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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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154 | 154 |
155 void ApplyAudioNetworkAdaptor(); | 155 void ApplyAudioNetworkAdaptor(); |
156 std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator( | 156 std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator( |
157 const std::string& config_string, | 157 const std::string& config_string, |
158 RtcEventLog* event_log, | 158 RtcEventLog* event_log, |
159 const Clock* clock) const; | 159 const Clock* clock) const; |
160 | 160 |
161 void MaybeUpdateUplinkBandwidth(); | 161 void MaybeUpdateUplinkBandwidth(); |
162 | 162 |
163 Config config_; | 163 Config config_; |
| 164 const bool send_side_bwe_with_overhead_; |
164 float packet_loss_rate_; | 165 float packet_loss_rate_; |
165 std::vector<int16_t> input_buffer_; | 166 std::vector<int16_t> input_buffer_; |
166 OpusEncInst* inst_; | 167 OpusEncInst* inst_; |
167 uint32_t first_timestamp_in_buffer_; | 168 uint32_t first_timestamp_in_buffer_; |
168 size_t num_channels_to_encode_; | 169 size_t num_channels_to_encode_; |
169 int next_frame_length_ms_; | 170 int next_frame_length_ms_; |
170 int complexity_; | 171 int complexity_; |
171 std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_; | 172 std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_; |
172 AudioNetworkAdaptorCreator audio_network_adaptor_creator_; | 173 AudioNetworkAdaptorCreator audio_network_adaptor_creator_; |
173 std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_; | 174 std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_; |
174 rtc::Optional<size_t> overhead_bytes_per_packet_; | 175 rtc::Optional<size_t> overhead_bytes_per_packet_; |
175 const std::unique_ptr<SmoothingFilter> bitrate_smoother_; | 176 const std::unique_ptr<SmoothingFilter> bitrate_smoother_; |
176 rtc::Optional<int64_t> bitrate_smoother_last_update_time_; | 177 rtc::Optional<int64_t> bitrate_smoother_last_update_time_; |
177 | 178 |
178 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); | 179 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); |
179 }; | 180 }; |
180 | 181 |
181 } // namespace webrtc | 182 } // namespace webrtc |
182 | 183 |
183 #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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