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 class AudioNetworkAdaptor {
22  public:
23   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   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_
62