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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 <stddef.h> // size_t | 11 #include <stddef.h> // size_t |
12 | 12 |
13 #include <memory> | 13 #include <memory> |
14 #include <string> | 14 #include <string> |
15 #include <vector> | 15 #include <vector> |
16 | 16 |
17 #include "webrtc/base/task_queue.h" | |
18 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" | 17 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" |
19 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h" | 18 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h" |
20 #include "webrtc/modules/audio_processing/test/debug_dump_replayer.h" | 19 #include "webrtc/modules/audio_processing/test/debug_dump_replayer.h" |
21 #include "webrtc/modules/audio_processing/test/test_utils.h" | 20 #include "webrtc/modules/audio_processing/test/test_utils.h" |
| 21 #include "webrtc/rtc_base/task_queue.h" |
22 #include "webrtc/test/gtest.h" | 22 #include "webrtc/test/gtest.h" |
23 #include "webrtc/test/testsupport/fileutils.h" | 23 #include "webrtc/test/testsupport/fileutils.h" |
24 | 24 |
25 | |
26 namespace webrtc { | 25 namespace webrtc { |
27 namespace test { | 26 namespace test { |
28 | 27 |
29 namespace { | 28 namespace { |
30 | 29 |
31 void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer, | 30 void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer, |
32 const StreamConfig& config) { | 31 const StreamConfig& config) { |
33 auto& buffer_ref = *buffer; | 32 auto& buffer_ref = *buffer; |
34 if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() || | 33 if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() || |
35 buffer_ref->num_channels() != config.num_channels()) { | 34 buffer_ref->num_channels() != config.num_channels()) { |
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592 config.Set<ExperimentalNs>(new ExperimentalNs(true)); | 591 config.Set<ExperimentalNs>(new ExperimentalNs(true)); |
593 DebugDumpGenerator generator(config, AudioProcessing::Config()); | 592 DebugDumpGenerator generator(config, AudioProcessing::Config()); |
594 generator.StartRecording(); | 593 generator.StartRecording(); |
595 generator.Process(100); | 594 generator.Process(100); |
596 generator.StopRecording(); | 595 generator.StopRecording(); |
597 VerifyDebugDump(generator.dump_file_name()); | 596 VerifyDebugDump(generator.dump_file_name()); |
598 } | 597 } |
599 | 598 |
600 } // namespace test | 599 } // namespace test |
601 } // namespace webrtc | 600 } // namespace webrtc |
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