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..f6e64ced5af9373c0886f613869de3652acfb70d 100644
--- a/webrtc/test/fake_encoder.cc
+++ b/webrtc/test/fake_encoder.cc
@@ -57,6 +57,11 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
     // at the display time of the previous frame.
     time_since_last_encode_ms = time_now_ms - last_encode_time_ms_;
   }
+  if (time_since_last_encode_ms > 1000 / config_.maxFramerate) {
+    // 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;

[mflodman, 2015/11/26 10:23:38]

This can still add too much, why isn't this the same as in the if statement
above?

[stefan-webrtc, 2015/11/26 11:23:38]

Very good question :)

It should be 3x in the if statement too.

[sprang_webrtc, 2015/11/26 13:04:22]

Done.

+  }
 
   size_t bits_available =
       static_cast<size_t>(target_bitrate_kbps_ * time_since_last_encode_ms);