Remove WebRtc audio source references for local audio tracks.

content/renderer/media/webrtc/processed_local_audio_source.cc

 // Copyright (c) 2012 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/processed_local_audio_source.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/stringprintf.h"
 #include "content/renderer/media/audio_device_factory.h"
@@ skipping 87 matching lines @@
   // Sanity-check that the constraints, plus the additional input effects are
   // valid when combined.
   const MediaAudioConstraints audio_constraints(
       constraints_, device_info().device.input.effects);
   if (!audio_constraints.IsValid()) {
     WebRtcLogMessage("ProcessedLocalAudioSource::EnsureSourceIsStarted() fails "
                      " because MediaAudioConstraints are not valid.");
     return false;
   }
 
-  // Build an AudioOptions by applying relevant constraints to it, and then use
-  // it to create a webrtc::AudioSourceInterface instance.
-  cricket::AudioOptions rtc_options;
-  rtc_options.echo_cancellation = ConstraintToOptional(
-      constraints_, &blink::WebMediaTrackConstraintSet::echoCancellation);
-  rtc_options.delay_agnostic_aec = ConstraintToOptional(
-      constraints_, &blink::WebMediaTrackConstraintSet::googDAEchoCancellation);
-  rtc_options.auto_gain_control = ConstraintToOptional(
-      constraints_, &blink::WebMediaTrackConstraintSet::googAutoGainControl);
-  rtc_options.experimental_agc = ConstraintToOptional(
-      constraints_,
-      &blink::WebMediaTrackConstraintSet::googExperimentalAutoGainControl);
-  rtc_options.noise_suppression = ConstraintToOptional(
-      constraints_, &blink::WebMediaTrackConstraintSet::googNoiseSuppression);
-  rtc_options.experimental_ns = ConstraintToOptional(
-      constraints_,
-      &blink::WebMediaTrackConstraintSet::googExperimentalNoiseSuppression);
-  rtc_options.highpass_filter = ConstraintToOptional(
-      constraints_, &blink::WebMediaTrackConstraintSet::googHighpassFilter);
-  rtc_options.typing_detection = ConstraintToOptional(
-      constraints_,
-      &blink::WebMediaTrackConstraintSet::googTypingNoiseDetection);
-  rtc_options.stereo_swapping = ConstraintToOptional(
-      constraints_, &blink::WebMediaTrackConstraintSet::googAudioMirroring);
-  MediaAudioConstraints::ApplyFixedAudioConstraints(&rtc_options);
   if (device_info().device.input.effects &
       media::AudioParameters::ECHO_CANCELLER) {
     // TODO(hta): Figure out if we should be looking at echoCancellation.
     // Previous code had googEchoCancellation only.
     const blink::BooleanConstraint& echoCancellation =
         constraints_.basic().googEchoCancellation;
     if (echoCancellation.hasExact() && !echoCancellation.exact()) {
       StreamDeviceInfo modified_device_info(device_info());
       modified_device_info.device.input.effects &=
           ~media::AudioParameters::ECHO_CANCELLER;
       SetDeviceInfo(modified_device_info);
     }
-    rtc_options.echo_cancellation = rtc::Optional<bool>(false);
-  }
-  rtc_source_ = pc_factory_->CreateLocalAudioSource(rtc_options);
-  if (rtc_source_->state() != webrtc::MediaSourceInterface::kLive) {
-    WebRtcLogMessage("ProcessedLocalAudioSource::EnsureSourceIsStarted() fails "
-                     " because the rtc LocalAudioSource is not live.");
-    return false;
   }
 
   // Create the MediaStreamAudioProcessor, bound to the WebRTC audio device
   // module.
   WebRtcAudioDeviceImpl* const rtc_audio_device =
       pc_factory_->GetWebRtcAudioDevice();
   if (!rtc_audio_device) {
     WebRtcLogMessage("ProcessedLocalAudioSource::EnsureSourceIsStarted() fails "
                      " because there is no WebRtcAudioDeviceImpl instance.");
     return false;
@@ skipping 235 matching lines @@
 
   // If the buffer size is missing from the StreamDeviceInfo, provide 10ms as a
   // fall-back.
   //
   // TODO(miu): Identify where/why the buffer size might be missing, fix the
   // code, and then require it here.
   return (sample_rate / 100);
 }
 
 }  // namespace content