Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180

Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.

TBR=solenberg

Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}

api/peerconnectioninterface.h

@@ -788,6 +788,21 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
       std::unique_ptr<rtc::BitrateAllocationStrategy>
           bitrate_allocation_strategy) {}
 
+  // Enable/disable playout of received audio streams. Enabled by default. Note
+  // that even if playout is enabled, streams will only be played out if the
+  // appropriate SDP is also applied. Setting |playout| to false will stop
+  // playout of the underlying audio device but starts a task which will poll
+  // for audio data every 10ms to ensure that audio processing happens and the
+  // audio statistics are updated.
+  // TODO(henrika): deprecate and remove this.
+  virtual void SetAudioPlayout(bool playout) {}
+
+  // Enable/disable recording of transmitted audio streams. Enabled by default.
+  // Note that even if recording is enabled, streams will only be recorded if
+  // the appropriate SDP is also applied.
+  // TODO(henrika): deprecate and remove this.
+  virtual void SetAudioRecording(bool recording) {}
+
   // Returns the current SignalingState.
   virtual SignalingState signaling_state() = 0;