Added configurable offsets to the per-packet overhead in ANA.

modules/audio_coding/audio_network_adaptor/bitrate_controller.cc

 /*
  *  Copyright (c) 2016 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.
  */
 
 #include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
 
 #include <algorithm>
 
 #include "rtc_base/checks.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 namespace audio_network_adaptor {
 
 BitrateController::Config::Config(int initial_bitrate_bps,
-                                  int initial_frame_length_ms)
+                                  int initial_frame_length_ms,
+                                  int fl_increase_overhead_offset,
+                                  int fl_decrease_overhead_offset)
     : initial_bitrate_bps(initial_bitrate_bps),
-      initial_frame_length_ms(initial_frame_length_ms) {}
+      initial_frame_length_ms(initial_frame_length_ms),
+      fl_increase_overhead_offset(fl_increase_overhead_offset),
+      fl_decrease_overhead_offset(fl_decrease_overhead_offset) {}
 
 BitrateController::Config::~Config() = default;
 
 BitrateController::BitrateController(const Config& config)
     : config_(config),
       bitrate_bps_(config_.initial_bitrate_bps),
       frame_length_ms_(config_.initial_frame_length_ms) {
   RTC_DCHECK_GT(bitrate_bps_, 0);
   RTC_DCHECK_GT(frame_length_ms_, 0);
 }
@@ skipping 12 matching lines @@
   // Decision on |bitrate_bps| should not have been made.
   RTC_DCHECK(!config->bitrate_bps);
   if (target_audio_bitrate_bps_ && overhead_bytes_per_packet_) {
     // Current implementation of BitrateController can only work when
     // |metrics.target_audio_bitrate_bps| includes overhead is enabled. This is
     // currently governed by the following field trial.
     RTC_DCHECK(
         webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead"));
     if (config->frame_length_ms)
       frame_length_ms_ = *config->frame_length_ms;
-    int overhead_rate_bps =
-        static_cast<int>(*overhead_bytes_per_packet_ * 8 * 1000 /
-            frame_length_ms_);
+    int offset = config->last_fl_change_increase
+                     ? config_.fl_increase_overhead_offset
+                     : config_.fl_decrease_overhead_offset;
+    // Check that
+    // -(*overhead_bytes_per_packet_) <= offset <= (*overhead_bytes_per_packet_)
+    RTC_DCHECK_GE(*overhead_bytes_per_packet_, -offset);
+    RTC_DCHECK_LE(offset, *overhead_bytes_per_packet_);
+    int overhead_rate_bps = static_cast<int>(
+        (*overhead_bytes_per_packet_ + offset) * 8 * 1000 / frame_length_ms_);
     bitrate_bps_ = std::max(0, *target_audio_bitrate_bps_ - overhead_rate_bps);
   }
   config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
 }
 
 }  // namespace audio_network_adaptor
 }  // namespace webrtc