1 /* 2 * Copyright (c) 2017 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_ 12 #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_ 13 14 #include <stddef.h> 15 16 #include "absl/types/optional.h" 17 18 namespace webrtc { 19 20 struct AudioEncoderRuntimeConfig { 21 AudioEncoderRuntimeConfig(); 22 AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other); 23 ~AudioEncoderRuntimeConfig(); 24 AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other); 25 bool operator==(const AudioEncoderRuntimeConfig& other) const; 26 absl::optional<int> bitrate_bps; 27 absl::optional<int> frame_length_ms; 28 // Note: This is what we tell the encoder. It doesn't have to reflect 29 // the actual NetworkMetrics; it's subject to our decision. 30 absl::optional<float> uplink_packet_loss_fraction; 31 absl::optional<bool> enable_fec; 32 absl::optional<bool> enable_dtx; 33 34 // Some encoders can encode fewer channels than the actual input to make 35 // better use of the bandwidth. |num_channels| sets the number of channels 36 // to encode. 37 absl::optional<size_t> num_channels; 38 39 // This is true if the last frame length change was an increase, and otherwise 40 // false. 41 // The value of this boolean is used to apply a different offset to the 42 // per-packet overhead that is reported by the BWE. The exact offset value 43 // is most important right after a frame length change, because the frame 44 // length change affects the overhead. In the steady state, the exact value is 45 // not important because the BWE will compensate. 46 bool last_fl_change_increase = false; 47 }; 48 49 } // namespace webrtc 50 51 #endif // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_ 52