1 /* 2 * Copyright 2019 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef API_TRANSPORT_RTP_RTP_SOURCE_H_ 12 #define API_TRANSPORT_RTP_RTP_SOURCE_H_ 13 14 #include <stdint.h> 15 16 #include "absl/types/optional.h" 17 #include "api/rtp_headers.h" 18 #include "rtc_base/checks.h" 19 20 namespace webrtc { 21 22 enum class RtpSourceType { 23 SSRC, 24 CSRC, 25 }; 26 27 class RtpSource { 28 public: 29 struct Extensions { 30 absl::optional<uint8_t> audio_level; 31 absl::optional<AbsoluteCaptureTime> absolute_capture_time; 32 }; 33 34 RtpSource() = delete; 35 36 // TODO(bugs.webrtc.org/10739): Remove this constructor once all clients 37 // migrate to the version with absolute capture time. RtpSource(int64_t timestamp_ms,uint32_t source_id,RtpSourceType source_type,absl::optional<uint8_t> audio_level,uint32_t rtp_timestamp)38 RtpSource(int64_t timestamp_ms, 39 uint32_t source_id, 40 RtpSourceType source_type, 41 absl::optional<uint8_t> audio_level, 42 uint32_t rtp_timestamp) 43 : RtpSource(timestamp_ms, 44 source_id, 45 source_type, 46 rtp_timestamp, 47 {audio_level, absl::nullopt}) {} 48 RtpSource(int64_t timestamp_ms,uint32_t source_id,RtpSourceType source_type,uint32_t rtp_timestamp,const RtpSource::Extensions & extensions)49 RtpSource(int64_t timestamp_ms, 50 uint32_t source_id, 51 RtpSourceType source_type, 52 uint32_t rtp_timestamp, 53 const RtpSource::Extensions& extensions) 54 : timestamp_ms_(timestamp_ms), 55 source_id_(source_id), 56 source_type_(source_type), 57 extensions_(extensions), 58 rtp_timestamp_(rtp_timestamp) {} 59 60 RtpSource(const RtpSource&) = default; 61 RtpSource& operator=(const RtpSource&) = default; 62 ~RtpSource() = default; 63 timestamp_ms()64 int64_t timestamp_ms() const { return timestamp_ms_; } update_timestamp_ms(int64_t timestamp_ms)65 void update_timestamp_ms(int64_t timestamp_ms) { 66 RTC_DCHECK_LE(timestamp_ms_, timestamp_ms); 67 timestamp_ms_ = timestamp_ms; 68 } 69 70 // The identifier of the source can be the CSRC or the SSRC. source_id()71 uint32_t source_id() const { return source_id_; } 72 73 // The source can be either a contributing source or a synchronization source. source_type()74 RtpSourceType source_type() const { return source_type_; } 75 audio_level()76 absl::optional<uint8_t> audio_level() const { 77 return extensions_.audio_level; 78 } 79 set_audio_level(const absl::optional<uint8_t> & level)80 void set_audio_level(const absl::optional<uint8_t>& level) { 81 extensions_.audio_level = level; 82 } 83 rtp_timestamp()84 uint32_t rtp_timestamp() const { return rtp_timestamp_; } 85 absolute_capture_time()86 absl::optional<AbsoluteCaptureTime> absolute_capture_time() const { 87 return extensions_.absolute_capture_time; 88 } 89 90 bool operator==(const RtpSource& o) const { 91 return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() && 92 source_type_ == o.source_type() && 93 extensions_.audio_level == o.extensions_.audio_level && 94 extensions_.absolute_capture_time == 95 o.extensions_.absolute_capture_time && 96 rtp_timestamp_ == o.rtp_timestamp(); 97 } 98 99 private: 100 int64_t timestamp_ms_; 101 uint32_t source_id_; 102 RtpSourceType source_type_; 103 RtpSource::Extensions extensions_; 104 uint32_t rtp_timestamp_; 105 }; 106 107 } // namespace webrtc 108 109 #endif // API_TRANSPORT_RTP_RTP_SOURCE_H_ 110