1 /* 2 * Copyright (c) 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 MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_ 12 #define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_ 13 14 #include "api/array_view.h" 15 #include "api/rtp_headers.h" 16 #include "api/units/time_delta.h" 17 #include "api/units/timestamp.h" 18 #include "rtc_base/synchronization/mutex.h" 19 #include "rtc_base/thread_annotations.h" 20 #include "system_wrappers/include/clock.h" 21 22 namespace webrtc { 23 24 // 25 // Helper class for sending the |AbsoluteCaptureTime| header extension. 26 // 27 // Supports the "timestamp interpolation" optimization: 28 // A sender SHOULD save bandwidth by not sending abs-capture-time with every 29 // RTP packet. It SHOULD still send them at regular intervals (e.g. every 30 // second) to help mitigate the impact of clock drift and packet loss. Mixers 31 // SHOULD always send abs-capture-time with the first RTP packet after 32 // changing capture system. 33 // 34 // Timestamp interpolation works fine as long as there’s reasonably low 35 // NTP/RTP clock drift. This is not always true. Senders that detect “jumps” 36 // between its NTP and RTP clock mappings SHOULD send abs-capture-time with 37 // the first RTP packet after such a thing happening. 38 // 39 // See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/ 40 // 41 class AbsoluteCaptureTimeSender { 42 public: 43 static constexpr TimeDelta kInterpolationMaxInterval = 44 TimeDelta::Millis(1000); 45 static constexpr TimeDelta kInterpolationMaxError = TimeDelta::Millis(1); 46 47 explicit AbsoluteCaptureTimeSender(Clock* clock); 48 49 // Returns the source (i.e. SSRC or CSRC) of the capture system. 50 static uint32_t GetSource(uint32_t ssrc, 51 rtc::ArrayView<const uint32_t> csrcs); 52 53 // Returns a header extension to be sent, or |absl::nullopt| if the header 54 // extension shouldn't be sent. 55 absl::optional<AbsoluteCaptureTime> OnSendPacket( 56 uint32_t source, 57 uint32_t rtp_timestamp, 58 uint32_t rtp_clock_frequency, 59 uint64_t absolute_capture_timestamp, 60 absl::optional<int64_t> estimated_capture_clock_offset); 61 62 private: 63 bool ShouldSendExtension( 64 Timestamp send_time, 65 uint32_t source, 66 uint32_t rtp_timestamp, 67 uint32_t rtp_clock_frequency, 68 uint64_t absolute_capture_timestamp, 69 absl::optional<int64_t> estimated_capture_clock_offset) const 70 RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); 71 72 Clock* const clock_; 73 74 Mutex mutex_; 75 76 Timestamp last_send_time_ RTC_GUARDED_BY(mutex_); 77 78 uint32_t last_source_ RTC_GUARDED_BY(mutex_); 79 uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_); 80 uint32_t last_rtp_clock_frequency_ RTC_GUARDED_BY(mutex_); 81 uint64_t last_absolute_capture_timestamp_ RTC_GUARDED_BY(mutex_); 82 absl::optional<int64_t> last_estimated_capture_clock_offset_ 83 RTC_GUARDED_BY(mutex_); 84 }; // AbsoluteCaptureTimeSender 85 86 } // namespace webrtc 87 88 #endif // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_ 89