Add rtc_use_memcheck flag, update MB and GN to handle it, and add gni files listing the runtime deps

webrtc/build/webrtc.gni

Index: webrtc/build/webrtc.gni
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index 5ab5d35973b4bc0412113064f938a79ca398227c..17405c3cb38f8204a15661ca143ff29ef5d412e5 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -118,6 +118,10 @@ declare_args() {
   # use file-based audio playout and record.
   rtc_use_dummy_audio_file_devices = false
 
+  # When set to true, test targets will declare the files needed to run memcheck
+  # as data dependencies. This is to enable memcheck execution on swarming bots.
+  rtc_use_memcheck = false
+
   # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
   # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
   # only be initialized once. Projects that initialize FFmpeg externally, such