Enable new getUserMedia audio constraints algorithm behind a flag.

content/renderer/media/webrtc/processed_local_audio_source.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 
+#include <string>
+
 #include "base/atomicops.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "content/common/media/media_stream_options.h"
 #include "content/renderer/media/media_stream_audio_level_calculator.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "media/base/audio_capturer_source.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
@@ skipping 11 matching lines @@
 // video conferencing call). Owns a media::AudioCapturerSource and the
 // MediaStreamProcessor that modifies its audio. Modified audio is delivered to
 // one or more MediaStreamAudioTracks.
 class CONTENT_EXPORT ProcessedLocalAudioSource final
     : NON_EXPORTED_BASE(public MediaStreamAudioSource),
       NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
  public:
   // |consumer_render_frame_id| references the RenderFrame that will consume the
   // audio data. Audio parameters and (optionally) a pre-existing audio session
   // ID are derived from |device_info|. |factory| must outlive this instance.
-  ProcessedLocalAudioSource(int consumer_render_frame_id,
-                            const StreamDeviceInfo& device_info,
-                            const blink::WebMediaConstraints& constraints,
-                            const ConstraintsCallback& started_callback,
-                            PeerConnectionDependencyFactory* factory);
+  ProcessedLocalAudioSource(
+      int consumer_render_frame_id,
+      const StreamDeviceInfo& device_info,
+      const AudioProcessingProperties& audio_processing_properties,
+      const ConstraintsCallback& started_callback,
+      PeerConnectionDependencyFactory* factory);
 
   ~ProcessedLocalAudioSource() final;
 
   // If |source| is an instance of ProcessedLocalAudioSource, return a
   // type-casted pointer to it. Otherwise, return null.
   static ProcessedLocalAudioSource* From(MediaStreamAudioSource* source);
 
   // Non-browser unit tests cannot provide RenderFrame implementations at
   // run-time. This is used to skip the otherwise mandatory check for a valid
   // render frame ID when the source is started.
   void SetAllowInvalidRenderFrameIdForTesting(bool allowed) {
     allow_invalid_render_frame_id_for_testing_ = allowed;
   }
 
-  // Gets/Sets source constraints. Using this is optional, but must be done
-  // before the first call to ConnectToTrack().
-  const blink::WebMediaConstraints& source_constraints() const {
-    return constraints_;
+  const AudioProcessingProperties& audio_processing_properties() const {
+    return audio_processing_properties_;
   }
 
   // The following accessors are not valid until after the source is started
   // (when the first track is connected).
   const scoped_refptr<MediaStreamAudioProcessor>& audio_processor() const {
     return audio_processor_;
   }
   const scoped_refptr<MediaStreamAudioLevelCalculator::Level>& audio_level()
       const {
     return level_calculator_.level();
@@ skipping 32 matching lines @@
   // The RenderFrame that will consume the audio data. Used when creating
   // AudioCapturerSources.
   const int consumer_render_frame_id_;
 
   PeerConnectionDependencyFactory* const pc_factory_;
 
   // In debug builds, check that all methods that could cause object graph
   // or data flow changes are being called on the main thread.
   base::ThreadChecker thread_checker_;
 
-  // Cached audio constraints for the capturer.
-  const blink::WebMediaConstraints constraints_;
+  AudioProcessingProperties audio_processing_properties_;
 
   // Callback that's called when the audio source has been initialized.
   ConstraintsCallback started_callback_;
 
   // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output
   // data is in a unit of 10 ms data chunk.
   scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
   // The device created by the AudioDeviceFactory in EnsureSourceIsStarted().
   scoped_refptr<media::AudioCapturerSource> source_;
 
   // Lock used to ensure thread-safe access to |source_| by SetVolume().
   mutable base::Lock source_lock_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   base::subtle::Atomic32 volume_;
 
   // Used to calculate the signal level that shows in the UI.
   MediaStreamAudioLevelCalculator level_calculator_;
 
   bool allow_invalid_render_frame_id_for_testing_;
 
   DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioSource);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_