110 lines
3.2 KiB
C
110 lines
3.2 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 "api/rtp_headers.h"
|
||
|
|
#include "rtc_base/checks.h"
|
||
|
|
|
||
|
|
namespace webrtc {
|
||
|
|
|
||
|
|
enum class RtpSourceType {
|
||
|
|
SSRC,
|
||
|
|
CSRC,
|
||
|
|
};
|
||
|
|
|
||
|
|
class RtpSource {
|
||
|
|
public:
|
||
|
|
struct Extensions {
|
||
|
|
absl::optional<uint8_t> audio_level;
|
||
|
|
absl::optional<AbsoluteCaptureTime> absolute_capture_time;
|
||
|
|
};
|
||
|
|
|
||
|
|
RtpSource() = delete;
|
||
|
|
|
||
|
|
// TODO(bugs.webrtc.org/10739): Remove this constructor once all clients
|
||
|
|
// migrate to the version with absolute capture time.
|
||
|
|
RtpSource(int64_t timestamp_ms,
|
||
|
|
uint32_t source_id,
|
||
|
|
RtpSourceType source_type,
|
||
|
|
absl::optional<uint8_t> audio_level,
|
||
|
|
uint32_t rtp_timestamp)
|
||
|
|
: RtpSource(timestamp_ms,
|
||
|
|
source_id,
|
||
|
|
source_type,
|
||
|
|
rtp_timestamp,
|
||
|
|
{audio_level, absl::nullopt}) {}
|
||
|
|
|
||
|
|
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;
|
||
|
|
}
|
||
|
|
|
||
|
|
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_
|