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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 |
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ | 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ |
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ | 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ |
13 | 13 |
14 #include <list> | 14 #include <list> |
15 #include <map> | 15 #include <map> |
16 #include <utility> | 16 #include <utility> |
17 #include <vector> | 17 #include <vector> |
18 | 18 |
| 19 #include "webrtc/base/random.h" |
19 #include "webrtc/base/thread_annotations.h" | 20 #include "webrtc/base/thread_annotations.h" |
20 #include "webrtc/common_types.h" | 21 #include "webrtc/common_types.h" |
21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
22 #include "webrtc/modules/rtp_rtcp/source/bitrate.h" | 23 #include "webrtc/modules/rtp_rtcp/source/bitrate.h" |
23 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" | 24 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
24 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" | 25 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" |
25 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" | 26 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" |
26 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 27 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
27 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" | 28 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" |
28 #include "webrtc/transport.h" | 29 #include "webrtc/transport.h" |
29 | 30 |
30 #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1. | |
31 | |
32 namespace webrtc { | 31 namespace webrtc { |
33 | 32 |
34 class BitrateAggregator; | 33 class BitrateAggregator; |
35 class CriticalSectionWrapper; | 34 class CriticalSectionWrapper; |
36 class RTPSenderAudio; | 35 class RTPSenderAudio; |
37 class RTPSenderVideo; | 36 class RTPSenderVideo; |
38 | 37 |
39 class RTPSenderInterface { | 38 class RTPSenderInterface { |
40 public: | 39 public: |
41 RTPSenderInterface() {} | 40 RTPSenderInterface() {} |
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380 | 379 |
381 void UpdateRtpStats(const uint8_t* buffer, | 380 void UpdateRtpStats(const uint8_t* buffer, |
382 size_t packet_length, | 381 size_t packet_length, |
383 const RTPHeader& header, | 382 const RTPHeader& header, |
384 bool is_rtx, | 383 bool is_rtx, |
385 bool is_retransmit); | 384 bool is_retransmit); |
386 bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const; | 385 bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const; |
387 | 386 |
388 Clock* clock_; | 387 Clock* clock_; |
389 int64_t clock_delta_ms_; | 388 int64_t clock_delta_ms_; |
| 389 Random random_ GUARDED_BY(send_critsect_); |
390 | 390 |
391 rtc::scoped_ptr<BitrateAggregator> bitrates_; | 391 rtc::scoped_ptr<BitrateAggregator> bitrates_; |
392 Bitrate total_bitrate_sent_; | 392 Bitrate total_bitrate_sent_; |
393 | 393 |
394 const bool audio_configured_; | 394 const bool audio_configured_; |
395 rtc::scoped_ptr<RTPSenderAudio> audio_; | 395 rtc::scoped_ptr<RTPSenderAudio> audio_; |
396 rtc::scoped_ptr<RTPSenderVideo> video_; | 396 rtc::scoped_ptr<RTPSenderVideo> video_; |
397 | 397 |
398 RtpPacketSender* const paced_sender_; | 398 RtpPacketSender* const paced_sender_; |
399 TransportSequenceNumberAllocator* const transport_sequence_number_allocator_; | 399 TransportSequenceNumberAllocator* const transport_sequence_number_allocator_; |
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462 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember | 462 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember |
463 // that by the time the function returns there is no guarantee | 463 // that by the time the function returns there is no guarantee |
464 // that the target bitrate is still valid. | 464 // that the target bitrate is still valid. |
465 rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_; | 465 rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_; |
466 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); | 466 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); |
467 }; | 467 }; |
468 | 468 |
469 } // namespace webrtc | 469 } // namespace webrtc |
470 | 470 |
471 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ | 471 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ |
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