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Convert common error and warning logs to info severity

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This commit is contained in:
Jim Gustafson 2022-03-15 18:34:43 -07:00 committed by GitHub
parent 3996f98d35
commit ad8014f0fd
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GPG key ID: 4AEE18F83AFDEB23
36 changed files with 125 additions and 122 deletions
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8
audio/audio_send_stream.cc
View file

@ -886,10 +886,10 @@ void AudioSendStream::RemoveBitrateObserver() {
absl::optional<AudioSendStream::TargetAudioBitrateConstraints>
AudioSendStream::GetMinMaxBitrateConstraints() const {
if (config_.min_bitrate_bps < 0 || config_.max_bitrate_bps < 0) {
RTC_LOG(LS_WARNING) << "Config is invalid: min_bitrate_bps="
<< config_.min_bitrate_bps
<< "; max_bitrate_bps=" << config_.max_bitrate_bps
<< "; both expected greater or equal to 0";
RTC_LOG(LS_INFO) << "Config is invalid: min_bitrate_bps="
<< config_.min_bitrate_bps
<< "; max_bitrate_bps=" << config_.max_bitrate_bps
<< "; both expected greater or equal to 0";
return absl::nullopt;
}
TargetAudioBitrateConstraints constraints{

4
call/rtp_video_sender.cc
View file

@ -104,7 +104,7 @@ bool ShouldDisableRedAndUlpfec(bool flexfec_enabled,
// Note that this is not the case with FlexFEC.
if (nack_enabled && IsUlpfecEnabled() &&
!PayloadTypeSupportsSkippingFecPackets(rtp_config.payload_name, trials)) {
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "Transmitting payload type without picture ID using "
"NACK+ULPFEC is a waste of bandwidth since ULPFEC packets "
"also have to be retransmitted. Disabling ULPFEC.";
@ -113,7 +113,7 @@ bool ShouldDisableRedAndUlpfec(bool flexfec_enabled,
// Verify payload types.
if (IsUlpfecEnabled() ^ IsRedEnabled()) {
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "Only RED or only ULPFEC enabled, but not both. Disabling both.";
should_disable_red_and_ulpfec = true;
}

8
common_video/video_render_frames.cc
View file

@ -55,7 +55,7 @@ int32_t VideoRenderFrames::AddFrame(VideoFrame&& new_frame) {
// really slow system never renders any frames.
if (!incoming_frames_.empty() &&
new_frame.render_time_ms() + kOldRenderTimestampMS < time_now) {
RTC_LOG(LS_WARNING) << "Too old frame, timestamp=" << new_frame.timestamp();
RTC_LOG(LS_INFO) << "Too old frame, timestamp=" << new_frame.timestamp();
++frames_dropped_;
return -1;
}
@ -68,9 +68,9 @@ int32_t VideoRenderFrames::AddFrame(VideoFrame&& new_frame) {
}
if (new_frame.render_time_ms() < last_render_time_ms_) {
RTC_LOG(LS_WARNING) << "Frame scheduled out of order, render_time="
<< new_frame.render_time_ms()
<< ", latest=" << last_render_time_ms_;
RTC_LOG(LS_INFO) << "Frame scheduled out of order, render_time="
<< new_frame.render_time_ms()
<< ", latest=" << last_render_time_ms_;
// For more details, see bug:
// https://bugs.chromium.org/p/webrtc/issues/detail?id=7253
++frames_dropped_;

4
media/engine/adm_helpers.cc
View file

@ -59,7 +59,7 @@ void Init(AudioDeviceModule* adm) {
RTC_LOG(LS_ERROR) << "Failed to query stereo playout.";
}
if (adm->SetStereoPlayout(available) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set stereo playout mode.";
RTC_LOG(LS_INFO) << "Failed to set stereo playout mode.";
}
}
@ -79,7 +79,7 @@ void Init(AudioDeviceModule* adm) {
RTC_LOG(LS_ERROR) << "Failed to query stereo recording.";
}
if (adm->SetStereoRecording(available) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set stereo recording mode.";
RTC_LOG(LS_INFO) << "Failed to set stereo recording mode.";
}
}
}

4
media/engine/simulcast.cc
View file

@ -313,8 +313,8 @@ size_t LimitSimulcastLayerCount(int width,
enable_lowres_bitrate_interpolation)
.max_layers);
if (layer_count > adaptive_layer_count) {
RTC_LOG(LS_WARNING) << "Reducing simulcast layer count from "
<< layer_count << " to " << adaptive_layer_count;
RTC_LOG(LS_INFO) << "Reducing simulcast layer count from "
<< layer_count << " to " << adaptive_layer_count;
layer_count = adaptive_layer_count;
}
}

8
media/engine/webrtc_video_engine.cc
View file

@ -3108,7 +3108,7 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::OnFrame(
estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms;
if (sink_ == NULL) {
RTC_LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoSink.";
RTC_LOG(LS_INFO) << "VideoReceiveStream not connected to a VideoSink.";
return;
}
@ -3481,9 +3481,9 @@ void WebRtcVideoChannel::ClearRecordableEncodedFrameCallback(uint32_t ssrc) {
if (stream) {
stream->ClearRecordableEncodedFrameCallback();
} else {
RTC_LOG(LS_ERROR) << "Absent receive stream; ignoring clearing encoded "
"frame sink for ssrc "
<< ssrc;
RTC_LOG(LS_INFO) << "Absent receive stream; ignoring clearing encoded "
"frame sink for ssrc "
<< ssrc;
}
}

2
media/engine/webrtc_voice_engine.cc
View file

@ -2077,7 +2077,7 @@ bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
__func__, ssrc, volume);
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
RTC_LOG(LS_WARNING) << rtc::StringFormat(
RTC_LOG(LS_INFO) << rtc::StringFormat(
"WRVMC::%s => (WARNING: no receive stream for SSRC %u)", __func__,
ssrc);
return false;

8
modules/audio_device/audio_device_generic.cc
View file

@ -15,7 +15,7 @@
namespace webrtc {
bool AudioDeviceGeneric::BuiltInAECIsAvailable() const {
RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
RTC_LOG_F(LS_INFO) << "Not supported on this platform";
return false;
}
@ -25,7 +25,7 @@ int32_t AudioDeviceGeneric::EnableBuiltInAEC(bool enable) {
}
bool AudioDeviceGeneric::BuiltInAGCIsAvailable() const {
RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
RTC_LOG_F(LS_INFO) << "Not supported on this platform";
return false;
}
@ -35,7 +35,7 @@ int32_t AudioDeviceGeneric::EnableBuiltInAGC(bool enable) {
}
bool AudioDeviceGeneric::BuiltInNSIsAvailable() const {
RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
RTC_LOG_F(LS_INFO) << "Not supported on this platform";
return false;
}
@ -45,7 +45,7 @@ int32_t AudioDeviceGeneric::EnableBuiltInNS(bool enable) {
}
int32_t AudioDeviceGeneric::GetPlayoutUnderrunCount() const {
RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
RTC_LOG_F(LS_INFO) << "Not supported on this platform";
return -1;
}

4
modules/audio_device/win/audio_device_core_win.cc
View file

@ -538,8 +538,8 @@ AudioDeviceWindowsCore::~AudioDeviceWindowsCore() {
<< "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
" failed to free the loaded Avrt DLL module correctly";
} else {
RTC_LOG(LS_WARNING) << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
" the Avrt DLL module is now unloaded";
RTC_LOG(LS_INFO) << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
" the Avrt DLL module is now unloaded";
}
}
}

3
modules/audio_device/win/core_audio_base_win.cc
View file

@ -440,14 +440,17 @@ bool CoreAudioBase::Init() {
audio_client.Get(), AUDCLNT_SHAREMODE_SHARED, &format_)) {
// RingRTC change to try again to match a format for multi-channel.
if (IsInput() && (params.channels() > 2)) {
RTC_LOG(LS_WARNING) << "Trying again to match for multi-channel";
format->nChannels = params.channels();
format->nBlockAlign = (format->wBitsPerSample / 8) * format->nChannels;
format->nAvgBytesPerSec = format->nSamplesPerSec * format->nBlockAlign;
if (!core_audio_utility::IsFormatSupported(
audio_client.Get(), AUDCLNT_SHAREMODE_SHARED, &format_)) {
RTC_LOG(LS_ERROR) << "No multi-channel format matched";
return false;
}
} else {
RTC_LOG(LS_ERROR) << "No format matched";
return false;
}
}

2
modules/congestion_controller/goog_cc/goog_cc_network_control.cc
View file

@ -343,7 +343,7 @@ void GoogCcNetworkController::ClampConstraints() {
min_data_rate_ = std::max(min_data_rate_, min_total_allocated_bitrate_);
}
if (max_data_rate_ < min_data_rate_) {
RTC_LOG(LS_WARNING) << "max bitrate smaller than min bitrate";
RTC_LOG(LS_INFO) << "max bitrate smaller than min bitrate";
max_data_rate_ = min_data_rate_;
}
if (starting_rate_ && starting_rate_ < min_data_rate_) {

6
modules/congestion_controller/rtp/transport_feedback_adapter.cc
View file

@ -261,9 +261,9 @@ TransportFeedbackAdapter::ProcessTransportFeedbackInner(
}
if (failed_lookups > 0) {
RTC_LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
<< " packet" << (failed_lookups > 1 ? "s" : "")
<< ". Send time history too small?";
RTC_LOG(LS_INFO) << "Failed to lookup send time for " << failed_lookups
<< " packet" << (failed_lookups > 1 ? "s" : "")
<< ". Send time history too small?";
}
if (ignored > 0) {
RTC_LOG(LS_INFO) << "Ignoring " << ignored

6
modules/pacing/pacing_controller.cc
View file

@ -315,9 +315,9 @@ TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) {
TimeDelta elapsed_time = now - last_process_time_;
last_process_time_ = now;
if (elapsed_time > kMaxElapsedTime) {
RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time.ms()
<< " ms) longer than expected, limiting to "
<< kMaxElapsedTime.ms();
RTC_LOG(LS_INFO) << "Elapsed time (" << elapsed_time.ms()
<< " ms) longer than expected, limiting to "
<< kMaxElapsedTime.ms();
elapsed_time = kMaxElapsedTime;
}
return elapsed_time;

2
modules/pacing/packet_router.cc
View file

@ -149,7 +149,7 @@ void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
uint32_t ssrc = packet->Ssrc();
auto kv = send_modules_map_.find(ssrc);
if (kv == send_modules_map_.end()) {
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "Failed to send packet, matching RTP module not found "
"or transport error. SSRC = "
<< packet->Ssrc() << ", sequence number " << packet->SequenceNumber();

6
modules/rtp_rtcp/source/rtcp_receiver.cc
View file

@ -354,10 +354,10 @@ absl::optional<TimeDelta> RTCPReceiver::OnPeriodicRttUpdate(
// Check for expired timers and if so, log and reset.
auto now = clock_->CurrentTime();
if (RtcpRrTimeoutLocked(now)) {
RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
RTC_LOG_F(LS_INFO) << "Timeout: No RTCP RR received.";
} else if (RtcpRrSequenceNumberTimeoutLocked(now)) {
RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended "
"highest sequence number.";
RTC_LOG_F(LS_INFO) << "Timeout: No increase in RTCP RR extended "
"highest sequence number.";
}
} else {
// Report rtt from receiver.

4
modules/rtp_rtcp/source/rtp_sender.cc
View file

@ -362,8 +362,8 @@ void RTPSender::OnReceivedNack(
const int32_t bytes_sent = ReSendPacket(seq_no);
if (bytes_sent < 0) {
// Failed to send one Sequence number. Give up the rest in this nack.
RTC_LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
<< ", Discard rest of packets.";
RTC_LOG(LS_INFO) << "Failed resending RTP packet " << seq_no
<< ", Discard rest of packets.";
break;
}
}

8
modules/video_coding/frame_buffer2.cc
View file

@ -444,9 +444,9 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
ClearFramesAndHistory();
last_continuous_frame_id = -1;
} else {
RTC_LOG(LS_WARNING) << "Frame " << frame->Id() << " inserted after frame "
<< *last_decoded_frame
<< " was handed off for decoding, dropping frame.";
RTC_LOG(LS_INFO) << "Frame " << frame->Id() << " inserted after frame "
<< *last_decoded_frame
<< " was handed off for decoding, dropping frame.";
return last_continuous_frame_id;
}
}
@ -580,7 +580,7 @@ bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const EncodedFrame& frame,
if (!decoded_frames_history_.WasDecoded(frame.references[i])) {
int64_t now_ms = clock_->TimeInMilliseconds();
if (last_log_non_decoded_ms_ + kLogNonDecodedIntervalMs < now_ms) {
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "Frame " << frame.Id()
<< " depends on a non-decoded frame more previous than the last "
"decoded frame, dropping frame.";

4
modules/video_coding/nack_requester.cc
View file

@ -354,8 +354,8 @@ std::vector<uint16_t> NackRequester::GetNackBatch(NackFilterOptions options) {
++it->second.retries;
it->second.sent_at_time = now.ms();
if (it->second.retries >= kMaxNackRetries) {
RTC_LOG(LS_WARNING) << "Sequence number " << it->second.seq_num
<< " removed from NACK list due to max retries.";
RTC_LOG(LS_INFO) << "Sequence number " << it->second.seq_num
<< " removed from NACK list due to max retries.";
it = nack_list_.erase(it);
} else {
++it;

10
modules/video_coding/rtp_vp8_ref_finder.cc
View file

@ -162,11 +162,11 @@ RtpVp8RefFinder::FrameDecision RtpVp8RefFinder::ManageFrameInternal(
if (!(AheadOf<uint16_t, kFrameIdLength>(frame->Id(),
layer_info_it->second[layer]))) {
RTC_LOG(LS_WARNING) << "Frame with picture id " << frame->Id()
<< " and packet range [" << frame->first_seq_num()
<< ", " << frame->last_seq_num()
<< "] already received, "
" dropping frame.";
RTC_LOG(LS_INFO) << "Frame with picture id " << frame->Id()
<< " and packet range [" << frame->first_seq_num()
<< ", " << frame->last_seq_num()
<< "] already received, "
" dropping frame.";
return kDrop;
}

2
p2p/base/basic_packet_socket_factory.cc
View file

@ -179,7 +179,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
}
if (socket->Connect(remote_address) < 0) {
RTC_LOG(LS_ERROR) << "TCP connect failed with error " << socket->GetError();
RTC_LOG(LS_INFO) << "TCP connect failed with error " << socket->GetError();
delete socket;
return NULL;
}

18
p2p/base/connection.cc
View file

@ -442,10 +442,10 @@ void Connection::OnSendStunPacket(const void* data,
auto err =
port_->SendTo(data, size, remote_candidate_.address(), options, false);
if (err < 0) {
RTC_LOG(LS_WARNING) << ToString()
<< ": Failed to send STUN ping "
" err="
<< err << " id=" << rtc::hex_encode(req->id());
RTC_LOG(LS_INFO) << ToString()
<< ": Failed to send STUN ping "
" err="
<< err << " id=" << rtc::hex_encode(req->id());
}
}
@ -712,10 +712,10 @@ void Connection::SendResponseMessage(const StunMessage& response) {
rtc::PacketType::kIceConnectivityCheckResponse;
auto err = port_->SendTo(buf.Data(), buf.Length(), addr, options, false);
if (err < 0) {
RTC_LOG(LS_ERROR) << ToString() << ": Failed to send "
<< StunMethodToString(response.type())
<< ", to=" << addr.ToSensitiveString() << ", err=" << err
<< ", id=" << rtc::hex_encode(response.transaction_id());
RTC_LOG(LS_INFO) << ToString() << ": Failed to send "
<< StunMethodToString(response.type())
<< ", to=" << addr.ToSensitiveString() << ", err=" << err
<< ", id=" << rtc::hex_encode(response.transaction_id());
} else {
// Log at LS_INFO if we send a stun ping response on an unwritable
// connection.
@ -1253,7 +1253,7 @@ void Connection::MaybeUpdateLocalCandidate(ConnectionRequest* request,
const StunAddressAttribute* addr =
response->GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS);
if (!addr) {
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "Connection::OnConnectionRequestResponse - "
"No MAPPED-ADDRESS or XOR-MAPPED-ADDRESS found in the "
"stun response message";

4
p2p/base/p2p_transport_channel.cc
View file

@ -1186,7 +1186,7 @@ void P2PTransportChannel::OnUnknownAddress(PortInterface* port,
// If we don't know that the remote ufrag and the session is shared between
// different transports, then don't create a peer reflexive candidate.
// Otherwise, each transport would end up with one.
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "Ignoring peer-refexive ICE candidate because the ufrag is unknown.";
return;
}
@ -1397,7 +1397,7 @@ void P2PTransportChannel::AddRemoteCandidate(const Candidate& candidate) {
} else {
// The candidate belongs to the next generation. Its pwd will be set
// when the new remote ICE credentials arrive.
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "A remote candidate arrives with an unknown ufrag: "
<< candidate.username();
}

6
p2p/base/port.cc
View file

@ -368,9 +368,9 @@ void Port::OnReadPacket(const char* data,
std::unique_ptr<IceMessage> msg;
std::string remote_username;
if (!GetStunMessage(data, size, addr, &msg, &remote_username)) {
RTC_LOG(LS_ERROR) << ToString()
<< ": Received non-STUN packet from unknown address: "
<< addr.ToSensitiveString();
RTC_LOG(LS_INFO) << ToString()
<< ": Received non-STUN packet from unknown address: "
<< addr.ToSensitiveString();
} else if (!msg) {
// STUN message handled already
} else if (msg->type() == STUN_BINDING_REQUEST) {

22
p2p/base/stun_port.cc
View file

@ -93,9 +93,9 @@ class StunBindingRequest : public StunRequest {
}
}
void OnTimeout() override {
RTC_LOG(LS_ERROR) << "Binding request timed out from "
<< port_->GetLocalAddress().ToSensitiveString() << " ("
<< port_->Network()->name() << ")";
RTC_LOG(LS_INFO) << "Binding request timed out from "
<< port_->GetLocalAddress().ToSensitiveString() << " ("
<< port_->Network()->name() << ")";
port_->OnStunBindingOrResolveRequestFailed(
server_addr_, SERVER_NOT_REACHABLE_ERROR,
"STUN allocate request timed out.");
@ -307,10 +307,10 @@ int UDPPort::SendTo(const void* data,
// TODO(webrtc:9622): Use general rate limiting mechanism once it exists.
if (send_error_count_ < kSendErrorLogLimit) {
++send_error_count_;
RTC_LOG(LS_ERROR) << ToString() << ": UDP send of " << size
<< " bytes to host " << addr.ToSensitiveString() << " ("
<< addr.ToResolvedSensitiveString()
<< ") failed with error " << error_;
RTC_LOG(LS_INFO) << ToString() << ": UDP send of " << size
<< " bytes to host " << addr.ToSensitiveString() << " ("
<< addr.ToResolvedSensitiveString()
<< ") failed with error " << error_;
}
} else {
send_error_count_ = 0;
@ -453,9 +453,9 @@ void UDPPort::OnResolveResult(const rtc::SocketAddress& input, int error) {
rtc::SocketAddress resolved;
if (error != 0 || !resolver_->GetResolvedAddress(
input, Network()->GetBestIP().family(), &resolved)) {
RTC_LOG(LS_WARNING) << ToString()
<< ": StunPort: stun host lookup received error "
<< error;
RTC_LOG(LS_INFO) << ToString()
<< ": StunPort: stun host lookup received error "
<< error;
OnStunBindingOrResolveRequestFailed(input, SERVER_NOT_REACHABLE_ERROR,
"STUN host lookup received error.");
return;
@ -599,7 +599,7 @@ 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_INFO, socket_->GetError())
<< "UDP send of " << size << " bytes to host "
<< sreq->server_addr().ToSensitiveString() << " ("
<< sreq->server_addr().ToResolvedSensitiveString()

40
p2p/base/turn_port.cc
View file

@ -376,7 +376,7 @@ void TurnPort::PrepareAddress() {
<< ProtoToString(server_address_.proto) << " @ "
<< server_address_.address.ToSensitiveString();
if (!CreateTurnClientSocket()) {
RTC_LOG(LS_ERROR) << "Failed to create TURN client socket";
RTC_LOG(LS_INFO) << "Failed to create TURN client socket";
OnAllocateError(SERVER_NOT_REACHABLE_ERROR,
"Failed to create TURN client socket.");
return;
@ -478,13 +478,13 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
return socket_address.ipaddr() == addr;
})) {
if (socket->GetLocalAddress().IsLoopbackIP()) {
RTC_LOG(LS_WARNING) << "Socket is bound to the address:"
<< socket_address.ipaddr().ToSensitiveString()
<< ", rather than an address associated with network:"
<< Network()->ToString()
<< ". Still allowing it since it's localhost.";
RTC_LOG(LS_INFO) << "Socket is bound to the address:"
<< socket_address.ipaddr().ToSensitiveString()
<< ", rather than an address associated with network:"
<< Network()->ToString()
<< ". Still allowing it since it's localhost.";
} else if (IPIsAny(Network()->GetBestIP())) {
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "Socket is bound to the address:"
<< socket_address.ipaddr().ToSensitiveString()
<< ", rather than an address associated with network:"
@ -492,10 +492,10 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
<< ". Still allowing it since it's the 'any' address"
", possibly caused by multiple_routes being disabled.";
} else {
RTC_LOG(LS_WARNING) << "Socket is bound to the address:"
<< socket_address.ipaddr().ToSensitiveString()
<< ", rather than an address associated with network:"
<< Network()->ToString() << ". Discarding TURN port.";
RTC_LOG(LS_INFO) << "Socket is bound to the address:"
<< socket_address.ipaddr().ToSensitiveString()
<< ", rather than an address associated with network:"
<< Network()->ToString() << ". Discarding TURN port.";
OnAllocateError(
STUN_ERROR_GLOBAL_FAILURE,
"Address not associated with the desired network interface.");
@ -515,9 +515,9 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
}
void TurnPort::OnSocketClose(rtc::AsyncPacketSocket* socket, int error) {
RTC_LOG(LS_WARNING) << ToString()
<< ": Connection with server failed with error: "
<< error;
RTC_LOG(LS_INFO) << ToString()
<< ": Connection with server failed with error: "
<< error;
RTC_DCHECK(socket == socket_);
Close();
}
@ -823,8 +823,8 @@ void TurnPort::OnResolveResult(rtc::AsyncResolverInterface* resolver) {
if (resolver_->GetError() != 0 ||
!resolver_->GetResolvedAddress(Network()->GetBestIP().family(),
&resolved_address)) {
RTC_LOG(LS_WARNING) << ToString() << ": TURN host lookup received error "
<< resolver_->GetError();
RTC_LOG(LS_INFO) << ToString() << ": TURN host lookup received error "
<< resolver_->GetError();
error_ = resolver_->GetError();
OnAllocateError(SERVER_NOT_REACHABLE_ERROR,
"TURN host lookup received error.");
@ -845,8 +845,8 @@ void TurnPort::OnSendStunPacket(const void* data,
options.info_signaled_after_sent.packet_type = rtc::PacketType::kTurnMessage;
CopyPortInformationToPacketInfo(&options.info_signaled_after_sent);
if (Send(data, size, options) < 0) {
RTC_LOG(LS_ERROR) << ToString() << ": Failed to send TURN message, error: "
<< socket_->GetError();
RTC_LOG(LS_INFO) << ToString() << ": Failed to send TURN message, error: "
<< socket_->GetError();
}
}
@ -1462,8 +1462,8 @@ void TurnAllocateRequest::OnErrorResponse(StunMessage* response) {
}
void TurnAllocateRequest::OnTimeout() {
RTC_LOG(LS_WARNING) << port_->ToString() << ": TURN allocate request "
<< rtc::hex_encode(id()) << " timeout";
RTC_LOG(LS_INFO) << port_->ToString() << ": TURN allocate request "
<< rtc::hex_encode(id()) << " timeout";
port_->OnAllocateRequestTimeout();
}

2
p2p/client/basic_port_allocator.cc
View file

@ -995,7 +995,7 @@ void BasicPortAllocatorSession::OnCandidateReady(Port* port,
// Discarding any candidate signal if port allocation status is
// already done with gathering.
if (!data->inprogress()) {
RTC_LOG(LS_WARNING)
RTC_LOG(LS_INFO)
<< "Discarding candidate because port is already done gathering.";
return;
}

2
pc/rtp_sender.cc
View file

@ -600,7 +600,7 @@ void VideoRtpSender::AttachTrack() {
}
rtc::scoped_refptr<DtmfSenderInterface> VideoRtpSender::GetDtmfSender() const {
RTC_LOG(LS_ERROR) << "Tried to get DTMF sender from video sender.";
RTC_LOG(LS_INFO) << "Tried to get DTMF sender from video sender.";
return nullptr;
}

8
pc/rtp_transport.cc
View file

@ -148,7 +148,7 @@ bool RtpTransport::SendPacket(bool rtcp,
options, flags);
if (ret != static_cast<int>(packet->size())) {
if (transport->GetError() == ENOTCONN) {
RTC_LOG(LS_WARNING) << "Got ENOTCONN from transport.";
RTC_LOG(LS_INFO) << "Got ENOTCONN from transport.";
SetReadyToSend(rtcp, false);
}
return false;
@ -192,8 +192,8 @@ void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet,
}
if (!rtp_demuxer_.OnRtpPacket(parsed_packet)) {
RTC_LOG(LS_WARNING) << "Failed to demux RTP packet: "
<< RtpDemuxer::DescribePacket(parsed_packet);
RTC_LOG(LS_INFO) << "Failed to demux RTP packet: "
<< RtpDemuxer::DescribePacket(parsed_packet);
}
}
@ -264,7 +264,7 @@ void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
// RingRTC change to avoid processing RTP packets too soon
if (!incoming_rtp_enabled_) {
if (packet_type == cricket::RtpPacketType::kRtcp) {
RTC_LOG(LS_WARNING) << "Dropping RTCP packet because incoming RTP is disabled; len: " << len;
RTC_LOG(LS_INFO) << "Dropping RTCP packet because incoming RTP is disabled; len: " << len;
return;
} else {
RTC_LOG(LS_INFO) << "Dropping RTP packet because incoming RTP is disabled; len: " << len;

4
pc/used_ids.h
View file

@ -52,8 +52,8 @@ class UsedIds {
if (IsIdUsed(original_id)) {
new_id = FindUnusedId();
RTC_LOG(LS_WARNING) << "Duplicate id found. Reassigning from "
<< original_id << " to " << new_id;
RTC_LOG(LS_INFO) << "Duplicate id found. Reassigning from "
<< original_id << " to " << new_id;
idstruct->id = new_id;
}
SetIdUsed(new_id);

4
rtc_base/physical_socket_server.cc
View file

@ -595,7 +595,7 @@ int PhysicalSocket::TranslateOption(Option opt, int* slevel, int* sopt) {
}
break;
#else
RTC_LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
RTC_LOG(LS_INFO) << "Socket::OPT_DSCP not supported.";
return -1;
#endif
case OPT_RTP_SENDTIME_EXTN_ID:
@ -1654,7 +1654,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
}
if ((wsaEvents.lNetworkEvents & FD_CONNECT) &&
wsaEvents.iErrorCode[FD_CONNECT_BIT] != 0) {
RTC_LOG(WARNING)
RTC_LOG(LS_INFO)
<< "PhysicalSocketServer got FD_CONNECT_BIT error "
<< wsaEvents.iErrorCode[FD_CONNECT_BIT];
}

8
sdk/objc/components/capturer/RTCCameraVideoCapturer.m
View file

@ -309,7 +309,7 @@ const int64_t kNanosecondsPerSecond = 1000000000;
// DroppedFrameReason unavailable on macOS.
CFStringRef droppedReason = nil;
#endif
RTCLogError(@"Dropped sample buffer. Reason: %@", (__bridge NSString *)droppedReason);
RTCLogInfo(@"Dropped sample buffer. Reason: %@", (__bridge NSString *)droppedReason);
}
#pragma mark - AVCaptureSession notifications
@ -344,7 +344,7 @@ const int64_t kNanosecondsPerSecond = 1000000000;
- (void)handleCaptureSessionRuntimeError:(NSNotification *)notification {
NSError *error = [notification.userInfo objectForKey:AVCaptureSessionErrorKey];
RTCLogError(@"Capture session runtime error: %@", error);
RTCLogInfo(@"Capture session runtime error: %@", error);
[RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
@ -380,11 +380,11 @@ const int64_t kNanosecondsPerSecond = 1000000000;
dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
if (!self.hasRetriedOnFatalError) {
RTCLogWarning(@"Attempting to recover from fatal capture error.");
RTCLogInfo(@"Attempting to recover from fatal capture error.");
[self handleNonFatalError];
self.hasRetriedOnFatalError = YES;
} else {
RTCLogError(@"Previous fatal error recovery failed.");
RTCLogInfo(@"Previous fatal error recovery failed.");
}
}];
}

14
sdk/objc/native/src/audio/audio_device_ios.mm
View file

@ -452,9 +452,9 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
const int64_t delta_time = now_time - last_playout_time_;
const int glitch_threshold = 1.6 * playout_parameters_.GetBufferSizeInMilliseconds();
if (delta_time > glitch_threshold) {
RTCLogWarning(@"Possible playout audio glitch detected.\n"
" Time since last OnGetPlayoutData was %lld ms.\n",
delta_time);
RTCLogInfo(@"Possible playout audio glitch detected.\n"
" Time since last OnGetPlayoutData was %lld ms.\n",
delta_time);
// Exclude extreme delta values since they do most likely not correspond
// to a real glitch. Instead, the most probable cause is that a headset
// has been plugged in or out. There are more direct ways to detect
@ -881,7 +881,7 @@ void AudioDeviceIOS::UnconfigureAudioSession() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTCLog(@"Unconfiguring audio session.");
if (!has_configured_session_) {
RTCLogWarning(@"Audio session already unconfigured.");
RTCLogInfo(@"Audio session already unconfigured.");
return;
}
RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
@ -1041,7 +1041,7 @@ int32_t AudioDeviceIOS::SpeakerMute(bool& enabled) const {
}
int32_t AudioDeviceIOS::SetPlayoutDevice(uint16_t index) {
RTC_LOG_F(LS_WARNING) << "Not implemented";
RTC_LOG_F(LS_INFO) << "Not implemented";
return 0;
}
@ -1094,7 +1094,7 @@ int32_t AudioDeviceIOS::StereoPlayoutIsAvailable(bool& available) {
}
int32_t AudioDeviceIOS::SetStereoPlayout(bool enable) {
RTC_LOG_F(LS_WARNING) << "Not implemented";
RTC_LOG_F(LS_INFO) << "Not implemented";
return -1;
}
@ -1143,7 +1143,7 @@ int32_t AudioDeviceIOS::RecordingDeviceName(uint16_t index,
}
int32_t AudioDeviceIOS::SetRecordingDevice(uint16_t index) {
RTC_LOG_F(LS_WARNING) << "Not implemented";
RTC_LOG_F(LS_INFO) << "Not implemented";
return 0;
}

2
sdk/objc/native/src/audio/audio_device_module_ios.mm
View file

@ -329,7 +329,7 @@ AudioDeviceModuleIOS::AudioDeviceModuleIOS(bool bypass_voice_processing)
return -1;
}
if (audio_device_->SetStereoPlayout(enable)) {
RTC_LOG(WARNING) << "stereo playout is not supported";
RTC_LOG(INFO) << "stereo playout is not supported";
return -1;
}
int8_t nChannels(1);

4
sdk/objc/native/src/audio/voice_processing_audio_unit.mm
View file

@ -280,8 +280,8 @@ bool VoiceProcessingAudioUnit::Initialize(Float64 sample_rate) {
UInt32 agc_is_enabled = 0;
result = GetAGCState(vpio_unit_, &agc_is_enabled);
if (result != noErr) {
RTCLogError(@"Failed to get AGC state (1st attempt). "
"Error=%ld.",
RTCLogInfo(@"Failed to get AGC state (1st attempt). "
"Error=%ld.",
(long)result);
// Example of error code: kAudioUnitErr_NoConnection (-10876).
// All error codes related to audio units are negative and are therefore

2
video/buffered_frame_decryptor.cc
View file

@ -40,7 +40,7 @@ void BufferedFrameDecryptor::ManageEncryptedFrame(
switch (DecryptFrame(encrypted_frame.get())) {
case FrameDecision::kStash:
if (stashed_frames_.size() >= kMaxStashedFrames) {
RTC_LOG(LS_WARNING) << "Encrypted frame stash full poping oldest item.";
RTC_LOG(LS_INFO) << "Encrypted frame stash full poping oldest item.";
stashed_frames_.pop_front();
}
stashed_frames_.push_back(std::move(encrypted_frame));

4
video/video_receive_stream2.cc
View file

@ -918,8 +918,8 @@ void VideoReceiveStream2::HandleFrameBufferTimeout(int64_t now_ms,
if (stream_is_active && !IsReceivingKeyFrame(now_ms) &&
(!config_.crypto_options.sframe.require_frame_encryption ||
rtp_video_stream_receiver_.IsDecryptable())) {
RTC_LOG(LS_WARNING) << "No decodable frame in " << wait_ms
<< " ms, requesting keyframe.";
RTC_LOG(LS_INFO) << "No decodable frame in " << wait_ms
<< " ms, requesting keyframe.";
RequestKeyFrame(now_ms);
}
}