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https://github.com/mollyim/webrtc.git
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f53597140f
Bug: webrtc:13757 Change-Id: I5d7da9c9aee489e4b57d361de174c59713cb2b14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317780 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40650}
129 lines
4.1 KiB
C++
129 lines
4.1 KiB
C++
/*
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* Copyright 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_TRANSPORT_RTP_RTP_SOURCE_H_
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#define API_TRANSPORT_RTP_RTP_SOURCE_H_
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "api/rtp_headers.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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enum class RtpSourceType {
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SSRC,
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CSRC,
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};
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class RtpSource {
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public:
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struct Extensions {
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absl::optional<uint8_t> audio_level;
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// Fields from the Absolute Capture Time header extension:
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// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
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absl::optional<AbsoluteCaptureTime> absolute_capture_time;
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// Clock offset between the local clock and the capturer's clock.
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// Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset`
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// which instead represents the clock offset between a remote sender and the
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// capturer. The following holds:
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// Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset
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absl::optional<TimeDelta> local_capture_clock_offset;
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};
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RtpSource() = delete;
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RtpSource(Timestamp timestamp,
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uint32_t source_id,
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RtpSourceType source_type,
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uint32_t rtp_timestamp,
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const RtpSource::Extensions& extensions)
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: timestamp_(timestamp),
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source_id_(source_id),
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source_type_(source_type),
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extensions_(extensions),
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rtp_timestamp_(rtp_timestamp) {}
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// TODO(bugs.webrtc.org/13757): deprecate when chromium stop using this
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// and remove after 2023-09-18
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RtpSource(int64_t timestamp_ms,
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uint32_t source_id,
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RtpSourceType source_type,
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uint32_t rtp_timestamp,
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const RtpSource::Extensions& extensions)
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: timestamp_(Timestamp::Millis(timestamp_ms)),
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source_id_(source_id),
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source_type_(source_type),
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extensions_(extensions),
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rtp_timestamp_(rtp_timestamp) {}
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RtpSource(const RtpSource&) = default;
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RtpSource& operator=(const RtpSource&) = default;
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~RtpSource() = default;
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Timestamp timestamp() const { return timestamp_; }
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// TODO(bugs.webrtc.org/13757): deprecate when chromium stop using this
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// and remove after 2023-09-18
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int64_t timestamp_ms() const { return timestamp_.ms(); }
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[[deprecated]] void update_timestamp_ms(int64_t timestamp_ms) {
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RTC_DCHECK_LE(timestamp_.ms(), timestamp_ms);
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timestamp_ = Timestamp::Millis(timestamp_ms);
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}
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// The identifier of the source can be the CSRC or the SSRC.
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uint32_t source_id() const { return source_id_; }
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// The source can be either a contributing source or a synchronization source.
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RtpSourceType source_type() const { return source_type_; }
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absl::optional<uint8_t> audio_level() const {
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return extensions_.audio_level;
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}
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void set_audio_level(const absl::optional<uint8_t>& level) {
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extensions_.audio_level = level;
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}
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uint32_t rtp_timestamp() const { return rtp_timestamp_; }
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absl::optional<AbsoluteCaptureTime> absolute_capture_time() const {
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return extensions_.absolute_capture_time;
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}
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absl::optional<TimeDelta> local_capture_clock_offset() const {
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return extensions_.local_capture_clock_offset;
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}
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bool operator==(const RtpSource& o) const {
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return timestamp_ == o.timestamp() && source_id_ == o.source_id() &&
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source_type_ == o.source_type() &&
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extensions_.audio_level == o.extensions_.audio_level &&
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extensions_.absolute_capture_time ==
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o.extensions_.absolute_capture_time &&
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rtp_timestamp_ == o.rtp_timestamp();
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}
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private:
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Timestamp timestamp_;
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uint32_t source_id_;
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RtpSourceType source_type_;
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RtpSource::Extensions extensions_;
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uint32_t rtp_timestamp_;
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};
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} // namespace webrtc
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#endif // API_TRANSPORT_RTP_RTP_SOURCE_H_
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