifo_->frames() < audio_bus->frames())
+      return 0;
+
+    fifo_->Consume(audio_bus, 0, audio_bus->frames());
+    return 1.0;
+  }
+
+  base::ThreadChecker create_thread_checker_;
+  base::ThreadChecker worker_thread_checker_;
+  media::AudioParameters source_params_;
+  media::AudioParameters sink_params_;
+
+  // TODO(xians): consider using SincResampler to save some memcpy.
+  // Handles mixing and resampling between input and output parameters.
+  media::AudioConverter audio_converter_;
+  scoped_ptr<media::AudioBus> audio_wrapper_;
+  scoped_ptr<media::AudioFifo> fifo_;
+};
+
+WebRtcAudioProcessor::WebRtcAudioProcessor(
+    const webrtc::MediaConstraintsInterface* constraints)
+    : render_delay_ms_(0) {
+  capture_thread_checker_.DetachFromThread();
+  render_thread_checker_.DetachFromThread();
+  InitializeAudioProcessingModule(constraints);
+}
+
+WebRtcAudioProcessor::~WebRtcAudioProcessor() {
+  DCHECK(main_thread_checker_.CalledOnValidThread());
+  StopAudioProcessing();
+}
+
+void WebRtcAudioProcessor::PushCaptureData(media::AudioBus* audio_source) {
+  DCHECK(capture_thread_checker_.CalledOnValidThread());
+  capture_converter_->Push(audio_source);
+}
+
+bool WebRtcAudioProcessor::ProcessAndConsumeData(
+    base::TimeDelta capture_delay, int volume, bool key_pressed,
+    int16** out) {
+  DCHECK(capture_thread_checker_.CalledOnValidThread());
+  TRACE_EVENT0("audio",
+               "WebRtcAudioProcessor::ProcessAndConsumeData");
+
+  if (!capture_converter_->Convert(&capture_frame_))
+    return false;
+
+  ProcessData(&capture_frame_, capture_delay, volume, key_pressed);
+  *out = capture_frame_.data_;
+
+  return true;
+}
+
+void WebRtcAudioProcessor::SetCaptureFormat(
+    const media::AudioParameters& source_params) {
+  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(source_params.IsValid());
+
+  // Create and initialize audio converter for the source data.
+  // When the webrtc AudioProcessing is enabled, the sink format of the
+  // converter will be the same as the post-processed data format, which is
+  // 32k mono for desktops and 16k mono for Android. When the AudioProcessing
+  // is disabled, the sink format will be the same as the source format.
+  const int sink_sample_rate = audio_processing_ ?
+      kAudioProcessingSampleRate : source_params.sample_rate();
+  const media::ChannelLayout sink_channel_layout = audio_processing_ ?
+      media::CHANNEL_LAYOUT_MONO : source_params.channel_layout();
+
+  // WebRtc is using 10ms data as its native packet size.
+  media::AudioParameters sink_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY, sink_channel_layout,
+      sink_sample_rate, 16, sink_sample_rate / 100);
+  capture_converter_.reset(
+      new WebRtcAudioConverter(source_params, sink_params));
+}
+
+const media::AudioParameters& WebRtcAudioProcessor::OutputFormat() const {
+  return capture_converter_->sink_parameters();
+}
+
+void WebRtcAudioProcessor::ProcessData(webrtc::AudioFrame* audio_frame,
+                                       base::TimeDelta capture_delay,
+                                       int volume,
+                                       bool key_pressed) {
+  DCHECK(capture_thread_checker_.CalledOnValidThread());
+  if (!audio_processing_)
+    return;
+
+  TRACE_EVENT0("audio", "WebRtcAudioProcessor::Process10MsData");
+  DCHECK_EQ(audio_processing_->sample_rate_hz(),
+            capture_converter_->sink_parameters().sample_rate());
+  DCHECK_EQ(audio_processing_->num_input_channels(),
+            capture_converter_->sink_parameters().channels());
+  DCHECK_EQ(audio_processing_->num_output_channels(),
+            capture_converter_->sink_parameters().channels());
+
+  base::subtle::Atomic32 render_delay_ms =
+      base::subtle::Acquire_Load(&render_delay_ms_);
+  int64 capture_delay_ms = capture_delay.InMilliseconds();
+  DCHECK_LT(capture_delay_ms,
+            std::numeric_limits<base::subtle::Atomic32>::max());
+  int total_delay_ms =  capture_delay_ms + render_delay_ms;
+  if (total_delay_ms > 1000) {
+    LOG(WARNING) << "Large audio delay, capture delay: " << capture_delay_ms
+                 << "ms; render delay: " << render_delay_ms << "ms";
+  }
+
+  audio_processing_->set_stream_delay_ms(total_delay_ms);
+  webrtc::GainControl* agc = audio_processing_->gain_control();
+  if (agc->set_stream_analog_level(volume))
+    NOTREACHED();
+  int err = audio_processing_->ProcessStream(audio_frame);
+  DCHECK(!err) << "ProcessStream() error: " << err;
+
+  // TODO(xians): Add support for AGC, typing detection, audio level
+  // calculation, stereo swapping.
+}
+
+void WebRtcAudioProcessor::PushRenderData(

[Henrik Grunell, 2013/11/19 08:52:22]

Just a step ahead: have you thought about guaranteeing that the user(s) of
objects of this class not calling these functions after of during destruction?
What will be the lifetime of objects of this class?

I just want to make sure this doesn't slip and is forgotten when starting using
the class.

[no longer working on chromium, 2013/11/21 15:59:26]

The owner of this class will gurantee this.

+    const int16* render_audio, int sample_rate, int number_of_channels,
+    int number_of_frames, base::TimeDelta render_delay) {
+  DCHECK(render_thread_checker_.CalledOnValidThread());
+
+  // Return immediately if the echo cancellation is off.
+  if (!audio_processing_ ||
+      !audio_processing_->echo_cancellation()->is_enabled())
+    return;
+
+  TRACE_EVENT0("audio",
+               "WebRtcAudioProcessor::FeedRenderDataToAudioProcessing");
+  int64 new_render_delay_ms = render_delay.InMilliseconds();
+  DCHECK_LT(new_render_delay_ms,
+            std::numeric_limits<base::subtle::Atomic32>::max());
+  base::subtle::Release_Store(&render_delay_ms_, new_render_delay_ms);
+
+  InitializeRenderConverterIfNeeded(sample_rate, number_of_channels,
+                                    number_of_frames);
+
+  // TODO(xians): Avoid this extra interleave/deinterleave.
+  render_data_bus_->FromInterleaved(render_audio,
+                                    render_data_bus_->frames(),
+                                    sizeof(render_audio[0]));
+  render_converter_->Push(render_data_bus_.get());
+  while (render_converter_->Convert(&render_frame_)) {
+    audio_processing_->AnalyzeReverseStream(&render_frame_);
+  }
+}
+
+void WebRtcAudioProcessor::InitializeAudioProcessingModule(
+    const webrtc::MediaConstraintsInterface* constraints) {
+  if (!CommandLine::ForCurrentProcess()->HasSwitch(
+          switches::kEnableAudioTrackProcessing)) {
+    return;
+  }
+
+  if (!constraints)
+    return;
+
+  const bool enable_aec = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kEchoCancellation);
+  const bool enable_ns = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kNoiseSuppression);
+  const bool enable_high_pass_filter = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kHighpassFilter);
+  const bool start_aec_dump = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kInternalAecDump);
+#if defined(IOS) || defined(ANDROID)
+  const bool enable_experimental_aec = false;
+  const bool enable_typing_detection = false;
+#else
+  const bool enable_experimental_aec = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kExperimentalEchoCancellation);
+  const bool enable_typing_detection = GetPropertyFromConstraints(
+      constraints, MediaConstraintsInterface::kTypingNoiseDetection);
+#endif
+
+  // Return immediately if no audio processing component is enabled.
+  if (!enable_aec && !enable_experimental_aec && !enable_ns &&
+      !enable_high_pass_filter && !enable_typing_detection) {
+    return;
+  }
+
+  // Create and configure the audio processing if it does not exist.
+  if (!audio_processing_)
+    audio_processing_.reset(webrtc::AudioProcessing::Create(0));
+
+  // Enable the audio processing components.
+  if (enable_aec) {
+    EnableEchoCancellation(audio_processing_.get());
+    if (enable_experimental_aec)
+      EnableExperimentalEchoCancellation(audio_processing_.get());
+  }
+
+  if (enable_ns)
+    EnableNoiseSuppression(audio_processing_.get());
+
+  if (enable_high_pass_filter)
+    EnableHighPassFilter(audio_processing_.get());
+
+  if (enable_typing_detection)
+    EnableTypingDetection(audio_processing_.get());
+
+  if (enable_aec && start_aec_dump)
+    StartAecDump(audio_processing_.get());
+
+  // Configure the audio format the audio processing is running on. This
+  // has to be done after all the needed components are enabled.
+  if (audio_processing_->set_sample_rate_hz(kAudioProcessingSampleRate))
+    NOTREACHED();
+  if (audio_processing_->set_num_channels(kAudioProcessingNumberOfChannel,
+                                          kAudioProcessingNumberOfChannel))
+    NOTREACHED();
+}
+
+void WebRtcAudioProcessor::InitializeRenderConverterIfNeeded(
+    int sample_rate, int number_of_channels, int frames_per_buffer) {
+  // TODO(xians): Figure out if we need to handle the buffer size change.
+  if (render_converter_.get() &&
+      render_converter_->source_parameters().sample_rate() == sample_rate &&
+      render_converter_->source_parameters().channels() == number_of_channels) {
+    // Do nothing if the |render_converter_| has been setup properly.
+    return;
+  }
+
+  media::AudioParameters source_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      media::GuessChannelLayout(number_of_channels), sample_rate, 16,
+      frames_per_buffer);
+  media::AudioParameters sink_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      media::CHANNEL_LAYOUT_MONO, kAudioProcessingSampleRate, 16,
+      kAudioProcessingSampleRate / 100);
+  render_converter_.reset(new WebRtcAudioConverter(source_params, sink_params));
+  render_data_bus_ = media::AudioBus::Create(number_of_channels,
+                                             frames_per_buffer);
+}
+
+void WebRtcAudioProcessor::StopAudioProcessing() {
+  if (!audio_processing_.get())
+    return;
+
+  // It is safe to stop the AEC dump even it is not started.
+  StopAecDump(audio_processing_.get());
+
+  audio_processing_.reset();
+}
+
+}  // namespace content