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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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107 bool GetDtx() const override; | 107 bool GetDtx() const override; |
108 | 108 |
109 bool SetApplication(Application application) override; | 109 bool SetApplication(Application application) override; |
110 void SetMaxPlaybackRate(int frequency_hz) override; | 110 void SetMaxPlaybackRate(int frequency_hz) override; |
111 bool EnableAudioNetworkAdaptor(const std::string& config_string, | 111 bool EnableAudioNetworkAdaptor(const std::string& config_string, |
112 RtcEventLog* event_log, | 112 RtcEventLog* event_log, |
113 const Clock* clock) override; | 113 const Clock* clock) override; |
114 void DisableAudioNetworkAdaptor() override; | 114 void DisableAudioNetworkAdaptor() override; |
115 void OnReceivedUplinkPacketLossFraction( | 115 void OnReceivedUplinkPacketLossFraction( |
116 float uplink_packet_loss_fraction) override; | 116 float uplink_packet_loss_fraction) override; |
| 117 void OnReceivedUplinkRecoverablePacketLossFraction( |
| 118 float uplink_recoverable_packet_loss_fraction) override; |
117 void OnReceivedUplinkBandwidth( | 119 void OnReceivedUplinkBandwidth( |
118 int target_audio_bitrate_bps, | 120 int target_audio_bitrate_bps, |
119 rtc::Optional<int64_t> probing_interval_ms) override; | 121 rtc::Optional<int64_t> probing_interval_ms) override; |
120 void OnReceivedRtt(int rtt_ms) override; | 122 void OnReceivedRtt(int rtt_ms) override; |
121 void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; | 123 void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; |
122 void SetReceiverFrameLengthRange(int min_frame_length_ms, | 124 void SetReceiverFrameLengthRange(int min_frame_length_ms, |
123 int max_frame_length_ms) override; | 125 int max_frame_length_ms) override; |
124 rtc::ArrayView<const int> supported_frame_lengths_ms() const { | 126 rtc::ArrayView<const int> supported_frame_lengths_ms() const { |
125 return config_.supported_frame_lengths_ms; | 127 return config_.supported_frame_lengths_ms; |
126 } | 128 } |
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175 rtc::Optional<size_t> overhead_bytes_per_packet_; | 177 rtc::Optional<size_t> overhead_bytes_per_packet_; |
176 const std::unique_ptr<SmoothingFilter> bitrate_smoother_; | 178 const std::unique_ptr<SmoothingFilter> bitrate_smoother_; |
177 rtc::Optional<int64_t> bitrate_smoother_last_update_time_; | 179 rtc::Optional<int64_t> bitrate_smoother_last_update_time_; |
178 | 180 |
179 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); | 181 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); |
180 }; | 182 }; |
181 | 183 |
182 } // namespace webrtc | 184 } // namespace webrtc |
183 | 185 |
184 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ | 186 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ |
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