Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 46 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/syncable.h b/third_party/libwebrtc/call/syncable.h
new file mode 100644
index 0000000000..6817be9c55
--- /dev/null
+++ b/third_party/libwebrtc/call/syncable.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+// Syncable is used by RtpStreamsSynchronizer in VideoReceiveStreamInterface,
+// and implemented by AudioReceiveStreamInterface.
+
+#ifndef CALL_SYNCABLE_H_
+#define CALL_SYNCABLE_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+class Syncable {
+ public:
+  struct Info {
+    int64_t latest_receive_time_ms = 0;
+    uint32_t latest_received_capture_timestamp = 0;
+    uint32_t capture_time_ntp_secs = 0;
+    uint32_t capture_time_ntp_frac = 0;
+    uint32_t capture_time_source_clock = 0;
+    int current_delay_ms = 0;
+  };
+
+  virtual ~Syncable();
+
+  virtual uint32_t id() const = 0;
+  virtual absl::optional<Info> GetInfo() const = 0;
+  virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                                      int64_t* time_ms) const = 0;
+  virtual bool SetMinimumPlayoutDelay(int delay_ms) = 0;
+  virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+                                                 int64_t time_ms) = 0;
+};
+}  // namespace webrtc
+
+#endif  // CALL_SYNCABLE_H_