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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 <memory> |
16 #include <utility> | 17 #include <utility> |
17 #include <vector> | 18 #include <vector> |
18 | 19 |
19 #include "webrtc/base/constructormagic.h" | 20 #include "webrtc/base/constructormagic.h" |
20 #include "webrtc/base/criticalsection.h" | 21 #include "webrtc/base/criticalsection.h" |
21 #include "webrtc/base/random.h" | 22 #include "webrtc/base/random.h" |
22 #include "webrtc/base/thread_annotations.h" | 23 #include "webrtc/base/thread_annotations.h" |
23 #include "webrtc/common_types.h" | 24 #include "webrtc/common_types.h" |
24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
25 #include "webrtc/modules/rtp_rtcp/source/bitrate.h" | 26 #include "webrtc/modules/rtp_rtcp/source/bitrate.h" |
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416 }; | 417 }; |
417 | 418 |
418 Clock* const clock_; | 419 Clock* const clock_; |
419 const int64_t clock_delta_ms_; | 420 const int64_t clock_delta_ms_; |
420 Random random_ GUARDED_BY(send_critsect_); | 421 Random random_ GUARDED_BY(send_critsect_); |
421 | 422 |
422 BitrateAggregator bitrates_; | 423 BitrateAggregator bitrates_; |
423 Bitrate total_bitrate_sent_; | 424 Bitrate total_bitrate_sent_; |
424 | 425 |
425 const bool audio_configured_; | 426 const bool audio_configured_; |
426 const rtc::scoped_ptr<RTPSenderAudio> audio_; | 427 const std::unique_ptr<RTPSenderAudio> audio_; |
427 const rtc::scoped_ptr<RTPSenderVideo> video_; | 428 const std::unique_ptr<RTPSenderVideo> video_; |
428 | 429 |
429 RtpPacketSender* const paced_sender_; | 430 RtpPacketSender* const paced_sender_; |
430 TransportSequenceNumberAllocator* const transport_sequence_number_allocator_; | 431 TransportSequenceNumberAllocator* const transport_sequence_number_allocator_; |
431 TransportFeedbackObserver* const transport_feedback_observer_; | 432 TransportFeedbackObserver* const transport_feedback_observer_; |
432 int64_t last_capture_time_ms_sent_; | 433 int64_t last_capture_time_ms_sent_; |
433 rtc::CriticalSection send_critsect_; | 434 rtc::CriticalSection send_critsect_; |
434 | 435 |
435 Transport *transport_; | 436 Transport *transport_; |
436 bool sending_media_ GUARDED_BY(send_critsect_); | 437 bool sending_media_ GUARDED_BY(send_critsect_); |
437 | 438 |
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492 // that the target bitrate is still valid. | 493 // that the target bitrate is still valid. |
493 rtc::CriticalSection target_bitrate_critsect_; | 494 rtc::CriticalSection target_bitrate_critsect_; |
494 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); | 495 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); |
495 | 496 |
496 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender); | 497 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender); |
497 }; | 498 }; |
498 | 499 |
499 } // namespace webrtc | 500 } // namespace webrtc |
500 | 501 |
501 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ | 502 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ |
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