Remove potential deadlock in RTPSenderAudio.

Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls
by doing a lot shorter locking which fetches a current state of
RTPSenderAudio variables before sending.

Thread annotates locked variables and removes one lock in
RTPSenderAudio, bonus fixes data races reported in voe_auto_test
--automated under TSan (DTMF data race).

Also includes some bonus cleanup of RTPSenderVideo which removes the
send critsect completely as all methods using it was always called
from RTPSender under its send_critsect.

R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
BUG=3001, chromium:454654

Review URL: https://webrtc-codereview.appspot.com/41869004

Cr-Commit-Position: refs/heads/master@{#8348}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -904,11 +904,6 @@ int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
   return rtp_sender_.SendTelephoneEvent(key, time_ms, level);
 }
 
-bool ModuleRtpRtcpImpl::SendTelephoneEventActive(
-    int8_t& telephone_event) const {
-  return rtp_sender_.SendTelephoneEventActive(&telephone_event);
-}
-
 // Set audio packet size, used to determine when it's time to send a DTMF
 // packet in silence (CNG).
 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(