Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(830)

Side by Side Diff: webrtc/modules/audio_coding/codecs/audio_encoder.h

Issue 2362703002: Adding audio network adaptor to AudioEncoderOpus. (Closed)
Patch Set: Created 4 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 144 matching lines...) Expand 10 before | Expand all | Expand 10 after
155 155
156 // Causes this encoder to let go of any other encoders it contains, and 156 // Causes this encoder to let go of any other encoders it contains, and
157 // returns a pointer to an array where they are stored (which is required to 157 // returns a pointer to an array where they are stored (which is required to
158 // live as long as this encoder). Unless the returned array is empty, you may 158 // live as long as this encoder). Unless the returned array is empty, you may
159 // not call any methods on this encoder afterwards, except for the 159 // not call any methods on this encoder afterwards, except for the
160 // destructor. The default implementation just returns an empty array. 160 // destructor. The default implementation just returns an empty array.
161 // NOTE: This method is subject to change. Do not call or override it. 161 // NOTE: This method is subject to change. Do not call or override it.
162 virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>> 162 virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
163 ReclaimContainedEncoders(); 163 ReclaimContainedEncoders();
164 164
165 // Provides uplink bandwidth to this encoder to allow it to adapt.
166 virtual void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps);
167
168 // Provides uplink packet loss fraction to this encoder to allow it to adapt.
169 virtual void OnReceivedUplinkPacketLossFraction(
170 float uplink_packet_loss_fraction);
171
172 // Provides target audio bitrate to this encoder to allow it to adapt.
173 virtual void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps);
174
175 // Provides RTT to this encoder to allow it to adapt.
176 virtual void OnReceivedRtt(int rtt_ms);
177
165 protected: 178 protected:
166 // Subclasses implement this to perform the actual encoding. Called by 179 // Subclasses implement this to perform the actual encoding. Called by
167 // Encode(). 180 // Encode().
168 virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp, 181 virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
169 rtc::ArrayView<const int16_t> audio, 182 rtc::ArrayView<const int16_t> audio,
170 rtc::Buffer* encoded) = 0; 183 rtc::Buffer* encoded) = 0;
171 }; 184 };
172 } // namespace webrtc 185 } // namespace webrtc
173 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_ 186 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698