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

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years ago
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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/main/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 "testing/gtest/include/gtest/gtest.h" 17 #include "testing/gtest/include/gtest/gtest.h"
18 #include "webrtc/common_types.h" 18 #include "webrtc/common_types.h"
19 #include "webrtc/engine_configurations.h" 19 #include "webrtc/engine_configurations.h"
20 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" 20 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
21 #include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs. h" 21 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
22 #include "webrtc/modules/audio_coding/main/test/TestStereo.h" 22 #include "webrtc/modules/audio_coding/test/TestStereo.h"
23 #include "webrtc/modules/audio_coding/main/test/utility.h" 23 #include "webrtc/modules/audio_coding/test/utility.h"
24 #include "webrtc/system_wrappers/include/trace.h" 24 #include "webrtc/system_wrappers/include/trace.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),
(...skipping 337 matching lines...) Expand 10 before | Expand all | Expand 10 after
371 out_file_.Open(file_name, 48000, "wb"); 371 out_file_.Open(file_name, 48000, "wb");
372 file_stream.str(""); 372 file_stream.str("");
373 file_name = file_stream.str(); 373 file_name = file_stream.str();
374 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" 374 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
375 << test_number << ".pcm"; 375 << test_number << ".pcm";
376 file_name = file_stream.str(); 376 file_name = file_stream.str();
377 out_file_standalone_.Open(file_name, 48000, "wb"); 377 out_file_standalone_.Open(file_name, 48000, "wb");
378 } 378 }
379 379
380 } // namespace webrtc 380 } // namespace webrtc
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