Adding FrameLengthController to audio network adaptor.

webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h

Index: webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
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index 0000000000000000000000000000000000000000..82baa60b8fa18fe355d162c136ba7f815f2565a4
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+++ b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
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+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+
+#include <map>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
+
+namespace webrtc {
+
+// Determines target frame length based on the network metrics and the decision
+// of FEC controller.
+class FrameLengthController final : public Controller {
+ public:
+  struct Config {
+    Config(const std::vector<int>& encoder_frame_lengths_ms,
+           int initial_frame_length_ms,
+           float fl_increasing_packet_loss_fraction,
+           float fl_decreasing_packet_loss_fraction,
+           int fl_20ms_to_60ms_bandwidth_bps,
+           int fl_60ms_to_20ms_bandwidth_bps);
+    Config(const Config& other);
+    ~Config();
+    std::vector<int> encoder_frame_lengths_ms;
+    int initial_frame_length_ms;
+    float fl_increasing_packet_loss_fraction;
+    float fl_decreasing_packet_loss_fraction;
+    int fl_20ms_to_60ms_bandwidth_bps;
+    int fl_60ms_to_20ms_bandwidth_bps;
+  };
+
+  explicit FrameLengthController(const Config& config);
+
+  ~FrameLengthController() override;
+
+  void MakeDecision(const NetworkMetrics& metrics,
+                    AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
+
+  void SetConstraints(const Constraints& constraints) override;
+
+ private:
+  friend class FrameLengthControllerForTest;
+
+  struct FrameLengthChange {
+    FrameLengthChange(int from_frame_length_ms, int to_frame_length_ms);
+    ~FrameLengthChange();
+    bool operator<(const FrameLengthChange& rhs) const;
+    int from_frame_length_ms;
+    int to_frame_length_ms;
+  };
+
+  void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                   int max_frame_length_ms);
+
+  bool FrameLengthIncreasingDecision(
+      const NetworkMetrics& metrics,
+      const AudioNetworkAdaptor::EncoderRuntimeConfig& config) const;
+
+  bool FrameLengthDecreasingDecision(
+      const NetworkMetrics& metrics,
+      const AudioNetworkAdaptor::EncoderRuntimeConfig& config) const;
+
+  const Config config_;
+
+  std::vector<int> run_time_frame_lengths_ms_;
+
+  std::vector<int>::iterator frame_length_ms_;
+
+  std::map<FrameLengthChange, int> frame_length_change_criteria_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(FrameLengthController);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_