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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.h

Issue 2089773002: Add EncodedImageCallback::OnEncodedImage(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: update SendOutgoingData() Created 4 years, 4 months 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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145 145
146 void SetMaxPayloadLength(size_t max_payload_length); 146 void SetMaxPayloadLength(size_t max_payload_length);
147 147
148 int32_t SendOutgoingData(FrameType frame_type, 148 int32_t SendOutgoingData(FrameType frame_type,
149 int8_t payload_type, 149 int8_t payload_type,
150 uint32_t timestamp, 150 uint32_t timestamp,
151 int64_t capture_time_ms, 151 int64_t capture_time_ms,
152 const uint8_t* payload_data, 152 const uint8_t* payload_data,
153 size_t payload_size, 153 size_t payload_size,
154 const RTPFragmentationHeader* fragmentation, 154 const RTPFragmentationHeader* fragmentation,
155 const RTPVideoHeader* rtp_hdr = NULL); 155 const RTPVideoHeader* rtp_hdr,
156 uint32_t* frame_id_out);
156 157
157 // RTP header extension 158 // RTP header extension
158 int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset); 159 int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset);
159 int32_t SetAbsoluteSendTime(uint32_t absolute_send_time); 160 int32_t SetAbsoluteSendTime(uint32_t absolute_send_time);
160 void SetVideoRotation(VideoRotation rotation); 161 void SetVideoRotation(VideoRotation rotation);
161 int32_t SetTransportSequenceNumber(uint16_t sequence_number); 162 int32_t SetTransportSequenceNumber(uint16_t sequence_number);
162 163
163 int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); 164 int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
164 bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override; 165 bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
165 int32_t DeregisterRtpHeaderExtension(RTPExtensionType type); 166 int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
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475 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_); 476 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
476 477
477 RateLimiter* const retransmission_rate_limiter_; 478 RateLimiter* const retransmission_rate_limiter_;
478 479
479 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender); 480 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
480 }; 481 };
481 482
482 } // namespace webrtc 483 } // namespace webrtc
483 484
484 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 485 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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