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/channel_send.cc b/audio/channel_send.cc
index 0c8be1f..8269fd5 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -103,7 +103,7 @@
~ChannelSend() override;
// Send using this encoder, with this payload type.
- bool SetEncoder(int payload_type,
+ void SetEncoder(int payload_type,
std::unique_ptr<AudioEncoder> encoder) override;
void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
modifier) override;
@@ -118,7 +118,7 @@
int GetBitrate() const override;
// Network
- bool ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
+ void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
// Muting, Volume and Level.
void SetInputMute(bool enable) override;
@@ -131,7 +131,7 @@
// DTMF.
bool SendTelephoneEventOutband(int event, int duration_ms) override;
- bool SetSendTelephoneEventPayloadType(int payload_type,
+ void SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) override;
// RTP+RTCP
@@ -789,7 +789,7 @@
_rtpRtcpModule->SetSendingMediaStatus(false);
}
-bool ChannelSend::SetEncoder(int payload_type,
+void ChannelSend::SetEncoder(int payload_type,
std::unique_ptr<AudioEncoder> encoder) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_GE(payload_type, 0);
@@ -811,7 +811,6 @@
media_transport_sampling_frequency_ = encoder->RtpTimestampRateHz();
}
audio_coding_->SetEncoder(std::move(encoder));
- return true;
}
void ChannelSend::ModifyEncoder(
@@ -879,13 +878,12 @@
});
}
-// TODO(nisse): Delete always-true return value.
-bool ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
+void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
// May be called on either worker thread or network thread.
if (media_transport_) {
// Ignore RTCP packets while media transport is used.
// Those packets should not arrive, but we are seeing occasional packets.
- return 0;
+ return;
}
// Deliver RTCP packet to RTP/RTCP module for parsing
@@ -894,7 +892,7 @@
int64_t rtt = GetRTT();
if (rtt == 0) {
// Waiting for valid RTT.
- return true;
+ return;
}
int64_t nack_window_ms = rtt;
@@ -906,7 +904,6 @@
retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
OnReceivedRtt(rtt);
- return true;
}
void ChannelSend::SetInputMute(bool enable) {
@@ -937,14 +934,13 @@
return true;
}
-bool ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
+void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
RTC_DCHECK_LE(0, payload_type);
RTC_DCHECK_GE(127, payload_type);
_rtpRtcpModule->RegisterAudioSendPayload(payload_type, "telephone-event",
payload_frequency, 0, 0);
- return true;
}
void ChannelSend::SetLocalSSRC(uint32_t ssrc) {