Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(750)

Side by Side Diff: webrtc/modules/audio_coding/test/opus_test.cc

Issue 2381013002: Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ (Closed)
Patch Set: rebase Created 4 years, 2 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) 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 "webrtc/modules/audio_coding/test/opus_test.h" 11 #include "webrtc/modules/audio_coding/test/opus_test.h"
12 12
13 #include <assert.h> 13 #include <assert.h>
14 14
15 #include <string> 15 #include <string>
16 16
17 #include "webrtc/test/gtest.h"
18 #include "webrtc/common_types.h" 17 #include "webrtc/common_types.h"
19 #include "webrtc/engine_configurations.h" 18 #include "webrtc/engine_configurations.h"
20 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" 19 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
21 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" 20 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
22 #include "webrtc/modules/audio_coding/test/TestStereo.h" 21 #include "webrtc/modules/audio_coding/test/TestStereo.h"
23 #include "webrtc/modules/audio_coding/test/utility.h" 22 #include "webrtc/modules/audio_coding/test/utility.h"
24 #include "webrtc/system_wrappers/include/trace.h" 23 #include "webrtc/system_wrappers/include/trace.h"
24 #include "webrtc/test/gtest.h"
25 #include "webrtc/test/testsupport/fileutils.h" 25 #include "webrtc/test/testsupport/fileutils.h"
26 26
27 namespace webrtc { 27 namespace webrtc {
28 28
29 OpusTest::OpusTest() 29 OpusTest::OpusTest()
30 : acm_receiver_(AudioCodingModule::Create(0)), 30 : acm_receiver_(AudioCodingModule::Create(0)),
31 channel_a2b_(NULL), 31 channel_a2b_(NULL),
32 counter_(0), 32 counter_(0),
33 payload_type_(255), 33 payload_type_(255),
34 rtp_timestamp_(0) {} 34 rtp_timestamp_(0) {}
(...skipping 342 matching lines...) Expand 10 before | Expand all | Expand 10 after
377 out_file_.Open(file_name, 48000, "wb"); 377 out_file_.Open(file_name, 48000, "wb");
378 file_stream.str(""); 378 file_stream.str("");
379 file_name = file_stream.str(); 379 file_name = file_stream.str();
380 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" 380 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
381 << test_number << ".pcm"; 381 << test_number << ".pcm";
382 file_name = file_stream.str(); 382 file_name = file_stream.str();
383 out_file_standalone_.Open(file_name, 48000, "wb"); 383 out_file_standalone_.Open(file_name, 48000, "wb");
384 } 384 }
385 385
386 } // namespace webrtc 386 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698