NetEq: Don't check sample rate and frame size upon error

webrtc/modules/audio_coding/neteq/neteq_impl.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.
  */
 
@@ skipping 188 matching lines @@
     // Always set kVadUnknown when receive VAD is inactive.
     audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
   }
 }
 }  // namespace
 
 int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted) {
   TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
   rtc::CritScope lock(&crit_sect_);
   int error = GetAudioInternal(audio_frame, muted);
-  RTC_DCHECK_EQ(
-      audio_frame->sample_rate_hz_,
-      rtc::checked_cast<int>(audio_frame->samples_per_channel_ * 100));
   if (error != 0) {
     error_code_ = error;
     return kFail;
   }
+  RTC_DCHECK_EQ(
+      audio_frame->sample_rate_hz_,
+      rtc::checked_cast<int>(audio_frame->samples_per_channel_ * 100));
   SetAudioFrameActivityAndType(vad_->enabled(), LastOutputType(),
                                last_vad_activity_, audio_frame);
   last_vad_activity_ = audio_frame->vad_activity_;
   last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
   RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
              last_output_sample_rate_hz_ == 16000 ||
              last_output_sample_rate_hz_ == 32000 ||
              last_output_sample_rate_hz_ == 48000)
       << "Unexpected sample rate " << last_output_sample_rate_hz_;
   return kOK;
@@ skipping 1913 matching lines @@
   }
 }
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(
       fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
       *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
       tick_timer_.get()));
 }
 }  // namespace webrtc