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| 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 SetReceiverFrameLengthRange(int min_frame_length_ms, |
| 47 int max_frame_length_ms) = 0; |
| 48 |
| 49 virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0; |
| 50 |
| 51 virtual void StartDebugDump(FILE* file_handle) = 0; |
| 52 }; |
| 53 |
| 54 } // namespace webrtc |
| 55 |
| 56 #endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWO
RK_ADAPTOR_H_ |
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