Revert of Adding BitrateController to audio network adaptor.

webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc

Index: webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
deleted file mode 100644
index 4cc49e8c0d7717f943d9b297ff2d85a8ce764443..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  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 "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
-
-#include <algorithm>
-
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-
-namespace {
-// TODO(minyue): consider passing this from a higher layer through
-// SetConstraints().
-// L2(14B) + IPv4(20B) + UDP(8B) + RTP(12B) + SRTP_AUTH(10B) = 64B = 512 bits
-constexpr int kPacketOverheadBits = 512;
-}
-
-BitrateController::Config::Config(int initial_bitrate_bps,
-                                  int initial_frame_length_ms)
-    : initial_bitrate_bps(initial_bitrate_bps),
-      initial_frame_length_ms(initial_frame_length_ms) {}
-
-BitrateController::Config::~Config() = default;
-
-BitrateController::BitrateController(const Config& config)
-    : config_(config),
-      bitrate_bps_(config_.initial_bitrate_bps),
-      overhead_rate_bps_(kPacketOverheadBits * 1000 /
-                         config_.initial_frame_length_ms) {
-  RTC_DCHECK_GT(bitrate_bps_, 0);
-  RTC_DCHECK_GT(overhead_rate_bps_, 0);
-}
-
-void BitrateController::MakeDecision(
-    const NetworkMetrics& metrics,
-    AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
-  // Decision on |bitrate_bps| should not have been made.
-  RTC_DCHECK(!config->bitrate_bps);
-
-  if (metrics.target_audio_bitrate_bps) {
-    int overhead_rate =
-        config->frame_length_ms
-            ? kPacketOverheadBits * 1000 / *config->frame_length_ms
-            : overhead_rate_bps_;
-    // If |metrics.target_audio_bitrate_bps| had included overhead, we would
-    // simply do:
-    //     bitrate_bps_ = metrics.target_audio_bitrate_bps - overhead_rate;
-    // Follow https://bugs.chromium.org/p/webrtc/issues/detail?id=6315 to track
-    // progress regarding this.
-    // Now we assume that |metrics.target_audio_bitrate_bps| can handle the
-    // overhead of most recent packets.
-    bitrate_bps_ = std::max(0, *metrics.target_audio_bitrate_bps +
-                                   overhead_rate_bps_ - overhead_rate);
-    // TODO(minyue): apply a smoothing on the |overhead_rate_bps_|.
-    overhead_rate_bps_ = overhead_rate;
-  }
-  config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
-}
-
-}  // namespace webrtc