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Issue 3007153003: Enable UBSan float-cast-overflow warnings and fix existing ones (Closed)
Patch Set: Fix/silence existing warnings Created 3 years, 3 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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42 for (float& v : x) { 42 for (float& v : x) {
43 v -= *dc_level; 43 v -= *dc_level;
44 } 44 }
45 } 45 }
46 46
47 float FrameEnergy(const AudioBuffer& audio) { 47 float FrameEnergy(const AudioBuffer& audio) {
48 float energy = 0.f; 48 float energy = 0.f;
49 for (size_t k = 0; k < audio.num_channels(); ++k) { 49 for (size_t k = 0; k < audio.num_channels(); ++k) {
50 float channel_energy = 50 float channel_energy =
51 std::accumulate(audio.channels_const_f()[k], 51 std::accumulate(audio.channels_const_f()[k],
52 audio.channels_const_f()[k] + audio.num_frames(), 0, 52 audio.channels_const_f()[k] + audio.num_frames(), 0.f,
saza WebRTC 2017/09/05 09:35:20 Good find. Fyi, aleloi@ or hlundin@ are the least
53 [](float a, float b) -> float { return a + b * b; }); 53 [](float a, float b) -> float { return a + b * b; });
54 energy = std::max(channel_energy, energy); 54 energy = std::max(channel_energy, energy);
55 } 55 }
56 return energy; 56 return energy;
57 } 57 }
58 58
59 float PeakLevel(const AudioBuffer& audio) { 59 float PeakLevel(const AudioBuffer& audio) {
60 float peak_level = 0.f; 60 float peak_level = 0.f;
61 for (size_t k = 0; k < audio.num_channels(); ++k) { 61 for (size_t k = 0; k < audio.num_channels(); ++k) {
62 auto* channel_peak_level = std::max_element( 62 auto* channel_peak_level = std::max_element(
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284 284
285 bool LevelController::Validate( 285 bool LevelController::Validate(
286 const AudioProcessing::Config::LevelController& config) { 286 const AudioProcessing::Config::LevelController& config) {
287 return (config.initial_peak_level_dbfs < 287 return (config.initial_peak_level_dbfs <
288 std::numeric_limits<float>::epsilon() && 288 std::numeric_limits<float>::epsilon() &&
289 config.initial_peak_level_dbfs > 289 config.initial_peak_level_dbfs >
290 -(100.f + std::numeric_limits<float>::epsilon())); 290 -(100.f + std::numeric_limits<float>::epsilon()));
291 } 291 }
292 292
293 } // namespace webrtc 293 } // namespace webrtc
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