Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(109)

Side by Side Diff: webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc

Issue 1417683006: modules: more interface -> include renames (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix last incorrectly wrapped pragma message Created 5 years, 1 month 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) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 <stdio.h> 11 #include <stdio.h>
12 12
13 #include "gflags/gflags.h" 13 #include "gflags/gflags.h"
14 #include "testing/gtest/include/gtest/gtest.h" 14 #include "testing/gtest/include/gtest/gtest.h"
15 #include "webrtc/base/scoped_ptr.h" 15 #include "webrtc/base/scoped_ptr.h"
16 #include "webrtc/common_types.h" 16 #include "webrtc/common_types.h"
17 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" 17 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
18 #include "webrtc/modules/audio_coding/main/test/Channel.h" 18 #include "webrtc/modules/audio_coding/main/test/Channel.h"
19 #include "webrtc/modules/audio_coding/main/test/PCMFile.h" 19 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
20 #include "webrtc/modules/interface/module_common_types.h" 20 #include "webrtc/modules/include/module_common_types.h"
21 #include "webrtc/system_wrappers/include/clock.h" 21 #include "webrtc/system_wrappers/include/clock.h"
22 #include "webrtc/test/testsupport/fileutils.h" 22 #include "webrtc/test/testsupport/fileutils.h"
23 23
24 // Codec. 24 // Codec.
25 DEFINE_string(codec, "opus", "Codec Name"); 25 DEFINE_string(codec, "opus", "Codec Name");
26 DEFINE_int32(codec_sample_rate_hz, 48000, "Sampling rate in Hertz."); 26 DEFINE_int32(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
27 DEFINE_int32(codec_channels, 1, "Number of channels of the codec."); 27 DEFINE_int32(codec_channels, 1, "Number of channels of the codec.");
28 28
29 // PCM input/output. 29 // PCM input/output.
30 DEFINE_string(input, "", "Input PCM file at 16 kHz."); 30 DEFINE_string(input, "", "Input PCM file at 16 kHz.");
(...skipping 267 matching lines...) Expand 10 before | Expand all | Expand 10 after
298 if (delay_log != NULL) { 298 if (delay_log != NULL) {
299 fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms); 299 fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
300 } 300 }
301 } 301 }
302 } 302 }
303 std::cout << std::endl; 303 std::cout << std::endl;
304 test.TearDown(); 304 test.TearDown();
305 if (delay_log != NULL) 305 if (delay_log != NULL)
306 fclose(delay_log); 306 fclose(delay_log);
307 } 307 }
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/main/test/RTPFile.h ('k') | webrtc/modules/audio_coding/main/test/target_delay_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698