/*
* 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 <optional>
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
namespace webrtc {
enum class RtpSourceType {
SSRC,
CSRC,
};
class RtpSource {
public:
struct Extensions {
std::optional<uint8_t> audio_level;
// Fields from the Absolute Capture Time header extension:
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
std::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
std::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) {}
RtpSource(
const RtpSource&) =
default;
RtpSource&
operator=(
const RtpSource&) =
default;
~RtpSource() =
default;
Timestamp timestamp()
const {
return timestamp_; }
// 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_; }
std::optional<uint8_t> audio_level()
const {
return extensions_.audio_level; }
void set_audio_level(
const std::optional<uint8_t>& level) {
extensions_.audio_level = level;
}
uint32_t rtp_timestamp()
const {
return rtp_timestamp_; }
std::optional<AbsoluteCaptureTime> absolute_capture_time()
const {
return extensions_.absolute_capture_time;
}
std::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_
