Adding AudioNetworkAdaptor interfaces.

webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h

+/*
+ *  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_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+
+#include "webrtc/base/optional.h"
+
+namespace webrtc {
+
+// An AudioNetworkAdaptor optimizes the audio experience by suggesting a
+// suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
+// encoder based on network metrics.
+class AudioNetworkAdaptor {
+ public:
+  struct EncoderRuntimeConfig {
+    EncoderRuntimeConfig();
+    EncoderRuntimeConfig(const EncoderRuntimeConfig& other);
+    ~EncoderRuntimeConfig();
+    rtc::Optional<int> bitrate_bps;
+    rtc::Optional<int> frame_length_ms;
+    rtc::Optional<float> uplink_packet_loss_fraction;
+    rtc::Optional<bool> enable_fec;
+    rtc::Optional<bool> enable_dtx;
+
+    // Some encoders can encode fewer channels than the actual input to make
+    // better use of the bandwidth. |num_channels| sets the number of channels
+    // to encode.
+    rtc::Optional<size_t> num_channels;
+  };
+
+  virtual ~AudioNetworkAdaptor() = default;
+
+  virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
+
+  virtual void SetUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) = 0;
+
+  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                           int max_frame_length_ms) = 0;
+
+  virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
+
+  virtual void StartDebugDump(FILE* file_handle) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_