Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(113)

Side by Side Diff: webrtc/test/rtp_file_reader.h

Issue 1235433005: Add pcap support to bwe tools. Allow filtering on SSRCs. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 5 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 #ifndef WEBRTC_TEST_RTP_FILE_READER_H_ 10 #ifndef WEBRTC_TEST_RTP_FILE_READER_H_
11 #define WEBRTC_TEST_RTP_FILE_READER_H_ 11 #define WEBRTC_TEST_RTP_FILE_READER_H_
12 12
13 #include <set>
13 #include <string> 14 #include <string>
14 15
15 #include "webrtc/common_types.h" 16 #include "webrtc/common_types.h"
16 17
17 namespace webrtc { 18 namespace webrtc {
18 namespace test { 19 namespace test {
19 20
20 struct RtpPacket { 21 struct RtpPacket {
21 // Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus 22 // Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
22 // some overhead. 23 // some overhead.
23 static const size_t kMaxPacketBufferSize = 3500; 24 static const size_t kMaxPacketBufferSize = 3500;
24 uint8_t data[kMaxPacketBufferSize]; 25 uint8_t data[kMaxPacketBufferSize];
25 size_t length; 26 size_t length;
26 // The length the packet had on wire. Will be different from |length| when 27 // The length the packet had on wire. Will be different from |length| when
27 // reading a header-only RTP dump. 28 // reading a header-only RTP dump.
28 size_t original_length; 29 size_t original_length;
29 30
30 uint32_t time_ms; 31 uint32_t time_ms;
31 }; 32 };
32 33
33 class RtpFileReader { 34 class RtpFileReader {
34 public: 35 public:
35 enum FileFormat { kPcap, kRtpDump, kLengthPacketInterleaved }; 36 enum FileFormat { kPcap, kRtpDump, kLengthPacketInterleaved };
36 37
37 virtual ~RtpFileReader() {} 38 virtual ~RtpFileReader() {}
38 static RtpFileReader* Create(FileFormat format, 39 static RtpFileReader* Create(FileFormat format,
39 const std::string& filename); 40 const std::string& filename);
41 static RtpFileReader* Create(FileFormat format,
42 const std::string& filename,
43 const std::set<uint32_t>& ssrc_filter);
40 44
41 virtual bool NextPacket(RtpPacket* packet) = 0; 45 virtual bool NextPacket(RtpPacket* packet) = 0;
42 }; 46 };
43 } // namespace test 47 } // namespace test
44 } // namespace webrtc 48 } // namespace webrtc
45 #endif // WEBRTC_TEST_RTP_FILE_READER_H_ 49 #endif // WEBRTC_TEST_RTP_FILE_READER_H_
OLDNEW
« no previous file with comments | « webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc ('k') | webrtc/test/rtp_file_reader.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698