| /* |
| * Copyright 2019 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef API_TRANSPORT_RTP_RTP_SOURCE_H_ |
| #define API_TRANSPORT_RTP_RTP_SOURCE_H_ |
| |
| #include <stdint.h> |
| |
| #include "absl/types/optional.h" |
| #include "api/rtp_headers.h" |
| #include "api/units/time_delta.h" |
| #include "api/units/timestamp.h" |
| #include "rtc_base/checks.h" |
| |
| namespace webrtc { |
| |
| enum class RtpSourceType { |
| SSRC, |
| CSRC, |
| }; |
| |
| class RtpSource { |
| public: |
| struct Extensions { |
| absl::optional<uint8_t> audio_level; |
| |
| // Fields from the Absolute Capture Time header extension: |
| // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time |
| absl::optional<AbsoluteCaptureTime> absolute_capture_time; |
| |
| // Clock offset between the local clock and the capturer's clock. |
| // Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset` |
| // which instead represents the clock offset between a remote sender and the |
| // capturer. The following holds: |
| // Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset |
| absl::optional<TimeDelta> local_capture_clock_offset; |
| }; |
| |
| RtpSource() = delete; |
| |
| RtpSource(Timestamp timestamp, |
| uint32_t source_id, |
| RtpSourceType source_type, |
| uint32_t rtp_timestamp, |
| const RtpSource::Extensions& extensions) |
| : timestamp_(timestamp), |
| source_id_(source_id), |
| source_type_(source_type), |
| extensions_(extensions), |
| rtp_timestamp_(rtp_timestamp) {} |
| |
| // TODO(bugs.webrtc.org/13757): Remove after 2023-09-18 |
| [[deprecated]] RtpSource(int64_t timestamp_ms, |
| uint32_t source_id, |
| RtpSourceType source_type, |
| uint32_t rtp_timestamp, |
| const RtpSource::Extensions& extensions) |
| : timestamp_(Timestamp::Millis(timestamp_ms)), |
| source_id_(source_id), |
| source_type_(source_type), |
| extensions_(extensions), |
| rtp_timestamp_(rtp_timestamp) {} |
| |
| RtpSource(const RtpSource&) = default; |
| RtpSource& operator=(const RtpSource&) = default; |
| ~RtpSource() = default; |
| |
| Timestamp timestamp() const { return timestamp_; } |
| |
| // TODO(bugs.webrtc.org/13757): Remove after 2023-09-18 |
| [[deprecated]] int64_t timestamp_ms() const { return timestamp_.ms(); } |
| [[deprecated]] void update_timestamp_ms(int64_t timestamp_ms) { |
| RTC_DCHECK_LE(timestamp_.ms(), timestamp_ms); |
| timestamp_ = Timestamp::Millis(timestamp_ms); |
| } |
| |
| // The identifier of the source can be the CSRC or the SSRC. |
| uint32_t source_id() const { return source_id_; } |
| |
| // The source can be either a contributing source or a synchronization source. |
| RtpSourceType source_type() const { return source_type_; } |
| |
| absl::optional<uint8_t> audio_level() const { |
| return extensions_.audio_level; |
| } |
| |
| void set_audio_level(const absl::optional<uint8_t>& level) { |
| extensions_.audio_level = level; |
| } |
| |
| uint32_t rtp_timestamp() const { return rtp_timestamp_; } |
| |
| absl::optional<AbsoluteCaptureTime> absolute_capture_time() const { |
| return extensions_.absolute_capture_time; |
| } |
| |
| absl::optional<TimeDelta> local_capture_clock_offset() const { |
| return extensions_.local_capture_clock_offset; |
| } |
| |
| bool operator==(const RtpSource& o) const { |
| return timestamp_ == o.timestamp() && source_id_ == o.source_id() && |
| source_type_ == o.source_type() && |
| extensions_.audio_level == o.extensions_.audio_level && |
| extensions_.absolute_capture_time == |
| o.extensions_.absolute_capture_time && |
| rtp_timestamp_ == o.rtp_timestamp(); |
| } |
| |
| private: |
| Timestamp timestamp_; |
| uint32_t source_id_; |
| RtpSourceType source_type_; |
| RtpSource::Extensions extensions_; |
| uint32_t rtp_timestamp_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_TRANSPORT_RTP_RTP_SOURCE_H_ |