Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq4/neteq_impl.h

@@ -166,8 +166,7 @@ class NetEqImpl : public webrtc::NetEq {
   // Disables post-decode VAD.
   virtual void DisableVad();
 
-  // Returns the RTP timestamp for the last sample delivered by GetAudio().
-  virtual uint32_t PlayoutTimestamp();
+  virtual bool GetPlayoutTimestamp(uint32_t* timestamp);
 
   virtual int SetTargetNumberOfChannels() { return kNotImplemented; }