Adding a some checks and switching out a few assert for RTC_[D]CHECK.

webrtc/voice_engine/utility.cc

 /*
  *  Copyright (c) 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.
  */
 
 #include "webrtc/voice_engine/utility.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
 namespace voe {
@@ skipping 22 matching lines @@
   // Downmix before resampling.
   if (num_channels == 2 && dst_frame->num_channels_ == 1) {
     AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
                                        mono_audio);
     audio_ptr = mono_audio;
     audio_ptr_num_channels = 1;
   }
 
   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
                                     audio_ptr_num_channels) == -1) {
-    LOG(LS_ERROR) << "InitializeIfNeeded failed: sample_rate_hz = "
-                  << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
-                  << dst_frame->sample_rate_hz_
-                  << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
-    assert(false);
+    FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
+            << ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
+            << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
   }
 
   const size_t src_length = samples_per_channel * audio_ptr_num_channels;
   int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
                                        AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
-    LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
-                  << ", src_length = " << src_length
-                  << ", dst_frame->data_ = " << dst_frame->data_;
-    assert(false);
+    FATAL() << "Resample failed: audio_ptr = " << audio_ptr
+            << ", src_length = " << src_length
+            << ", dst_frame->data_ = " << dst_frame->data_;
   }
   dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
 
   // Upmix after resampling.
   if (num_channels == 1 && dst_frame->num_channels_ == 2) {
     // The audio in dst_frame really is mono at this point; MonoToStereo will
     // set this back to stereo.
     dst_frame->num_channels_ = 1;
     AudioFrameOperations::MonoToStereo(dst_frame);
   }
 }
 
 void MixWithSat(int16_t target[],
                 size_t target_channel,
                 const int16_t source[],
                 size_t source_channel,
                 size_t source_len) {
-  assert(target_channel == 1 || target_channel == 2);
-  assert(source_channel == 1 || source_channel == 2);
+  RTC_DCHECK_GE(target_channel, 1u);
+  RTC_DCHECK_LE(target_channel, 2u);
+  RTC_DCHECK_GE(source_channel, 1u);
+  RTC_DCHECK_LE(source_channel, 2u);
 
   if (target_channel == 2 && source_channel == 1) {
     // Convert source from mono to stereo.
     int32_t left = 0;
     int32_t right = 0;
     for (size_t i = 0; i < source_len; ++i) {
       left = source[i] + target[i * 2];
       right = source[i] + target[i * 2 + 1];
       target[i * 2] = WebRtcSpl_SatW32ToW16(left);
       target[i * 2 + 1] = WebRtcSpl_SatW32ToW16(right);
     }
   } else if (target_channel == 1 && source_channel == 2) {
     // Convert source from stereo to mono.
     int32_t temp = 0;
     for (size_t i = 0; i < source_len / 2; ++i) {
       temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   } else {
     int32_t temp = 0;
     for (size_t i = 0; i < source_len; ++i) {
       temp = source[i] + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   }
 }
 
 }  // namespace voe
 }  // namespace webrtc