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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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79 | 79 |
80 void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction( | 80 void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction( |
81 float uplink_recoverable_packet_loss_fraction) {} | 81 float uplink_recoverable_packet_loss_fraction) {} |
82 | 82 |
83 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) { | 83 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) { |
84 OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>()); | 84 OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>()); |
85 } | 85 } |
86 | 86 |
87 void AudioEncoder::OnReceivedUplinkBandwidth( | 87 void AudioEncoder::OnReceivedUplinkBandwidth( |
88 int target_audio_bitrate_bps, | 88 int target_audio_bitrate_bps, |
89 rtc::Optional<int64_t> probing_interval_ms) {} | 89 rtc::Optional<int64_t> bwe_period_ms) {} |
90 | 90 |
91 void AudioEncoder::OnReceivedRtt(int rtt_ms) {} | 91 void AudioEncoder::OnReceivedRtt(int rtt_ms) {} |
92 | 92 |
93 void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {} | 93 void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {} |
94 | 94 |
95 void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms, | 95 void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms, |
96 int max_frame_length_ms) {} | 96 int max_frame_length_ms) {} |
97 | 97 |
98 } // namespace webrtc | 98 } // namespace webrtc |
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