1 files changed, 111 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/transport/rtp/rtp_source.h b/third_party/libwebrtc/api/transport/rtp/rtp_source.h
new file mode 100644
index 0000000000..e51dcd70b6
--- /dev/null
+++ b/third_party/libwebrtc/api/transport/rtp/rtp_source.h
@@ -0,0 +1,111 @@
+/*
+ *  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_