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Side by Side Diff: webrtc/modules/audio_coding/neteq/include/neteq.h

Issue 1429943004: AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix log message Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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140 // method. 140 // method.
141 static NetEq* Create(const NetEq::Config& config); 141 static NetEq* Create(const NetEq::Config& config);
142 142
143 virtual ~NetEq() {} 143 virtual ~NetEq() {}
144 144
145 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication 145 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
146 // of the time when the packet was received, and should be measured with 146 // of the time when the packet was received, and should be measured with
147 // the same tick rate as the RTP timestamp of the current payload. 147 // the same tick rate as the RTP timestamp of the current payload.
148 // Returns 0 on success, -1 on failure. 148 // Returns 0 on success, -1 on failure.
149 virtual int InsertPacket(const WebRtcRTPHeader& rtp_header, 149 virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
150 const uint8_t* payload, 150 rtc::ArrayView<const uint8_t> payload,
151 size_t length_bytes,
152 uint32_t receive_timestamp) = 0; 151 uint32_t receive_timestamp) = 0;
153 152
154 // Inserts a sync-packet into packet queue. Sync-packets are decoded to 153 // Inserts a sync-packet into packet queue. Sync-packets are decoded to
155 // silence and are intended to keep AV-sync intact in an event of long packet 154 // silence and are intended to keep AV-sync intact in an event of long packet
156 // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq 155 // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
157 // might insert sync-packet when they observe that buffer level of NetEq is 156 // might insert sync-packet when they observe that buffer level of NetEq is
158 // decreasing below a certain threshold, defined by the application. 157 // decreasing below a certain threshold, defined by the application.
159 // Sync-packets should have the same payload type as the last audio payload 158 // Sync-packets should have the same payload type as the last audio payload
160 // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change 159 // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
161 // can be implied by inserting a sync-packet. 160 // can be implied by inserting a sync-packet.
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288 287
289 protected: 288 protected:
290 NetEq() {} 289 NetEq() {}
291 290
292 private: 291 private:
293 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq); 292 RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
294 }; 293 };
295 294
296 } // namespace webrtc 295 } // namespace webrtc
297 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_ 296 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
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