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Side by Side Diff: webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc

Issue 1697823002: Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@up-codecs
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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
11 #include <assert.h> 11 #include <assert.h>
12 #include <stdio.h> 12 #include <stdio.h>
13
14 #include <memory>
13 #include <vector> 15 #include <vector>
14 16
15 #include "gflags/gflags.h" 17 #include "gflags/gflags.h"
16 #include "webrtc/base/scoped_ptr.h"
17 #include "webrtc/modules/audio_coding/neteq/tools/packet.h" 18 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
18 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" 19 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
19 20
20 // Flag validator. 21 // Flag validator.
21 static bool ValidatePayloadType(const char* flagname, int32_t value) { 22 static bool ValidatePayloadType(const char* flagname, int32_t value) {
22 if (value >= 0 && value <= 127) // Value is ok. 23 if (value >= 0 && value <= 127) // Value is ok.
23 return true; 24 return true;
24 printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value)); 25 printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
25 return false; 26 return false;
26 } 27 }
(...skipping 29 matching lines...) Expand all
56 google::SetUsageMessage(usage); 57 google::SetUsageMessage(usage);
57 google::ParseCommandLineFlags(&argc, &argv, true); 58 google::ParseCommandLineFlags(&argc, &argv, true);
58 59
59 if (argc != 2 && argc != 3) { 60 if (argc != 2 && argc != 3) {
60 // Print usage information. 61 // Print usage information.
61 printf("%s", google::ProgramUsage()); 62 printf("%s", google::ProgramUsage());
62 return 0; 63 return 0;
63 } 64 }
64 65
65 printf("Input file: %s\n", argv[1]); 66 printf("Input file: %s\n", argv[1]);
66 rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source( 67 std::unique_ptr<webrtc::test::RtpFileSource> file_source(
67 webrtc::test::RtpFileSource::Create(argv[1])); 68 webrtc::test::RtpFileSource::Create(argv[1]));
68 assert(file_source.get()); 69 assert(file_source.get());
69 // Set RTP extension IDs. 70 // Set RTP extension IDs.
70 bool print_audio_level = false; 71 bool print_audio_level = false;
71 if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) { 72 if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) {
72 print_audio_level = true; 73 print_audio_level = true;
73 file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, 74 file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
74 FLAGS_audio_level); 75 FLAGS_audio_level);
75 } 76 }
76 bool print_abs_send_time = false; 77 bool print_abs_send_time = false;
(...skipping 20 matching lines...) Expand all
97 if (print_audio_level) { 98 if (print_audio_level) {
98 fprintf(out_file, " AuLvl (V)"); 99 fprintf(out_file, " AuLvl (V)");
99 } 100 }
100 if (print_abs_send_time) { 101 if (print_abs_send_time) {
101 fprintf(out_file, " AbsSendTime"); 102 fprintf(out_file, " AbsSendTime");
102 } 103 }
103 fprintf(out_file, "\n"); 104 fprintf(out_file, "\n");
104 105
105 uint32_t max_abs_send_time = 0; 106 uint32_t max_abs_send_time = 0;
106 int cycles = -1; 107 int cycles = -1;
107 rtc::scoped_ptr<webrtc::test::Packet> packet; 108 std::unique_ptr<webrtc::test::Packet> packet;
108 while (true) { 109 while (true) {
109 packet.reset(file_source->NextPacket()); 110 packet.reset(file_source->NextPacket());
110 if (!packet.get()) { 111 if (!packet.get()) {
111 // End of file reached. 112 // End of file reached.
112 break; 113 break;
113 } 114 }
114 // Write packet data to file. Use virtual_packet_length_bytes so that the 115 // Write packet data to file. Use virtual_packet_length_bytes so that the
115 // correct packet sizes are printed also for RTP header-only dumps. 116 // correct packet sizes are printed also for RTP header-only dumps.
116 fprintf(out_file, 117 fprintf(out_file,
117 "%5u %10u %10u %5i %5i %2i %#08X", 118 "%5u %10u %10u %5i %5i %2i %#08X",
(...skipping 56 matching lines...) Expand 10 before | Expand all | Expand 10 after
174 red_headers.pop_front(); 175 red_headers.pop_front();
175 delete red; 176 delete red;
176 } 177 }
177 } 178 }
178 } 179 }
179 180
180 fclose(out_file); 181 fclose(out_file);
181 182
182 return 0; 183 return 0;
183 } 184 }
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