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

Issue 2303283002: Introduce helpers to RtpSender to propagate RtpPacketToSend. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: nit Created 4 years, 3 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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196 196
197 // RTX. 197 // RTX.
198 void SetRtxStatus(int mode); 198 void SetRtxStatus(int mode);
199 int RtxStatus() const; 199 int RtxStatus() const;
200 200
201 uint32_t RtxSsrc() const; 201 uint32_t RtxSsrc() const;
202 void SetRtxSsrc(uint32_t ssrc); 202 void SetRtxSsrc(uint32_t ssrc);
203 203
204 void SetRtxPayloadType(int payload_type, int associated_payload_type); 204 void SetRtxPayloadType(int payload_type, int associated_payload_type);
205 205
206 // Create empty packet, fills ssrc, csrcs and reserve place for header
207 // extensions RtpSender updates before sending.
208 std::unique_ptr<RtpPacketToSend> AllocatePacket() const;
209 // Allocate sequence number for provided packet.
210 // Save packet's fields to generate padding that doesn't break media stream.
211 // Return false if sending was turned off.
212 bool AssignSequenceNumber(RtpPacketToSend* packet);
213
206 // Functions wrapping RTPSenderInterface. 214 // Functions wrapping RTPSenderInterface.
207 int32_t BuildRTPheader(uint8_t* data_buffer, 215 int32_t BuildRTPheader(uint8_t* data_buffer,
208 int8_t payload_type, 216 int8_t payload_type,
209 bool marker_bit, 217 bool marker_bit,
210 uint32_t capture_timestamp, 218 uint32_t capture_timestamp,
211 int64_t capture_time_ms, 219 int64_t capture_time_ms,
212 bool timestamp_provided = true, 220 bool timestamp_provided = true,
213 bool inc_sequence_number = true); 221 bool inc_sequence_number = true);
214 int32_t BuildRtpHeader(uint8_t* data_buffer, 222 int32_t BuildRtpHeader(uint8_t* data_buffer,
215 int8_t payload_type, 223 int8_t payload_type,
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419 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_); 427 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
420 428
421 RateLimiter* const retransmission_rate_limiter_; 429 RateLimiter* const retransmission_rate_limiter_;
422 430
423 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender); 431 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
424 }; 432 };
425 433
426 } // namespace webrtc 434 } // namespace webrtc
427 435
428 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 436 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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