Declare LERROR deprecated and remove all usage in webrtc

Bug: webrtc:13362
Change-Id: I1c6c6eccd950d73be616b34f96db7832ff94377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238804
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35416}
This commit is contained in:
Harald Alvestrand 2021-11-24 10:01:32 +00:00 committed by WebRTC LUCI CQ
parent 4e8e36c745
commit 5f34130f26
10 changed files with 28 additions and 30 deletions

View file

@ -64,7 +64,7 @@ rtc::IPAddress QueryDefaultLocalAddress(int family) {
std::unique_ptr<rtc::Socket> socket(
thread->socketserver()->CreateSocket(family, SOCK_DGRAM));
if (!socket) {
RTC_LOG_ERR(LERROR) << "Socket creation failed";
RTC_LOG_ERR(LS_ERROR) << "Socket creation failed";
return rtc::IPAddress();
}
@ -308,7 +308,7 @@ void AndroidVoipClient::StartSession(JNIEnv* env) {
rtp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(),
rtp_local_address_));
if (!rtp_socket_) {
RTC_LOG_ERR(LERROR) << "Socket creation failed";
RTC_LOG_ERR(LS_ERROR) << "Socket creation failed";
Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_,
/*isSuccessful=*/false);
return;
@ -319,7 +319,7 @@ void AndroidVoipClient::StartSession(JNIEnv* env) {
rtcp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(),
rtcp_local_address_));
if (!rtcp_socket_) {
RTC_LOG_ERR(LERROR) << "Socket creation failed";
RTC_LOG_ERR(LS_ERROR) << "Socket creation failed";
Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_,
/*isSuccessful=*/false);
return;

View file

@ -269,7 +269,7 @@ void SimplePeerConnection::OnSuccess(
}
void SimplePeerConnection::OnFailure(webrtc::RTCError error) {
RTC_LOG(LERROR) << ToString(error.type()) << ": " << error.message();
RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
// TODO(hta): include error.type in the message
if (OnFailureMessage)

View file

@ -66,14 +66,14 @@ AcmReceiver::~AcmReceiver() = default;
int AcmReceiver::SetMinimumDelay(int delay_ms) {
if (neteq_->SetMinimumDelay(delay_ms))
return 0;
RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
return -1;
}
int AcmReceiver::SetMaximumDelay(int delay_ms) {
if (neteq_->SetMaximumDelay(delay_ms))
return 0;
RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
return -1;
}
@ -134,9 +134,9 @@ int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
} // `mutex_` is released.
if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
RTC_LOG(LERROR) << "AcmReceiver::InsertPacket "
<< static_cast<int>(rtp_header.payloadType)
<< " Failed to insert packet";
RTC_LOG(LS_ERROR) << "AcmReceiver::InsertPacket "
<< static_cast<int>(rtp_header.payloadType)
<< " Failed to insert packet";
return -1;
}
return 0;
@ -150,7 +150,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
int current_sample_rate_hz = 0;
if (neteq_->GetAudio(audio_frame, muted, &current_sample_rate_hz) !=
NetEq::kOK) {
RTC_LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
return -1;
}
@ -170,8 +170,8 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
temp_output);
if (samples_per_channel_int < 0) {
RTC_LOG(LERROR) << "AcmReceiver::GetAudio - "
"Resampling last_audio_buffer_ failed.";
RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - "
"Resampling last_audio_buffer_ failed.";
return -1;
}
}
@ -185,7 +185,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
audio_frame->mutable_data());
if (samples_per_channel_int < 0) {
RTC_LOG(LERROR)
RTC_LOG(LS_ERROR)
<< "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
return -1;
}

View file

@ -142,7 +142,7 @@ int32_t AudioDeviceModuleImpl::CheckPlatform() {
RTC_LOG(LS_INFO) << "current platform is Mac";
#endif
if (platform == kPlatformNotSupported) {
RTC_LOG(LERROR)
RTC_LOG(LS_ERROR)
<< "current platform is not supported => this module will self "
"destruct!";
return -1;
@ -546,7 +546,7 @@ int32_t AudioDeviceModuleImpl::SetStereoRecording(bool enable) {
RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->RecordingIsInitialized()) {
RTC_LOG(LERROR)
RTC_LOG(LS_ERROR)
<< "unable to set stereo mode after recording is initialized";
return -1;
}
@ -592,7 +592,7 @@ int32_t AudioDeviceModuleImpl::SetStereoPlayout(bool enable) {
RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->PlayoutIsInitialized()) {
RTC_LOG(LERROR)
RTC_LOG(LS_ERROR)
<< "unable to set stereo mode while playing side is initialized";
return -1;
}
@ -856,7 +856,7 @@ int32_t AudioDeviceModuleImpl::PlayoutDelay(uint16_t* delayMS) const {
CHECKinitialized_();
uint16_t delay = 0;
if (audio_device_->PlayoutDelay(delay) == -1) {
RTC_LOG(LERROR) << "failed to retrieve the playout delay";
RTC_LOG(LS_ERROR) << "failed to retrieve the playout delay";
return -1;
}
*delayMS = delay;

View file

@ -577,7 +577,7 @@ void UDPPort::OnSendPacket(const void* data, size_t size, StunRequest* req) {
options.info_signaled_after_sent.packet_type = rtc::PacketType::kStunMessage;
CopyPortInformationToPacketInfo(&options.info_signaled_after_sent);
if (socket_->SendTo(data, size, sreq->server_addr(), options) < 0) {
RTC_LOG_ERR_EX(LERROR, socket_->GetError())
RTC_LOG_ERR_EX(LS_ERROR, socket_->GetError())
<< "UDP send of " << size << " bytes to host "
<< sreq->server_addr().ToSensitiveString() << " ("
<< sreq->server_addr().ToResolvedSensitiveString()

View file

@ -147,9 +147,9 @@ bool StunRequestManager::CheckResponse(StunMessage* msg) {
} else if (msg->type() == GetStunErrorResponseType(request->type())) {
request->OnErrorResponse(msg);
} else {
RTC_LOG(LERROR) << "Received response with wrong type: " << msg->type()
<< " (expecting "
<< GetStunSuccessResponseType(request->type()) << ")";
RTC_LOG(LS_ERROR) << "Received response with wrong type: " << msg->type()
<< " (expecting "
<< GetStunSuccessResponseType(request->type()) << ")";
return false;
}

View file

@ -94,8 +94,7 @@ enum LoggingSeverity {
INFO [[deprecated("Use LS_INFO")]] = LS_INFO,
// WARNING [[deprecated("Use LS_WARNING")]] = LS_WARNING,
WARNING = LS_WARNING,
// LERROR [[deprecated("Use LS_ERROR")]] = LS_ERROR
LERROR = LS_ERROR
LERROR [[deprecated("Use LS_ERROR")]] = LS_ERROR
};
// LogErrorContext assists in interpreting the meaning of an error value.

View file

@ -652,8 +652,8 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
struct ifaddrs* interfaces;
int error = getifaddrs(&interfaces);
if (error != 0) {
RTC_LOG_ERR(LERROR) << "getifaddrs failed to gather interface data: "
<< error;
RTC_LOG_ERR(LS_ERROR) << "getifaddrs failed to gather interface data: "
<< error;
return false;
}
@ -983,7 +983,7 @@ IPAddress BasicNetworkManager::QueryDefaultLocalAddress(int family) const {
std::unique_ptr<Socket> socket(
socket_factory->CreateSocket(family, SOCK_DGRAM));
if (!socket) {
RTC_LOG_ERR(LERROR) << "Socket creation failed";
RTC_LOG_ERR(LS_ERROR) << "Socket creation failed";
return IPAddress();
}

View file

@ -939,7 +939,7 @@ class Signaler : public Dispatcher {
std::array<int, 2> afd = {-1, -1};
if (pipe(afd.data()) < 0) {
RTC_LOG(LERROR) << "pipe failed";
RTC_LOG(LS_ERROR) << "pipe failed";
}
return afd;
}()),

View file

@ -324,8 +324,7 @@ AudioDeviceModuleIOS::AudioDeviceModuleIOS(bool bypass_voice_processing)
RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->PlayoutIsInitialized()) {
RTC_LOG(LERROR)
<< "unable to set stereo mode while playing side is initialized";
RTC_LOG(LS_ERROR) << "unable to set stereo mode while playing side is initialized";
return -1;
}
if (audio_device_->SetStereoPlayout(enable)) {
@ -588,7 +587,7 @@ AudioDeviceModuleIOS::AudioDeviceModuleIOS(bool bypass_voice_processing)
CHECKinitialized_();
uint16_t delay = 0;
if (audio_device_->PlayoutDelay(delay) == -1) {
RTC_LOG(LERROR) << "failed to retrieve the playout delay";
RTC_LOG(LS_ERROR) << "failed to retrieve the playout delay";
return -1;
}
*delayMS = delay;