Deprecate and remove usage for WARNING log level
Bug: webrtc:13362
Change-Id: Ida112158e4ac5f667e533a0ebfedb400c84df4d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239124
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35425}
diff --git a/examples/peerconnection/client/peer_connection_client.cc b/examples/peerconnection/client/peer_connection_client.cc
index e4d2df4..c0de4ff 100644
--- a/examples/peerconnection/client/peer_connection_client.cc
+++ b/examples/peerconnection/client/peer_connection_client.cc
@@ -77,7 +77,7 @@
RTC_DCHECK(!client_name.empty());
if (state_ != NOT_CONNECTED) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "The client must not be connected before you can call Connect()";
callback_->OnServerConnectionFailure();
return;
@@ -479,7 +479,7 @@
}
} else {
if (socket == control_socket_.get()) {
- RTC_LOG(WARNING) << "Connection refused; retrying in 2 seconds";
+ RTC_LOG(LS_WARNING) << "Connection refused; retrying in 2 seconds";
rtc::Thread::Current()->PostDelayed(RTC_FROM_HERE, kReconnectDelay, this,
0);
} else {
diff --git a/examples/unityplugin/simple_peer_connection.cc b/examples/unityplugin/simple_peer_connection.cc
index 0503a26..8a82718 100644
--- a/examples/unityplugin/simple_peer_connection.cc
+++ b/examples/unityplugin/simple_peer_connection.cc
@@ -343,9 +343,9 @@
webrtc::SessionDescriptionInterface* session_description(
webrtc::CreateSessionDescription(desc_type, remote_desc, &error));
if (!session_description) {
- RTC_LOG(WARNING) << "Can't parse received session description message. "
- "SdpParseError was: "
- << error.description;
+ RTC_LOG(LS_WARNING) << "Can't parse received session description message. "
+ "SdpParseError was: "
+ << error.description;
return false;
}
RTC_LOG(LS_INFO) << " Received session description :" << remote_desc;
@@ -365,13 +365,13 @@
std::unique_ptr<webrtc::IceCandidateInterface> ice_candidate(
webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate, &error));
if (!ice_candidate.get()) {
- RTC_LOG(WARNING) << "Can't parse received candidate message. "
- "SdpParseError was: "
- << error.description;
+ RTC_LOG(LS_WARNING) << "Can't parse received candidate message. "
+ "SdpParseError was: "
+ << error.description;
return false;
}
if (!peer_connection_->AddIceCandidate(ice_candidate.get())) {
- RTC_LOG(WARNING) << "Failed to apply the received candidate";
+ RTC_LOG(LS_WARNING) << "Failed to apply the received candidate";
return false;
}
RTC_LOG(LS_INFO) << " Received candidate :" << candidate;
diff --git a/modules/audio_device/android/aaudio_player.cc b/modules/audio_device/android/aaudio_player.cc
index a5a3675..5257b2b 100644
--- a/modules/audio_device/android/aaudio_player.cc
+++ b/modules/audio_device/android/aaudio_player.cc
@@ -142,7 +142,7 @@
if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
// The stream is disconnected and any attempt to use it will return
// AAUDIO_ERROR_DISCONNECTED.
- RTC_LOG(WARNING) << "Output stream disconnected";
+ RTC_LOG(LS_WARNING) << "Output stream disconnected";
// AAudio documentation states: "You should not close or reopen the stream
// from the callback, use another thread instead". A message is therefore
// sent to the main thread to do the restart operation.
diff --git a/modules/audio_device/android/aaudio_recorder.cc b/modules/audio_device/android/aaudio_recorder.cc
index d91fb9e..4757cf8 100644
--- a/modules/audio_device/android/aaudio_recorder.cc
+++ b/modules/audio_device/android/aaudio_recorder.cc
@@ -137,7 +137,7 @@
if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
// The stream is disconnected and any attempt to use it will return
// AAUDIO_ERROR_DISCONNECTED..
- RTC_LOG(WARNING) << "Input stream disconnected => restart is required";
+ RTC_LOG(LS_WARNING) << "Input stream disconnected => restart is required";
// AAudio documentation states: "You should not close or reopen the stream
// from the callback, use another thread instead". A message is therefore
// sent to the main thread to do the restart operation.
diff --git a/modules/audio_device/android/aaudio_wrapper.cc b/modules/audio_device/android/aaudio_wrapper.cc
index 82860e3..3d824b5 100644
--- a/modules/audio_device/android/aaudio_wrapper.cc
+++ b/modules/audio_device/android/aaudio_wrapper.cc
@@ -91,8 +91,8 @@
aaudio_result_t error) {
RTC_DCHECK(user_data);
AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
- RTC_LOG(WARNING) << "ErrorCallback: "
- << DirectionToString(aaudio_wrapper->direction());
+ RTC_LOG(LS_WARNING) << "ErrorCallback: "
+ << DirectionToString(aaudio_wrapper->direction());
RTC_DCHECK(aaudio_wrapper->observer());
aaudio_wrapper->observer()->OnErrorCallback(error);
}
diff --git a/modules/audio_device/android/audio_device_template.h b/modules/audio_device/android/audio_device_template.h
index a1510d3..999c587 100644
--- a/modules/audio_device/android/audio_device_template.h
+++ b/modules/audio_device/android/audio_device_template.h
@@ -171,7 +171,7 @@
int32_t StartPlayout() override {
RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!audio_manager_->IsCommunicationModeEnabled()) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "The application should use MODE_IN_COMMUNICATION audio mode!";
}
return output_.StartPlayout();
@@ -194,7 +194,7 @@
int32_t StartRecording() override {
RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!audio_manager_->IsCommunicationModeEnabled()) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "The application should use MODE_IN_COMMUNICATION audio mode!";
}
return input_.StartRecording();
diff --git a/modules/audio_device/android/audio_manager.cc b/modules/audio_device/android/audio_manager.cc
index e75bd4d..0b55496 100644
--- a/modules/audio_device/android/audio_manager.cc
+++ b/modules/audio_device/android/audio_manager.cc
@@ -123,7 +123,8 @@
// If one already has been created, return existing object instead of
// creating a new.
if (engine_object_.Get() != nullptr) {
- RTC_LOG(WARNING) << "The OpenSL ES engine object has already been created";
+ RTC_LOG(LS_WARNING)
+ << "The OpenSL ES engine object has already been created";
return engine_object_.Get();
}
// Create the engine object in thread safe mode.
diff --git a/modules/audio_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index 73a8210..d393a88 100644
--- a/modules/audio_device/audio_device_buffer.cc
+++ b/modules/audio_device/audio_device_buffer.cc
@@ -64,7 +64,7 @@
RTC_LOG(LS_INFO) << "AudioDeviceBuffer::ctor";
#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
phase_ = 0.0;
- RTC_LOG(WARNING) << "AUDIO_DEVICE_PLAYS_SINUS_TONE is defined!";
+ RTC_LOG(LS_WARNING) << "AUDIO_DEVICE_PLAYS_SINUS_TONE is defined!";
#endif
}
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
index 200259a..01a8a25 100644
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -251,7 +251,7 @@
// - kPlatformDefaultAudio => ALSA, and
// - kLinuxAlsaAudio => ALSA, and
// - kLinuxPulseAudio => Invalid selection.
- RTC_LOG(WARNING) << "PulseAudio is disabled using build flag.";
+ RTC_LOG(LS_WARNING) << "PulseAudio is disabled using build flag.";
if ((audio_layer == kLinuxAlsaAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
audio_device_.reset(new AudioDeviceLinuxALSA());
@@ -271,7 +271,7 @@
RTC_LOG(LS_INFO) << "Linux PulseAudio APIs will be utilized";
} else if (audio_layer == kLinuxAlsaAudio) {
audio_device_.reset(new AudioDeviceLinuxALSA());
- RTC_LOG(WARNING) << "Linux ALSA APIs will be utilized.";
+ RTC_LOG(LS_WARNING) << "Linux ALSA APIs will be utilized.";
}
#endif // #if !defined(WEBRTC_ENABLE_LINUX_PULSE)
#endif // #if defined(WEBRTC_LINUX)
@@ -552,7 +552,7 @@
}
if (audio_device_->SetStereoRecording(enable) == -1) {
if (enable) {
- RTC_LOG(WARNING) << "failed to enable stereo recording";
+ RTC_LOG(LS_WARNING) << "failed to enable stereo recording";
}
return -1;
}
@@ -597,7 +597,7 @@
return -1;
}
if (audio_device_->SetStereoPlayout(enable)) {
- RTC_LOG(WARNING) << "stereo playout is not supported";
+ RTC_LOG(LS_WARNING) << "stereo playout is not supported";
return -1;
}
int8_t nChannels(1);
diff --git a/modules/audio_device/win/core_audio_base_win.cc b/modules/audio_device/win/core_audio_base_win.cc
index 40645d5..f43c068 100644
--- a/modules/audio_device/win/core_audio_base_win.cc
+++ b/modules/audio_device/win/core_audio_base_win.cc
@@ -506,9 +506,9 @@
RTC_DLOG(LS_INFO) << "preferred_frames_per_buffer: "
<< preferred_frames_per_buffer;
if (preferred_frames_per_buffer % params.frames_per_buffer()) {
- RTC_LOG(WARNING) << "Buffer size of " << params.frames_per_buffer()
- << " is not an even divisor of "
- << preferred_frames_per_buffer;
+ RTC_LOG(LS_WARNING) << "Buffer size of " << params.frames_per_buffer()
+ << " is not an even divisor of "
+ << preferred_frames_per_buffer;
}
// Create an AudioSessionControl interface given the initialized client.
diff --git a/modules/audio_device/win/core_audio_input_win.cc b/modules/audio_device/win/core_audio_input_win.cc
index 6ad5f46..17790da 100644
--- a/modules/audio_device/win/core_audio_input_win.cc
+++ b/modules/audio_device/win/core_audio_input_win.cc
@@ -187,7 +187,7 @@
// Release resources allocated in InitRecording() and then return if this
// method is called without any active input audio.
if (!Recording()) {
- RTC_DLOG(WARNING) << "No input stream is active";
+ RTC_DLOG(LS_WARNING) << "No input stream is active";
ReleaseCOMObjects();
initialized_ = false;
return 0;
@@ -387,7 +387,7 @@
if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
HandleStreamDisconnected();
} else {
- RTC_DLOG(WARNING) << "Unsupported error type";
+ RTC_DLOG(LS_WARNING) << "Unsupported error type";
}
return true;
}
diff --git a/modules/audio_device/win/core_audio_output_win.cc b/modules/audio_device/win/core_audio_output_win.cc
index 7522922..c92fedf 100644
--- a/modules/audio_device/win/core_audio_output_win.cc
+++ b/modules/audio_device/win/core_audio_output_win.cc
@@ -188,7 +188,7 @@
// Release resources allocated in InitPlayout() and then return if this
// method is called without any active output audio.
if (!Playing()) {
- RTC_DLOG(WARNING) << "No output stream is active";
+ RTC_DLOG(LS_WARNING) << "No output stream is active";
ReleaseCOMObjects();
initialized_ = false;
return 0;
@@ -273,7 +273,7 @@
if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
HandleStreamDisconnected();
} else {
- RTC_DLOG(WARNING) << "Unsupported error type";
+ RTC_DLOG(LS_WARNING) << "Unsupported error type";
}
return true;
}
diff --git a/modules/audio_device/win/core_audio_utility_win.cc b/modules/audio_device/win/core_audio_utility_win.cc
index 976edc8..c5a3520 100644
--- a/modules/audio_device/win/core_audio_utility_win.cc
+++ b/modules/audio_device/win/core_audio_utility_win.cc
@@ -1020,7 +1020,7 @@
// Log a warning for the rare case where `mix_format` only contains a
// stand-alone WAVEFORMATEX structure but don't return.
if (!wrapped_format.IsExtensible()) {
- RTC_DLOG(WARNING)
+ RTC_DLOG(LS_WARNING)
<< "The returned format contains no extended information. "
"The size is "
<< wrapped_format.size() << " bytes.";
diff --git a/modules/congestion_controller/goog_cc/trendline_estimator.cc b/modules/congestion_controller/goog_cc/trendline_estimator.cc
index 1008bad..7fdf66c 100644
--- a/modules/congestion_controller/goog_cc/trendline_estimator.cc
+++ b/modules/congestion_controller/goog_cc/trendline_estimator.cc
@@ -44,7 +44,7 @@
if (parsed_values == 1) {
if (window_size > 1)
return window_size;
- RTC_LOG(WARNING) << "Window size must be greater than 1.";
+ RTC_LOG(LS_WARNING) << "Window size must be greater than 1.";
}
RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweWindowSizeInPackets"
" experiment from field trial string. Using default.";
diff --git a/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/modules/desktop_capture/win/screen_capturer_win_gdi.cc
index 231754c..57b1f71 100644
--- a/modules/desktop_capture/win/screen_capturer_win_gdi.cc
+++ b/modules/desktop_capture/win/screen_capturer_win_gdi.cc
@@ -86,7 +86,7 @@
PrepareCaptureResources();
if (!CaptureImage()) {
- RTC_LOG(WARNING) << "Failed to capture screen by GDI.";
+ RTC_LOG(LS_WARNING) << "Failed to capture screen by GDI.";
callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
return;
}
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.cc b/modules/remote_bitrate_estimator/aimd_rate_control.cc
index 408f208..a3da2b5 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control.cc
+++ b/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -53,9 +53,9 @@
sscanf(experiment_string.c_str(), "Enabled-%lf", &backoff_factor);
if (parsed_values == 1) {
if (backoff_factor >= 1.0) {
- RTC_LOG(WARNING) << "Back-off factor must be less than 1.";
+ RTC_LOG(LS_WARNING) << "Back-off factor must be less than 1.";
} else if (backoff_factor <= 0.0) {
- RTC_LOG(WARNING) << "Back-off factor must be greater than 0.";
+ RTC_LOG(LS_WARNING) << "Back-off factor must be greater than 0.";
} else {
return backoff_factor;
}
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
index af2ed0c..5c41b48 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
@@ -191,14 +191,15 @@
reinterpret_cast<const char*>(rtp_payload.data()), rtp_payload.size());
uint8_t aggregation_header;
if (!payload.ReadUInt8(&aggregation_header)) {
- RTC_DLOG(WARNING) << "Failed to find aggregation header in the packet.";
+ RTC_DLOG(LS_WARNING)
+ << "Failed to find aggregation header in the packet.";
return {};
}
// Z-bit: 1 if the first OBU contained in the packet is a continuation of a
// previous OBU.
bool continues_obu = RtpStartsWithFragment(aggregation_header);
if (continues_obu != expect_continues_obu) {
- RTC_DLOG(WARNING) << "Unexpected Z-bit " << continues_obu;
+ RTC_DLOG(LS_WARNING) << "Unexpected Z-bit " << continues_obu;
return {};
}
int num_expected_obus = RtpNumObus(aggregation_header);
@@ -206,7 +207,8 @@
// rtp packet has just the aggregation header. That may be valid only when
// there is exactly one fragment in the packet of size 0.
if (num_expected_obus != 1) {
- RTC_DLOG(WARNING) << "Invalid packet with just an aggregation header.";
+ RTC_DLOG(LS_WARNING)
+ << "Invalid packet with just an aggregation header.";
return {};
}
if (!continues_obu) {
@@ -228,16 +230,16 @@
bool has_fragment_size = (obu_index != num_expected_obus);
if (has_fragment_size) {
if (!payload.ReadUVarint(&fragment_size)) {
- RTC_DLOG(WARNING) << "Failed to read fragment size for obu #"
- << obu_index << "/" << num_expected_obus;
+ RTC_DLOG(LS_WARNING) << "Failed to read fragment size for obu #"
+ << obu_index << "/" << num_expected_obus;
return {};
}
if (fragment_size > payload.Length()) {
// Malformed input: written size is larger than remaining buffer.
- RTC_DLOG(WARNING) << "Malformed fragment size " << fragment_size
- << " is larger than remaining size "
- << payload.Length() << " while reading obu #"
- << obu_index << "/" << num_expected_obus;
+ RTC_DLOG(LS_WARNING) << "Malformed fragment size " << fragment_size
+ << " is larger than remaining size "
+ << payload.Length() << " while reading obu #"
+ << obu_index << "/" << num_expected_obus;
return {};
}
} else {
@@ -254,7 +256,7 @@
expect_continues_obu = RtpEndsWithFragment(aggregation_header);
}
if (expect_continues_obu) {
- RTC_DLOG(WARNING) << "Last packet shouldn't have last obu fragmented.";
+ RTC_DLOG(LS_WARNING) << "Last packet shouldn't have last obu fragmented.";
return {};
}
return obu_infos;
@@ -278,7 +280,7 @@
// Returns false if obu found to be misformed.
bool CalculateObuSizes(ObuInfo* obu_info) {
if (obu_info->data.empty()) {
- RTC_DLOG(WARNING) << "Invalid bitstream: empty obu provided.";
+ RTC_DLOG(LS_WARNING) << "Invalid bitstream: empty obu provided.";
return false;
}
auto it = obu_info->data.begin();
@@ -305,7 +307,7 @@
uint8_t leb128_byte;
do {
if (it == obu_info->data.end() || size_of_obu_size_bytes >= 8) {
- RTC_DLOG(WARNING)
+ RTC_DLOG(LS_WARNING)
<< "Failed to read obu_size. obu_size field is too long: "
<< size_of_obu_size_bytes << " bytes processed.";
return false;
@@ -321,8 +323,9 @@
obu_info->data.size() - obu_info->prefix_size - size_of_obu_size_bytes;
if (obu_size_bytes != obu_info->payload_size) {
// obu_size was present in the bitstream and mismatches calculated size.
- RTC_DLOG(WARNING) << "Mismatch in obu_size. signaled: " << obu_size_bytes
- << ", actual: " << obu_info->payload_size;
+ RTC_DLOG(LS_WARNING) << "Mismatch in obu_size. signaled: "
+ << obu_size_bytes
+ << ", actual: " << obu_info->payload_size;
return false;
}
}
diff --git a/modules/video_coding/fec_controller_default.cc b/modules/video_coding/fec_controller_default.cc
index 88315a8..f204b01 100644
--- a/modules/video_coding/fec_controller_default.cc
+++ b/modules/video_coding/fec_controller_default.cc
@@ -70,8 +70,9 @@
<< overhead_threshold;
return overhead_threshold;
} else if (overhead_threshold < 0 || overhead_threshold > 1) {
- RTC_LOG(WARNING) << "ProtectionOverheadRateThreshold field trial is set to "
- "an invalid value, expecting a value between (0, 1].";
+ RTC_LOG(LS_WARNING)
+ << "ProtectionOverheadRateThreshold field trial is set to "
+ "an invalid value, expecting a value between (0, 1].";
}
// WebRTC-ProtectionOverheadRateThreshold field trial string is not found, use
// the default value.
diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc
index 655490b..f6a3858 100644
--- a/p2p/base/p2p_transport_channel.cc
+++ b/p2p/base/p2p_transport_channel.cc
@@ -1569,8 +1569,8 @@
if (val < 0) {
// Because this also occurs deferred, probably no point in reporting an
// error
- RTC_LOG(WARNING) << "SetOption(" << opt << ", " << value
- << ") failed: " << port->GetError();
+ RTC_LOG(LS_WARNING) << "SetOption(" << opt << ", " << value
+ << ") failed: " << port->GetError();
}
}
return 0;
diff --git a/p2p/base/pseudo_tcp.cc b/p2p/base/pseudo_tcp.cc
index f168227..eff86e8 100644
--- a/p2p/base/pseudo_tcp.cc
+++ b/p2p/base/pseudo_tcp.cc
@@ -355,7 +355,7 @@
bool PseudoTcp::NotifyPacket(const char* buffer, size_t len) {
if (len > MAX_PACKET) {
- RTC_LOG_F(WARNING) << "packet too large";
+ RTC_LOG_F(LS_WARNING) << "packet too large";
return false;
}
return parse(reinterpret_cast<const uint8_t*>(buffer), uint32_t(len));
@@ -1240,7 +1240,7 @@
// Window scale factor.
// http://www.ietf.org/rfc/rfc1323.txt
if (len != 1) {
- RTC_LOG_F(WARNING) << "Invalid window scale option received.";
+ RTC_LOG_F(LS_WARNING) << "Invalid window scale option received.";
return;
}
applyWindowScaleOption(data[0]);
diff --git a/pc/ice_server_parsing.cc b/pc/ice_server_parsing.cc
index b70322e..88f77bf 100644
--- a/pc/ice_server_parsing.cc
+++ b/pc/ice_server_parsing.cc
@@ -211,19 +211,19 @@
}
if (hoststring.find('@') != std::string::npos) {
- RTC_LOG(WARNING) << "Invalid url: " << uri_without_transport;
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING) << "Invalid url: " << uri_without_transport;
+ RTC_LOG(LS_WARNING)
<< "Note that user-info@ in turn:-urls is long-deprecated.";
return RTCErrorType::SYNTAX_ERROR;
}
std::string address;
if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
- RTC_LOG(WARNING) << "Invalid hostname format: " << uri_without_transport;
+ RTC_LOG(LS_WARNING) << "Invalid hostname format: " << uri_without_transport;
return RTCErrorType::SYNTAX_ERROR;
}
if (port <= 0 || port > 0xffff) {
- RTC_LOG(WARNING) << "Invalid port: " << port;
+ RTC_LOG(LS_WARNING) << "Invalid port: " << port;
return RTCErrorType::SYNTAX_ERROR;
}
diff --git a/rtc_base/http_common.cc b/rtc_base/http_common.cc
index 1dd4a20..0d78322 100644
--- a/rtc_base/http_common.cc
+++ b/rtc_base/http_common.cc
@@ -373,7 +373,7 @@
if (DsMakeSpn("HTTP", server.HostAsURIString().c_str(), nullptr,
server.port(),
0, &len, spn) != ERROR_SUCCESS) {
- RTC_LOG_F(WARNING) << "(Negotiate) - DsMakeSpn failed";
+ RTC_LOG_F(LS_WARNING) << "(Negotiate) - DsMakeSpn failed";
return HAR_IGNORE;
}
#else
@@ -413,8 +413,8 @@
if (neg) {
const size_t max_steps = 10;
if (++neg->steps >= max_steps) {
- RTC_LOG(WARNING) << "AsyncHttpsProxySocket::Authenticate(Negotiate) "
- "too many retries";
+ RTC_LOG(LS_WARNING) << "AsyncHttpsProxySocket::Authenticate(Negotiate) "
+ "too many retries";
return HAR_ERROR;
}
steps = neg->steps;
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index 98729ef..3ac12d7 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -92,8 +92,7 @@
// Compatibility aliases, to be deleted.
// TODO(bugs.webrtc.org/13362): Remove usage and delete.
INFO [[deprecated("Use LS_INFO")]] = LS_INFO,
- // WARNING [[deprecated("Use LS_WARNING")]] = LS_WARNING,
- WARNING = LS_WARNING,
+ WARNING [[deprecated("Use LS_WARNING")]] = LS_WARNING,
LERROR [[deprecated("Use LS_ERROR")]] = LS_ERROR
};
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index 61acf54..33ebb69 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -1686,31 +1686,31 @@
{
if ((wsaEvents.lNetworkEvents & FD_READ) &&
wsaEvents.iErrorCode[FD_READ_BIT] != 0) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "PhysicalSocketServer got FD_READ_BIT error "
<< wsaEvents.iErrorCode[FD_READ_BIT];
}
if ((wsaEvents.lNetworkEvents & FD_WRITE) &&
wsaEvents.iErrorCode[FD_WRITE_BIT] != 0) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "PhysicalSocketServer got FD_WRITE_BIT error "
<< wsaEvents.iErrorCode[FD_WRITE_BIT];
}
if ((wsaEvents.lNetworkEvents & FD_CONNECT) &&
wsaEvents.iErrorCode[FD_CONNECT_BIT] != 0) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "PhysicalSocketServer got FD_CONNECT_BIT error "
<< wsaEvents.iErrorCode[FD_CONNECT_BIT];
}
if ((wsaEvents.lNetworkEvents & FD_ACCEPT) &&
wsaEvents.iErrorCode[FD_ACCEPT_BIT] != 0) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "PhysicalSocketServer got FD_ACCEPT_BIT error "
<< wsaEvents.iErrorCode[FD_ACCEPT_BIT];
}
if ((wsaEvents.lNetworkEvents & FD_CLOSE) &&
wsaEvents.iErrorCode[FD_CLOSE_BIT] != 0) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "PhysicalSocketServer got FD_CLOSE_BIT error "
<< wsaEvents.iErrorCode[FD_CLOSE_BIT];
}
diff --git a/rtc_base/win/scoped_com_initializer.cc b/rtc_base/win/scoped_com_initializer.cc
index e791adc..4b56772 100644
--- a/rtc_base/win/scoped_com_initializer.cc
+++ b/rtc_base/win/scoped_com_initializer.cc
@@ -49,7 +49,7 @@
RTC_DLOG(LS_INFO)
<< "The COM library was initialized successfully on this thread";
} else if (hr_ == S_FALSE) {
- RTC_DLOG(WARNING)
+ RTC_DLOG(LS_WARNING)
<< "The COM library is already initialized on this thread";
}
}
diff --git a/sdk/android/src/jni/audio_device/aaudio_player.cc b/sdk/android/src/jni/audio_device/aaudio_player.cc
index 29bcfae..ae8fcb9 100644
--- a/sdk/android/src/jni/audio_device/aaudio_player.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_player.cc
@@ -158,7 +158,7 @@
if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
// The stream is disconnected and any attempt to use it will return
// AAUDIO_ERROR_DISCONNECTED.
- RTC_LOG(WARNING) << "Output stream disconnected";
+ RTC_LOG(LS_WARNING) << "Output stream disconnected";
// AAudio documentation states: "You should not close or reopen the stream
// from the callback, use another thread instead". A message is therefore
// sent to the main thread to do the restart operation.
diff --git a/sdk/android/src/jni/audio_device/aaudio_recorder.cc b/sdk/android/src/jni/audio_device/aaudio_recorder.cc
index 8ab097d..d66c1d0 100644
--- a/sdk/android/src/jni/audio_device/aaudio_recorder.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_recorder.cc
@@ -148,7 +148,7 @@
if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
// The stream is disconnected and any attempt to use it will return
// AAUDIO_ERROR_DISCONNECTED..
- RTC_LOG(WARNING) << "Input stream disconnected => restart is required";
+ RTC_LOG(LS_WARNING) << "Input stream disconnected => restart is required";
// AAudio documentation states: "You should not close or reopen the stream
// from the callback, use another thread instead". A message is therefore
// sent to the main thread to do the restart operation.
diff --git a/sdk/android/src/jni/audio_device/aaudio_wrapper.cc b/sdk/android/src/jni/audio_device/aaudio_wrapper.cc
index 8fc8e78..6c20703 100644
--- a/sdk/android/src/jni/audio_device/aaudio_wrapper.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_wrapper.cc
@@ -92,8 +92,8 @@
aaudio_result_t error) {
RTC_DCHECK(user_data);
AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
- RTC_LOG(WARNING) << "ErrorCallback: "
- << DirectionToString(aaudio_wrapper->direction());
+ RTC_LOG(LS_WARNING) << "ErrorCallback: "
+ << DirectionToString(aaudio_wrapper->direction());
RTC_DCHECK(aaudio_wrapper->observer());
aaudio_wrapper->observer()->OnErrorCallback(error);
}
diff --git a/sdk/android/src/jni/audio_device/audio_device_module.cc b/sdk/android/src/jni/audio_device/audio_device_module.cc
index 21c644f..2e75db9 100644
--- a/sdk/android/src/jni/audio_device/audio_device_module.cc
+++ b/sdk/android/src/jni/audio_device/audio_device_module.cc
@@ -454,7 +454,7 @@
// to call this method if that same state is not modified.
bool available = is_stereo_playout_supported_;
if (enable != available) {
- RTC_LOG(WARNING) << "changing stereo playout not supported";
+ RTC_LOG(LS_WARNING) << "changing stereo playout not supported";
return -1;
}
return 0;
@@ -481,7 +481,7 @@
// to call this method if that same state is not modified.
bool available = is_stereo_record_supported_;
if (enable != available) {
- RTC_LOG(WARNING) << "changing stereo recording not supported";
+ RTC_LOG(LS_WARNING) << "changing stereo recording not supported";
return -1;
}
return 0;
diff --git a/sdk/android/src/jni/audio_device/opensles_common.cc b/sdk/android/src/jni/audio_device/opensles_common.cc
index abc415d..300019a 100644
--- a/sdk/android/src/jni/audio_device/opensles_common.cc
+++ b/sdk/android/src/jni/audio_device/opensles_common.cc
@@ -113,7 +113,8 @@
// If one already has been created, return existing object instead of
// creating a new.
if (engine_object_.Get() != nullptr) {
- RTC_LOG(WARNING) << "The OpenSL ES engine object has already been created";
+ RTC_LOG(LS_WARNING)
+ << "The OpenSL ES engine object has already been created";
return engine_object_.Get();
}
// Create the engine object in thread safe mode.
diff --git a/sdk/objc/native/src/audio/audio_device_module_ios.mm b/sdk/objc/native/src/audio/audio_device_module_ios.mm
index d0049c3..33ba926 100644
--- a/sdk/objc/native/src/audio/audio_device_module_ios.mm
+++ b/sdk/objc/native/src/audio/audio_device_module_ios.mm
@@ -291,7 +291,7 @@
RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (enable) {
- RTC_LOG(WARNING) << "recording in stereo is not supported";
+ RTC_LOG(LS_WARNING) << "recording in stereo is not supported";
}
return -1;
}
@@ -328,7 +328,7 @@
return -1;
}
if (audio_device_->SetStereoPlayout(enable)) {
- RTC_LOG(WARNING) << "stereo playout is not supported";
+ RTC_LOG(LS_WARNING) << "stereo playout is not supported";
return -1;
}
int8_t nChannels(1);
diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
index ab5cb54..da49d67 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -289,7 +289,7 @@
auto it = captured_frames_in_flight_.find(frame_id);
if (it == captured_frames_in_flight_.end()) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "The encoding of video frame with id [" << frame_id << "] for peer ["
<< peer_name << "] finished after all receivers rendered this frame. "
<< "It can be OK for simulcast/SVC if higher quality stream is not "
@@ -427,9 +427,10 @@
reason = kSkipRenderedFrameReasonDropped;
}
}
- RTC_LOG(WARNING) << "Peer " << peer_name
- << "; Received frame out of order: received frame with id "
- << frame.id() << " which was " << reason << " before";
+ RTC_LOG(LS_WARNING)
+ << "Peer " << peer_name
+ << "; Received frame out of order: received frame with id "
+ << frame.id() << " which was " << reason << " before";
return;
}
diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
index 669fcfe..0b9e2cd 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
@@ -34,8 +34,8 @@
frames_in_flight_.insert(frame_id);
frames_to_stream_label_.insert({frame_id, stream_label});
} else {
- RTC_LOG(WARNING) << "Meet new frame with the same id: " << frame_id
- << ". Assumes old one as dropped";
+ RTC_LOG(LS_WARNING) << "Meet new frame with the same id: " << frame_id
+ << ". Assumes old one as dropped";
// We needn't insert frame to frames_in_flight_, because it is already
// there.
++frames_dropped_;
diff --git a/test/pc/e2e/test_activities_executor.cc b/test/pc/e2e/test_activities_executor.cc
index ded3992..68f6760 100644
--- a/test/pc/e2e/test_activities_executor.cc
+++ b/test/pc/e2e/test_activities_executor.cc
@@ -75,8 +75,8 @@
? TimeDelta::Zero()
: activity.initial_delay_since_start - (Now() - start_time);
if (remaining_delay < TimeDelta::Zero()) {
- RTC_LOG(WARNING) << "Executing late task immediately, late by="
- << ToString(remaining_delay.Abs());
+ RTC_LOG(LS_WARNING) << "Executing late task immediately, late by="
+ << ToString(remaining_delay.Abs());
remaining_delay = TimeDelta::Zero();
}
