1 files changed, 68 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
new file mode 100644
index 0000000000..55505be630
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -0,0 +1,68 @@
+/*
+ *  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 MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+namespace test {
+
+class RtpFileReader;
+
+class RtpFileSource : public PacketSource {
+ public:
+  // Creates an RtpFileSource reading from `file_name`. If the file cannot be
+  // opened, or has the wrong format, NULL will be returned.
+  static RtpFileSource* Create(
+      absl::string_view file_name,
+      absl::optional<uint32_t> ssrc_filter = absl::nullopt);
+
+  // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
+  static bool ValidRtpDump(absl::string_view file_name);
+  static bool ValidPcap(absl::string_view file_name);
+
+  ~RtpFileSource() override;
+
+  RtpFileSource(const RtpFileSource&) = delete;
+  RtpFileSource& operator=(const RtpFileSource&) = delete;
+
+  // Registers an RTP header extension and binds it to `id`.
+  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
+
+  std::unique_ptr<Packet> NextPacket() override;
+
+ private:
+  static const int kFirstLineLength = 40;
+  static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
+  static const size_t kPacketHeaderSize = 8;
+
+  explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter);
+
+  bool OpenFile(absl::string_view file_name);
+
+  std::unique_ptr<RtpFileReader> rtp_reader_;
+  const absl::optional<uint32_t> ssrc_filter_;
+  RtpHeaderExtensionMap rtp_header_extension_map_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_