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}
diff --git a/webrtc/video_engine_tests.isolate b/webrtc/video_engine_tests.isolate
index 5aa9623..f2f961f 100644
--- a/webrtc/video_engine_tests.isolate
+++ b/webrtc/video_engine_tests.isolate
@@ -11,6 +11,7 @@
'variables': {
'files': [
'<(DEPTH)/resources/foreman_cif_short.yuv',
+ '<(DEPTH)/resources/voice_engine/audio_long16.pcm',
],
},
}],
