Fix bug in calculation of averge queue time in paced sender.

webrtc/test/fake_encoder.cc

Index: webrtc/test/fake_encoder.cc
diff --git a/webrtc/test/fake_encoder.cc b/webrtc/test/fake_encoder.cc
index 44fb1c5882c3fccd48f2a196a639bfa4be6c2cbf..530e43c04fd414d1ec36e2e95e382365b3adaaa2 100644
--- a/webrtc/test/fake_encoder.cc
+++ b/webrtc/test/fake_encoder.cc
@@ -50,6 +50,11 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
                             const std::vector<FrameType>* frame_types) {
   assert(config_.maxFramerate > 0);
   int64_t time_since_last_encode_ms = 1000 / config_.maxFramerate;
+  if (time_since_last_encode_ms > 1000 / config_.maxFramerate) {

[stefan-webrtc, 2015/11/25 18:14:22]

I don't think this does much... :)

You probably want this at line 65 or 64.

[sprang_webrtc, 2015/11/25 19:40:42]

Oops. Done.

+    // Rudimentary check to make sure we don't widely overshoot bitrate target
+    // when resuming encoding after a suspension.
+    time_since_last_encode_ms = 3 * 1000 / config_.maxFramerate;
+  }
   int64_t time_now_ms = clock_->TimeInMilliseconds();
   const bool first_encode = last_encode_time_ms_ == 0;
   if (!first_encode) {