1 files changed, 133 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/test/RTPFile.h b/third_party/libwebrtc/modules/audio_coding/test/RTPFile.h
new file mode 100644
index 0000000000..b796491da9
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+++ b/third_party/libwebrtc/modules/audio_coding/test/RTPFile.h
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2012 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_TEST_RTPFILE_H_
+#define MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+
+#include <stdio.h>
+
+#include <queue>
+
+#include "absl/strings/string_view.h"
+#include "api/rtp_headers.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class RTPStream {
+ public:
+  virtual ~RTPStream() {}
+
+  virtual void Write(uint8_t payloadType,
+                     uint32_t timeStamp,
+                     int16_t seqNo,
+                     const uint8_t* payloadData,
+                     size_t payloadSize,
+                     uint32_t frequency) = 0;
+
+  // Returns the packet's payload size. Zero should be treated as an
+  // end-of-stream (in the case that EndOfFile() is true) or an error.
+  virtual size_t Read(RTPHeader* rtp_Header,
+                      uint8_t* payloadData,
+                      size_t payloadSize,
+                      uint32_t* offset) = 0;
+  virtual bool EndOfFile() const = 0;
+
+ protected:
+  void MakeRTPheader(uint8_t* rtpHeader,
+                     uint8_t payloadType,
+                     int16_t seqNo,
+                     uint32_t timeStamp,
+                     uint32_t ssrc);
+
+  void ParseRTPHeader(RTPHeader* rtp_header, const uint8_t* rtpHeader);
+};
+
+class RTPPacket {
+ public:
+  RTPPacket(uint8_t payloadType,
+            uint32_t timeStamp,
+            int16_t seqNo,
+            const uint8_t* payloadData,
+            size_t payloadSize,
+            uint32_t frequency);
+
+  ~RTPPacket();
+
+  uint8_t payloadType;
+  uint32_t timeStamp;
+  int16_t seqNo;
+  uint8_t* payloadData;
+  size_t payloadSize;
+  uint32_t frequency;
+};
+
+class RTPBuffer : public RTPStream {
+ public:
+  RTPBuffer() = default;
+
+  ~RTPBuffer() = default;
+
+  void Write(uint8_t payloadType,
+             uint32_t timeStamp,
+             int16_t seqNo,
+             const uint8_t* payloadData,
+             size_t payloadSize,
+             uint32_t frequency) override;
+
+  size_t Read(RTPHeader* rtp_header,
+              uint8_t* payloadData,
+              size_t payloadSize,
+              uint32_t* offset) override;
+
+  bool EndOfFile() const override;
+
+ private:
+  mutable Mutex mutex_;
+  std::queue<RTPPacket*> _rtpQueue RTC_GUARDED_BY(&mutex_);
+};
+
+class RTPFile : public RTPStream {
+ public:
+  ~RTPFile() {}
+
+  RTPFile() : _rtpFile(NULL), _rtpEOF(false) {}
+
+  void Open(absl::string_view outFilename, absl::string_view mode);
+
+  void Close();
+
+  void WriteHeader();
+
+  void ReadHeader();
+
+  void Write(uint8_t payloadType,
+             uint32_t timeStamp,
+             int16_t seqNo,
+             const uint8_t* payloadData,
+             size_t payloadSize,
+             uint32_t frequency) override;
+
+  size_t Read(RTPHeader* rtp_header,
+              uint8_t* payloadData,
+              size_t payloadSize,
+              uint32_t* offset) override;
+
+  bool EndOfFile() const override { return _rtpEOF; }
+
+ private:
+  FILE* _rtpFile;
+  bool _rtpEOF;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_RTPFILE_H_