|  | /* | 
|  | *  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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ | 
|  | #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ | 
|  |  | 
|  | #include <stdio.h> | 
|  |  | 
|  | #include <memory> | 
|  | #include <string> | 
|  |  | 
|  | #include "webrtc/base/constructormagic.h" | 
|  | #include "webrtc/common_types.h" | 
|  | #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" | 
|  | #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class RtpHeaderParser; | 
|  |  | 
|  | 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(const std::string& file_name); | 
|  |  | 
|  | // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file. | 
|  | static bool ValidRtpDump(const std::string& file_name); | 
|  | static bool ValidPcap(const std::string& file_name); | 
|  |  | 
|  | virtual ~RtpFileSource(); | 
|  |  | 
|  | // 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; | 
|  |  | 
|  | RtpFileSource(); | 
|  |  | 
|  | bool OpenFile(const std::string& file_name); | 
|  |  | 
|  | std::unique_ptr<RtpFileReader> rtp_reader_; | 
|  | std::unique_ptr<RtpHeaderParser> parser_; | 
|  |  | 
|  | RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource); | 
|  | }; | 
|  |  | 
|  | }  // namespace test | 
|  | }  // namespace webrtc | 
|  | #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ |