Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
      12             : #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
      13             : 
      14             : #include "webrtc/base/optional.h"
      15             : 
      16             : namespace webrtc {
      17             : 
      18             : // An AudioNetworkAdaptor optimizes the audio experience by suggesting a
      19             : // suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
      20             : // encoder based on network metrics.
      21           0 : class AudioNetworkAdaptor {
      22             :  public:
      23           0 :   struct EncoderRuntimeConfig {
      24             :     EncoderRuntimeConfig();
      25             :     EncoderRuntimeConfig(const EncoderRuntimeConfig& other);
      26             :     ~EncoderRuntimeConfig();
      27             :     rtc::Optional<int> bitrate_bps;
      28             :     rtc::Optional<int> frame_length_ms;
      29             :     rtc::Optional<float> uplink_packet_loss_fraction;
      30             :     rtc::Optional<bool> enable_fec;
      31             :     rtc::Optional<bool> enable_dtx;
      32             : 
      33             :     // Some encoders can encode fewer channels than the actual input to make
      34             :     // better use of the bandwidth. |num_channels| sets the number of channels
      35             :     // to encode.
      36             :     rtc::Optional<size_t> num_channels;
      37             :   };
      38             : 
      39           0 :   virtual ~AudioNetworkAdaptor() = default;
      40             : 
      41             :   virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
      42             : 
      43             :   virtual void SetUplinkPacketLossFraction(
      44             :       float uplink_packet_loss_fraction) = 0;
      45             : 
      46             :   virtual void SetRtt(int rtt_ms) = 0;
      47             : 
      48             :   virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0;
      49             : 
      50             :   virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0;
      51             : 
      52             :   virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
      53             : 
      54             :   virtual void StartDebugDump(FILE* file_handle) = 0;
      55             : 
      56             :   virtual void StopDebugDump() = 0;
      57             : };
      58             : 
      59             : }  // namespace webrtc
      60             : 
      61             : #endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_