Add audio streams to CallTest and a first A/V call test.

Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1542653002 .

Cr-Commit-Position: refs/heads/master@{#11171}

webrtc/video_engine_tests.isolate

@@ -11,6 +11,7 @@
       'variables': {
         'files': [
           '<(DEPTH)/resources/foreman_cif_short.yuv',
+          '<(DEPTH)/resources/voice_engine/audio_long16.pcm',
         ],
       },
     }],