Move ownership of voe::Channel into Audio[Receive|Send]Stream.

* VoEBase contains only stub methods (until downstream code is
  updated).

* voe::Channel and ChannelProxy classes remain, but are now created
  internally to the streams. As a result,
  internal::Audio[Receive|Send]Stream can have a ChannelProxy injected
  for testing.

* Stream classes share Call::module_process_thread_ for their RtpRtcp
  modules, rather than using a separate thread shared only among audio
  streams.

* voe::Channel instances use Call::worker_queue_ for encoding packets,
  rather than having a separate queue for audio (send) streams.

Bug: webrtc:4690
Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d
Reviewed-on: https://webrtc-review.googlesource.com/34640
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21578}

call/audio_state.h

@@ -21,19 +21,12 @@ class AudioProcessing;
 class AudioTransport;
 class VoiceEngine;
 
-// WORK IN PROGRESS
-// This class is under development and is not yet intended for for use outside
-// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
-// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
-
 // AudioState holds the state which must be shared between multiple instances of
 // webrtc::Call for audio processing purposes.
 class AudioState : public rtc::RefCountInterface {
  public:
   struct Config {
-    // VoiceEngine used for audio streams and audio/video synchronization.
-    // AudioState will tickle the VoE refcount to keep it alive for as long as
-    // the AudioState itself.
+    // TODO(solenberg): Remove once clients don't use it anymore.
     VoiceEngine* voice_engine = nullptr;
 
     // The audio mixer connected to active receive streams. One per