Repo created

TMessagesProj/jni/voip/webrtc/pc/rtp_receiver.cc

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+/*
+ *  Copyright 2015 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.
+ */
+
+#include "pc/rtp_receiver.h"
+
+#include <stddef.h>
+
+#include <atomic>
+#include <utility>
+#include <vector>
+
+#include "pc/media_stream.h"
+#include "pc/media_stream_proxy.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+// This function is only expected to be called on the signalling thread.
+// On the other hand, some test or even production setups may use
+// several signaling threads.
+int RtpReceiverInternal::GenerateUniqueId() {
+  static std::atomic<int> g_unique_id{0};
+
+  return ++g_unique_id;
+}
+
+std::vector<rtc::scoped_refptr<MediaStreamInterface>>
+RtpReceiverInternal::CreateStreamsFromIds(std::vector<std::string> stream_ids) {
+  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams(
+      stream_ids.size());
+  for (size_t i = 0; i < stream_ids.size(); ++i) {
+    streams[i] = MediaStreamProxy::Create(
+        rtc::Thread::Current(), MediaStream::Create(std::move(stream_ids[i])));
+  }
+  return streams;
+}
+
+}  // namespace webrtc