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

Issue 2800033003: Fixing sample-rate dependent band-split filter issues in AEC3 (Closed)
Patch Set: Reverted the changes relating to the number of bands used internally inside AEC3 Created 3 years, 8 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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61 constexpr float kFixedEchoPathGain = 100; 61 constexpr float kFixedEchoPathGain = 100;
62 62
63 constexpr size_t kRenderDelayBufferSize = 63 constexpr size_t kRenderDelayBufferSize =
64 (3 * kDownsampledRenderBufferSize) / (4 * kSubBlockSize); 64 (3 * kDownsampledRenderBufferSize) / (4 * kSubBlockSize);
65 65
66 constexpr size_t kMaxApiCallsJitterBlocks = 20; 66 constexpr size_t kMaxApiCallsJitterBlocks = 20;
67 constexpr size_t kRenderTransferQueueSize = kMaxApiCallsJitterBlocks / 2; 67 constexpr size_t kRenderTransferQueueSize = kMaxApiCallsJitterBlocks / 2;
68 static_assert(2 * kRenderTransferQueueSize >= kMaxApiCallsJitterBlocks, 68 static_assert(2 * kRenderTransferQueueSize >= kMaxApiCallsJitterBlocks,
69 "Requirement to ensure buffer overflow detection"); 69 "Requirement to ensure buffer overflow detection");
70 70
71 // TODO(peah): Integrate this with how it is done inside audio_processing_impl.
71 constexpr size_t NumBandsForRate(int sample_rate_hz) { 72 constexpr size_t NumBandsForRate(int sample_rate_hz) {
72 return static_cast<size_t>(sample_rate_hz == 8000 ? 1 73 return static_cast<size_t>(sample_rate_hz == 8000 ? 1
73 : sample_rate_hz / 16000); 74 : sample_rate_hz / 16000);
74 } 75 }
75 constexpr int LowestBandRate(int sample_rate_hz) { 76 constexpr int LowestBandRate(int sample_rate_hz) {
76 return sample_rate_hz == 8000 ? sample_rate_hz : 16000; 77 return sample_rate_hz == 8000 ? sample_rate_hz : 16000;
77 } 78 }
78 79
79 constexpr bool ValidFullBandRate(int sample_rate_hz) { 80 constexpr bool ValidFullBandRate(int sample_rate_hz) {
80 return sample_rate_hz == 8000 || sample_rate_hz == 16000 || 81 return sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
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104 static_assert(ValidFullBandRate(32000), 105 static_assert(ValidFullBandRate(32000),
105 "Test that 32 kHz is a valid sample rate"); 106 "Test that 32 kHz is a valid sample rate");
106 static_assert(ValidFullBandRate(48000), 107 static_assert(ValidFullBandRate(48000),
107 "Test that 48 kHz is a valid sample rate"); 108 "Test that 48 kHz is a valid sample rate");
108 static_assert(!ValidFullBandRate(8001), 109 static_assert(!ValidFullBandRate(8001),
109 "Test that 8001 Hz is not a valid sample rate"); 110 "Test that 8001 Hz is not a valid sample rate");
110 111
111 } // namespace webrtc 112 } // namespace webrtc
112 113
113 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_ 114 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
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