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

webrtc/test/fake_encoder.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 39 matching lines @@
                             const std::vector<FrameType>* frame_types) {
   assert(config_.maxFramerate > 0);
   int64_t time_since_last_encode_ms = 1000 / config_.maxFramerate;
   int64_t time_now_ms = clock_->TimeInMilliseconds();
   const bool first_encode = last_encode_time_ms_ == 0;
   if (!first_encode) {
     // For all frames but the first we can estimate the display time by looking
     // 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 > 3 * 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;
+  }
 
   size_t bits_available =
       static_cast<size_t>(target_bitrate_kbps_ * time_since_last_encode_ms);
   size_t min_bits = static_cast<size_t>(
       config_.simulcastStream[0].minBitrate * time_since_last_encode_ms);
   if (bits_available < min_bits)
     bits_available = min_bits;
   size_t max_bits =
       static_cast<size_t>(max_target_bitrate_kbps_ * time_since_last_encode_ms);
   if (max_bits > 0 && max_bits < bits_available)
@@ skipping 118 matching lines @@
       delay_ms_(delay_ms) {}
 
 int32_t DelayedEncoder::Encode(const VideoFrame& input_image,
                                const CodecSpecificInfo* codec_specific_info,
                                const std::vector<FrameType>* frame_types) {
   SleepMs(delay_ms_);
   return FakeEncoder::Encode(input_image, codec_specific_info, frame_types);
 }
 }  // namespace test
 }  // namespace webrtc