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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/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/common_types.h" | 17 #include "webrtc/common_types.h" |
18 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" | 18 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" |
19 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" | 19 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" |
20 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" | 20 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" |
21 #include "webrtc/modules/audio_coding/test/TestStereo.h" | 21 #include "webrtc/modules/audio_coding/test/TestStereo.h" |
22 #include "webrtc/modules/audio_coding/test/utility.h" | 22 #include "webrtc/modules/audio_coding/test/utility.h" |
23 #include "webrtc/system_wrappers/include/trace.h" | |
24 #include "webrtc/test/gtest.h" | 23 #include "webrtc/test/gtest.h" |
25 #include "webrtc/test/testsupport/fileutils.h" | 24 #include "webrtc/test/testsupport/fileutils.h" |
26 #include "webrtc/typedefs.h" | 25 #include "webrtc/typedefs.h" |
27 | 26 |
28 namespace webrtc { | 27 namespace webrtc { |
29 | 28 |
30 OpusTest::OpusTest() | 29 OpusTest::OpusTest() |
31 : acm_receiver_(AudioCodingModule::Create(0)), | 30 : acm_receiver_(AudioCodingModule::Create(0)), |
32 channel_a2b_(NULL), | 31 channel_a2b_(NULL), |
33 counter_(0), | 32 counter_(0), |
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382 out_file_.Open(file_name, 48000, "wb"); | 381 out_file_.Open(file_name, 48000, "wb"); |
383 file_stream.str(""); | 382 file_stream.str(""); |
384 file_name = file_stream.str(); | 383 file_name = file_stream.str(); |
385 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" | 384 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" |
386 << test_number << ".pcm"; | 385 << test_number << ".pcm"; |
387 file_name = file_stream.str(); | 386 file_name = file_stream.str(); |
388 out_file_standalone_.Open(file_name, 48000, "wb"); | 387 out_file_standalone_.Open(file_name, 48000, "wb"); |
389 } | 388 } |
390 | 389 |
391 } // namespace webrtc | 390 } // namespace webrtc |
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