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
new file mode 100644
index 0000000000000000000000000000000000000000..4cc49e8c0d7717f943d9b297ff2d85a8ce764443
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+++ b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -0,0 +1,68 @@
+/*
+ *  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