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Side by Side Diff: webrtc/modules/audio_processing/agc/agc_manager_direct.cc

Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 "webrtc/modules/audio_processing/agc/agc_manager_direct.h" 11 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
12 12
13 #include <cassert> 13 #include <cassert>
14 #include <cmath> 14 #include <cmath>
15 15
16 #ifdef WEBRTC_AGC_DEBUG_DUMP 16 #ifdef WEBRTC_AGC_DEBUG_DUMP
17 #include <cstdio> 17 #include <cstdio>
18 #endif 18 #endif
19 19
20 #include "webrtc/modules/audio_processing/agc/gain_map_internal.h" 20 #include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
21 #include "webrtc/modules/audio_processing/gain_control_impl.h" 21 #include "webrtc/modules/audio_processing/gain_control_impl.h"
22 #include "webrtc/modules/interface/module_common_types.h" 22 #include "webrtc/modules/interface/module_common_types.h"
23 #include "webrtc/system_wrappers/interface/logging.h" 23 #include "webrtc/system_wrappers/include/logging.h"
24 24
25 namespace webrtc { 25 namespace webrtc {
26 26
27 namespace { 27 namespace {
28 28
29 // Lowest the microphone level can be lowered due to clipping. 29 // Lowest the microphone level can be lowered due to clipping.
30 const int kClippedLevelMin = 170; 30 const int kClippedLevelMin = 170;
31 // Amount the microphone level is lowered with every clipping event. 31 // Amount the microphone level is lowered with every clipping event.
32 const int kClippedLevelStep = 15; 32 const int kClippedLevelStep = 15;
33 // Proportion of clipped samples required to declare a clipping event. 33 // Proportion of clipped samples required to declare a clipping event.
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433 if (new_compression != compression_) { 433 if (new_compression != compression_) {
434 compression_ = new_compression; 434 compression_ = new_compression;
435 compression_accumulator_ = new_compression; 435 compression_accumulator_ = new_compression;
436 if (gctrl_->set_compression_gain_db(compression_) != 0) { 436 if (gctrl_->set_compression_gain_db(compression_) != 0) {
437 LOG_FERR1(LS_ERROR, set_compression_gain_db, compression_); 437 LOG_FERR1(LS_ERROR, set_compression_gain_db, compression_);
438 } 438 }
439 } 439 }
440 } 440 }
441 441
442 } // namespace webrtc 442 } // namespace webrtc
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