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Unified Diff: webrtc/video/vie_sync_module.cc

Issue 1703833002: Remove ignored return code from modules. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 10 months ago
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Index: webrtc/video/vie_sync_module.cc
diff --git a/webrtc/video/vie_sync_module.cc b/webrtc/video/vie_sync_module.cc
index dc924fb62fa7e4badbf8285e74ba7af7bcf7987a..f8376e53d1f6d0bf49b715ad09596bfd7201f556 100644
--- a/webrtc/video/vie_sync_module.cc
+++ b/webrtc/video/vie_sync_module.cc
@@ -88,14 +88,14 @@ int64_t ViESyncModule::TimeUntilNextProcess() {
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());
@@ -105,7 +105,7 @@ int32_t ViESyncModule::Process() {
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;
@@ -114,26 +114,26 @@ int32_t ViESyncModule::Process() {
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);
@@ -147,7 +147,7 @@ int32_t ViESyncModule::Process() {
current_audio_delay_ms,
&target_audio_delay_ms,
&target_video_delay_ms)) {
- return 0;
+ return;
}
if (voe_sync_interface_->SetMinimumPlayoutDelay(
@@ -155,7 +155,6 @@ int32_t ViESyncModule::Process() {
LOG(LS_ERROR) << "Error setting voice delay.";
}
vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
- return 0;
}
} // namespace webrtc
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