Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.

TEST=ACM unit test is added, also a manual integration test is writen. 
Review URL: https://webrtc-codereview.appspot.com/1097009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/interface/audio_coding_module.h

@@ -940,6 +940,22 @@ class AudioCodingModule: public Module {
   //
   virtual WebRtc_Word32 NetworkStatistics(
       ACMNetworkStatistics& network_statistics) const = 0;
+
+  //
+  // Set an initial delay for playout.
+  // An initial delay yields ACM playout silence until equivalent of |delay_ms|
+  // audio payload is accumulated in NetEq jitter. Thereafter, ACM pulls audio
+  // from NetEq in its regular fashion, and the given delay is maintained as
+  // "minimum playout delay."
+  //
+  // Input:
+  //   -delay_ms           : delay in milliseconds.
+  //
+  // Return values:
+  //   -1 if failed to set the delay.
+  //    0 if delay is set successfully.
+  //
+  virtual int SetInitialPlayoutDelay(int delay_ms) = 0;
 };
 
 }  // namespace webrtc