From 45cc890560ef5aa678753cb07b59d0b299a8841a Mon Sep 17 00:00:00 2001 From: Jonas Olsson Date: Tue, 13 Feb 2018 10:37:07 +0100 Subject: [PATCH] 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 Reviewed-by: Fredrik Solenberg Reviewed-by: Karl Wiberg Reviewed-by: Henrik Grunell Cr-Commit-Position: refs/heads/master@{#21996} --- media/base/rtpdataengine.cc | 13 ---------- p2p/base/fakeicetransport.h | 2 +- p2p/base/p2ptransportchannel.cc | 3 +-- p2p/base/port.cc | 3 +-- p2p/base/relayport.cc | 2 +- p2p/base/stunport.cc | 2 +- pc/datachannel.cc | 17 +++++++----- pc/dtlssrtptransport.cc | 2 +- pc/dtmfsender.cc | 7 ++--- pc/mediasession.cc | 4 +-- pc/peerconnection.cc | 37 ++++++++++++++------------- pc/rtpsender.cc | 4 +-- pc/srtpfilter.cc | 12 ++++++--- pc/srtpsession.cc | 3 ++- pc/srtptransport.cc | 9 +++---- pc/webrtcsdp.cc | 9 ++++--- pc/webrtcsessiondescriptionfactory.cc | 4 +-- rtc_base/asyncudpsocket.cc | 4 +-- rtc_base/httpcommon.cc | 7 ----- rtc_base/logging.h | 4 --- rtc_base/socketadapters.cc | 2 +- 21 files changed, 67 insertions(+), 83 deletions(-) diff --git a/media/base/rtpdataengine.cc b/media/base/rtpdataengine.cc index 7cb5fa8585..191645b28a 100644 --- a/media/base/rtpdataengine.cc +++ b/media/base/rtpdataengine.cc @@ -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; } diff --git a/p2p/base/fakeicetransport.h b/p2p/base/fakeicetransport.h index f3a8313792..556a1cda03 100644 --- a/p2p/base/fakeicetransport.h +++ b/p2p/base/fakeicetransport.h @@ -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); diff --git a/p2p/base/p2ptransportchannel.cc b/p2p/base/p2ptransportchannel.cc index 0e6deeecbf..71af657c3b 100644 --- a/p2p/base/p2ptransportchannel.cc +++ b/p2p/base/p2ptransportchannel.cc @@ -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); diff --git a/p2p/base/port.cc b/p2p/base/port.cc index 2f77aa7cc9..5f5153facb 100644 --- a/p2p/base/port.cc +++ b/p2p/base/port.cc @@ -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); } } diff --git a/p2p/base/relayport.cc b/p2p/base/relayport.cc index 045f0c3644..a789a1b16b 100644 --- a/p2p/base/relayport.cc +++ b/p2p/base/relayport.cc @@ -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); } diff --git a/p2p/base/stunport.cc b/p2p/base/stunport.cc index d450d8e94d..69196cf130 100644 --- a/p2p/base/stunport.cc +++ b/p2p/base/stunport.cc @@ -517,7 +517,7 @@ void UDPPort::OnSendPacket(const void* data, size_t size, StunRequest* req) { StunBindingRequest* sreq = static_cast(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 950fbeaea8..525b6f9508 100644 --- a/pc/datachannel.cc +++ b/pc/datachannel.cc @@ -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; diff --git a/pc/dtlssrtptransport.cc b/pc/dtlssrtptransport.cc index bc5d3aa900..03771f4988 100644 --- a/pc/dtlssrtptransport.cc +++ b/pc/dtlssrtptransport.cc @@ -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; diff --git a/pc/dtmfsender.cc b/pc/dtmfsender.cc index 82644aacf0..7a98bc32dd 100644 --- a/pc/dtmfsender.cc +++ b/pc/dtmfsender.cc @@ -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; } diff --git a/pc/mediasession.cc b/pc/mediasession.cc index a2f47dcb27..a1bc01f03e 100644 --- a/pc/mediasession.cc +++ b/pc/mediasession.cc @@ -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; diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc index e565e572a7..14c2a88d3a 100644 --- a/pc/peerconnection.cc +++ b/pc/peerconnection.cc @@ -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 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( @@ -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; } diff --git a/pc/rtpsender.cc b/pc/rtpsender.cc index 6095418788..fb940e3438 100644 --- a/pc/rtpsender.cc +++ b/pc/rtpsender.cc @@ -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(RTC_FROM_HERE, [&] { diff --git a/pc/srtpfilter.cc b/pc/srtpfilter.cc index 157fdebff3..4fe9dae00b 100644 --- a/pc/srtpfilter.cc +++ b/pc/srtpfilter.cc @@ -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; } diff --git a/pc/srtpsession.cc b/pc/srtpsession.cc index a07848d475..347b099a96 100644 --- a/pc/srtpsession.cc +++ b/pc/srtpsession.cc @@ -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; } diff --git a/pc/srtptransport.cc b/pc/srtptransport.cc index 149651cdda..625034dfab 100644 --- a/pc/srtptransport.cc +++ b/pc/srtptransport.cc @@ -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; } diff --git a/pc/webrtcsdp.cc b/pc/webrtcsdp.cc index 535ed23082..e796f4c87c 100644 --- a/pc/webrtcsdp.cc +++ b/pc/webrtcsdp.cc @@ -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 " - << " of the m-line: " << line; + " of the m-line: " + << line; return true; } const std::string& encoder = fields[1]; diff --git a/pc/webrtcsessiondescriptionfactory.cc b/pc/webrtcsessiondescriptionfactory.cc index 36927bba0c..f6e5b96438 100644 --- a/pc/webrtcsessiondescriptionfactory.cc +++ b/pc/webrtcsessiondescriptionfactory.cc @@ -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|. diff --git a/rtc_base/asyncudpsocket.cc b/rtc_base/asyncudpsocket.cc index 0896e50f49..5a50ae3855 100644 --- a/rtc_base/asyncudpsocket.cc +++ b/rtc_base/asyncudpsocket.cc @@ -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; } diff --git a/rtc_base/httpcommon.cc b/rtc_base/httpcommon.cc index 5f2112ac1a..345b4aa28b 100644 --- a/rtc_base/httpcommon.cc +++ b/rtc_base/httpcommon.cc @@ -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) << "$$$ InitializeSecurityContext @ " << - // TimeSince(now); if (FAILED(ret)) { RTC_LOG(LS_ERROR) << "InitializeSecurityContext returned: " << ErrorName(ret, SECURITY_ERRORS); @@ -931,7 +929,6 @@ HttpAuthResult HttpAuthenticate( ret = AcquireCredentialsHandleA( 0, const_cast(want_negotiate ? NEGOSSP_NAME_A : NTLMSP_NAME_A), SECPKG_CRED_OUTBOUND, 0, pauth_id, 0, 0, &cred, &lifetime); - // RTC_LOG(INFO) << "$$$ 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(" "); diff --git a/rtc_base/logging.h b/rtc_base/logging.h index 91b71a63e6..6445fe1263 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 @@ 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 diff --git a/rtc_base/socketadapters.cc b/rtc_base/socketadapters.cc index 4cde690782..48e7898d60 100644 --- a/rtc_base/socketadapters.cc +++ b/rtc_base/socketadapters.cc @@ -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; }