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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 <map> | 14 #include <map> |
15 #include <memory> | 15 #include <memory> |
16 #include <utility> | 16 #include <utility> |
17 #include <vector> | 17 #include <vector> |
18 | 18 |
19 #include "webrtc/base/constructormagic.h" | 19 #include "webrtc/base/constructormagic.h" |
20 #include "webrtc/base/criticalsection.h" | 20 #include "webrtc/base/criticalsection.h" |
21 #include "webrtc/base/deprecation.h" | 21 #include "webrtc/base/deprecation.h" |
22 #include "webrtc/base/random.h" | 22 #include "webrtc/base/random.h" |
23 #include "webrtc/base/rate_statistics.h" | 23 #include "webrtc/base/rate_statistics.h" |
24 #include "webrtc/base/thread_annotations.h" | 24 #include "webrtc/base/thread_annotations.h" |
25 #include "webrtc/common_types.h" | 25 #include "webrtc/common_types.h" |
| 26 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h" |
26 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 27 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
27 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h" | 28 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h" |
28 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" | 29 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
29 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" | 30 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" |
30 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" | 31 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" |
31 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 32 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
32 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" | 33 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" |
33 #include "webrtc/transport.h" | 34 #include "webrtc/transport.h" |
34 | 35 |
35 namespace webrtc { | 36 namespace webrtc { |
36 | 37 |
37 class RateLimiter; | 38 class RateLimiter; |
38 class RtcEventLog; | 39 class RtcEventLog; |
39 class RtpPacketToSend; | 40 class RtpPacketToSend; |
40 class RTPSenderAudio; | 41 class RTPSenderAudio; |
41 class RTPSenderVideo; | 42 class RTPSenderVideo; |
42 | 43 |
43 class RTPSender { | 44 class RTPSender { |
44 public: | 45 public: |
45 RTPSender(bool audio, | 46 RTPSender(bool audio, |
46 Clock* clock, | 47 Clock* clock, |
47 Transport* transport, | 48 Transport* transport, |
48 RtpPacketSender* paced_sender, | 49 RtpPacketSender* paced_sender, |
| 50 // TODO(brandtr): Remove |flexfec_sender| when that is hooked up |
| 51 // to PacedSender instead. |
| 52 FlexfecSender* flexfec_sender, |
49 TransportSequenceNumberAllocator* sequence_number_allocator, | 53 TransportSequenceNumberAllocator* sequence_number_allocator, |
50 TransportFeedbackObserver* transport_feedback_callback, | 54 TransportFeedbackObserver* transport_feedback_callback, |
51 BitrateStatisticsObserver* bitrate_callback, | 55 BitrateStatisticsObserver* bitrate_callback, |
52 FrameCountObserver* frame_count_observer, | 56 FrameCountObserver* frame_count_observer, |
53 SendSideDelayObserver* send_side_delay_observer, | 57 SendSideDelayObserver* send_side_delay_observer, |
54 RtcEventLog* event_log, | 58 RtcEventLog* event_log, |
55 SendPacketObserver* send_packet_observer, | 59 SendPacketObserver* send_packet_observer, |
56 RateLimiter* nack_rate_limiter); | 60 RateLimiter* nack_rate_limiter); |
57 | 61 |
58 ~RTPSender(); | 62 ~RTPSender(); |
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318 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_); | 322 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_); |
319 | 323 |
320 RateLimiter* const retransmission_rate_limiter_; | 324 RateLimiter* const retransmission_rate_limiter_; |
321 | 325 |
322 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender); | 326 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender); |
323 }; | 327 }; |
324 | 328 |
325 } // namespace webrtc | 329 } // namespace webrtc |
326 | 330 |
327 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ | 331 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ |
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