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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 <math.h> | 11 #include <math.h> |
12 #include <stdio.h> | 12 #include <stdio.h> |
13 | 13 |
14 #include <algorithm> | 14 #include <algorithm> |
15 #include <limits> | 15 #include <limits> |
16 #include <memory> | 16 #include <memory> |
17 #include <queue> | 17 #include <queue> |
18 | 18 |
19 #include "webrtc/base/arraysize.h" | 19 #include "webrtc/base/arraysize.h" |
20 #include "webrtc/base/checks.h" | 20 #include "webrtc/base/checks.h" |
21 #include "webrtc/base/gtest_prod_util.h" | 21 #include "webrtc/base/gtest_prod_util.h" |
22 #include "webrtc/base/ignore_wundef.h" | 22 #include "webrtc/base/ignore_wundef.h" |
23 #include "webrtc/base/protobuf_utils.h" | 23 #include "webrtc/base/protobuf_utils.h" |
| 24 #include "webrtc/base/safe_minmax.h" |
24 #include "webrtc/common_audio/include/audio_util.h" | 25 #include "webrtc/common_audio/include/audio_util.h" |
25 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 26 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
26 #include "webrtc/common_audio/resampler/push_sinc_resampler.h" | 27 #include "webrtc/common_audio/resampler/push_sinc_resampler.h" |
27 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" | 28 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" |
28 #include "webrtc/modules/audio_processing/audio_processing_impl.h" | 29 #include "webrtc/modules/audio_processing/audio_processing_impl.h" |
29 #include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
" | 30 #include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
" |
30 #include "webrtc/modules/audio_processing/common.h" | 31 #include "webrtc/modules/audio_processing/common.h" |
31 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 32 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
32 #include "webrtc/modules/audio_processing/level_controller/level_controller_cons
tants.h" | 33 #include "webrtc/modules/audio_processing/level_controller/level_controller_cons
tants.h" |
33 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" | 34 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" |
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671 | 672 |
672 rewind(near_file_); | 673 rewind(near_file_); |
673 while (!frame_queue.empty()) { | 674 while (!frame_queue.empty()) { |
674 AudioFrame* frame = frame_queue.front(); | 675 AudioFrame* frame = frame_queue.front(); |
675 frame_queue.pop(); | 676 frame_queue.pop(); |
676 delete frame; | 677 delete frame; |
677 } | 678 } |
678 // Calculate expected delay estimate and acceptable regions. Further, | 679 // Calculate expected delay estimate and acceptable regions. Further, |
679 // limit them w.r.t. AEC delay estimation support. | 680 // limit them w.r.t. AEC delay estimation support. |
680 const size_t samples_per_ms = | 681 const size_t samples_per_ms = |
681 std::min(static_cast<size_t>(16), frame_->samples_per_channel_ / 10); | 682 rtc::SafeMin<size_t>(16u, frame_->samples_per_channel_ / 10); |
682 int expected_median = std::min(std::max(delay_ms - system_delay_ms, | 683 int expected_median = std::min(std::max(delay_ms - system_delay_ms, |
683 delay_min), delay_max); | 684 delay_min), delay_max); |
684 int expected_median_high = std::min( | 685 int expected_median_high = std::min( |
685 std::max(expected_median + static_cast<int>(96 / samples_per_ms), | 686 std::max(expected_median + static_cast<int>(96 / samples_per_ms), |
686 delay_min), | 687 delay_min), |
687 delay_max); | 688 delay_max); |
688 int expected_median_low = std::min( | 689 int expected_median_low = std::min( |
689 std::max(expected_median - static_cast<int>(96 / samples_per_ms), | 690 std::max(expected_median - static_cast<int>(96 / samples_per_ms), |
690 delay_min), | 691 delay_min), |
691 delay_max); | 692 delay_max); |
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2877 // TODO(peah): Remove the testing for | 2878 // TODO(peah): Remove the testing for |
2878 // apm->capture_nonlocked_.level_controller_enabled once the value in config_ | 2879 // apm->capture_nonlocked_.level_controller_enabled once the value in config_ |
2879 // is instead used to activate the level controller. | 2880 // is instead used to activate the level controller. |
2880 EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled); | 2881 EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled); |
2881 EXPECT_NEAR(kTargetLcPeakLeveldBFS, | 2882 EXPECT_NEAR(kTargetLcPeakLeveldBFS, |
2882 apm->config_.level_controller.initial_peak_level_dbfs, | 2883 apm->config_.level_controller.initial_peak_level_dbfs, |
2883 std::numeric_limits<float>::epsilon()); | 2884 std::numeric_limits<float>::epsilon()); |
2884 } | 2885 } |
2885 | 2886 |
2886 } // namespace webrtc | 2887 } // namespace webrtc |
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