Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

1 files changed, 51 insertions, 0 deletions

diff --git a/third_party/libwebrtc/test/rtp_file_reader.h b/third_party/libwebrtc/test/rtp_file_reader.h
new file mode 100644
index 0000000000..4e26c71baa
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_file_reader.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2014 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 TEST_RTP_FILE_READER_H_
+#define TEST_RTP_FILE_READER_H_
+
+#include <set>
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+struct RtpPacket {
+  // Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
+  // some overhead.
+  static const size_t kMaxPacketBufferSize = 3500;
+  uint8_t data[kMaxPacketBufferSize];
+  size_t length;
+  // The length the packet had on wire. Will be different from `length` when
+  // reading a header-only RTP dump.
+  size_t original_length;
+
+  uint32_t time_ms;
+};
+
+class RtpFileReader {
+ public:
+  enum FileFormat { kPcap, kRtpDump, kLengthPacketInterleaved };
+
+  virtual ~RtpFileReader() {}
+  static RtpFileReader* Create(FileFormat format,
+                               const uint8_t* data,
+                               size_t size,
+                               const std::set<uint32_t>& ssrc_filter);
+  static RtpFileReader* Create(FileFormat format, absl::string_view filename);
+  static RtpFileReader* Create(FileFormat format,
+                               absl::string_view filename,
+                               const std::set<uint32_t>& ssrc_filter);
+  virtual bool NextPacket(RtpPacket* packet) = 0;
+};
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_RTP_FILE_READER_H_