Revert of Fix simulator issue where chokes didn't apply to non-congested packets.

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.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 413 matching lines @@
 void ChokeFilter::SetCapacity(uint32_t kbps) {
   BWE_TEST_LOGGING_ENABLE(false);
   BWE_TEST_LOGGING_LOG1("BitrateChoke", "%d kbps", kbps);
   kbps_ = kbps;
 }
 
 void ChokeFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (PacketsIt it = in_out->begin(); it != in_out->end(); ) {
     int64_t earliest_send_time_us =
-        std::max(last_send_time_us_, (*it)->send_time_us());
-    int64_t new_send_time_us = earliest_send_time_us +
+        last_send_time_us_ +
         ((*it)->payload_size() * 8 * 1000 + kbps_ / 2) / kbps_;
+    int64_t new_send_time_us =
+        std::max((*it)->send_time_us(), earliest_send_time_us);
     if (delay_cap_helper_->ShouldSendPacket(new_send_time_us,
                                             (*it)->send_time_us())) {
       (*it)->set_send_time_us(new_send_time_us);
       last_send_time_us_ = new_send_time_us;
       ++it;
     } else {
       delete *it;
       it = in_out->erase(it);
     }
   }
@@ skipping 266 matching lines @@
 uint32_t PeriodicKeyFrameSource::NextPacketSize(uint32_t frame_size,
                                                 uint32_t remaining_payload) {
   uint32_t fragments =
       (frame_size + (kMaxPayloadSizeBytes - 1)) / kMaxPayloadSizeBytes;
   uint32_t avg_size = (frame_size + fragments - 1) / fragments;
   return std::min(avg_size, remaining_payload);
 }
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc