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Deprecate and remove usage for WARNING log level

Browse source 
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}
This commit is contained in:
Harald Alvestrand 2021-11-27 21:31:08 +00:00 committed by WebRTC LUCI CQ
parent 150503566c
commit ef5b21e637
34 changed files with 91 additions and 85 deletions
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4
examples/peerconnection/client/peer_connection_client.cc
View file

@ -77,7 +77,7 @@ void PeerConnectionClient::Connect(const std::string& server,
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 @@ void PeerConnectionClient::OnClose(rtc::Socket* socket, int err) {
}
} 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 {

6
examples/unityplugin/simple_peer_connection.cc
View file

@ -343,7 +343,7 @@ bool SimplePeerConnection::SetRemoteDescription(const char* type,
webrtc::SessionDescriptionInterface* session_description(
webrtc::CreateSessionDescription(desc_type, remote_desc, &error));
if (!session_description) {
RTC_LOG(WARNING) << "Can't parse received session description message. "
RTC_LOG(LS_WARNING) << "Can't parse received session description message. "
"SdpParseError was: "
<< error.description;
return false;
@ -365,13 +365,13 @@ bool SimplePeerConnection::AddIceCandidate(const char* candidate,
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. "
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;

2
modules/audio_device/android/aaudio_player.cc
View file

@ -142,7 +142,7 @@ void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {
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.

2
modules/audio_device/android/aaudio_recorder.cc
View file

@ -137,7 +137,7 @@ void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
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.

2
modules/audio_device/android/aaudio_wrapper.cc
View file

@ -91,7 +91,7 @@ void ErrorCallback(AAudioStream* stream,
aaudio_result_t error) {
RTC_DCHECK(user_data);
AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
RTC_LOG(WARNING) << "ErrorCallback: "
RTC_LOG(LS_WARNING) << "ErrorCallback: "
<< DirectionToString(aaudio_wrapper->direction());
RTC_DCHECK(aaudio_wrapper->observer());
aaudio_wrapper->observer()->OnErrorCallback(error);

4
modules/audio_device/android/audio_device_template.h
View file

@ -171,7 +171,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
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 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
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();

3
modules/audio_device/android/audio_manager.cc
View file

@ -123,7 +123,8 @@ SLObjectItf AudioManager::GetOpenSLEngine() {
// 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.

2
modules/audio_device/audio_device_buffer.cc
View file

@ -64,7 +64,7 @@ AudioDeviceBuffer::AudioDeviceBuffer(TaskQueueFactory* task_queue_factory)
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
}

8
modules/audio_device/audio_device_impl.cc
View file

@ -251,7 +251,7 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
// - 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 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
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 @@ int32_t AudioDeviceModuleImpl::SetStereoRecording(bool enable) {
}
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 @@ int32_t AudioDeviceModuleImpl::SetStereoPlayout(bool enable) {
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);

2
modules/audio_device/win/core_audio_base_win.cc
View file

@ -506,7 +506,7 @@ bool CoreAudioBase::Init() {
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()
RTC_LOG(LS_WARNING) << "Buffer size of " << params.frames_per_buffer()
<< " is not an even divisor of "
<< preferred_frames_per_buffer;
}

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

@ -187,7 +187,7 @@ int CoreAudioInput::StopRecording() {
// 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 @@ bool CoreAudioInput::OnErrorCallback(ErrorType error) {
if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
HandleStreamDisconnected();
} else {
RTC_DLOG(WARNING) << "Unsupported error type";
RTC_DLOG(LS_WARNING) << "Unsupported error type";
}
return true;
}

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

@ -188,7 +188,7 @@ int CoreAudioOutput::StopPlayout() {
// 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 @@ bool CoreAudioOutput::OnErrorCallback(ErrorType error) {
if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
HandleStreamDisconnected();
} else {
RTC_DLOG(WARNING) << "Unsupported error type";
RTC_DLOG(LS_WARNING) << "Unsupported error type";
}
return true;
}

2
modules/audio_device/win/core_audio_utility_win.cc
View file

@ -1020,7 +1020,7 @@ HRESULT GetSharedModeMixFormat(IAudioClient* client,
// 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.";

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

@ -44,7 +44,7 @@ size_t ReadTrendlineFilterWindowSize(
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.";

2
modules/desktop_capture/win/screen_capturer_win_gdi.cc
View file

@ -86,7 +86,7 @@ void ScreenCapturerWinGdi::CaptureFrame() {
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;
}

4
modules/remote_bitrate_estimator/aimd_rate_control.cc
View file

@ -53,9 +53,9 @@ double ReadBackoffFactor(const WebRtcKeyValueConfig& key_value_config) {
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;
}

21
modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
View file

@ -191,14 +191,15 @@ VectorObuInfo ParseObus(
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 @@ VectorObuInfo ParseObus(
// 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) {
@ -228,13 +230,13 @@ VectorObuInfo ParseObus(
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 #"
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
RTC_DLOG(LS_WARNING) << "Malformed fragment size " << fragment_size
<< " is larger than remaining size "
<< payload.Length() << " while reading obu #"
<< obu_index << "/" << num_expected_obus;
@ -254,7 +256,7 @@ VectorObuInfo ParseObus(
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 @@ int WriteLeb128(uint32_t value, uint8_t* buffer) {
// 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 @@ bool CalculateObuSizes(ObuInfo* obu_info) {
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;
@ -321,7 +323,8 @@ bool CalculateObuSizes(ObuInfo* obu_info) {
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
RTC_DLOG(LS_WARNING) << "Mismatch in obu_size. signaled: "
<< obu_size_bytes
<< ", actual: " << obu_info->payload_size;
return false;
}

3
modules/video_coding/fec_controller_default.cc
View file

@ -70,7 +70,8 @@ float FecControllerDefault::GetProtectionOverheadRateThreshold() {
<< overhead_threshold;
return overhead_threshold;
} else if (overhead_threshold < 0 || overhead_threshold > 1) {
RTC_LOG(WARNING) << "ProtectionOverheadRateThreshold field trial is set to "
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

2
p2p/base/p2p_transport_channel.cc
View file

@ -1569,7 +1569,7 @@ int P2PTransportChannel::SetOption(rtc::Socket::Option opt, int value) {
if (val < 0) {
// Because this also occurs deferred, probably no point in reporting an
// error
RTC_LOG(WARNING) << "SetOption(" << opt << ", " << value
RTC_LOG(LS_WARNING) << "SetOption(" << opt << ", " << value
<< ") failed: " << port->GetError();
}
}

4
p2p/base/pseudo_tcp.cc
View file

@ -355,7 +355,7 @@ void PseudoTcp::NotifyClock(uint32_t now) {
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 @@ void PseudoTcp::applyOption(char kind, const char* data, uint32_t len) {
// 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]);

8
pc/ice_server_parsing.cc
View file

@ -211,19 +211,19 @@ static RTCErrorType ParseIceServerUrl(
}
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;
}

4
rtc_base/http_common.cc
View file

@ -373,7 +373,7 @@ HttpAuthResult HttpAuthenticate(const char* challenge,
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
@ -413,7 +413,7 @@ HttpAuthResult HttpAuthenticate(const char* challenge,
if (neg) {
const size_t max_steps = 10;
if (++neg->steps >= max_steps) {
RTC_LOG(WARNING) << "AsyncHttpsProxySocket::Authenticate(Negotiate) "
RTC_LOG(LS_WARNING) << "AsyncHttpsProxySocket::Authenticate(Negotiate) "
"too many retries";
return HAR_ERROR;
}

3
rtc_base/logging.h
View file

@ -92,8 +92,7 @@ enum LoggingSeverity {
// 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
};

10
rtc_base/physical_socket_server.cc
View file

@ -1686,31 +1686,31 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
{
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];
}

2
rtc_base/win/scoped_com_initializer.cc
View file

@ -49,7 +49,7 @@ void ScopedCOMInitializer::Initialize(COINIT init) {
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";
}
}

2
sdk/android/src/jni/audio_device/aaudio_player.cc
View file

@ -158,7 +158,7 @@ void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {
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.

2
sdk/android/src/jni/audio_device/aaudio_recorder.cc
View file

@ -148,7 +148,7 @@ void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
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.

2
sdk/android/src/jni/audio_device/aaudio_wrapper.cc
View file

@ -92,7 +92,7 @@ void ErrorCallback(AAudioStream* stream,
aaudio_result_t error) {
RTC_DCHECK(user_data);
AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
RTC_LOG(WARNING) << "ErrorCallback: "
RTC_LOG(LS_WARNING) << "ErrorCallback: "
<< DirectionToString(aaudio_wrapper->direction());
RTC_DCHECK(aaudio_wrapper->observer());
aaudio_wrapper->observer()->OnErrorCallback(error);

4
sdk/android/src/jni/audio_device/audio_device_module.cc
View file

@ -454,7 +454,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
// 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 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
// 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;

3
sdk/android/src/jni/audio_device/opensles_common.cc
View file

@ -113,7 +113,8 @@ SLObjectItf OpenSLEngineManager::GetOpenSLEngine() {
// 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.

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

@ -291,7 +291,7 @@ AudioDeviceModuleIOS::AudioDeviceModuleIOS(bool bypass_voice_processing)
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 @@ AudioDeviceModuleIOS::AudioDeviceModuleIOS(bool bypass_voice_processing)
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);

5
test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
View file

@ -289,7 +289,7 @@ void DefaultVideoQualityAnalyzer::OnFrameEncoded(
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 "
@ -427,7 +427,8 @@ void DefaultVideoQualityAnalyzer::OnFrameRendered(
reason = kSkipRenderedFrameReasonDropped;
}
}
RTC_LOG(WARNING) << "Peer " << peer_name
RTC_LOG(LS_WARNING)
<< "Peer " << peer_name
<< "; Received frame out of order: received frame with id "
<< frame.id() << " which was " << reason << " before";
return;

2
test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
View file

@ -34,7 +34,7 @@ uint16_t ExampleVideoQualityAnalyzer::OnFrameCaptured(
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
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.

2
test/pc/e2e/test_activities_executor.cc
View file

@ -75,7 +75,7 @@ void TestActivitiesExecutor::PostActivity(ScheduledActivity activity) {
? TimeDelta::Zero()
: activity.initial_delay_since_start - (Now() - start_time);
if (remaining_delay < TimeDelta::Zero()) {
RTC_LOG(WARNING) << "Executing late task immediately, late by="
RTC_LOG(LS_WARNING) << "Executing late task immediately, late by="
<< ToString(remaining_delay.Abs());
remaining_delay = TimeDelta::Zero();
}