webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()

webrtc/modules/audio_processing/agc/agc_manager_direct.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
 
-#include <cassert>
 #include <cmath>
 
 #ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <cstdio>
 #endif
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
 namespace {
 
 // Lowest the microphone level can be lowered due to clipping.
@@ skipping 24 matching lines @@
 
 // Maximum additional gain allowed to compensate for microphone level
 // restrictions from clipping events.
 const int kSurplusCompressionGain = 6;
 
 int ClampLevel(int mic_level) {
   return std::min(std::max(kMinMicLevel, mic_level), kMaxMicLevel);
 }
 
 int LevelFromGainError(int gain_error, int level) {
-  assert(level >= 0 && level <= kMaxMicLevel);
+  RTC_DCHECK_GE(level, 0);
+  RTC_DCHECK_LE(level, kMaxMicLevel);
   if (gain_error == 0) {
     return level;
   }
   // TODO(ajm): Could be made more efficient with a binary search.
   int new_level = level;
   if (gain_error > 0) {
     while (kGainMap[new_level] - kGainMap[level] < gain_error &&
           new_level < kMaxMicLevel) {
       ++new_level;
     }
   } else {
     while (kGainMap[new_level] - kGainMap[level] > gain_error &&
           new_level > kMinMicLevel) {
       --new_level;
     }
   }
   return new_level;
 }
 
 }  // namespace
 
 // Facility for dumping debug audio files. All methods are no-ops in the
 // default case where WEBRTC_AGC_DEBUG_DUMP is undefined.
 class DebugFile {
 #ifdef WEBRTC_AGC_DEBUG_DUMP
  public:
   explicit DebugFile(const char* filename)
       : file_(fopen(filename, "wb")) {
-    assert(file_);
+    RTC_DCHECK(file_);
   }
   ~DebugFile() {
     fclose(file_);
   }
   void Write(const int16_t* data, size_t length_samples) {
     fwrite(data, 1, length_samples * sizeof(int16_t), file_);
   }
  private:
   FILE* file_;
 #else
@@ skipping 134 matching lines @@
 
   if (check_volume_on_next_process_) {
     check_volume_on_next_process_ = false;
     // We have to wait until the first process call to check the volume,
     // because Chromium doesn't guarantee it to be valid any earlier.
     CheckVolumeAndReset();
   }
 
   if (agc_->Process(audio, length, sample_rate_hz) != 0) {
     LOG(LS_ERROR) << "Agc::Process failed";
-    assert(false);
+    RTC_NOTREACHED();
   }
 
   UpdateGain();
   UpdateCompressor();
 
   file_postproc_->Write(audio, length);
 }
 
 void AgcManagerDirect::SetLevel(int new_level) {
   int voe_level = volume_callbacks_->GetMicVolume();
@@ skipping 31 matching lines @@
   }
 
   volume_callbacks_->SetMicVolume(new_level);
   LOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
                << "level_=" << level_ << ", "
                << "new_level=" << new_level;
   level_ = new_level;
 }
 
 void AgcManagerDirect::SetMaxLevel(int level) {
-  assert(level >= kClippedLevelMin);
+  RTC_DCHECK_GE(level, kClippedLevelMin);
   max_level_ = level;
   // Scale the |kSurplusCompressionGain| linearly across the restricted
   // level range.
   max_compression_gain_ = kMaxCompressionGain + std::floor(
       (1.f * kMaxMicLevel - max_level_) / (kMaxMicLevel - kClippedLevelMin) *
       kSurplusCompressionGain + 0.5f);
   LOG(LS_INFO) << "[agc] max_level_=" << max_level_
                << ", max_compression_gain_="  << max_compression_gain_;
 }
 
@@ skipping 123 matching lines @@
     compression_ = new_compression;
     compression_accumulator_ = new_compression;
     if (gctrl_->set_compression_gain_db(compression_) != 0) {
       LOG(LS_ERROR) << "set_compression_gain_db(" << compression_
                     << ") failed.";
     }
   }
 }
 
 }  // namespace webrtc