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Side by Side Diff: webrtc/modules/audio_processing/aec3/aec3_common.h

Issue 2974883002: Robustification of the AEC3 echo removal in the first part of the call (Closed)
Patch Set: Rebase Created 3 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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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53 constexpr size_t kNumMatchedFilters = 4; 53 constexpr size_t kNumMatchedFilters = 4;
54 constexpr size_t kMatchedFilterWindowSizeSubBlocks = 32; 54 constexpr size_t kMatchedFilterWindowSizeSubBlocks = 32;
55 constexpr size_t kMatchedFilterAlignmentShiftSizeSubBlocks = 55 constexpr size_t kMatchedFilterAlignmentShiftSizeSubBlocks =
56 kMatchedFilterWindowSizeSubBlocks * 3 / 4; 56 kMatchedFilterWindowSizeSubBlocks * 3 / 4;
57 constexpr size_t kDownsampledRenderBufferSize = 57 constexpr size_t kDownsampledRenderBufferSize =
58 kSubBlockSize * 58 kSubBlockSize *
59 (kMatchedFilterAlignmentShiftSizeSubBlocks * kNumMatchedFilters + 59 (kMatchedFilterAlignmentShiftSizeSubBlocks * kNumMatchedFilters +
60 kMatchedFilterWindowSizeSubBlocks + 60 kMatchedFilterWindowSizeSubBlocks +
61 1); 61 1);
62 62
63 constexpr float kFixedEchoPathGain = 100;
64
65 constexpr size_t kRenderDelayBufferSize = 63 constexpr size_t kRenderDelayBufferSize =
66 (3 * kDownsampledRenderBufferSize) / (4 * kSubBlockSize); 64 (3 * kDownsampledRenderBufferSize) / (4 * kSubBlockSize);
67 65
68 constexpr size_t kMaxApiCallsJitterBlocks = 30; 66 constexpr size_t kMaxApiCallsJitterBlocks = 30;
69 constexpr size_t kRenderTransferQueueSize = kMaxApiCallsJitterBlocks / 2; 67 constexpr size_t kRenderTransferQueueSize = kMaxApiCallsJitterBlocks / 2;
70 static_assert(2 * kRenderTransferQueueSize >= kMaxApiCallsJitterBlocks, 68 static_assert(2 * kRenderTransferQueueSize >= kMaxApiCallsJitterBlocks,
71 "Requirement to ensure buffer overflow detection"); 69 "Requirement to ensure buffer overflow detection");
72 70
73 // TODO(peah): Integrate this with how it is done inside audio_processing_impl. 71 // TODO(peah): Integrate this with how it is done inside audio_processing_impl.
74 constexpr size_t NumBandsForRate(int sample_rate_hz) { 72 constexpr size_t NumBandsForRate(int sample_rate_hz) {
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107 static_assert(ValidFullBandRate(32000), 105 static_assert(ValidFullBandRate(32000),
108 "Test that 32 kHz is a valid sample rate"); 106 "Test that 32 kHz is a valid sample rate");
109 static_assert(ValidFullBandRate(48000), 107 static_assert(ValidFullBandRate(48000),
110 "Test that 48 kHz is a valid sample rate"); 108 "Test that 48 kHz is a valid sample rate");
111 static_assert(!ValidFullBandRate(8001), 109 static_assert(!ValidFullBandRate(8001),
112 "Test that 8001 Hz is not a valid sample rate"); 110 "Test that 8001 Hz is not a valid sample rate");
113 111
114 } // namespace webrtc 112 } // namespace webrtc
115 113
116 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_ 114 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
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