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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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