1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ 13 14 #include <stdio.h> 15 #include <string> 16 17 #include "webrtc/base/constructormagic.h" 18 #include "webrtc/base/scoped_ptr.h" 19 #include "webrtc/common_types.h" 20 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" 21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 22 23 namespace webrtc { 24 25 class RtpHeaderParser; 26 27 namespace test { 28 29 class RtpFileReader; 30 31 class RtpFileSource : public PacketSource { 32 public: 33 // Creates an RtpFileSource reading from |file_name|. If the file cannot be 34 // opened, or has the wrong format, NULL will be returned. 35 static RtpFileSource* Create(const std::string& file_name); 36 37 // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file. 38 static bool ValidRtpDump(const std::string& file_name); 39 static bool ValidPcap(const std::string& file_name); 40 41 virtual ~RtpFileSource(); 42 43 // Registers an RTP header extension and binds it to |id|. 44 virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); 45 46 // Returns a pointer to the next packet. Returns NULL if end of file was 47 // reached, or if a the data was corrupt. 48 Packet* NextPacket() override; 49 50 private: 51 static const int kFirstLineLength = 40; 52 static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2; 53 static const size_t kPacketHeaderSize = 8; 54 55 RtpFileSource(); 56 57 bool OpenFile(const std::string& file_name); 58 59 rtc::scoped_ptr<RtpFileReader> rtp_reader_; 60 rtc::scoped_ptr<RtpHeaderParser> parser_; 61 62 RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource); 63 }; 64 65 } // namespace test 66 } // namespace webrtc 67 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ 68