Delete a few return values from audio streams and video send streams.
Bug: webrtc:10198
Change-Id: I583dbb717aea26c9d282a3786062d285121fbf66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125723
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26986}diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index bb79149..8157e6a 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -353,9 +353,9 @@
int event,
int duration_ms) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- return channel_send_->SetSendTelephoneEventPayloadType(payload_type,
- payload_frequency) &&
- channel_send_->SendTelephoneEventOutband(event, duration_ms);
+ channel_send_->SetSendTelephoneEventPayloadType(payload_type,
+ payload_frequency);
+ return channel_send_->SendTelephoneEventOutband(event, duration_ms);
}
void AudioSendStream::SetMuted(bool muted) {
@@ -422,12 +422,12 @@
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
}
-bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+void AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
- return channel_send_->ReceivedRTCPPacket(packet, length);
+ channel_send_->ReceivedRTCPPacket(packet, length);
}
uint32_t AudioSendStream::OnBitrateUpdated(BitrateAllocationUpdate update) {
