Added a kEchoCancellation constraint to turn off the audio processing.

content/renderer/media/media_stream_audio_processor.cc

 // 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/media_stream_audio_processor.h"
 
 #include "base/command_line.h"
 #include "base/debug/trace_event.h"
 #include "base/metrics/field_trial.h"
 #include "base/metrics/histogram.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/rtc_media_constraints.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"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 #include "third_party/webrtc/modules/audio_processing/typing_detection.h"
 
 namespace content {
 
 namespace {
 
 using webrtc::AudioProcessing;
-using webrtc::MediaConstraintsInterface;
 
 #if defined(OS_ANDROID)
 const int kAudioProcessingSampleRate = 16000;
 #else
 const int kAudioProcessingSampleRate = 32000;
 #endif
 const int kAudioProcessingNumberOfChannels = 1;
 
 const int kMaxNumberOfBuffersInFifo = 2;
 
@@ skipping 112 matching lines @@
   // 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_;
 };
 
 MediaStreamAudioProcessor::MediaStreamAudioProcessor(
     const blink::WebMediaConstraints& constraints,
     int effects,
-    MediaStreamType type,
     WebRtcPlayoutDataSource* playout_data_source)
     : render_delay_ms_(0),
       playout_data_source_(playout_data_source),
-      audio_mirroring_(false),
-      typing_detected_(false) {
+      goog_audio_mirroring_(false),
+      goog_typing_detected_(false) {
   capture_thread_checker_.DetachFromThread();
   render_thread_checker_.DetachFromThread();
-  InitializeAudioProcessingModule(constraints, effects, type);
+  InitializeAudioProcessingModule(constraints, effects);
 }
 
 MediaStreamAudioProcessor::~MediaStreamAudioProcessor() {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   StopAudioProcessing();
 }
 
 void MediaStreamAudioProcessor::OnCaptureFormatChanged(
     const media::AudioParameters& source_params) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   // There is no need to hold a lock here since the caller guarantees that
   // there is no more PushCaptureData() and ProcessAndConsumeData() callbacks
   // on the capture thread.
   InitializeCaptureConverter(source_params);
 
   // Reset the |capture_thread_checker_| since the capture data will come from
   // a new capture thread.
   capture_thread_checker_.DetachFromThread();
 }
 
 void MediaStreamAudioProcessor::PushCaptureData(media::AudioBus* audio_source) {
   DCHECK(capture_thread_checker_.CalledOnValidThread());
   DCHECK_EQ(audio_source->channels(),
             capture_converter_->source_parameters().channels());
   DCHECK_EQ(audio_source->frames(),
             capture_converter_->source_parameters().frames_per_buffer());
 
-  if (audio_mirroring_ &&
+  if (goog_audio_mirroring_ &&
       capture_converter_->source_parameters().channel_layout() ==
           media::CHANNEL_LAYOUT_STEREO) {
     // Swap the first and second channels.
     audio_source->SwapChannels(0, 1);
   }
 
   capture_converter_->Push(audio_source);
 }
 
 bool MediaStreamAudioProcessor::ProcessAndConsumeData(
@@ skipping 57 matching lines @@
 void MediaStreamAudioProcessor::OnPlayoutDataSourceChanged() {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   // There is no need to hold a lock here since the caller guarantees that
   // there is no more OnPlayoutData() callback on the render thread.
   render_thread_checker_.DetachFromThread();
   render_converter_.reset();
 }
 
 void MediaStreamAudioProcessor::GetStats(AudioProcessorStats* stats) {
   stats->typing_noise_detected =
-      (base::subtle::Acquire_Load(&typing_detected_) != false);
+      (base::subtle::Acquire_Load(&goog_typing_detected_) != false);
   GetAecStats(audio_processing_.get(), stats);
 }
 
 void MediaStreamAudioProcessor::InitializeAudioProcessingModule(
-    const blink::WebMediaConstraints& constraints, int effects,
-    MediaStreamType type) {
+    const blink::WebMediaConstraints& constraints, int effects) {
   DCHECK(!audio_processing_);
 
-  RTCMediaConstraints native_constraints(constraints);
+  MediaAudioConstraints audio_constraints(constraints, effects);
+  DCHECK(audio_constraints.IsValid());

[perkj_chrome, 2014/04/14 12:15:19]

I don't think you should crash if a user user insert invalid constraints from
JS. That is something you need to handle according to the spec.

[no longer working on chromium, 2014/04/14 14:40:50]

I was thinking WebRtcAudioCapture::Initialize() will safe guard the IsValid(),
but taking it a second look, it unfortunately created the processor before the
Initialize() method.

DCHECK is removed.

 
   // Audio mirroring can be enabled even though audio processing is otherwise
   // disabled.
-  audio_mirroring_ = GetPropertyFromConstraints(
-      &native_constraints, webrtc::MediaConstraintsInterface::kAudioMirroring);
+  goog_audio_mirroring_ = audio_constraints.GetProperty(
+      MediaAudioConstraints::kGoogAudioMirroring);
 
   if (!IsAudioTrackProcessingEnabled()) {
     RecordProcessingState(AUDIO_PROCESSING_IN_WEBRTC);
     return;
   }
 
-  // Only apply the fixed constraints for gUM of MEDIA_DEVICE_AUDIO_CAPTURE.
-  DCHECK(IsAudioMediaType(type));
-  if (type == MEDIA_DEVICE_AUDIO_CAPTURE)
-    ApplyFixedAudioConstraints(&native_constraints);
-
-  if (effects & media::AudioParameters::ECHO_CANCELLER) {
-    // If platform echo canceller is enabled, disable the software AEC.
-    native_constraints.AddMandatory(
-        MediaConstraintsInterface::kEchoCancellation,
-        MediaConstraintsInterface::kValueFalse, true);
+  // |kEchoCancellation| is used as a master control on enabling/disabling
+  // the audio processing.
+  const bool echo_cancellation = audio_constraints.GetProperty(
+      MediaAudioConstraints::kEchoCancellation);
+  if (!echo_cancellation) {
+    RecordProcessingState(AUDIO_PROCESSING_DISABLED);
+    return;
   }
 
 #if defined(OS_IOS)
   // On iOS, VPIO provides built-in AEC and AGC.
-  const bool enable_aec = false;
-  const bool enable_agc = false;
+  const bool goog_aec = false;
+  const bool goog_agc = false;
 #else
-  const bool enable_aec = GetPropertyFromConstraints(
-      &native_constraints, MediaConstraintsInterface::kEchoCancellation);
-  const bool enable_agc = GetPropertyFromConstraints(
-      &native_constraints, webrtc::MediaConstraintsInterface::kAutoGainControl);
+  // TODO(xians): goog_aec should be just echo_cancellation.
+  const bool goog_aec = audio_constraints.GetProperty(
+      MediaAudioConstraints::kGoogEchoCancellation);
+  const bool goog_agc = audio_constraints.GetProperty(
+      MediaAudioConstraints::kGoogAutoGainControl);
 #endif
 
 #if defined(OS_IOS) || defined(OS_ANDROID)
-  const bool enable_experimental_aec = false;
-  const bool enable_typing_detection = false;
+  const bool goog_experimental_aec = false;
+  const bool goog_typing_detection = false;
 #else
-  const bool enable_experimental_aec = GetPropertyFromConstraints(
-      &native_constraints,
-      MediaConstraintsInterface::kExperimentalEchoCancellation);
-  const bool enable_typing_detection = GetPropertyFromConstraints(
-      &native_constraints, MediaConstraintsInterface::kTypingNoiseDetection);
+  const bool goog_experimental_aec = audio_constraints.GetProperty(
+      MediaAudioConstraints::kGoogExperimentalEchoCancellation);
+  const bool goog_typing_detection = audio_constraints.GetProperty(
+      MediaAudioConstraints::kGoogTypingNoiseDetection);
 #endif
 
-  const bool enable_ns = GetPropertyFromConstraints(
-      &native_constraints, MediaConstraintsInterface::kNoiseSuppression);
-  const bool enable_experimental_ns = GetPropertyFromConstraints(
-        &native_constraints,
-        MediaConstraintsInterface::kExperimentalNoiseSuppression);
-  const bool enable_high_pass_filter = GetPropertyFromConstraints(
-      &native_constraints, MediaConstraintsInterface::kHighpassFilter);
+  const bool goog_ns = audio_constraints.GetProperty(
+      MediaAudioConstraints::kGoogNoiseSuppression);
+  const bool goog_experimental_ns = audio_constraints.GetProperty(
+      MediaAudioConstraints::kGoogExperimentalNoiseSuppression);
+ const bool goog_high_pass_filter = audio_constraints.GetProperty(
+     MediaAudioConstraints::kGoogHighpassFilter);
 
-  // Return immediately if no audio processing component is enabled.
-  if (!enable_aec && !enable_experimental_aec && !enable_ns &&
-      !enable_high_pass_filter && !enable_typing_detection && !enable_agc &&
-      !enable_experimental_ns) {
+  // Return immediately if no goog constraint is enabled.
+  if (!goog_aec && !goog_experimental_aec && !goog_ns &&
+      !goog_high_pass_filter && !goog_typing_detection &&
+      !goog_agc && !goog_experimental_ns) {
     RecordProcessingState(AUDIO_PROCESSING_DISABLED);
     return;
   }
 
   // Create and configure the webrtc::AudioProcessing.
   audio_processing_.reset(webrtc::AudioProcessing::Create(0));
 
   // Enable the audio processing components.
-  if (enable_aec) {
+  if (goog_aec) {
     EnableEchoCancellation(audio_processing_.get());
-    if (enable_experimental_aec)
+
+    if (goog_experimental_aec)
       EnableExperimentalEchoCancellation(audio_processing_.get());
 
     if (playout_data_source_)
       playout_data_source_->AddPlayoutSink(this);
   }
 
-  if (enable_ns)
+  if (goog_ns)
     EnableNoiseSuppression(audio_processing_.get());
 
-  if (enable_experimental_ns)
+  if (goog_experimental_ns)
     EnableExperimentalNoiseSuppression(audio_processing_.get());
 
-  if (enable_high_pass_filter)
+  if (goog_high_pass_filter)
     EnableHighPassFilter(audio_processing_.get());
 
-  if (enable_typing_detection) {
+  if (goog_typing_detection) {
     // TODO(xians): Remove this |typing_detector_| after the typing suppression
     // is enabled by default.
     typing_detector_.reset(new webrtc::TypingDetection());
     EnableTypingDetection(audio_processing_.get(), typing_detector_.get());
   }
 
-  if (enable_agc)
+  if (goog_agc)
     EnableAutomaticGainControl(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.
   CHECK_EQ(0,
            audio_processing_->set_sample_rate_hz(kAudioProcessingSampleRate));
   CHECK_EQ(0, audio_processing_->set_num_channels(
       kAudioProcessingNumberOfChannels, kAudioProcessingNumberOfChannels));
 
   RecordProcessingState(AUDIO_PROCESSING_ENABLED);
@@ skipping 96 matching lines @@
   audio_processing_->set_stream_key_pressed(key_pressed);
 
   err = audio_processing_->ProcessStream(audio_frame);
   DCHECK_EQ(err, 0) << "ProcessStream() error: " << err;
 
   if (typing_detector_ &&
       audio_frame->vad_activity_ != webrtc::AudioFrame::kVadUnknown) {
     bool vad_active =
         (audio_frame->vad_activity_ == webrtc::AudioFrame::kVadActive);
     bool typing_detected = typing_detector_->Process(key_pressed, vad_active);
-    base::subtle::Release_Store(&typing_detected_, typing_detected);
+    base::subtle::Release_Store(&goog_typing_detected_, typing_detected);
   }
 
   // Return 0 if the volume has not been changed, otherwise return the new
   // volume.
   return (agc->stream_analog_level() == volume) ?
       0 : agc->stream_analog_level();
 }
 
 void MediaStreamAudioProcessor::StopAudioProcessing() {
   if (!audio_processing_.get())
     return;
 
   StopAecDump();
 
   if (playout_data_source_)
     playout_data_source_->RemovePlayoutSink(this);
 
   audio_processing_.reset();
 }
 
 bool MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() const {
   const std::string group_name =
       base::FieldTrialList::FindFullName("MediaStreamAudioTrackProcessing");
   return group_name == "Enabled" || CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kEnableAudioTrackProcessing);
 }
 
 }  // namespace content