Replace ASSERT by RTC_DCHECK in all non-test code.

webrtc/api/dtmfsender.cc

 /*
  *  Copyright 2012 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.
  */
 
@@ skipping 63 matching lines @@
 
 DtmfSender::DtmfSender(AudioTrackInterface* track,
                        rtc::Thread* signaling_thread,
                        DtmfProviderInterface* provider)
     : track_(track),
       observer_(NULL),
       signaling_thread_(signaling_thread),
       provider_(provider),
       duration_(kDtmfDefaultDurationMs),
       inter_tone_gap_(kDtmfDefaultGapMs) {
-  ASSERT(track_ != NULL);
-  ASSERT(signaling_thread_ != NULL);
+  RTC_DCHECK(track_ != NULL);
+  RTC_DCHECK(signaling_thread_ != NULL);
   // TODO(deadbeef): Once we can use shared_ptr and weak_ptr,
   // do that instead of relying on a "destroyed" signal.
   if (provider_) {
-    ASSERT(provider_->GetOnDestroyedSignal() != NULL);
+    RTC_DCHECK(provider_->GetOnDestroyedSignal() != NULL);
     provider_->GetOnDestroyedSignal()->connect(
         this, &DtmfSender::OnProviderDestroyed);
   }
 }
 
 DtmfSender::~DtmfSender() {
   StopSending();
 }
 
 void DtmfSender::RegisterObserver(DtmfSenderObserverInterface* observer) {
   observer_ = observer;
 }
 
 void DtmfSender::UnregisterObserver() {
   observer_ = NULL;
 }
 
 bool DtmfSender::CanInsertDtmf() {
-  ASSERT(signaling_thread_->IsCurrent());
+  RTC_DCHECK(signaling_thread_->IsCurrent());
   if (!provider_) {
     return false;
   }
   return provider_->CanInsertDtmf(track_->id());
 }
 
 bool DtmfSender::InsertDtmf(const std::string& tones, int duration,
                             int inter_tone_gap) {
-  ASSERT(signaling_thread_->IsCurrent());
+  RTC_DCHECK(signaling_thread_->IsCurrent());
 
   if (duration > kDtmfMaxDurationMs ||
       duration < kDtmfMinDurationMs ||
       inter_tone_gap < kDtmfMinGapMs) {
     LOG(LS_ERROR) << "InsertDtmf is called with invalid duration or tones gap. "
         << "The duration cannot be more than " << kDtmfMaxDurationMs
         << "ms or less than " << kDtmfMinDurationMs << "ms. "
         << "The gap between tones must be at least " << kDtmfMinGapMs << "ms.";
     return false;
   }
@@ skipping 37 matching lines @@
       break;
     }
     default: {
       RTC_NOTREACHED();
       break;
     }
   }
 }
 
 void DtmfSender::DoInsertDtmf() {
-  ASSERT(signaling_thread_->IsCurrent());
+  RTC_DCHECK(signaling_thread_->IsCurrent());
 
   // Get the first DTMF tone from the tone buffer. Unrecognized characters will
   // be ignored and skipped.
   size_t first_tone_pos = tones_.find_first_of(kDtmfValidTones);
   int code = 0;
   if (first_tone_pos == std::string::npos) {
     tones_.clear();
     // Fire a “OnToneChange” event with an empty string and stop.
     if (observer_) {
       observer_->OnToneChange(std::string());
@@ skipping 45 matching lines @@
   LOG(LS_INFO) << "The Dtmf provider is deleted. Clear the sending queue.";
   StopSending();
   provider_ = NULL;
 }
 
 void DtmfSender::StopSending() {
   signaling_thread_->Clear(this);
 }
 
 }  // namespace webrtc