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Side by Side Diff: webrtc/modules/audio_processing/aec/aec_core.cc

Issue 1991723002: AEC: Add UMA logging of buffer re-alignment (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
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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
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26 #include "webrtc/common_audio/ring_buffer.h" 26 #include "webrtc/common_audio/ring_buffer.h"
27 } 27 }
28 #include "webrtc/base/checks.h" 28 #include "webrtc/base/checks.h"
29 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar y.h" 29 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar y.h"
30 #include "webrtc/modules/audio_processing/aec/aec_common.h" 30 #include "webrtc/modules/audio_processing/aec/aec_common.h"
31 #include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h" 31 #include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
32 #include "webrtc/modules/audio_processing/aec/aec_rdft.h" 32 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
33 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" 33 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
34 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h" 34 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
35 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h" 35 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
36 #include "webrtc/system_wrappers/include/metrics.h"
36 #include "webrtc/typedefs.h" 37 #include "webrtc/typedefs.h"
37 38
38 namespace webrtc { 39 namespace webrtc {
40 namespace {
41 enum class DelaySource {
42 kSystemDelay, // The delay values come from the OS.
43 kDelayAgnostic, // The delay values come from the DA-AEC.
44 };
45
46 constexpr int kMinDelayLogValue = -200;
47 constexpr int kMaxDelayLogValue = 200;
48 constexpr int kNumDelayLogBuckets = 100;
49
50 void MaybeLogDelayAdjustment(int moved_elements, DelaySource source) {
51 if (moved_elements == 0)
52 return;
53 switch (source) {
54 case DelaySource::kSystemDelay:
55 RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecDelayAdjustmentSystemValue",
56 moved_elements, kMinDelayLogValue,
57 kMaxDelayLogValue, kNumDelayLogBuckets);
58 return;
59 case DelaySource::kDelayAgnostic:
60 RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecDelayAdjustmentAgnosticValue",
61 moved_elements, kMinDelayLogValue,
62 kMaxDelayLogValue, kNumDelayLogBuckets);
63 return;
64 }
peah-webrtc 2016/05/18 08:22:11 Is a default case needed, or is it fine to use lik
hlundin-webrtc 2016/05/18 10:15:44 I believe it is fine. The compiler complains when
65 }
66 } // namespace
39 67
40 // Buffer size (samples) 68 // Buffer size (samples)
41 static const size_t kBufSizePartitions = 250; // 1 second of audio in 16 kHz. 69 static const size_t kBufSizePartitions = 250; // 1 second of audio in 16 kHz.
42 70
43 // Metrics 71 // Metrics
44 static const size_t kSubCountLen = 4; 72 static const size_t kSubCountLen = 4;
45 static const size_t kCountLen = 50; 73 static const size_t kCountLen = 50;
46 static const int kDelayMetricsAggregationWindow = 1250; // 5 seconds at 16 kHz. 74 static const int kDelayMetricsAggregationWindow = 1250; // 5 seconds at 16 kHz.
47 75
48 // Divergence metric is based on audio level, which gets updated every 76 // Divergence metric is based on audio level, which gets updated every
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1778 1806
1779 // TODO(bjornv): Investigate how we should round the delay difference; 1807 // TODO(bjornv): Investigate how we should round the delay difference;
1780 // right now we know that incoming |knownDelay| is underestimated when 1808 // right now we know that incoming |knownDelay| is underestimated when
1781 // it's less than |aec->knownDelay|. We therefore, round (-32) in that 1809 // it's less than |aec->knownDelay|. We therefore, round (-32) in that
1782 // direction. In the other direction, we don't have this situation, but 1810 // direction. In the other direction, we don't have this situation, but
1783 // might flush one partition too little. This can cause non-causality, 1811 // might flush one partition too little. This can cause non-causality,
1784 // which should be investigated. Maybe, allow for a non-symmetric 1812 // which should be investigated. Maybe, allow for a non-symmetric
1785 // rounding, like -16. 1813 // rounding, like -16.
1786 int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN; 1814 int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
1787 int moved_elements = WebRtc_MoveReadPtr(aec->far_time_buf, move_elements); 1815 int moved_elements = WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
1816 MaybeLogDelayAdjustment(moved_elements, DelaySource::kSystemDelay);
hlundin-webrtc 2016/05/18 08:04:30 What is the unit for moved_elements? Should probab
peah-webrtc 2016/05/18 08:22:11 For moved_elements below, each element corresponds
hlundin-webrtc 2016/05/18 10:15:44 Done.
1788 aec->knownDelay -= moved_elements * PART_LEN; 1817 aec->knownDelay -= moved_elements * PART_LEN;
1789 } else { 1818 } else {
1790 // 2 b) Apply signal based delay correction. 1819 // 2 b) Apply signal based delay correction.
1791 int move_elements = SignalBasedDelayCorrection(aec); 1820 int move_elements = SignalBasedDelayCorrection(aec);
1792 int moved_elements = WebRtc_MoveReadPtr(aec->far_time_buf, move_elements); 1821 int moved_elements = WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
1822 MaybeLogDelayAdjustment(moved_elements, DelaySource::kDelayAgnostic);
peah-webrtc 2016/05/18 08:22:11 There is also delay adjustments being applied in e
hlundin-webrtc 2016/05/18 10:15:44 Acknowledged. We'll keep that as a separate exerci
1793 int far_near_buffer_diff = 1823 int far_near_buffer_diff =
1794 WebRtc_available_read(aec->far_time_buf) - 1824 WebRtc_available_read(aec->far_time_buf) -
1795 WebRtc_available_read(aec->nearFrBuf) / PART_LEN; 1825 WebRtc_available_read(aec->nearFrBuf) / PART_LEN;
1796 WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements); 1826 WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
1797 WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend, 1827 WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
1798 moved_elements); 1828 moved_elements);
1799 aec->signal_delay_correction += moved_elements; 1829 aec->signal_delay_correction += moved_elements;
1800 // If we rely on reported system delay values only, a buffer underrun here 1830 // If we rely on reported system delay values only, a buffer underrun here
1801 // can never occur since we've taken care of that in 1) above. Here, we 1831 // can never occur since we've taken care of that in 1) above. Here, we
1802 // apply signal based delay correction and can therefore end up with 1832 // apply signal based delay correction and can therefore end up with
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1937 1967
1938 int WebRtcAec_system_delay(AecCore* self) { 1968 int WebRtcAec_system_delay(AecCore* self) {
1939 return self->system_delay; 1969 return self->system_delay;
1940 } 1970 }
1941 1971
1942 void WebRtcAec_SetSystemDelay(AecCore* self, int delay) { 1972 void WebRtcAec_SetSystemDelay(AecCore* self, int delay) {
1943 assert(delay >= 0); 1973 assert(delay >= 0);
1944 self->system_delay = delay; 1974 self->system_delay = delay;
1945 } 1975 }
1946 } // namespace webrtc 1976 } // namespace webrtc
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