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Issue 2746333009: OnReceivedUplinkPacketLossFraction() receives [const rtc::Optional<float>&] (Closed)
Patch Set: Rebased Created 3 years, 9 months ago
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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
(...skipping 95 matching lines...)
106 bool SetDtx(bool enable) override; 106 bool SetDtx(bool enable) override;
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 const rtc::Optional<float>& uplink_packet_loss_fraction) override;
117 void OnReceivedUplinkBandwidth( 117 void OnReceivedUplinkBandwidth(
118 int target_audio_bitrate_bps, 118 int target_audio_bitrate_bps,
119 rtc::Optional<int64_t> probing_interval_ms) override; 119 rtc::Optional<int64_t> probing_interval_ms) override;
120 void OnReceivedRtt(int rtt_ms) override; 120 void OnReceivedRtt(int rtt_ms) override;
121 void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; 121 void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
122 void SetReceiverFrameLengthRange(int min_frame_length_ms, 122 void SetReceiverFrameLengthRange(int min_frame_length_ms,
123 int max_frame_length_ms) override; 123 int max_frame_length_ms) override;
124 rtc::ArrayView<const int> supported_frame_lengths_ms() const { 124 rtc::ArrayView<const int> supported_frame_lengths_ms() const {
125 return config_.supported_frame_lengths_ms; 125 return config_.supported_frame_lengths_ms;
126 } 126 }
(...skipping 48 matching lines...)
175 rtc::Optional<size_t> overhead_bytes_per_packet_; 175 rtc::Optional<size_t> overhead_bytes_per_packet_;
176 const std::unique_ptr<SmoothingFilter> bitrate_smoother_; 176 const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
177 rtc::Optional<int64_t> bitrate_smoother_last_update_time_; 177 rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
178 178
179 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); 179 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
180 }; 180 };
181 181
182 } // namespace webrtc 182 } // namespace webrtc
183 183
184 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ 184 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
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