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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/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) {} |
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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 |
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