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/*
* 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 "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(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type,
uint32_t rtp_timestamp,
const RtpSource::Extensions& extensions)
: timestamp_ms_(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;
int64_t timestamp_ms() const { return timestamp_ms_; }
void update_timestamp_ms(int64_t timestamp_ms) {
RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
timestamp_ms_ = 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_ms_ == o.timestamp_ms() && 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:
int64_t timestamp_ms_;
uint32_t source_id_;
RtpSourceType source_type_;
RtpSource::Extensions extensions_;
uint32_t rtp_timestamp_;
};
} // namespace webrtc
#endif // API_TRANSPORT_RTP_RTP_SOURCE_H_
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