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Side by Side Diff: webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h

Issue 1427923003: Re-enable PCAP reading in neteq_rtpplay (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 16 matching lines...) Expand all
27 namespace test { 27 namespace test {
28 28
29 class RtpFileReader; 29 class RtpFileReader;
30 30
31 class RtpFileSource : public PacketSource { 31 class RtpFileSource : public PacketSource {
32 public: 32 public:
33 // Creates an RtpFileSource reading from |file_name|. If the file cannot be 33 // Creates an RtpFileSource reading from |file_name|. If the file cannot be
34 // opened, or has the wrong format, NULL will be returned. 34 // opened, or has the wrong format, NULL will be returned.
35 static RtpFileSource* Create(const std::string& file_name); 35 static RtpFileSource* Create(const std::string& file_name);
36 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
37 virtual ~RtpFileSource(); 41 virtual ~RtpFileSource();
38 42
39 // Registers an RTP header extension and binds it to |id|. 43 // Registers an RTP header extension and binds it to |id|.
40 virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); 44 virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
41 45
42 // Returns a pointer to the next packet. Returns NULL if end of file was 46 // Returns a pointer to the next packet. Returns NULL if end of file was
43 // reached, or if a the data was corrupt. 47 // reached, or if a the data was corrupt.
44 Packet* NextPacket() override; 48 Packet* NextPacket() override;
45 49
46 private: 50 private:
47 static const int kFirstLineLength = 40; 51 static const int kFirstLineLength = 40;
48 static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2; 52 static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
49 static const size_t kPacketHeaderSize = 8; 53 static const size_t kPacketHeaderSize = 8;
50 54
51 RtpFileSource(); 55 RtpFileSource();
52 56
53 bool OpenFile(const std::string& file_name); 57 bool OpenFile(const std::string& file_name);
54 58
55 rtc::scoped_ptr<RtpFileReader> rtp_reader_; 59 rtc::scoped_ptr<RtpFileReader> rtp_reader_;
56 rtc::scoped_ptr<RtpHeaderParser> parser_; 60 rtc::scoped_ptr<RtpHeaderParser> parser_;
57 61
58 RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource); 62 RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
59 }; 63 };
60 64
61 } // namespace test 65 } // namespace test
62 } // namespace webrtc 66 } // namespace webrtc
63 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_ 67 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
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