Assorted logging pedantry
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}diff --git a/media/base/rtpdataengine.cc b/media/base/rtpdataengine.cc
index 7cb5fa8..191645b 100644
--- a/media/base/rtpdataengine.cc
+++ b/media/base/rtpdataengine.cc
@@ -202,18 +202,11 @@
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 @@
}
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;
}
diff --git a/p2p/base/fakeicetransport.h b/p2p/base/fakeicetransport.h
index f3a8313..556a1cd 100644
--- a/p2p/base/fakeicetransport.h
+++ b/p2p/base/fakeicetransport.h
@@ -221,7 +221,7 @@
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);
diff --git a/p2p/base/p2ptransportchannel.cc b/p2p/base/p2ptransportchannel.cc
index 0e6deee..71af657 100644
--- a/p2p/base/p2ptransportchannel.cc
+++ b/p2p/base/p2ptransportchannel.cc
@@ -2237,8 +2237,7 @@
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);
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index 2f77aa7..5f5153f 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -1107,8 +1107,7 @@
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);
}
}
diff --git a/p2p/base/relayport.cc b/p2p/base/relayport.cc
index 045f0c3..a789a1b 100644
--- a/p2p/base/relayport.cc
+++ b/p2p/base/relayport.cc
@@ -678,7 +678,7 @@
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);
}
diff --git a/p2p/base/stunport.cc b/p2p/base/stunport.cc
index d450d8e..69196cf 100644
--- a/p2p/base/stunport.cc
+++ b/p2p/base/stunport.cc
@@ -517,7 +517,7 @@
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 {
diff --git a/pc/datachannel.cc b/pc/datachannel.cc
index 950fbea..525b6f9 100644
--- a/pc/datachannel.cc
+++ b/pc/datachannel.cc
@@ -151,7 +151,7 @@
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 @@
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 @@
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 @@
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 @@
// 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 @@
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;
diff --git a/pc/dtlssrtptransport.cc b/pc/dtlssrtptransport.cc
index bc5d3aa..03771f4 100644
--- a/pc/dtlssrtptransport.cc
+++ b/pc/dtlssrtptransport.cc
@@ -62,7 +62,7 @@
// 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;
diff --git a/pc/dtmfsender.cc b/pc/dtmfsender.cc
index 82644aa..7a98bc3 100644
--- a/pc/dtmfsender.cc
+++ b/pc/dtmfsender.cc
@@ -120,9 +120,10 @@
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;
}
diff --git a/pc/mediasession.cc b/pc/mediasession.cc
index a2f47dc..a1bc01f 100644
--- a/pc/mediasession.cc
+++ b/pc/mediasession.cc
@@ -468,8 +468,8 @@
} 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;
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index e565e57..14c2a88 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -802,14 +802,14 @@
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 @@
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 @@
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 @@
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 @@
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 @@
// 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 @@
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 @@
}
} 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 @@
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 @@
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 @@
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::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 @@
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 @@
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 @@
if (valid) {
RTC_LOG(LS_INFO)
<< "UseCandidatesInSessionDescription: Not ready to use "
- << "candidate.";
+ "candidate.";
}
continue;
}
diff --git a/pc/rtpsender.cc b/pc/rtpsender.cc
index 6095418..fb940e3 100644
--- a/pc/rtpsender.cc
+++ b/pc/rtpsender.cc
@@ -111,11 +111,11 @@
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, [&] {
diff --git a/pc/srtpfilter.cc b/pc/srtpfilter.cc
index 157fdeb..4fe9dae 100644
--- a/pc/srtpfilter.cc
+++ b/pc/srtpfilter.cc
@@ -184,7 +184,8 @@
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 @@
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 @@
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 @@
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;
}
diff --git a/pc/srtpsession.cc b/pc/srtpsession.cc
index a07848d..347b099 100644
--- a/pc/srtpsession.cc
+++ b/pc/srtpsession.cc
@@ -248,6 +248,7 @@
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 @@
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;
}
diff --git a/pc/srtptransport.cc b/pc/srtptransport.cc
index 149651c..625034d 100644
--- a/pc/srtptransport.cc
+++ b/pc/srtptransport.cc
@@ -228,9 +228,8 @@
}
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 @@
}
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;
}
diff --git a/pc/webrtcsdp.cc b/pc/webrtcsdp.cc
index 535ed23..e796f4c 100644
--- a/pc/webrtcsdp.cc
+++ b/pc/webrtcsdp.cc
@@ -1483,7 +1483,7 @@
} 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 @@
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 @@
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];
diff --git a/pc/webrtcsessiondescriptionfactory.cc b/pc/webrtcsessiondescriptionfactory.cc
index 36927bb..f6e5b96 100644
--- a/pc/webrtcsessiondescriptionfactory.cc
+++ b/pc/webrtcsessiondescriptionfactory.cc
@@ -170,8 +170,8 @@
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|.
diff --git a/rtc_base/asyncudpsocket.cc b/rtc_base/asyncudpsocket.cc
index 0896e50..5a50ae3 100644
--- a/rtc_base/asyncudpsocket.cc
+++ b/rtc_base/asyncudpsocket.cc
@@ -112,8 +112,8 @@
// 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;
}
diff --git a/rtc_base/httpcommon.cc b/rtc_base/httpcommon.cc
index 5f2112a..345b4aa 100644
--- a/rtc_base/httpcommon.cc
+++ b/rtc_base/httpcommon.cc
@@ -863,8 +863,6 @@
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) << "$$$ InitializeSecurityContext @ " <<
- // TimeSince(now);
if (FAILED(ret)) {
RTC_LOG(LS_ERROR) << "InitializeSecurityContext returned: "
<< ErrorName(ret, SECURITY_ERRORS);
@@ -931,7 +929,6 @@
ret = AcquireCredentialsHandleA(
0, const_cast<char*>(want_negotiate ? NEGOSSP_NAME_A : NTLMSP_NAME_A),
SECPKG_CRED_OUTBOUND, 0, pauth_id, 0, 0, &cred, &lifetime);
- // RTC_LOG(INFO) << "$$$ AcquireCredentialsHandle @ " << TimeSince(now);
if (ret != SEC_E_OK) {
RTC_LOG(LS_ERROR) << "AcquireCredentialsHandle error: "
<< ErrorName(ret, SECURITY_ERRORS);
@@ -942,7 +939,6 @@
CtxtHandle ctx;
ret = InitializeSecurityContextA(&cred, 0, spn, flags, 0, SECURITY_NATIVE_DREP, 0, 0, &ctx, &out_buf_desc, &ret_flags, &lifetime);
- // RTC_LOG(INFO) << "$$$ InitializeSecurityContext @ " << TimeSince(now);
if (FAILED(ret)) {
RTC_LOG(LS_ERROR) << "InitializeSecurityContext returned: "
<< ErrorName(ret, SECURITY_ERRORS);
@@ -958,7 +954,6 @@
if ((ret == SEC_I_COMPLETE_NEEDED) || (ret == SEC_I_COMPLETE_AND_CONTINUE)) {
ret = CompleteAuthToken(&neg->ctx, &out_buf_desc);
- // RTC_LOG(INFO) << "$$$ CompleteAuthToken @ " << TimeSince(now);
RTC_LOG(LS_VERBOSE) << "CompleteAuthToken returned: "
<< ErrorName(ret, SECURITY_ERRORS);
if (FAILED(ret)) {
@@ -966,8 +961,6 @@
}
}
- // RTC_LOG(INFO) << "$$$ NEGOTIATE took " << TimeSince(now) << "ms";
-
std::string decoded(out_buf, out_buf + out_sec.cbBuffer);
response = auth_method;
response.append(" ");
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index 91b71a6..6445fe1 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -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 @@
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
diff --git a/rtc_base/socketadapters.cc b/rtc_base/socketadapters.cc
index 4cde690..48e7898 100644
--- a/rtc_base/socketadapters.cc
+++ b/rtc_base/socketadapters.cc
@@ -104,7 +104,7 @@
}
if (data_len_ >= buffer_size_) {
- RTC_LOG(INFO) << "Input buffer overflow";
+ RTC_LOG(LS_ERROR) << "Input buffer overflow";
RTC_NOTREACHED();
data_len_ = 0;
}
