a837030f8f
And moved declaration into a new api directory, as api/transport/rtp/rtp_source.h. Bug: webrtc:8733 Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29039}
81 lines
2.3 KiB
C++
81 lines
2.3 KiB
C++
/*
|
|
* 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 "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
|
|
enum class RtpSourceType {
|
|
SSRC,
|
|
CSRC,
|
|
};
|
|
|
|
class RtpSource {
|
|
public:
|
|
RtpSource() = delete;
|
|
|
|
RtpSource(int64_t timestamp_ms,
|
|
uint32_t source_id,
|
|
RtpSourceType source_type,
|
|
absl::optional<uint8_t> audio_level,
|
|
uint32_t rtp_timestamp)
|
|
: timestamp_ms_(timestamp_ms),
|
|
source_id_(source_id),
|
|
source_type_(source_type),
|
|
audio_level_(audio_level),
|
|
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 audio_level_; }
|
|
void set_audio_level(const absl::optional<uint8_t>& level) {
|
|
audio_level_ = level;
|
|
}
|
|
|
|
uint32_t rtp_timestamp() const { return rtp_timestamp_; }
|
|
|
|
bool operator==(const RtpSource& o) const {
|
|
return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
|
|
source_type_ == o.source_type() && audio_level_ == o.audio_level_ &&
|
|
rtp_timestamp_ == o.rtp_timestamp();
|
|
}
|
|
|
|
private:
|
|
int64_t timestamp_ms_;
|
|
uint32_t source_id_;
|
|
RtpSourceType source_type_;
|
|
absl::optional<uint8_t> audio_level_;
|
|
uint32_t rtp_timestamp_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // API_TRANSPORT_RTP_RTP_SOURCE_H_
|