Remove ignored return code from modules.

webrtc/video/vie_sync_module.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 70 matching lines @@
   video_rtp_rtcp_ = video_rtcp_module;
   sync_.reset(
       new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
 }
 
 int64_t ViESyncModule::TimeUntilNextProcess() {
   const int64_t kSyncIntervalMs = 1000;
   return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
 }
 
-int32_t ViESyncModule::Process() {
+void ViESyncModule::Process() {
   rtc::CritScope lock(&data_cs_);
   last_sync_time_ = TickTime::Now();
 
   const int current_video_delay_ms = vcm_->Delay();
 
   if (voe_channel_id_ == -1) {
-    return 0;
+    return;
   }
   assert(video_rtp_rtcp_ && voe_sync_interface_);
   assert(sync_.get());
 
   int audio_jitter_buffer_delay_ms = 0;
   int playout_buffer_delay_ms = 0;
   if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
                                             &audio_jitter_buffer_delay_ms,
                                             &playout_buffer_delay_ms) != 0) {
-    return 0;
+    return;
   }
   const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
       playout_buffer_delay_ms;
 
   RtpRtcp* voice_rtp_rtcp = NULL;
   RtpReceiver* voice_receiver = NULL;
   if (0 != voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
                                            &voice_receiver)) {
-    return 0;
+    return;
   }
   assert(voice_rtp_rtcp);
   assert(voice_receiver);
 
   if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
                          *video_receiver_) != 0) {
-    return 0;
+    return;
   }
 
   if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
                          *voice_receiver) != 0) {
-    return 0;
+    return;
   }
 
   int relative_delay_ms;
   // Calculate how much later or earlier the audio stream is compared to video.
   if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
                                    &relative_delay_ms)) {
-    return 0;
+    return;
   }
 
   TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
   TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
   TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
   int target_audio_delay_ms = 0;
   int target_video_delay_ms = current_video_delay_ms;
   // Calculate the necessary extra audio delay and desired total video
   // delay to get the streams in sync.
   if (!sync_->ComputeDelays(relative_delay_ms,
                             current_audio_delay_ms,
                             &target_audio_delay_ms,
                             &target_video_delay_ms)) {
-    return 0;
+    return;
   }
 
   if (voe_sync_interface_->SetMinimumPlayoutDelay(
       voe_channel_id_, target_audio_delay_ms) == -1) {
     LOG(LS_ERROR) << "Error setting voice delay.";
   }
   vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
-  return 0;
 }
 
 }  // namespace webrtc