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Assorted logging pedantry

Browse source 
This cl fixes various minor issues found during a quick scan of the current log
usage.

Bug: webrtc:8529
Change-Id: I1e1eb02ef220177dbb327203509736ad7f70cc1c
Reviewed-on: https://webrtc-review.googlesource.com/52262
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21996}
This commit is contained in:
Jonas Olsson 2018-02-13 10:37:07 +01:00 committed by Commit Bot
parent a6cc0f94bf
commit 45cc890560
21 changed files with 67 additions and 83 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

13
media/base/rtpdataengine.cc
View file

@ -202,18 +202,11 @@ void RtpDataMediaChannel::OnPacketReceived(
rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
RtpHeader header;
if (!GetRtpHeader(packet->cdata(), packet->size(), &header)) {
// Don't want to log for every corrupt packet.
// RTC_LOG(LS_WARNING) << "Could not read rtp header from packet of length "
// << packet->length() << ".";
return;
}
size_t header_length;
if (!GetRtpHeaderLen(packet->cdata(), packet->size(), &header_length)) {
// Don't want to log for every corrupt packet.
// RTC_LOG(LS_WARNING) << "Could not read rtp header"
// << length from packet of length "
// << packet->length() << ".";
return;
}
const char* data =
@ -227,12 +220,6 @@ void RtpDataMediaChannel::OnPacketReceived(
}
if (!FindCodecById(recv_codecs_, header.payload_type)) {
// For bundling, this will be logged for every message.
// So disable this logging.
// RTC_LOG(LS_WARNING) << "Not receiving packet "
// << header.ssrc << ":" << header.seq_num
// << " (" << data_len << ")"
// << " because unknown payload id: " << header.payload_type;
return;
}

2
p2p/base/fakeicetransport.h
View file

@ -221,7 +221,7 @@ class FakeIceTransport : public IceTransportInternal {
if (writable_ == writable) {
return;
}
RTC_LOG(INFO) << "set_writable from:" << writable_ << " to " << writable;
RTC_LOG(INFO) << "Change writable_ to " << writable;
writable_ = writable;
if (writable_) {
SignalReadyToSend(this);

3
p2p/base/p2ptransportchannel.cc
View file

@ -2237,8 +2237,7 @@ void P2PTransportChannel::set_writable(bool writable) {
if (writable_ == writable) {
return;
}
LOG_J(LS_VERBOSE, this) << "set_writable from:" << writable_ << " to "
<< writable;
LOG_J(LS_VERBOSE, this) << "Changed writable_ to " << writable;
writable_ = writable;
if (writable_) {
SignalReadyToSend(this);

3
p2p/base/port.cc
View file

@ -1107,8 +1107,7 @@ void Connection::set_connected(bool value) {
bool old_value = connected_;
connected_ = value;
if (value != old_value) {
LOG_J(LS_VERBOSE, this) << "set_connected from: " << old_value << " to "
<< value;
LOG_J(LS_VERBOSE, this) << "Change connected_ to " << value;
SignalStateChange(this);
}
}

2
p2p/base/relayport.cc
View file

@ -678,7 +678,7 @@ void RelayEntry::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
void RelayEntry::OnSocketClose(rtc::AsyncPacketSocket* socket,
int error) {
RTC_PLOG(LERROR, error) << "Relay connection failed: socket closed";
RTC_LOG_ERR_EX(LERROR, error) << "Relay connection failed: socket closed";
HandleConnectFailure(socket);
}

2
p2p/base/stunport.cc
View file

@ -517,7 +517,7 @@ void UDPPort::OnSendPacket(const void* data, size_t size, StunRequest* req) {
StunBindingRequest* sreq = static_cast<StunBindingRequest*>(req);
rtc::PacketOptions options(DefaultDscpValue());
if (socket_->SendTo(data, size, sreq->server_addr(), options) < 0)
RTC_PLOG(LERROR, socket_->GetError()) << "sendto";
RTC_LOG_ERR_EX(LERROR, socket_->GetError()) << "sendto";
}
bool UDPPort::HasCandidateWithAddress(const rtc::SocketAddress& addr) const {

17
pc/datachannel.cc
View file

@ -151,7 +151,7 @@ bool DataChannel::Init(const InternalDataChannelInit& config) {
config.maxRetransmits != -1 ||
config.maxRetransmitTime != -1) {
RTC_LOG(LS_ERROR) << "Failed to initialize the RTP data channel due to "
<< "invalid DataChannelInit.";
"invalid DataChannelInit.";
return false;
}
handshake_state_ = kHandshakeReady;
@ -160,7 +160,7 @@ bool DataChannel::Init(const InternalDataChannelInit& config) {
config.maxRetransmits < -1 ||
config.maxRetransmitTime < -1) {
RTC_LOG(LS_ERROR) << "Failed to initialize the SCTP data channel due to "
<< "invalid DataChannelInit.";
"invalid DataChannelInit.";
return false;
}
if (config.maxRetransmits != -1 && config.maxRetransmitTime != -1) {
@ -344,8 +344,9 @@ void DataChannel::OnDataReceived(const cricket::ReceiveDataParams& params,
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
if (handshake_state_ != kHandshakeWaitingForAck) {
// Ignore it if we are not expecting an ACK message.
RTC_LOG(LS_WARNING) << "DataChannel received unexpected CONTROL message, "
<< "sid = " << params.sid;
RTC_LOG(LS_WARNING)
<< "DataChannel received unexpected CONTROL message, sid = "
<< params.sid;
return;
}
if (ParseDataChannelOpenAckMessage(payload)) {
@ -551,7 +552,7 @@ bool DataChannel::SendDataMessage(const DataBuffer& buffer,
send_params.ordered = true;
RTC_LOG(LS_VERBOSE)
<< "Sending data as ordered for unordered DataChannel "
<< "because the OPEN_ACK message has not been received.";
"because the OPEN_ACK message has not been received.";
}
send_params.max_rtx_count = config_.maxRetransmits;
@ -583,7 +584,8 @@ bool DataChannel::SendDataMessage(const DataBuffer& buffer,
// Close the channel if the error is not SDR_BLOCK, or if queuing the
// message failed.
RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to send data, "
<< "send_result = " << send_result;
"send_result = "
<< send_result;
Close();
return false;
@ -649,7 +651,8 @@ bool DataChannel::SendControlMessage(const rtc::CopyOnWriteBuffer& buffer) {
QueueControlMessage(buffer);
} else {
RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to send"
<< " the CONTROL message, send_result = " << send_result;
" the CONTROL message, send_result = "
<< send_result;
Close();
}
return retval;

2
pc/dtlssrtptransport.cc
View file

@ -62,7 +62,7 @@ void DtlsSrtpTransport::SetDtlsTransports(
// allowed according to the BUNDLE spec.
RTC_CHECK(!(IsActive()))
<< "Setting RTCP for DTLS/SRTP after the DTLS is active "
<< "should never happen.";
"should never happen.";
RTC_LOG(LS_INFO) << "Setting RTCP Transport on " << transport_name
<< " transport " << rtcp_dtls_transport;

7
pc/dtmfsender.cc
View file

@ -120,9 +120,10 @@ bool DtmfSender::InsertDtmf(const std::string& tones, int duration,
inter_tone_gap < kDtmfMinGapMs) {
RTC_LOG(LS_ERROR)
<< "InsertDtmf is called with invalid duration or tones gap. "
<< "The duration cannot be more than " << kDtmfMaxDurationMs
<< "ms or less than " << kDtmfMinDurationMs << "ms. "
<< "The gap between tones must be at least " << kDtmfMinGapMs << "ms.";
"The duration cannot be more than "
<< kDtmfMaxDurationMs << "ms or less than " << kDtmfMinDurationMs
<< "ms. The gap between tones must be at least "
<< kDtmfMinGapMs << "ms.";
return false;
}

4
pc/mediasession.cc
View file

@ -468,8 +468,8 @@ static bool AddStreamParams(
} else if (!ssrcs.empty()) {
RTC_LOG(LS_WARNING)
<< "Our FlexFEC implementation only supports protecting "
<< "a single media streams. This session has multiple "
<< "media streams however, so no FlexFEC SSRC will be generated.";
"a single media streams. This session has multiple "
"media streams however, so no FlexFEC SSRC will be generated.";
}
}
stream_param.cname = rtcp_cname;

37
pc/peerconnection.cc
View file

@ -802,14 +802,14 @@ bool PeerConnection::Initialize(
if (!allocator) {
RTC_LOG(LS_ERROR)
<< "PeerConnection initialized without a PortAllocator? "
<< "This shouldn't happen if using PeerConnectionFactory.";
"This shouldn't happen if using PeerConnectionFactory.";
return false;
}
if (!observer) {
// TODO(deadbeef): Why do we do this?
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
<< "PeerConnectionObserver";
"PeerConnectionObserver";
return false;
}
observer_ = observer;
@ -2593,7 +2593,7 @@ bool PeerConnection::AddIceCandidate(
if (!remote_description()) {
RTC_LOG(LS_ERROR) << "ProcessIceMessage: ICE candidates can't be added "
<< "without any remote session description.";
"without any remote session description.";
return false;
}
@ -2627,7 +2627,7 @@ bool PeerConnection::RemoveIceCandidates(
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
if (!remote_description()) {
RTC_LOG(LS_ERROR) << "RemoveRemoteIceCandidates: ICE candidates can't be "
<< "removed without any remote session description.";
"removed without any remote session description.";
return false;
}
@ -2641,7 +2641,8 @@ bool PeerConnection::RemoveIceCandidates(
if (number_removed != candidates.size()) {
RTC_LOG(LS_ERROR)
<< "RemoveRemoteIceCandidates: Failed to remove candidates. "
<< "Requested " << candidates.size() << " but only " << number_removed
"Requested "
<< candidates.size() << " but only " << number_removed
<< " are removed.";
}
@ -3833,7 +3834,7 @@ void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
if (sender->media_type() != media_type) {
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
<< " description with an unexpected media type.";
" description with an unexpected media type.";
return;
}
@ -3855,7 +3856,7 @@ void PeerConnection::OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
// match with the calls to CreateSender, AddStream and RemoveStream.
if (sender->media_type() != media_type) {
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
<< " description with an unexpected media type.";
" description with an unexpected media type.";
return;
}
@ -3944,7 +3945,7 @@ void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
InternalCreateDataChannel(label, nullptr));
if (!channel.get()) {
RTC_LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
<< "CreateDataChannel failed.";
"CreateDataChannel failed.";
return;
}
channel->SetReceiveSsrc(remote_ssrc);
@ -3977,7 +3978,7 @@ rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
}
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
RTC_LOG(LS_ERROR) << "Failed to create a SCTP data channel "
<< "because the id is already in use or out of range.";
"because the id is already in use or out of range.";
return nullptr;
}
}
@ -4313,12 +4314,12 @@ bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
<< "Local and Remote descriptions must be applied to get the "
<< "SSL Role of the SCTP transport.";
"SSL Role of the SCTP transport.";
return false;
}
if (!sctp_transport_) {
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
<< "SSL Role of the SCTP transport.";
"SSL Role of the SCTP transport.";
return false;
}
@ -4330,7 +4331,7 @@ bool PeerConnection::GetSslRole(const std::string& content_name,
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
<< "Local and Remote descriptions must be applied to get the "
<< "SSL Role of the session.";
"SSL Role of the session.";
return false;
}
@ -4681,7 +4682,7 @@ bool PeerConnection::SendData(const cricket::SendDataParams& params,
cricket::SendDataResult* result) {
if (!rtp_data_channel_ && !sctp_transport_) {
RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
<< "and sctp_transport_ are NULL.";
"and sctp_transport_ are NULL.";
return false;
}
return rtp_data_channel_
@ -4746,7 +4747,7 @@ void PeerConnection::AddSctpDataStream(int sid) {
void PeerConnection::RemoveSctpDataStream(int sid) {
if (!sctp_transport_) {
RTC_LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
<< "NULL.";
"NULL.";
return;
}
network_thread()->Invoke<void>(
@ -4861,12 +4862,12 @@ void PeerConnection::OnTransportControllerConnectionState(
break;
case cricket::kIceConnectionConnected:
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
<< "all transports are writable.";
"all transports are writable.";
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
break;
case cricket::kIceConnectionCompleted:
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
<< "all transports are complete.";
"all transports are complete.";
if (ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionConnected) {
// If jumping directly from "checking" to "connected",
@ -4914,7 +4915,7 @@ void PeerConnection::OnTransportControllerCandidatesRemoved(
for (const cricket::Candidate& candidate : candidates) {
if (candidate.transport_name().empty()) {
RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
<< "empty content name in candidate "
"empty content name in candidate "
<< candidate.ToString();
return;
}
@ -4984,7 +4985,7 @@ bool PeerConnection::UseCandidatesInSessionDescription(
if (valid) {
RTC_LOG(LS_INFO)
<< "UseCandidatesInSessionDescription: Not ready to use "
<< "candidate.";
"candidate.";
}
continue;
}

4
pc/rtpsender.cc
View file

@ -111,11 +111,11 @@ bool AudioRtpSender::CanInsertDtmf() {
bool AudioRtpSender::InsertDtmf(int code, int duration) {
if (!media_channel_) {
RTC_LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
RTC_LOG(LS_ERROR) << "InsertDtmf: No audio channel exists.";
return false;
}
if (!ssrc_) {
RTC_LOG(LS_ERROR) << "CanInsertDtmf: Sender does not have SSRC.";
RTC_LOG(LS_ERROR) << "InsertDtmf: Sender does not have SSRC.";
return false;
}
bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {

12
pc/srtpfilter.cc
View file

@ -184,7 +184,8 @@ bool SrtpFilter::ApplySendParams(const CryptoParams& send_params) {
send_cipher_suite_ = rtc::SrtpCryptoSuiteFromName(send_params.cipher_suite);
if (send_cipher_suite_ == rtc::SRTP_INVALID_CRYPTO_SUITE) {
RTC_LOG(LS_WARNING) << "Unknown crypto suite(s) received:"
<< " send cipher_suite " << send_params.cipher_suite;
" send cipher_suite "
<< send_params.cipher_suite;
return false;
}
@ -192,7 +193,8 @@ bool SrtpFilter::ApplySendParams(const CryptoParams& send_params) {
if (!rtc::GetSrtpKeyAndSaltLengths(*send_cipher_suite_, &send_key_len,
&send_salt_len)) {
RTC_LOG(LS_WARNING) << "Could not get lengths for crypto suite(s):"
<< " send cipher_suite " << send_params.cipher_suite;
" send cipher_suite "
<< send_params.cipher_suite;
return false;
}
@ -213,7 +215,8 @@ bool SrtpFilter::ApplyRecvParams(const CryptoParams& recv_params) {
recv_cipher_suite_ = rtc::SrtpCryptoSuiteFromName(recv_params.cipher_suite);
if (recv_cipher_suite_ == rtc::SRTP_INVALID_CRYPTO_SUITE) {
RTC_LOG(LS_WARNING) << "Unknown crypto suite(s) received:"
<< " recv cipher_suite " << recv_params.cipher_suite;
" recv cipher_suite "
<< recv_params.cipher_suite;
return false;
}
@ -221,7 +224,8 @@ bool SrtpFilter::ApplyRecvParams(const CryptoParams& recv_params) {
if (!rtc::GetSrtpKeyAndSaltLengths(*recv_cipher_suite_, &recv_key_len,
&recv_salt_len)) {
RTC_LOG(LS_WARNING) << "Could not get lengths for crypto suite(s):"
<< " recv cipher_suite " << recv_params.cipher_suite;
" recv cipher_suite "
<< recv_params.cipher_suite;
return false;
}

3
pc/srtpsession.cc
View file

@ -248,6 +248,7 @@ bool SrtpSession::DoSetKey(int type,
if (!rtc::GetSrtpKeyAndSaltLengths(cs, &expected_key_len,
&expected_salt_len)) {
// This should never happen.
RTC_NOTREACHED();
RTC_LOG(LS_WARNING)
<< "Failed to " << (session_ ? "update" : "create")
<< " SRTP session: unsupported cipher_suite without length information"
@ -314,7 +315,7 @@ bool SrtpSession::SetKey(int type,
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (session_) {
RTC_LOG(LS_ERROR) << "Failed to create SRTP session: "
<< "SRTP session already created";
"SRTP session already created";
return false;
}

9
pc/srtptransport.cc
View file

@ -228,9 +228,8 @@ bool SrtpTransport::SetRtpParams(int send_cs,
}
RTC_LOG(LS_INFO) << "SRTP " << (new_sessions ? "activated" : "updated")
<< " with negotiated parameters:"
<< " send cipher_suite " << send_cs << " recv cipher_suite "
<< recv_cs;
<< " with negotiated parameters: send cipher_suite "
<< send_cs << " recv cipher_suite " << recv_cs;
return true;
}
@ -262,8 +261,8 @@ bool SrtpTransport::SetRtcpParams(int send_cs,
}
RTC_LOG(LS_INFO) << "SRTCP activated with negotiated parameters:"
<< " send cipher_suite " << send_cs << " recv cipher_suite "
<< recv_cs;
" send cipher_suite "
<< send_cs << " recv cipher_suite " << recv_cs;
return true;
}

9
pc/webrtcsdp.cc
View file

@ -1483,7 +1483,7 @@ void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
} else if (streams.size() > 1u) {
RTC_LOG(LS_WARNING)
<< "Trying to serialize Unified Plan SDP with more than "
<< "one track in a media section. Omitting 'a=msid'.";
"one track in a media section. Omitting 'a=msid'.";
}
}
@ -2459,8 +2459,8 @@ bool ParseMediaDescription(const std::string& message,
bundle_only = false;
RTC_LOG(LS_WARNING)
<< "a=bundle-only attribute observed with a nonzero "
<< "port; this usage is unspecified so the attribute is being "
<< "ignored.";
"port; this usage is unspecified so the attribute is being "
"ignored.";
}
} else {
// If not using bundle-only, interpret port 0 in the normal way; the m=
@ -3176,7 +3176,8 @@ bool ParseRtpmapAttribute(const std::string& line,
if (std::find(payload_types.begin(), payload_types.end(), payload_type) ==
payload_types.end()) {
RTC_LOG(LS_WARNING) << "Ignore rtpmap line that did not appear in the "
<< "<fmt> of the m-line: " << line;
"<fmt> of the m-line: "
<< line;
return true;
}
const std::string& encoder = fields[1];

4
pc/webrtcsessiondescriptionfactory.cc
View file

@ -170,8 +170,8 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
rtc::KeyParams key_params = rtc::KeyParams();
RTC_LOG(LS_VERBOSE)
<< "DTLS-SRTP enabled; sending DTLS identity request (key "
<< "type: " << key_params.type() << ").";
<< "DTLS-SRTP enabled; sending DTLS identity request (key type: "
<< key_params.type() << ").";
// Request certificate. This happens asynchronously, so that the caller gets
// a chance to connect to |SignalCertificateReady|.

4
rtc_base/asyncudpsocket.cc
View file

@ -112,8 +112,8 @@ void AsyncUDPSocket::OnReadEvent(AsyncSocket* socket) {
// TODO: Do something better like forwarding the error to the user.
SocketAddress local_addr = socket_->GetLocalAddress();
RTC_LOG(LS_INFO) << "AsyncUDPSocket[" << local_addr.ToSensitiveString()
<< "] "
<< "receive failed with error " << socket_->GetError();
<< "] receive failed with error "
<< socket_->GetError();
return;
}

7
rtc_base/httpcommon.cc
View file

@ -863,8 +863,6 @@ HttpAuthResult HttpAuthenticate(
in_buf_desc.pBuffers = &in_sec;
ret = InitializeSecurityContextA(&neg->cred, &neg->ctx, spn, flags, 0, SECURITY_NATIVE_DREP, &in_buf_desc, 0, &neg->ctx, &out_buf_desc, &ret_flags, &lifetime);
// RTC_LOG(INFO) << "$$$ InitializeSecurityContext @ " <<
// TimeSince(now);
if (FAILED(ret)) {
RTC_LOG(LS_ERROR) << "InitializeSecurityContext returned: "
<< ErrorName(ret, SECURITY_ERRORS);
@ -931,7 +929,6 @@ HttpAuthResult HttpAuthenticate(
ret = AcquireCredentialsHandleA(
0, const_cast<char*>(want_negotiate ? NEGOSSP_NAME_A : NTLMSP_NAME_A),
SECPKG_CRED_OUTBOUND, 0, pauth_id, 0, 0, &cred, &lifetime);
// RTC_LOG(INFO) << "$$$ AcquireCredentialsHandle @ " << TimeSince(now);
if (ret != SEC_E_OK) {
RTC_LOG(LS_ERROR) << "AcquireCredentialsHandle error: "
<< ErrorName(ret, SECURITY_ERRORS);
@ -942,7 +939,6 @@ HttpAuthResult HttpAuthenticate(
CtxtHandle ctx;
ret = InitializeSecurityContextA(&cred, 0, spn, flags, 0, SECURITY_NATIVE_DREP, 0, 0, &ctx, &out_buf_desc, &ret_flags, &lifetime);
// RTC_LOG(INFO) << "$$$ InitializeSecurityContext @ " << TimeSince(now);
if (FAILED(ret)) {
RTC_LOG(LS_ERROR) << "InitializeSecurityContext returned: "
<< ErrorName(ret, SECURITY_ERRORS);
@ -958,7 +954,6 @@ HttpAuthResult HttpAuthenticate(
if ((ret == SEC_I_COMPLETE_NEEDED) || (ret == SEC_I_COMPLETE_AND_CONTINUE)) {
ret = CompleteAuthToken(&neg->ctx, &out_buf_desc);
// RTC_LOG(INFO) << "$$$ CompleteAuthToken @ " << TimeSince(now);
RTC_LOG(LS_VERBOSE) << "CompleteAuthToken returned: "
<< ErrorName(ret, SECURITY_ERRORS);
if (FAILED(ret)) {
@ -966,8 +961,6 @@ HttpAuthResult HttpAuthenticate(
}
}
// RTC_LOG(INFO) << "$$$ NEGOTIATE took " << TimeSince(now) << "ms";
std::string decoded(out_buf, out_buf + out_sec.cbBuffer);
response = auth_method;
response.append(" ");

4
rtc_base/logging.h
View file

@ -41,7 +41,6 @@
// RTC_LOG_CHECK_LEVEL(sev) (and RTC_LOG_CHECK_LEVEL_V(sev)) can be used as a
// test before performing expensive or sensitive operations whose sole
// purpose is to output logging data at the desired level.
// Lastly, RTC_PLOG(sev, err) is an alias for RTC_LOG_ERR_EX.
#ifndef RTC_BASE_LOGGING_H_
#define RTC_BASE_LOGGING_H_
@ -343,9 +342,6 @@ inline bool LogCheckLevel(LoggingSeverity sev) {
RTC_LOG_SEVERITY_PRECONDITION(sev) \
rtc::LogMessage(nullptr, 0, sev, tag).stream()
#define RTC_PLOG(sev, err) \
RTC_LOG_ERR_EX(sev, err)
// The RTC_DLOG macros are equivalent to their RTC_LOG counterparts except that
// they only generate code in debug builds.
#if RTC_DLOG_IS_ON

2
rtc_base/socketadapters.cc
View file

@ -104,7 +104,7 @@ void BufferedReadAdapter::OnReadEvent(AsyncSocket * socket) {
}
if (data_len_ >= buffer_size_) {
RTC_LOG(INFO) << "Input buffer overflow";
RTC_LOG(LS_ERROR) << "Input buffer overflow";
RTC_NOTREACHED();
data_len_ = 0;
}