Remove usage of INFO alias for LS_INFO in log messages
Bug: webrtc:13362
Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35310}
diff --git a/api/video_codecs/video_encoder_software_fallback_wrapper.cc b/api/video_codecs/video_encoder_software_fallback_wrapper.cc
index 8202217..72e08a7 100644
--- a/api/video_codecs/video_encoder_software_fallback_wrapper.cc
+++ b/api/video_codecs/video_encoder_software_fallback_wrapper.cc
@@ -358,8 +358,8 @@
fallback_encoder_->GetEncoderInfo().supports_native_handle) {
return fallback_encoder_->Encode(frame, frame_types);
} else {
- RTC_LOG(INFO) << "Fallback encoder does not support native handle - "
- "converting frame to I420";
+ RTC_LOG(LS_INFO) << "Fallback encoder does not support native handle - "
+ "converting frame to I420";
rtc::scoped_refptr<I420BufferInterface> src_buffer =
frame.video_frame_buffer()->ToI420();
if (!src_buffer) {
diff --git a/api/voip/voip_engine_factory.cc b/api/voip/voip_engine_factory.cc
index 88f63f9..8da53ce 100644
--- a/api/voip/voip_engine_factory.cc
+++ b/api/voip/voip_engine_factory.cc
@@ -24,7 +24,7 @@
RTC_CHECK(config.audio_device_module);
if (!config.audio_processing) {
- RTC_DLOG(INFO) << "No audio processing functionality provided.";
+ RTC_DLOG(LS_INFO) << "No audio processing functionality provided.";
}
return std::make_unique<VoipCore>(std::move(config.encoder_factory),
diff --git a/audio/audio_state.cc b/audio/audio_state.cc
index 0e60f03..9e5b63b 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -123,7 +123,7 @@
}
void AudioState::SetPlayout(bool enabled) {
- RTC_LOG(INFO) << "SetPlayout(" << enabled << ")";
+ RTC_LOG(LS_INFO) << "SetPlayout(" << enabled << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
if (playout_enabled_ != enabled) {
playout_enabled_ = enabled;
@@ -140,7 +140,7 @@
}
void AudioState::SetRecording(bool enabled) {
- RTC_LOG(INFO) << "SetRecording(" << enabled << ")";
+ RTC_LOG(LS_INFO) << "SetRecording(" << enabled << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
if (recording_enabled_ != enabled) {
recording_enabled_ = enabled;
diff --git a/call/adaptation/resource_adaptation_processor.cc b/call/adaptation/resource_adaptation_processor.cc
index 3c06675..d95cd75 100644
--- a/call/adaptation/resource_adaptation_processor.cc
+++ b/call/adaptation/resource_adaptation_processor.cc
@@ -129,7 +129,7 @@
resources_.push_back(resource);
}
resource->SetResourceListener(resource_listener_delegate_);
- RTC_LOG(INFO) << "Registered resource \"" << resource->Name() << "\".";
+ RTC_LOG(LS_INFO) << "Registered resource \"" << resource->Name() << "\".";
}
std::vector<rtc::scoped_refptr<Resource>>
@@ -141,7 +141,7 @@
void ResourceAdaptationProcessor::RemoveResource(
rtc::scoped_refptr<Resource> resource) {
RTC_DCHECK(resource);
- RTC_LOG(INFO) << "Removing resource \"" << resource->Name() << "\".";
+ RTC_LOG(LS_INFO) << "Removing resource \"" << resource->Name() << "\".";
resource->SetResourceListener(nullptr);
{
MutexLock crit(&resources_lock_);
@@ -188,10 +188,11 @@
RTC_DCHECK_EQ(adapt_to.status(), Adaptation::Status::kValid);
stream_adapter_->ApplyAdaptation(adapt_to, nullptr);
- RTC_LOG(INFO) << "Most limited resource removed. Restoring restrictions to "
- "next most limited restrictions: "
- << most_limited.restrictions.ToString() << " with counters "
- << most_limited.counters.ToString();
+ RTC_LOG(LS_INFO)
+ << "Most limited resource removed. Restoring restrictions to "
+ "next most limited restrictions: "
+ << most_limited.restrictions.ToString() << " with counters "
+ << most_limited.counters.ToString();
}
}
@@ -204,8 +205,8 @@
{
MutexLock crit(&resources_lock_);
if (absl::c_find(resources_, resource) == resources_.end()) {
- RTC_LOG(INFO) << "Ignoring signal from removed resource \""
- << resource->Name() << "\".";
+ RTC_LOG(LS_INFO) << "Ignoring signal from removed resource \""
+ << resource->Name() << "\".";
return;
}
}
@@ -226,9 +227,9 @@
// successfully adapted since - don't log to avoid spam.
return;
}
- RTC_LOG(INFO) << "Resource \"" << resource->Name() << "\" signalled "
- << ResourceUsageStateToString(usage_state) << ". "
- << result_and_message.message;
+ RTC_LOG(LS_INFO) << "Resource \"" << resource->Name() << "\" signalled "
+ << ResourceUsageStateToString(usage_state) << ". "
+ << result_and_message.message;
if (result_and_message.result == MitigationResult::kAdaptationApplied) {
previous_mitigation_results_.clear();
} else {
diff --git a/call/adaptation/video_stream_adapter.cc b/call/adaptation/video_stream_adapter.cc
index 49a4d6a..ca62502 100644
--- a/call/adaptation/video_stream_adapter.cc
+++ b/call/adaptation/video_stream_adapter.cc
@@ -234,7 +234,7 @@
void VideoStreamAdapter::ClearRestrictions() {
RTC_DCHECK_RUN_ON(&sequence_checker_);
// Invalidate any previously returned Adaptation.
- RTC_LOG(INFO) << "Resetting restrictions";
+ RTC_LOG(LS_INFO) << "Resetting restrictions";
++adaptation_validation_id_;
current_restrictions_ = {VideoSourceRestrictions(),
VideoAdaptationCounters()};
@@ -333,8 +333,8 @@
if (!constraint->IsAdaptationUpAllowed(input_state,
current_restrictions_.restrictions,
restrictions.restrictions)) {
- RTC_LOG(INFO) << "Not adapting up because constraint \""
- << constraint->Name() << "\" disallowed it";
+ RTC_LOG(LS_INFO) << "Not adapting up because constraint \""
+ << constraint->Name() << "\" disallowed it";
step = Adaptation::Status::kRejectedByConstraint;
}
}
diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc
index d3b1d21..5f213de 100644
--- a/examples/peerconnection/client/conductor.cc
+++ b/examples/peerconnection/client/conductor.cc
@@ -61,10 +61,10 @@
static DummySetSessionDescriptionObserver* Create() {
return new rtc::RefCountedObject<DummySetSessionDescriptionObserver>();
}
- virtual void OnSuccess() { RTC_LOG(INFO) << __FUNCTION__; }
+ virtual void OnSuccess() { RTC_LOG(LS_INFO) << __FUNCTION__; }
virtual void OnFailure(webrtc::RTCError error) {
- RTC_LOG(INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
- << error.message();
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
+ << error.message();
}
};
@@ -221,23 +221,23 @@
rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
streams) {
- RTC_LOG(INFO) << __FUNCTION__ << " " << receiver->id();
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
main_wnd_->QueueUIThreadCallback(NEW_TRACK_ADDED,
receiver->track().release());
}
void Conductor::OnRemoveTrack(
rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) {
- RTC_LOG(INFO) << __FUNCTION__ << " " << receiver->id();
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
main_wnd_->QueueUIThreadCallback(TRACK_REMOVED, receiver->track().release());
}
void Conductor::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
- RTC_LOG(INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
// For loopback test. To save some connecting delay.
if (loopback_) {
if (!peer_connection_->AddIceCandidate(candidate)) {
- RTC_LOG(WARNING) << "Failed to apply the received candidate";
+ RTC_LOG(LS_WARNING) << "Failed to apply the received candidate";
}
return;
}
@@ -261,12 +261,12 @@
//
void Conductor::OnSignedIn() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
main_wnd_->SwitchToPeerList(client_->peers());
}
void Conductor::OnDisconnected() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
DeletePeerConnection();
@@ -275,16 +275,16 @@
}
void Conductor::OnPeerConnected(int id, const std::string& name) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
// Refresh the list if we're showing it.
if (main_wnd_->current_ui() == MainWindow::LIST_PEERS)
main_wnd_->SwitchToPeerList(client_->peers());
}
void Conductor::OnPeerDisconnected(int id) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
if (id == peer_id_) {
- RTC_LOG(INFO) << "Our peer disconnected";
+ RTC_LOG(LS_INFO) << "Our peer disconnected";
main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_CLOSED, NULL);
} else {
// Refresh the list if we're showing it.
@@ -308,7 +308,7 @@
}
} else if (peer_id != peer_id_) {
RTC_DCHECK(peer_id_ != -1);
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "Received a message from unknown peer while already in a "
"conversation with a different peer.";
return;
@@ -317,7 +317,7 @@
Json::Reader reader;
Json::Value jmessage;
if (!reader.parse(message, jmessage)) {
- RTC_LOG(WARNING) << "Received unknown message. " << message;
+ RTC_LOG(LS_WARNING) << "Received unknown message. " << message;
return;
}
std::string type_str;
@@ -346,19 +346,21 @@
std::string sdp;
if (!rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionSdpName,
&sdp)) {
- RTC_LOG(WARNING) << "Can't parse received session description message.";
+ RTC_LOG(LS_WARNING)
+ << "Can't parse received session description message.";
return;
}
webrtc::SdpParseError error;
std::unique_ptr<webrtc::SessionDescriptionInterface> session_description =
webrtc::CreateSessionDescription(type, sdp, &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;
}
- RTC_LOG(INFO) << " Received session description :" << message;
+ RTC_LOG(LS_INFO) << " Received session description :" << message;
peer_connection_->SetRemoteDescription(
DummySetSessionDescriptionObserver::Create(),
session_description.release());
@@ -375,23 +377,23 @@
!rtc::GetIntFromJsonObject(jmessage, kCandidateSdpMlineIndexName,
&sdp_mlineindex) ||
!rtc::GetStringFromJsonObject(jmessage, kCandidateSdpName, &sdp)) {
- RTC_LOG(WARNING) << "Can't parse received message.";
+ RTC_LOG(LS_WARNING) << "Can't parse received message.";
return;
}
webrtc::SdpParseError error;
std::unique_ptr<webrtc::IceCandidateInterface> candidate(
webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, sdp, &error));
if (!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;
}
if (!peer_connection_->AddIceCandidate(candidate.get())) {
- RTC_LOG(WARNING) << "Failed to apply the received candidate";
+ RTC_LOG(LS_WARNING) << "Failed to apply the received candidate";
return;
}
- RTC_LOG(INFO) << " Received candidate :" << message;
+ RTC_LOG(LS_INFO) << " Received candidate :" << message;
}
}
@@ -475,7 +477,7 @@
}
void Conductor::DisconnectFromCurrentPeer() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
if (peer_connection_.get()) {
client_->SendHangUp(peer_id_);
DeletePeerConnection();
@@ -488,7 +490,7 @@
void Conductor::UIThreadCallback(int msg_id, void* data) {
switch (msg_id) {
case PEER_CONNECTION_CLOSED:
- RTC_LOG(INFO) << "PEER_CONNECTION_CLOSED";
+ RTC_LOG(LS_INFO) << "PEER_CONNECTION_CLOSED";
DeletePeerConnection();
if (main_wnd_->IsWindow()) {
@@ -503,7 +505,7 @@
break;
case SEND_MESSAGE_TO_PEER: {
- RTC_LOG(INFO) << "SEND_MESSAGE_TO_PEER";
+ RTC_LOG(LS_INFO) << "SEND_MESSAGE_TO_PEER";
std::string* msg = reinterpret_cast<std::string*>(data);
if (msg) {
// For convenience, we always run the message through the queue.
@@ -579,7 +581,7 @@
}
void Conductor::OnFailure(webrtc::RTCError error) {
- RTC_LOG(LERROR) << ToString(error.type()) << ": " << error.message();
+ RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
}
void Conductor::SendMessage(const std::string& json_object) {
diff --git a/examples/peerconnection/client/linux/main_wnd.cc b/examples/peerconnection/client/linux/main_wnd.cc
index 7dcfa89..fe29d62 100644
--- a/examples/peerconnection/client/linux/main_wnd.cc
+++ b/examples/peerconnection/client/linux/main_wnd.cc
@@ -252,7 +252,7 @@
}
void GtkMainWnd::SwitchToConnectUI() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(IsWindow());
RTC_DCHECK(vbox_ == NULL);
@@ -308,7 +308,7 @@
}
void GtkMainWnd::SwitchToPeerList(const Peers& peers) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
if (!peer_list_) {
gtk_container_set_border_width(GTK_CONTAINER(window_), 0);
@@ -345,7 +345,7 @@
}
void GtkMainWnd::SwitchToStreamingUI() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(draw_area_ == NULL);
diff --git a/examples/peerconnection/client/peer_connection_client.cc b/examples/peerconnection/client/peer_connection_client.cc
index 7132593..b76b925 100644
--- a/examples/peerconnection/client/peer_connection_client.cc
+++ b/examples/peerconnection/client/peer_connection_client.cc
@@ -297,7 +297,7 @@
bool ret = false;
size_t i = data->find("\r\n\r\n");
if (i != std::string::npos) {
- RTC_LOG(INFO) << "Headers received";
+ RTC_LOG(LS_INFO) << "Headers received";
if (GetHeaderValue(*data, i, "\r\nContent-Length: ", content_length)) {
size_t total_response_size = (i + 4) + *content_length;
if (data->length() >= total_response_size) {
@@ -374,7 +374,7 @@
}
void PeerConnectionClient::OnHangingGetRead(rtc::Socket* socket) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
size_t content_length = 0;
if (ReadIntoBuffer(socket, ¬ification_data_, &content_length)) {
size_t peer_id = 0, eoh = 0;
@@ -472,7 +472,7 @@
}
void PeerConnectionClient::OnClose(rtc::Socket* socket, int err) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
socket->Close();
diff --git a/examples/unityplugin/simple_peer_connection.cc b/examples/unityplugin/simple_peer_connection.cc
index e1c9c40..4e16c00 100644
--- a/examples/unityplugin/simple_peer_connection.cc
+++ b/examples/unityplugin/simple_peer_connection.cc
@@ -101,10 +101,10 @@
static DummySetSessionDescriptionObserver* Create() {
return new rtc::RefCountedObject<DummySetSessionDescriptionObserver>();
}
- virtual void OnSuccess() { RTC_LOG(INFO) << __FUNCTION__; }
+ virtual void OnSuccess() { RTC_LOG(LS_INFO) << __FUNCTION__; }
virtual void OnFailure(webrtc::RTCError error) {
- RTC_LOG(INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
- << error.message();
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
+ << error.message();
}
protected:
@@ -278,7 +278,7 @@
void SimplePeerConnection::OnIceCandidate(
const webrtc::IceCandidateInterface* candidate) {
- RTC_LOG(INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
std::string sdp;
if (!candidate->ToString(&sdp)) {
@@ -348,7 +348,7 @@
<< error.description;
return false;
}
- RTC_LOG(INFO) << " Received session description :" << remote_desc;
+ RTC_LOG(LS_INFO) << " Received session description :" << remote_desc;
peer_connection_->SetRemoteDescription(
DummySetSessionDescriptionObserver::Create(), session_description);
@@ -374,7 +374,7 @@
RTC_LOG(WARNING) << "Failed to apply the received candidate";
return false;
}
- RTC_LOG(INFO) << " Received candidate :" << candidate;
+ RTC_LOG(LS_INFO) << " Received candidate :" << candidate;
return true;
}
@@ -409,7 +409,7 @@
void SimplePeerConnection::OnAddStream(
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {
- RTC_LOG(INFO) << __FUNCTION__ << " " << stream->id();
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << stream->id();
remote_stream_ = stream;
if (remote_video_observer_ && !remote_stream_->GetVideoTracks().empty()) {
remote_stream_->GetVideoTracks()[0]->AddOrUpdateSink(
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
index 2bd7507..6422ea0 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
@@ -674,7 +674,7 @@
encoder->set_payload_type(codec.payload_type);
if (event.config().codecs.size() > 1) {
- RTC_LOG(WARNING)
+ RTC_LOG(LS_WARNING)
<< "LogVideoSendStreamConfig currently only supports one "
"codec. Logging codec :"
<< codec.payload_name;
diff --git a/media/sctp/usrsctp_transport_reliability_unittest.cc b/media/sctp/usrsctp_transport_reliability_unittest.cc
index 104e320..a06d8d8 100644
--- a/media/sctp/usrsctp_transport_reliability_unittest.cc
+++ b/media/sctp/usrsctp_transport_reliability_unittest.cc
@@ -259,8 +259,8 @@
}
if (num_messages_received_ % kLogPerMessagesCount == 0) {
- RTC_LOG(INFO) << receiver_id_ << " receiver got "
- << num_messages_received_ << " messages";
+ RTC_LOG(LS_INFO) << receiver_id_ << " receiver got "
+ << num_messages_received_ << " messages";
}
}
diff --git a/modules/audio_device/android/aaudio_player.cc b/modules/audio_device/android/aaudio_player.cc
index 7f63512..a5a3675 100644
--- a/modules/audio_device/android/aaudio_player.cc
+++ b/modules/audio_device/android/aaudio_player.cc
@@ -27,19 +27,19 @@
AAudioPlayer::AAudioPlayer(AudioManager* audio_manager)
: main_thread_(rtc::Thread::Current()),
aaudio_(audio_manager, AAUDIO_DIRECTION_OUTPUT, this) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
thread_checker_aaudio_.Detach();
}
AAudioPlayer::~AAudioPlayer() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
Terminate();
- RTC_LOG(INFO) << "#detected underruns: " << underrun_count_;
+ RTC_LOG(LS_INFO) << "#detected underruns: " << underrun_count_;
}
int AAudioPlayer::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
if (aaudio_.audio_parameters().channels() == 2) {
RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
@@ -48,14 +48,14 @@
}
int AAudioPlayer::Terminate() {
- RTC_LOG(INFO) << "Terminate";
+ RTC_LOG(LS_INFO) << "Terminate";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
StopPlayout();
return 0;
}
int AAudioPlayer::InitPlayout() {
- RTC_LOG(INFO) << "InitPlayout";
+ RTC_LOG(LS_INFO) << "InitPlayout";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
RTC_DCHECK(!initialized_);
RTC_DCHECK(!playing_);
@@ -72,7 +72,7 @@
}
int AAudioPlayer::StartPlayout() {
- RTC_LOG(INFO) << "StartPlayout";
+ RTC_LOG(LS_INFO) << "StartPlayout";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
RTC_DCHECK(!playing_);
if (!initialized_) {
@@ -93,7 +93,7 @@
}
int AAudioPlayer::StopPlayout() {
- RTC_LOG(INFO) << "StopPlayout";
+ RTC_LOG(LS_INFO) << "StopPlayout";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
if (!initialized_ || !playing_) {
return 0;
@@ -114,7 +114,7 @@
}
void AAudioPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_DLOG(INFO) << "AttachAudioBuffer";
+ RTC_DLOG(LS_INFO) << "AttachAudioBuffer";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
audio_device_buffer_ = audioBuffer;
const AudioParameters audio_parameters = aaudio_.audio_parameters();
@@ -157,9 +157,9 @@
// Log device id in first data callback to ensure that a valid device is
// utilized.
if (first_data_callback_) {
- RTC_LOG(INFO) << "--- First output data callback: "
- "device id="
- << aaudio_.device_id();
+ RTC_LOG(LS_INFO) << "--- First output data callback: "
+ "device id="
+ << aaudio_.device_id();
first_data_callback_ = false;
}
@@ -179,8 +179,8 @@
latency_millis_ = aaudio_.EstimateLatencyMillis();
// TODO(henrika): use for development only.
if (aaudio_.frames_written() % (1000 * aaudio_.frames_per_burst()) == 0) {
- RTC_DLOG(INFO) << "output latency: " << latency_millis_
- << ", num_frames: " << num_frames;
+ RTC_DLOG(LS_INFO) << "output latency: " << latency_millis_
+ << ", num_frames: " << num_frames;
}
// Read audio data from the WebRTC source using the FineAudioBuffer object
@@ -215,7 +215,7 @@
void AAudioPlayer::HandleStreamDisconnected() {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
- RTC_DLOG(INFO) << "HandleStreamDisconnected";
+ RTC_DLOG(LS_INFO) << "HandleStreamDisconnected";
if (!initialized_ || !playing_) {
return;
}
diff --git a/modules/audio_device/android/aaudio_recorder.cc b/modules/audio_device/android/aaudio_recorder.cc
index 68c9cee..d91fb9e 100644
--- a/modules/audio_device/android/aaudio_recorder.cc
+++ b/modules/audio_device/android/aaudio_recorder.cc
@@ -28,19 +28,19 @@
AAudioRecorder::AAudioRecorder(AudioManager* audio_manager)
: main_thread_(rtc::Thread::Current()),
aaudio_(audio_manager, AAUDIO_DIRECTION_INPUT, this) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
thread_checker_aaudio_.Detach();
}
AAudioRecorder::~AAudioRecorder() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
Terminate();
- RTC_LOG(INFO) << "detected owerflows: " << overflow_count_;
+ RTC_LOG(LS_INFO) << "detected owerflows: " << overflow_count_;
}
int AAudioRecorder::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK(thread_checker_.IsCurrent());
if (aaudio_.audio_parameters().channels() == 2) {
RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
@@ -49,14 +49,14 @@
}
int AAudioRecorder::Terminate() {
- RTC_LOG(INFO) << "Terminate";
+ RTC_LOG(LS_INFO) << "Terminate";
RTC_DCHECK(thread_checker_.IsCurrent());
StopRecording();
return 0;
}
int AAudioRecorder::InitRecording() {
- RTC_LOG(INFO) << "InitRecording";
+ RTC_LOG(LS_INFO) << "InitRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!initialized_);
RTC_DCHECK(!recording_);
@@ -68,7 +68,7 @@
}
int AAudioRecorder::StartRecording() {
- RTC_LOG(INFO) << "StartRecording";
+ RTC_LOG(LS_INFO) << "StartRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(initialized_);
RTC_DCHECK(!recording_);
@@ -85,7 +85,7 @@
}
int AAudioRecorder::StopRecording() {
- RTC_LOG(INFO) << "StopRecording";
+ RTC_LOG(LS_INFO) << "StopRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
if (!initialized_ || !recording_) {
return 0;
@@ -100,7 +100,7 @@
}
void AAudioRecorder::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(INFO) << "AttachAudioBuffer";
+ RTC_LOG(LS_INFO) << "AttachAudioBuffer";
RTC_DCHECK(thread_checker_.IsCurrent());
audio_device_buffer_ = audioBuffer;
const AudioParameters audio_parameters = aaudio_.audio_parameters();
@@ -114,19 +114,19 @@
}
int AAudioRecorder::EnableBuiltInAEC(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInAEC: " << enable;
+ RTC_LOG(LS_INFO) << "EnableBuiltInAEC: " << enable;
RTC_LOG(LS_ERROR) << "Not implemented";
return -1;
}
int AAudioRecorder::EnableBuiltInAGC(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInAGC: " << enable;
+ RTC_LOG(LS_INFO) << "EnableBuiltInAGC: " << enable;
RTC_LOG(LS_ERROR) << "Not implemented";
return -1;
}
int AAudioRecorder::EnableBuiltInNS(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInNS: " << enable;
+ RTC_LOG(LS_INFO) << "EnableBuiltInNS: " << enable;
RTC_LOG(LS_ERROR) << "Not implemented";
return -1;
}
@@ -154,14 +154,14 @@
int32_t num_frames) {
// TODO(henrika): figure out why we sometimes hit this one.
// RTC_DCHECK(thread_checker_aaudio_.IsCurrent());
- // RTC_LOG(INFO) << "OnDataCallback: " << num_frames;
+ // RTC_LOG(LS_INFO) << "OnDataCallback: " << num_frames;
// Drain the input buffer at first callback to ensure that it does not
// contain any old data. Will also ensure that the lowest possible latency
// is obtained.
if (first_data_callback_) {
- RTC_LOG(INFO) << "--- First input data callback: "
- "device id="
- << aaudio_.device_id();
+ RTC_LOG(LS_INFO) << "--- First input data callback: "
+ "device id="
+ << aaudio_.device_id();
aaudio_.ClearInputStream(audio_data, num_frames);
first_data_callback_ = false;
}
@@ -177,8 +177,8 @@
latency_millis_ = aaudio_.EstimateLatencyMillis();
// TODO(henrika): use for development only.
if (aaudio_.frames_read() % (1000 * aaudio_.frames_per_burst()) == 0) {
- RTC_DLOG(INFO) << "input latency: " << latency_millis_
- << ", num_frames: " << num_frames;
+ RTC_DLOG(LS_INFO) << "input latency: " << latency_millis_
+ << ", num_frames: " << num_frames;
}
// Copy recorded audio in `audio_data` to the WebRTC sink using the
// FineAudioBuffer object.
@@ -204,7 +204,7 @@
void AAudioRecorder::HandleStreamDisconnected() {
RTC_DCHECK_RUN_ON(&thread_checker_);
- RTC_LOG(INFO) << "HandleStreamDisconnected";
+ RTC_LOG(LS_INFO) << "HandleStreamDisconnected";
if (!initialized_ || !recording_) {
return;
}
diff --git a/modules/audio_device/android/aaudio_wrapper.cc b/modules/audio_device/android/aaudio_wrapper.cc
index ab12784..82860e3 100644
--- a/modules/audio_device/android/aaudio_wrapper.cc
+++ b/modules/audio_device/android/aaudio_wrapper.cc
@@ -134,23 +134,23 @@
aaudio_direction_t direction,
AAudioObserverInterface* observer)
: direction_(direction), observer_(observer) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
RTC_DCHECK(observer_);
direction_ == AAUDIO_DIRECTION_OUTPUT
? audio_parameters_ = audio_manager->GetPlayoutAudioParameters()
: audio_parameters_ = audio_manager->GetRecordAudioParameters();
aaudio_thread_checker_.Detach();
- RTC_LOG(INFO) << audio_parameters_.ToString();
+ RTC_LOG(LS_INFO) << audio_parameters_.ToString();
}
AAudioWrapper::~AAudioWrapper() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!stream_);
}
bool AAudioWrapper::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK(thread_checker_.IsCurrent());
// Creates a stream builder which can be used to open an audio stream.
ScopedStreamBuilder builder;
@@ -174,7 +174,7 @@
}
bool AAudioWrapper::Start() {
- RTC_LOG(INFO) << "Start";
+ RTC_LOG(LS_INFO) << "Start";
RTC_DCHECK(thread_checker_.IsCurrent());
// TODO(henrika): this state check might not be needed.
aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
@@ -190,7 +190,7 @@
}
bool AAudioWrapper::Stop() {
- RTC_LOG(INFO) << "Stop: " << DirectionToString(direction());
+ RTC_LOG(LS_INFO) << "Stop: " << DirectionToString(direction());
RTC_DCHECK(thread_checker_.IsCurrent());
// Asynchronous request for the stream to stop.
RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
@@ -240,7 +240,7 @@
// Returns new buffer size or a negative error value if buffer size could not
// be increased.
bool AAudioWrapper::IncreaseOutputBufferSize() {
- RTC_LOG(INFO) << "IncreaseBufferSize";
+ RTC_LOG(LS_INFO) << "IncreaseBufferSize";
RTC_DCHECK(stream_);
RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
@@ -255,20 +255,20 @@
<< ") is higher than max: " << max_buffer_size;
return false;
}
- RTC_LOG(INFO) << "Updating buffer size to: " << buffer_size
- << " (max=" << max_buffer_size << ")";
+ RTC_LOG(LS_INFO) << "Updating buffer size to: " << buffer_size
+ << " (max=" << max_buffer_size << ")";
buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
if (buffer_size < 0) {
RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
<< AAudio_convertResultToText(buffer_size);
return false;
}
- RTC_LOG(INFO) << "Buffer size changed to: " << buffer_size;
+ RTC_LOG(LS_INFO) << "Buffer size changed to: " << buffer_size;
return true;
}
void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
- RTC_LOG(INFO) << "ClearInputStream";
+ RTC_LOG(LS_INFO) << "ClearInputStream";
RTC_DCHECK(stream_);
RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
@@ -357,7 +357,7 @@
}
void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
- RTC_LOG(INFO) << "SetStreamConfiguration";
+ RTC_LOG(LS_INFO) << "SetStreamConfiguration";
RTC_DCHECK(builder);
RTC_DCHECK(thread_checker_.IsCurrent());
// Request usage of default primary output/input device.
@@ -390,7 +390,7 @@
}
bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
- RTC_LOG(INFO) << "OpenStream";
+ RTC_LOG(LS_INFO) << "OpenStream";
RTC_DCHECK(builder);
AAudioStream* stream = nullptr;
RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
@@ -400,7 +400,7 @@
}
void AAudioWrapper::CloseStream() {
- RTC_LOG(INFO) << "CloseStream";
+ RTC_LOG(LS_INFO) << "CloseStream";
RTC_DCHECK(stream_);
LOG_ON_ERROR(AAudioStream_close(stream_));
stream_ = nullptr;
@@ -419,16 +419,16 @@
ss << ", direction=" << DirectionToString(direction());
ss << ", device id=" << AAudioStream_getDeviceId(stream_);
ss << ", frames per callback=" << frames_per_callback();
- RTC_LOG(INFO) << ss.str();
+ RTC_LOG(LS_INFO) << ss.str();
}
void AAudioWrapper::LogStreamState() {
- RTC_LOG(INFO) << "AAudio stream state: "
- << AAudio_convertStreamStateToText(stream_state());
+ RTC_LOG(LS_INFO) << "AAudio stream state: "
+ << AAudio_convertStreamStateToText(stream_state());
}
bool AAudioWrapper::VerifyStreamConfiguration() {
- RTC_LOG(INFO) << "VerifyStreamConfiguration";
+ RTC_LOG(LS_INFO) << "VerifyStreamConfiguration";
RTC_DCHECK(stream_);
// TODO(henrika): should we verify device ID as well?
if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
@@ -466,16 +466,16 @@
}
bool AAudioWrapper::OptimizeBuffers() {
- RTC_LOG(INFO) << "OptimizeBuffers";
+ RTC_LOG(LS_INFO) << "OptimizeBuffers";
RTC_DCHECK(stream_);
// Maximum number of frames that can be filled without blocking.
- RTC_LOG(INFO) << "max buffer capacity in frames: "
- << buffer_capacity_in_frames();
+ RTC_LOG(LS_INFO) << "max buffer capacity in frames: "
+ << buffer_capacity_in_frames();
// Query the number of frames that the application should read or write at
// one time for optimal performance.
int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
- RTC_LOG(INFO) << "frames per burst for optimal performance: "
- << frames_per_burst;
+ RTC_LOG(LS_INFO) << "frames per burst for optimal performance: "
+ << frames_per_burst;
frames_per_burst_ = frames_per_burst;
if (direction() == AAUDIO_DIRECTION_INPUT) {
// There is no point in calling setBufferSizeInFrames() for input streams
@@ -492,7 +492,7 @@
return false;
}
// Maximum number of frames that can be filled without blocking.
- RTC_LOG(INFO) << "buffer burst size in frames: " << buffer_size;
+ RTC_LOG(LS_INFO) << "buffer burst size in frames: " << buffer_size;
return true;
}
diff --git a/modules/audio_device/android/audio_device_template.h b/modules/audio_device/android/audio_device_template.h
index 3ea248f..a1510d3 100644
--- a/modules/audio_device/android/audio_device_template.h
+++ b/modules/audio_device/android/audio_device_template.h
@@ -39,22 +39,22 @@
output_(audio_manager_),
input_(audio_manager_),
initialized_(false) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_CHECK(audio_manager);
audio_manager_->SetActiveAudioLayer(audio_layer);
}
- virtual ~AudioDeviceTemplate() { RTC_LOG(INFO) << __FUNCTION__; }
+ virtual ~AudioDeviceTemplate() { RTC_LOG(LS_INFO) << __FUNCTION__; }
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
audioLayer = audio_layer_;
return 0;
}
InitStatus Init() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!initialized_);
if (!audio_manager_->Init()) {
@@ -74,7 +74,7 @@
}
int32_t Terminate() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.IsCurrent());
int32_t err = input_.Terminate();
err |= output_.Terminate();
@@ -85,18 +85,18 @@
}
bool Initialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.IsCurrent());
return initialized_;
}
int16_t PlayoutDevices() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return 1;
}
int16_t RecordingDevices() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return 1;
}
@@ -115,7 +115,7 @@
int32_t SetPlayoutDevice(uint16_t index) override {
// OK to use but it has no effect currently since device selection is
// done using Andoid APIs instead.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return 0;
}
@@ -127,7 +127,7 @@
int32_t SetRecordingDevice(uint16_t index) override {
// OK to use but it has no effect currently since device selection is
// done using Andoid APIs instead.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return 0;
}
@@ -137,39 +137,39 @@
}
int32_t PlayoutIsAvailable(bool& available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
available = true;
return 0;
}
int32_t InitPlayout() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_.InitPlayout();
}
bool PlayoutIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_.PlayoutIsInitialized();
}
int32_t RecordingIsAvailable(bool& available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
available = true;
return 0;
}
int32_t InitRecording() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return input_.InitRecording();
}
bool RecordingIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return input_.RecordingIsInitialized();
}
int32_t StartPlayout() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!audio_manager_->IsCommunicationModeEnabled()) {
RTC_LOG(WARNING)
<< "The application should use MODE_IN_COMMUNICATION audio mode!";
@@ -181,18 +181,18 @@
// Avoid using audio manger (JNI/Java cost) if playout was inactive.
if (!Playing())
return 0;
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
int32_t err = output_.StopPlayout();
return err;
}
bool Playing() const override {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
return output_.Playing();
}
int32_t StartRecording() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!audio_manager_->IsCommunicationModeEnabled()) {
RTC_LOG(WARNING)
<< "The application should use MODE_IN_COMMUNICATION audio mode!";
@@ -202,7 +202,7 @@
int32_t StopRecording() override {
// Avoid using audio manger (JNI/Java cost) if recording was inactive.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!Recording())
return 0;
int32_t err = input_.StopRecording();
@@ -212,47 +212,47 @@
bool Recording() const override { return input_.Recording(); }
int32_t InitSpeaker() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return 0;
}
bool SpeakerIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return true;
}
int32_t InitMicrophone() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return 0;
}
bool MicrophoneIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return true;
}
int32_t SpeakerVolumeIsAvailable(bool& available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_.SpeakerVolumeIsAvailable(available);
}
int32_t SetSpeakerVolume(uint32_t volume) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_.SetSpeakerVolume(volume);
}
int32_t SpeakerVolume(uint32_t& volume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_.SpeakerVolume(volume);
}
int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_.MaxSpeakerVolume(maxVolume);
}
int32_t MinSpeakerVolume(uint32_t& minVolume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_.MinSpeakerVolume(minVolume);
}
@@ -299,13 +299,13 @@
// Returns true if the audio manager has been configured to support stereo
// and false otherwised. Default is mono.
int32_t StereoPlayoutIsAvailable(bool& available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
available = audio_manager_->IsStereoPlayoutSupported();
return 0;
}
int32_t SetStereoPlayout(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
bool available = audio_manager_->IsStereoPlayoutSupported();
// Android does not support changes between mono and stero on the fly.
// Instead, the native audio layer is configured via the audio manager
@@ -320,13 +320,13 @@
}
int32_t StereoRecordingIsAvailable(bool& available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
available = audio_manager_->IsStereoRecordSupported();
return 0;
}
int32_t SetStereoRecording(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
bool available = audio_manager_->IsStereoRecordSupported();
// Android does not support changes between mono and stero on the fly.
// Instead, the native audio layer is configured via the audio manager
@@ -336,7 +336,7 @@
}
int32_t StereoRecording(bool& enabled) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
enabled = audio_manager_->IsStereoRecordSupported();
return 0;
}
@@ -349,7 +349,7 @@
}
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
output_.AttachAudioBuffer(audioBuffer);
input_.AttachAudioBuffer(audioBuffer);
}
@@ -367,13 +367,13 @@
// a "Not Implemented" log will be filed. This non-perfect state will remain
// until I have added full support for audio effects based on OpenSL ES APIs.
bool BuiltInAECIsAvailable() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return audio_manager_->IsAcousticEchoCancelerSupported();
}
// TODO(henrika): add implementation for OpenSL ES based audio as well.
int32_t EnableBuiltInAEC(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK(BuiltInAECIsAvailable()) << "HW AEC is not available";
return input_.EnableBuiltInAEC(enable);
}
@@ -383,13 +383,13 @@
// TODO(henrika): add implementation for OpenSL ES based audio as well.
// In addition, see comments for BuiltInAECIsAvailable().
bool BuiltInAGCIsAvailable() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return audio_manager_->IsAutomaticGainControlSupported();
}
// TODO(henrika): add implementation for OpenSL ES based audio as well.
int32_t EnableBuiltInAGC(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK(BuiltInAGCIsAvailable()) << "HW AGC is not available";
return input_.EnableBuiltInAGC(enable);
}
@@ -399,13 +399,13 @@
// TODO(henrika): add implementation for OpenSL ES based audio as well.
// In addition, see comments for BuiltInAECIsAvailable().
bool BuiltInNSIsAvailable() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return audio_manager_->IsNoiseSuppressorSupported();
}
// TODO(henrika): add implementation for OpenSL ES based audio as well.
int32_t EnableBuiltInNS(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK(BuiltInNSIsAvailable()) << "HW NS is not available";
return input_.EnableBuiltInNS(enable);
}
diff --git a/modules/audio_device/android/audio_manager.cc b/modules/audio_device/android/audio_manager.cc
index 7de2065..e75bd4d 100644
--- a/modules/audio_device/android/audio_manager.cc
+++ b/modules/audio_device/android/audio_manager.cc
@@ -33,11 +33,11 @@
is_device_blacklisted_for_open_sles_usage_(
native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
"()Z")) {
- RTC_LOG(INFO) << "JavaAudioManager::ctor";
+ RTC_LOG(LS_INFO) << "JavaAudioManager::ctor";
}
AudioManager::JavaAudioManager::~JavaAudioManager() {
- RTC_LOG(INFO) << "JavaAudioManager::~dtor";
+ RTC_LOG(LS_INFO) << "JavaAudioManager::~dtor";
}
bool AudioManager::JavaAudioManager::Init() {
@@ -68,7 +68,7 @@
low_latency_playout_(false),
low_latency_record_(false),
delay_estimate_in_milliseconds_(0) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
RTC_CHECK(j_environment_);
JNINativeMethod native_methods[] = {
{"nativeCacheAudioParameters", "(IIIZZZZZZZIIJ)V",
@@ -83,14 +83,14 @@
}
AudioManager::~AudioManager() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
Close();
}
void AudioManager::SetActiveAudioLayer(
AudioDeviceModule::AudioLayer audio_layer) {
- RTC_LOG(INFO) << "SetActiveAudioLayer: " << audio_layer;
+ RTC_LOG(LS_INFO) << "SetActiveAudioLayer: " << audio_layer;
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!initialized_);
// Store the currently utilized audio layer.
@@ -103,18 +103,18 @@
(audio_layer == AudioDeviceModule::kAndroidJavaAudio)
? kHighLatencyModeDelayEstimateInMilliseconds
: kLowLatencyModeDelayEstimateInMilliseconds;
- RTC_LOG(INFO) << "delay_estimate_in_milliseconds: "
- << delay_estimate_in_milliseconds_;
+ RTC_LOG(LS_INFO) << "delay_estimate_in_milliseconds: "
+ << delay_estimate_in_milliseconds_;
}
SLObjectItf AudioManager::GetOpenSLEngine() {
- RTC_LOG(INFO) << "GetOpenSLEngine";
+ RTC_LOG(LS_INFO) << "GetOpenSLEngine";
RTC_DCHECK(thread_checker_.IsCurrent());
// Only allow usage of OpenSL ES if such an audio layer has been specified.
if (audio_layer_ != AudioDeviceModule::kAndroidOpenSLESAudio &&
audio_layer_ !=
AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) {
- RTC_LOG(INFO)
+ RTC_LOG(LS_INFO)
<< "Unable to create OpenSL engine for the current audio layer: "
<< audio_layer_;
return nullptr;
@@ -149,7 +149,7 @@
}
bool AudioManager::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!initialized_);
RTC_DCHECK_NE(audio_layer_, AudioDeviceModule::kPlatformDefaultAudio);
@@ -162,7 +162,7 @@
}
bool AudioManager::Close() {
- RTC_LOG(INFO) << "Close";
+ RTC_LOG(LS_INFO) << "Close";
RTC_DCHECK(thread_checker_.IsCurrent());
if (!initialized_)
return true;
@@ -273,7 +273,7 @@
jboolean a_audio,
jint output_buffer_size,
jint input_buffer_size) {
- RTC_LOG(INFO)
+ RTC_LOG(LS_INFO)
<< "OnCacheAudioParameters: "
"hardware_aec: "
<< static_cast<bool>(hardware_aec)
diff --git a/modules/audio_device/android/audio_record_jni.cc b/modules/audio_device/android/audio_record_jni.cc
index 2c28ab2..9d7bf73 100644
--- a/modules/audio_device/android/audio_record_jni.cc
+++ b/modules/audio_device/android/audio_record_jni.cc
@@ -34,7 +34,7 @@
~ScopedHistogramTimer() {
const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
- RTC_LOG(INFO) << histogram_name_ << ": " << life_time_ms;
+ RTC_LOG(LS_INFO) << histogram_name_ << ": " << life_time_ms;
}
private:
@@ -93,7 +93,7 @@
initialized_(false),
recording_(false),
audio_device_buffer_(nullptr) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
RTC_DCHECK(audio_parameters_.is_valid());
RTC_CHECK(j_environment_);
JNINativeMethod native_methods[] = {
@@ -115,26 +115,26 @@
}
AudioRecordJni::~AudioRecordJni() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
Terminate();
}
int32_t AudioRecordJni::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK(thread_checker_.IsCurrent());
return 0;
}
int32_t AudioRecordJni::Terminate() {
- RTC_LOG(INFO) << "Terminate";
+ RTC_LOG(LS_INFO) << "Terminate";
RTC_DCHECK(thread_checker_.IsCurrent());
StopRecording();
return 0;
}
int32_t AudioRecordJni::InitRecording() {
- RTC_LOG(INFO) << "InitRecording";
+ RTC_LOG(LS_INFO) << "InitRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!initialized_);
RTC_DCHECK(!recording_);
@@ -147,7 +147,7 @@
return -1;
}
frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
- RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_;
+ RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
frames_per_buffer_ * bytes_per_frame);
@@ -157,7 +157,7 @@
}
int32_t AudioRecordJni::StartRecording() {
- RTC_LOG(INFO) << "StartRecording";
+ RTC_LOG(LS_INFO) << "StartRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!recording_);
if (!initialized_) {
@@ -175,7 +175,7 @@
}
int32_t AudioRecordJni::StopRecording() {
- RTC_LOG(INFO) << "StopRecording";
+ RTC_LOG(LS_INFO) << "StopRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
if (!initialized_ || !recording_) {
return 0;
@@ -195,24 +195,24 @@
}
void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(INFO) << "AttachAudioBuffer";
+ RTC_LOG(LS_INFO) << "AttachAudioBuffer";
RTC_DCHECK(thread_checker_.IsCurrent());
audio_device_buffer_ = audioBuffer;
const int sample_rate_hz = audio_parameters_.sample_rate();
- RTC_LOG(INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
+ RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
const size_t channels = audio_parameters_.channels();
- RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")";
+ RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
audio_device_buffer_->SetRecordingChannels(channels);
total_delay_in_milliseconds_ =
audio_manager_->GetDelayEstimateInMilliseconds();
RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
- RTC_LOG(INFO) << "total_delay_in_milliseconds: "
- << total_delay_in_milliseconds_;
+ RTC_LOG(LS_INFO) << "total_delay_in_milliseconds: "
+ << total_delay_in_milliseconds_;
}
int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInAEC(" << enable << ")";
+ RTC_LOG(LS_INFO) << "EnableBuiltInAEC(" << enable << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
}
@@ -223,7 +223,7 @@
}
int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInNS(" << enable << ")";
+ RTC_LOG(LS_INFO) << "EnableBuiltInNS(" << enable << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
}
@@ -240,12 +240,12 @@
void AudioRecordJni::OnCacheDirectBufferAddress(JNIEnv* env,
jobject byte_buffer) {
- RTC_LOG(INFO) << "OnCacheDirectBufferAddress";
+ RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!direct_buffer_address_);
direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
- RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
+ RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
}
@@ -274,7 +274,7 @@
// of `playDelayMs` and `recDelayMs`, hence the distributions does not matter.
audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_, 0);
if (audio_device_buffer_->DeliverRecordedData() == -1) {
- RTC_LOG(INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
+ RTC_LOG(LS_INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
}
}
diff --git a/modules/audio_device/android/audio_track_jni.cc b/modules/audio_device/android/audio_track_jni.cc
index daaeeca..178ccad 100644
--- a/modules/audio_device/android/audio_track_jni.cc
+++ b/modules/audio_device/android/audio_track_jni.cc
@@ -103,7 +103,7 @@
initialized_(false),
playing_(false),
audio_device_buffer_(nullptr) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
RTC_DCHECK(audio_parameters_.is_valid());
RTC_CHECK(j_environment_);
JNINativeMethod native_methods[] = {
@@ -125,26 +125,26 @@
}
AudioTrackJni::~AudioTrackJni() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
Terminate();
}
int32_t AudioTrackJni::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK(thread_checker_.IsCurrent());
return 0;
}
int32_t AudioTrackJni::Terminate() {
- RTC_LOG(INFO) << "Terminate";
+ RTC_LOG(LS_INFO) << "Terminate";
RTC_DCHECK(thread_checker_.IsCurrent());
StopPlayout();
return 0;
}
int32_t AudioTrackJni::InitPlayout() {
- RTC_LOG(INFO) << "InitPlayout";
+ RTC_LOG(LS_INFO) << "InitPlayout";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!initialized_);
RTC_DCHECK(!playing_);
@@ -158,7 +158,7 @@
}
int32_t AudioTrackJni::StartPlayout() {
- RTC_LOG(INFO) << "StartPlayout";
+ RTC_LOG(LS_INFO) << "StartPlayout";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!playing_);
if (!initialized_) {
@@ -175,7 +175,7 @@
}
int32_t AudioTrackJni::StopPlayout() {
- RTC_LOG(INFO) << "StopPlayout";
+ RTC_LOG(LS_INFO) << "StopPlayout";
RTC_DCHECK(thread_checker_.IsCurrent());
if (!initialized_ || !playing_) {
return 0;
@@ -200,7 +200,7 @@
}
int AudioTrackJni::SetSpeakerVolume(uint32_t volume) {
- RTC_LOG(INFO) << "SetSpeakerVolume(" << volume << ")";
+ RTC_LOG(LS_INFO) << "SetSpeakerVolume(" << volume << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
return j_audio_track_->SetStreamVolume(volume) ? 0 : -1;
}
@@ -220,20 +220,20 @@
int AudioTrackJni::SpeakerVolume(uint32_t& volume) const {
RTC_DCHECK(thread_checker_.IsCurrent());
volume = j_audio_track_->GetStreamVolume();
- RTC_LOG(INFO) << "SpeakerVolume: " << volume;
+ RTC_LOG(LS_INFO) << "SpeakerVolume: " << volume;
return 0;
}
// TODO(henrika): possibly add stereo support.
void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(INFO) << "AttachAudioBuffer";
+ RTC_LOG(LS_INFO) << "AttachAudioBuffer";
RTC_DCHECK(thread_checker_.IsCurrent());
audio_device_buffer_ = audioBuffer;
const int sample_rate_hz = audio_parameters_.sample_rate();
- RTC_LOG(INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")";
+ RTC_LOG(LS_INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")";
audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
const size_t channels = audio_parameters_.channels();
- RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")";
+ RTC_LOG(LS_INFO) << "SetPlayoutChannels(" << channels << ")";
audio_device_buffer_->SetPlayoutChannels(channels);
}
@@ -249,16 +249,16 @@
void AudioTrackJni::OnCacheDirectBufferAddress(JNIEnv* env,
jobject byte_buffer) {
- RTC_LOG(INFO) << "OnCacheDirectBufferAddress";
+ RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!direct_buffer_address_);
direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
- RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
+ RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / bytes_per_frame;
- RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_;
+ RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
}
JNI_FUNCTION_ALIGN
diff --git a/modules/audio_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index 572982e..73a8210 100644
--- a/modules/audio_device/audio_device_buffer.cc
+++ b/modules/audio_device/audio_device_buffer.cc
@@ -61,7 +61,7 @@
play_start_time_(0),
only_silence_recorded_(true),
log_stats_(false) {
- RTC_LOG(INFO) << "AudioDeviceBuffer::ctor";
+ 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!";
@@ -72,13 +72,13 @@
RTC_DCHECK_RUN_ON(&main_thread_checker_);
RTC_DCHECK(!playing_);
RTC_DCHECK(!recording_);
- RTC_LOG(INFO) << "AudioDeviceBuffer::~dtor";
+ RTC_LOG(LS_INFO) << "AudioDeviceBuffer::~dtor";
}
int32_t AudioDeviceBuffer::RegisterAudioCallback(
AudioTransport* audio_callback) {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (playing_ || recording_) {
RTC_LOG(LS_ERROR) << "Failed to set audio transport since media was active";
return -1;
@@ -95,7 +95,7 @@
if (playing_) {
return;
}
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Clear members tracking playout stats and do it on the task queue.
task_queue_.PostTask([this] { ResetPlayStats(); });
// Start a periodic timer based on task queue if not already done by the
@@ -114,7 +114,7 @@
if (recording_) {
return;
}
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Clear members tracking recording stats and do it on the task queue.
task_queue_.PostTask([this] { ResetRecStats(); });
// Start a periodic timer based on task queue if not already done by the
@@ -136,13 +136,14 @@
if (!playing_) {
return;
}
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
playing_ = false;
// Stop periodic logging if no more media is active.
if (!recording_) {
StopPeriodicLogging();
}
- RTC_LOG(INFO) << "total playout time: " << rtc::TimeSince(play_start_time_);
+ RTC_LOG(LS_INFO) << "total playout time: "
+ << rtc::TimeSince(play_start_time_);
}
void AudioDeviceBuffer::StopRecording() {
@@ -150,7 +151,7 @@
if (!recording_) {
return;
}
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
recording_ = false;
// Stop periodic logging if no more media is active.
if (!playing_) {
@@ -170,20 +171,20 @@
if (time_since_start > kMinValidCallTimeTimeInMilliseconds) {
const int only_zeros = static_cast<int>(only_silence_recorded_);
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros", only_zeros);
- RTC_LOG(INFO) << "HISTOGRAM(WebRTC.Audio.RecordedOnlyZeros): "
- << only_zeros;
+ RTC_LOG(LS_INFO) << "HISTOGRAM(WebRTC.Audio.RecordedOnlyZeros): "
+ << only_zeros;
}
- RTC_LOG(INFO) << "total recording time: " << time_since_start;
+ RTC_LOG(LS_INFO) << "total recording time: " << time_since_start;
}
int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
- RTC_LOG(INFO) << "SetRecordingSampleRate(" << fsHz << ")";
+ RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << fsHz << ")";
rec_sample_rate_ = fsHz;
return 0;
}
int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
- RTC_LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
+ RTC_LOG(LS_INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
play_sample_rate_ = fsHz;
return 0;
}
@@ -197,13 +198,13 @@
}
int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
- RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")";
+ RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
rec_channels_ = channels;
return 0;
}
int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
- RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")";
+ RTC_LOG(LS_INFO) << "SetPlayoutChannels(" << channels << ")";
play_channels_ = channels;
return 0;
}
@@ -408,21 +409,21 @@
((100.0f * std::abs(rate - rec_sample_rate)) / rec_sample_rate));
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.RecordSampleRateOffsetInPercent",
abs_diff_rate_in_percent);
- RTC_LOG(INFO) << "[REC : " << time_since_last << "msec, "
- << rec_sample_rate / 1000 << "kHz] callbacks: "
- << stats.rec_callbacks - last_stats_.rec_callbacks
- << ", "
- "samples: "
- << diff_samples
- << ", "
- "rate: "
- << static_cast<int>(rate + 0.5)
- << ", "
- "rate diff: "
- << abs_diff_rate_in_percent
- << "%, "
- "level: "
- << stats.max_rec_level;
+ RTC_LOG(LS_INFO) << "[REC : " << time_since_last << "msec, "
+ << rec_sample_rate / 1000 << "kHz] callbacks: "
+ << stats.rec_callbacks - last_stats_.rec_callbacks
+ << ", "
+ "samples: "
+ << diff_samples
+ << ", "
+ "rate: "
+ << static_cast<int>(rate + 0.5)
+ << ", "
+ "rate diff: "
+ << abs_diff_rate_in_percent
+ << "%, "
+ "level: "
+ << stats.max_rec_level;
}
diff_samples = stats.play_samples - last_stats_.play_samples;
@@ -434,21 +435,21 @@
((100.0f * std::abs(rate - play_sample_rate)) / play_sample_rate));
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.PlayoutSampleRateOffsetInPercent",
abs_diff_rate_in_percent);
- RTC_LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
- << play_sample_rate / 1000 << "kHz] callbacks: "
- << stats.play_callbacks - last_stats_.play_callbacks
- << ", "
- "samples: "
- << diff_samples
- << ", "
- "rate: "
- << static_cast<int>(rate + 0.5)
- << ", "
- "rate diff: "
- << abs_diff_rate_in_percent
- << "%, "
- "level: "
- << stats.max_play_level;
+ RTC_LOG(LS_INFO) << "[PLAY: " << time_since_last << "msec, "
+ << play_sample_rate / 1000 << "kHz] callbacks: "
+ << stats.play_callbacks - last_stats_.play_callbacks
+ << ", "
+ "samples: "
+ << diff_samples
+ << ", "
+ "rate: "
+ << static_cast<int>(rate + 0.5)
+ << ", "
+ "rate diff: "
+ << abs_diff_rate_in_percent
+ << "%, "
+ "level: "
+ << stats.max_play_level;
}
}
last_stats_ = stats;
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
index 84460ff..9d540f9 100644
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -73,7 +73,7 @@
rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
AudioLayer audio_layer,
TaskQueueFactory* task_queue_factory) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return AudioDeviceModule::CreateForTest(audio_layer, task_queue_factory);
}
@@ -81,7 +81,7 @@
rtc::scoped_refptr<AudioDeviceModuleForTest> AudioDeviceModule::CreateForTest(
AudioLayer audio_layer,
TaskQueueFactory* task_queue_factory) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own
// dedicated factory method which should be used instead.
@@ -118,28 +118,28 @@
AudioLayer audio_layer,
TaskQueueFactory* task_queue_factory)
: audio_layer_(audio_layer), audio_device_buffer_(task_queue_factory) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
}
int32_t AudioDeviceModuleImpl::CheckPlatform() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Ensure that the current platform is supported
PlatformType platform(kPlatformNotSupported);
#if defined(_WIN32)
platform = kPlatformWin32;
- RTC_LOG(INFO) << "current platform is Win32";
+ RTC_LOG(LS_INFO) << "current platform is Win32";
#elif defined(WEBRTC_ANDROID)
platform = kPlatformAndroid;
- RTC_LOG(INFO) << "current platform is Android";
+ RTC_LOG(LS_INFO) << "current platform is Android";
#elif defined(WEBRTC_LINUX)
platform = kPlatformLinux;
- RTC_LOG(INFO) << "current platform is Linux";
+ RTC_LOG(LS_INFO) << "current platform is Linux";
#elif defined(WEBRTC_IOS)
platform = kPlatformIOS;
- RTC_LOG(INFO) << "current platform is IOS";
+ RTC_LOG(LS_INFO) << "current platform is IOS";
#elif defined(WEBRTC_MAC)
platform = kPlatformMac;
- RTC_LOG(INFO) << "current platform is Mac";
+ RTC_LOG(LS_INFO) << "current platform is Mac";
#endif
if (platform == kPlatformNotSupported) {
RTC_LOG(LERROR)
@@ -152,19 +152,19 @@
}
int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
// Dummy ADM implementations if build flags are set.
#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
audio_device_.reset(new AudioDeviceDummy());
- RTC_LOG(INFO) << "Dummy Audio APIs will be utilized";
+ RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized";
#elif defined(WEBRTC_DUMMY_FILE_DEVICES)
audio_device_.reset(FileAudioDeviceFactory::CreateFileAudioDevice());
if (audio_device_) {
- RTC_LOG(INFO) << "Will use file-playing dummy device.";
+ RTC_LOG(LS_INFO) << "Will use file-playing dummy device.";
} else {
// Create a dummy device instead.
audio_device_.reset(new AudioDeviceDummy());
- RTC_LOG(INFO) << "Dummy Audio APIs will be utilized";
+ RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized";
}
// Real (non-dummy) ADM implementations.
@@ -174,10 +174,10 @@
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
if ((audio_layer == kWindowsCoreAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
- RTC_LOG(INFO) << "Attempting to use the Windows Core Audio APIs...";
+ RTC_LOG(LS_INFO) << "Attempting to use the Windows Core Audio APIs...";
if (AudioDeviceWindowsCore::CoreAudioIsSupported()) {
audio_device_.reset(new AudioDeviceWindowsCore());
- RTC_LOG(INFO) << "Windows Core Audio APIs will be utilized";
+ RTC_LOG(LS_INFO) << "Windows Core Audio APIs will be utilized";
}
}
#endif // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
@@ -255,7 +255,7 @@
if ((audio_layer == kLinuxAlsaAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
audio_device_.reset(new AudioDeviceLinuxALSA());
- RTC_LOG(INFO) << "Linux ALSA APIs will be utilized.";
+ RTC_LOG(LS_INFO) << "Linux ALSA APIs will be utilized.";
}
#else
// Build flag 'rtc_include_pulse_audio' is set to true (default). In this
@@ -263,12 +263,12 @@
// - kPlatformDefaultAudio => PulseAudio, and
// - kLinuxPulseAudio => PulseAudio, and
// - kLinuxAlsaAudio => ALSA (supported but not default).
- RTC_LOG(INFO) << "PulseAudio support is enabled.";
+ RTC_LOG(LS_INFO) << "PulseAudio support is enabled.";
if ((audio_layer == kLinuxPulseAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
// Linux PulseAudio implementation is default.
audio_device_.reset(new AudioDeviceLinuxPulse());
- RTC_LOG(INFO) << "Linux PulseAudio APIs will be utilized";
+ 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.";
@@ -281,7 +281,7 @@
if (audio_layer == kPlatformDefaultAudio) {
audio_device_.reset(
new ios_adm::AudioDeviceIOS(/*bypass_voice_processing=*/false));
- RTC_LOG(INFO) << "iPhone Audio APIs will be utilized.";
+ RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
}
// END #if defined(WEBRTC_IOS)
@@ -289,14 +289,14 @@
#elif defined(WEBRTC_MAC)
if (audio_layer == kPlatformDefaultAudio) {
audio_device_.reset(new AudioDeviceMac());
- RTC_LOG(INFO) << "Mac OS X Audio APIs will be utilized.";
+ RTC_LOG(LS_INFO) << "Mac OS X Audio APIs will be utilized.";
}
#endif // WEBRTC_MAC
// Dummy ADM implementation.
if (audio_layer == kDummyAudio) {
audio_device_.reset(new AudioDeviceDummy());
- RTC_LOG(INFO) << "Dummy Audio APIs will be utilized.";
+ RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized.";
}
#endif // if defined(WEBRTC_DUMMY_AUDIO_BUILD)
@@ -309,17 +309,17 @@
}
int32_t AudioDeviceModuleImpl::AttachAudioBuffer() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
audio_device_->AttachAudioBuffer(&audio_device_buffer_);
return 0;
}
AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
}
int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
AudioLayer activeAudio;
if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
return -1;
@@ -329,7 +329,7 @@
}
int32_t AudioDeviceModuleImpl::Init() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
if (initialized_)
return 0;
RTC_CHECK(audio_device_);
@@ -346,7 +346,7 @@
}
int32_t AudioDeviceModuleImpl::Terminate() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return 0;
if (audio_device_->Terminate() == -1) {
@@ -357,65 +357,65 @@
}
bool AudioDeviceModuleImpl::Initialized() const {
- RTC_LOG(INFO) << __FUNCTION__ << ": " << initialized_;
+ RTC_LOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
return initialized_;
}
int32_t AudioDeviceModuleImpl::InitSpeaker() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->InitSpeaker();
}
int32_t AudioDeviceModuleImpl::InitMicrophone() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->InitMicrophone();
}
int32_t AudioDeviceModuleImpl::SpeakerVolumeIsAvailable(bool* available) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetSpeakerVolume(uint32_t volume) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
CHECKinitialized_();
return audio_device_->SetSpeakerVolume(volume);
}
int32_t AudioDeviceModuleImpl::SpeakerVolume(uint32_t* volume) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
uint32_t level = 0;
if (audio_device_->SpeakerVolume(level) == -1) {
return -1;
}
*volume = level;
- RTC_LOG(INFO) << "output: " << *volume;
+ RTC_LOG(LS_INFO) << "output: " << *volume;
return 0;
}
bool AudioDeviceModuleImpl::SpeakerIsInitialized() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isInitialized = audio_device_->SpeakerIsInitialized();
- RTC_LOG(INFO) << "output: " << isInitialized;
+ RTC_LOG(LS_INFO) << "output: " << isInitialized;
return isInitialized;
}
bool AudioDeviceModuleImpl::MicrophoneIsInitialized() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isInitialized = audio_device_->MicrophoneIsInitialized();
- RTC_LOG(INFO) << "output: " << isInitialized;
+ RTC_LOG(LS_INFO) << "output: " << isInitialized;
return isInitialized;
}
@@ -440,110 +440,110 @@
}
int32_t AudioDeviceModuleImpl::SpeakerMuteIsAvailable(bool* available) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetSpeakerMute(bool enable) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return audio_device_->SetSpeakerMute(enable);
}
int32_t AudioDeviceModuleImpl::SpeakerMute(bool* enabled) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool muted = false;
if (audio_device_->SpeakerMute(muted) == -1) {
return -1;
}
*enabled = muted;
- RTC_LOG(INFO) << "output: " << muted;
+ RTC_LOG(LS_INFO) << "output: " << muted;
return 0;
}
int32_t AudioDeviceModuleImpl::MicrophoneMuteIsAvailable(bool* available) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetMicrophoneMute(bool enable) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return (audio_device_->SetMicrophoneMute(enable));
}
int32_t AudioDeviceModuleImpl::MicrophoneMute(bool* enabled) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool muted = false;
if (audio_device_->MicrophoneMute(muted) == -1) {
return -1;
}
*enabled = muted;
- RTC_LOG(INFO) << "output: " << muted;
+ RTC_LOG(LS_INFO) << "output: " << muted;
return 0;
}
int32_t AudioDeviceModuleImpl::MicrophoneVolumeIsAvailable(bool* available) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetMicrophoneVolume(uint32_t volume) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
CHECKinitialized_();
return (audio_device_->SetMicrophoneVolume(volume));
}
int32_t AudioDeviceModuleImpl::MicrophoneVolume(uint32_t* volume) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
uint32_t level = 0;
if (audio_device_->MicrophoneVolume(level) == -1) {
return -1;
}
*volume = level;
- RTC_LOG(INFO) << "output: " << *volume;
+ RTC_LOG(LS_INFO) << "output: " << *volume;
return 0;
}
int32_t AudioDeviceModuleImpl::StereoRecordingIsAvailable(
bool* available) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetStereoRecording(bool enable) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->RecordingIsInitialized()) {
RTC_LOG(LERROR)
@@ -565,31 +565,31 @@
}
int32_t AudioDeviceModuleImpl::StereoRecording(bool* enabled) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool stereo = false;
if (audio_device_->StereoRecording(stereo) == -1) {
return -1;
}
*enabled = stereo;
- RTC_LOG(INFO) << "output: " << stereo;
+ RTC_LOG(LS_INFO) << "output: " << stereo;
return 0;
}
int32_t AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetStereoPlayout(bool enable) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->PlayoutIsInitialized()) {
RTC_LOG(LERROR)
@@ -609,38 +609,38 @@
}
int32_t AudioDeviceModuleImpl::StereoPlayout(bool* enabled) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool stereo = false;
if (audio_device_->StereoPlayout(stereo) == -1) {
return -1;
}
*enabled = stereo;
- RTC_LOG(INFO) << "output: " << stereo;
+ RTC_LOG(LS_INFO) << "output: " << stereo;
return 0;
}
int32_t AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::RecordingIsAvailable(bool* available) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
@@ -665,21 +665,21 @@
}
int16_t AudioDeviceModuleImpl::PlayoutDevices() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
- RTC_LOG(INFO) << "output: " << nPlayoutDevices;
+ RTC_LOG(LS_INFO) << "output: " << nPlayoutDevices;
return (int16_t)(nPlayoutDevices);
}
int32_t AudioDeviceModuleImpl::SetPlayoutDevice(uint16_t index) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
CHECKinitialized_();
return audio_device_->SetPlayoutDevice(index);
}
int32_t AudioDeviceModuleImpl::SetPlayoutDevice(WindowsDeviceType device) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->SetPlayoutDevice(device);
}
@@ -688,7 +688,7 @@
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
CHECKinitialized_();
if (name == NULL) {
return -1;
@@ -697,10 +697,10 @@
return -1;
}
if (name != NULL) {
- RTC_LOG(INFO) << "output: name = " << name;
+ RTC_LOG(LS_INFO) << "output: name = " << name;
}
if (guid != NULL) {
- RTC_LOG(INFO) << "output: guid = " << guid;
+ RTC_LOG(LS_INFO) << "output: guid = " << guid;
}
return 0;
}
@@ -709,7 +709,7 @@
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
CHECKinitialized_();
if (name == NULL) {
return -1;
@@ -718,137 +718,137 @@
return -1;
}
if (name != NULL) {
- RTC_LOG(INFO) << "output: name = " << name;
+ RTC_LOG(LS_INFO) << "output: name = " << name;
}
if (guid != NULL) {
- RTC_LOG(INFO) << "output: guid = " << guid;
+ RTC_LOG(LS_INFO) << "output: guid = " << guid;
}
return 0;
}
int16_t AudioDeviceModuleImpl::RecordingDevices() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
uint16_t nRecordingDevices = audio_device_->RecordingDevices();
- RTC_LOG(INFO) << "output: " << nRecordingDevices;
+ RTC_LOG(LS_INFO) << "output: " << nRecordingDevices;
return (int16_t)nRecordingDevices;
}
int32_t AudioDeviceModuleImpl::SetRecordingDevice(uint16_t index) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
CHECKinitialized_();
return audio_device_->SetRecordingDevice(index);
}
int32_t AudioDeviceModuleImpl::SetRecordingDevice(WindowsDeviceType device) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->SetRecordingDevice(device);
}
int32_t AudioDeviceModuleImpl::InitPlayout() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
if (PlayoutIsInitialized()) {
return 0;
}
int32_t result = audio_device_->InitPlayout();
- RTC_LOG(INFO) << "output: " << result;
+ RTC_LOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleImpl::InitRecording() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
if (RecordingIsInitialized()) {
return 0;
}
int32_t result = audio_device_->InitRecording();
- RTC_LOG(INFO) << "output: " << result;
+ RTC_LOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleImpl::PlayoutIsInitialized() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->PlayoutIsInitialized();
}
bool AudioDeviceModuleImpl::RecordingIsInitialized() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->RecordingIsInitialized();
}
int32_t AudioDeviceModuleImpl::StartPlayout() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
if (Playing()) {
return 0;
}
audio_device_buffer_.StartPlayout();
int32_t result = audio_device_->StartPlayout();
- RTC_LOG(INFO) << "output: " << result;
+ RTC_LOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleImpl::StopPlayout() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopPlayout();
audio_device_buffer_.StopPlayout();
- RTC_LOG(INFO) << "output: " << result;
+ RTC_LOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleImpl::Playing() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->Playing();
}
int32_t AudioDeviceModuleImpl::StartRecording() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
if (Recording()) {
return 0;
}
audio_device_buffer_.StartRecording();
int32_t result = audio_device_->StartRecording();
- RTC_LOG(INFO) << "output: " << result;
+ RTC_LOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleImpl::StopRecording() {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopRecording();
audio_device_buffer_.StopRecording();
- RTC_LOG(INFO) << "output: " << result;
+ RTC_LOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleImpl::Recording() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->Recording();
}
int32_t AudioDeviceModuleImpl::RegisterAudioCallback(
AudioTransport* audioCallback) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
return audio_device_buffer_.RegisterAudioCallback(audioCallback);
}
@@ -864,87 +864,87 @@
}
bool AudioDeviceModuleImpl::BuiltInAECIsAvailable() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInAECIsAvailable();
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInAEC(enable);
- RTC_LOG(INFO) << "output: " << ok;
+ RTC_LOG(LS_INFO) << "output: " << ok;
return ok;
}
bool AudioDeviceModuleImpl::BuiltInAGCIsAvailable() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleImpl::EnableBuiltInAGC(bool enable) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInAGC(enable);
- RTC_LOG(INFO) << "output: " << ok;
+ RTC_LOG(LS_INFO) << "output: " << ok;
return ok;
}
bool AudioDeviceModuleImpl::BuiltInNSIsAvailable() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInNSIsAvailable();
- RTC_LOG(INFO) << "output: " << isAvailable;
+ RTC_LOG(LS_INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleImpl::EnableBuiltInNS(bool enable) {
- RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInNS(enable);
- RTC_LOG(INFO) << "output: " << ok;
+ RTC_LOG(LS_INFO) << "output: " << ok;
return ok;
}
int32_t AudioDeviceModuleImpl::GetPlayoutUnderrunCount() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t underrunCount = audio_device_->GetPlayoutUnderrunCount();
- RTC_LOG(INFO) << "output: " << underrunCount;
+ RTC_LOG(LS_INFO) << "output: " << underrunCount;
return underrunCount;
}
#if defined(WEBRTC_IOS)
int AudioDeviceModuleImpl::GetPlayoutAudioParameters(
AudioParameters* params) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
int r = audio_device_->GetPlayoutAudioParameters(params);
- RTC_LOG(INFO) << "output: " << r;
+ RTC_LOG(LS_INFO) << "output: " << r;
return r;
}
int AudioDeviceModuleImpl::GetRecordAudioParameters(
AudioParameters* params) const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
int r = audio_device_->GetRecordAudioParameters(params);
- RTC_LOG(INFO) << "output: " << r;
+ RTC_LOG(LS_INFO) << "output: " << r;
return r;
}
#endif // WEBRTC_IOS
AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
return platform_type_;
}
AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer()
const {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
return audio_layer_;
}
diff --git a/modules/audio_device/fine_audio_buffer.cc b/modules/audio_device/fine_audio_buffer.cc
index 4f3f48c..86240da 100644
--- a/modules/audio_device/fine_audio_buffer.cc
+++ b/modules/audio_device/fine_audio_buffer.cc
@@ -29,21 +29,21 @@
playout_channels_(audio_device_buffer->PlayoutChannels()),
record_channels_(audio_device_buffer->RecordingChannels()) {
RTC_DCHECK(audio_device_buffer_);
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (IsReadyForPlayout()) {
- RTC_DLOG(INFO) << "playout_samples_per_channel_10ms: "
- << playout_samples_per_channel_10ms_;
- RTC_DLOG(INFO) << "playout_channels: " << playout_channels_;
+ RTC_DLOG(LS_INFO) << "playout_samples_per_channel_10ms: "
+ << playout_samples_per_channel_10ms_;
+ RTC_DLOG(LS_INFO) << "playout_channels: " << playout_channels_;
}
if (IsReadyForRecord()) {
- RTC_DLOG(INFO) << "record_samples_per_channel_10ms: "
- << record_samples_per_channel_10ms_;
- RTC_DLOG(INFO) << "record_channels: " << record_channels_;
+ RTC_DLOG(LS_INFO) << "record_samples_per_channel_10ms: "
+ << record_samples_per_channel_10ms_;
+ RTC_DLOG(LS_INFO) << "record_channels: " << record_channels_;
}
}
FineAudioBuffer::~FineAudioBuffer() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
}
void FineAudioBuffer::ResetPlayout() {
diff --git a/modules/audio_device/include/audio_device_factory.cc b/modules/audio_device/include/audio_device_factory.cc
index d5b3810..130e096 100644
--- a/modules/audio_device/include/audio_device_factory.cc
+++ b/modules/audio_device/include/audio_device_factory.cc
@@ -27,7 +27,7 @@
rtc::scoped_refptr<AudioDeviceModule> CreateWindowsCoreAudioAudioDeviceModule(
TaskQueueFactory* task_queue_factory,
bool automatic_restart) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return CreateWindowsCoreAudioAudioDeviceModuleForTest(task_queue_factory,
automatic_restart);
}
@@ -36,7 +36,7 @@
CreateWindowsCoreAudioAudioDeviceModuleForTest(
TaskQueueFactory* task_queue_factory,
bool automatic_restart) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Returns NULL if Core Audio is not supported or if COM has not been
// initialized correctly using ScopedCOMInitializer.
if (!webrtc_win::core_audio_utility::IsSupported()) {
diff --git a/modules/audio_device/win/audio_device_core_win.cc b/modules/audio_device/win/audio_device_core_win.cc
index 6c08fcc..f1b6b93 100644
--- a/modules/audio_device/win/audio_device_core_win.cc
+++ b/modules/audio_device/win/audio_device_core_win.cc
@@ -1887,18 +1887,18 @@
break;
} else {
if (pWfxClosestMatch) {
- RTC_LOG(INFO) << "nChannels=" << Wfx.nChannels
- << ", nSamplesPerSec=" << Wfx.nSamplesPerSec
- << " is not supported. Closest match: "
- "nChannels="
- << pWfxClosestMatch->nChannels << ", nSamplesPerSec="
- << pWfxClosestMatch->nSamplesPerSec;
+ RTC_LOG(LS_INFO) << "nChannels=" << Wfx.nChannels
+ << ", nSamplesPerSec=" << Wfx.nSamplesPerSec
+ << " is not supported. Closest match: "
+ "nChannels="
+ << pWfxClosestMatch->nChannels << ", nSamplesPerSec="
+ << pWfxClosestMatch->nSamplesPerSec;
CoTaskMemFree(pWfxClosestMatch);
pWfxClosestMatch = NULL;
} else {
- RTC_LOG(INFO) << "nChannels=" << Wfx.nChannels
- << ", nSamplesPerSec=" << Wfx.nSamplesPerSec
- << " is not supported. No closest match.";
+ RTC_LOG(LS_INFO) << "nChannels=" << Wfx.nChannels
+ << ", nSamplesPerSec=" << Wfx.nSamplesPerSec
+ << " is not supported. No closest match.";
}
}
}
@@ -2208,18 +2208,18 @@
break;
} else {
if (pWfxClosestMatch) {
- RTC_LOG(INFO) << "nChannels=" << Wfx.Format.nChannels
- << ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
- << " is not supported. Closest match: "
- "nChannels="
- << pWfxClosestMatch->nChannels << ", nSamplesPerSec="
- << pWfxClosestMatch->nSamplesPerSec;
+ RTC_LOG(LS_INFO) << "nChannels=" << Wfx.Format.nChannels
+ << ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
+ << " is not supported. Closest match: "
+ "nChannels="
+ << pWfxClosestMatch->nChannels << ", nSamplesPerSec="
+ << pWfxClosestMatch->nSamplesPerSec;
CoTaskMemFree(pWfxClosestMatch);
pWfxClosestMatch = NULL;
} else {
- RTC_LOG(INFO) << "nChannels=" << Wfx.Format.nChannels
- << ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
- << " is not supported. No closest match.";
+ RTC_LOG(LS_INFO) << "nChannels=" << Wfx.Format.nChannels
+ << ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
+ << " is not supported. No closest match.";
}
}
}
diff --git a/modules/audio_device/win/audio_device_module_win.cc b/modules/audio_device/win/audio_device_module_win.cc
index ad26953..6643d84 100644
--- a/modules/audio_device/win/audio_device_module_win.cc
+++ b/modules/audio_device/win/audio_device_module_win.cc
@@ -95,12 +95,12 @@
task_queue_factory_(task_queue_factory) {
RTC_CHECK(input_);
RTC_CHECK(output_);
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
}
~WindowsAudioDeviceModule() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
Terminate();
}
@@ -110,7 +110,7 @@
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer* audioLayer) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
// TODO(henrika): it might be possible to remove this unique signature.
*audioLayer = AudioDeviceModule::kWindowsCoreAudio2;
@@ -118,14 +118,14 @@
}
int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(audio_device_buffer_);
RTC_DCHECK_RUN_ON(&thread_checker_);
return audio_device_buffer_->RegisterAudioCallback(audioCallback);
}
int32_t Init() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
RETURN_IF_INPUT_RESTARTS(0);
@@ -153,7 +153,7 @@
}
int32_t Terminate() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
RETURN_IF_INPUT_RESTARTS(0);
@@ -172,14 +172,14 @@
}
int16_t PlayoutDevices() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
return output_->NumDevices();
}
int16_t RecordingDevices() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_INPUT_RESTARTS(0);
return input_->NumDevices();
@@ -188,7 +188,7 @@
int32_t PlayoutDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
std::string name_str, guid_str;
@@ -205,7 +205,7 @@
int32_t RecordingDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_INPUT_RESTARTS(0);
std::string name_str, guid_str;
@@ -221,7 +221,7 @@
}
int32_t SetPlayoutDevice(uint16_t index) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
return output_->SetDevice(index);
@@ -229,33 +229,33 @@
int32_t SetPlayoutDevice(
AudioDeviceModule::WindowsDeviceType device) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
return output_->SetDevice(device);
}
int32_t SetRecordingDevice(uint16_t index) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return input_->SetDevice(index);
}
int32_t SetRecordingDevice(
AudioDeviceModule::WindowsDeviceType device) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return input_->SetDevice(device);
}
int32_t PlayoutIsAvailable(bool* available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
*available = true;
return 0;
}
int32_t InitPlayout() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
RETURN_IF_OUTPUT_IS_INITIALIZED(0);
@@ -263,21 +263,21 @@
}
bool PlayoutIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(true);
return output_->PlayoutIsInitialized();
}
int32_t RecordingIsAvailable(bool* available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
*available = true;
return 0;
}
int32_t InitRecording() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_INPUT_RESTARTS(0);
RETURN_IF_INPUT_IS_INITIALIZED(0);
@@ -285,14 +285,14 @@
}
bool RecordingIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_INPUT_RESTARTS(true);
return input_->RecordingIsInitialized();
}
int32_t StartPlayout() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
RETURN_IF_OUTPUT_IS_ACTIVE(0);
@@ -300,21 +300,21 @@
}
int32_t StopPlayout() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(-1);
return output_->StopPlayout();
}
bool Playing() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(true);
return output_->Playing();
}
int32_t StartRecording() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_INPUT_RESTARTS(0);
RETURN_IF_INPUT_IS_ACTIVE(0);
@@ -322,41 +322,41 @@
}
int32_t StopRecording() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_INPUT_RESTARTS(-1);
return input_->StopRecording();
}
bool Recording() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RETURN_IF_INPUT_RESTARTS(true);
return input_->Recording();
}
int32_t InitSpeaker() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DLOG(LS_WARNING) << "This method has no effect";
return initialized_ ? 0 : -1;
}
bool SpeakerIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DLOG(LS_WARNING) << "This method has no effect";
return initialized_;
}
int32_t InitMicrophone() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DLOG(LS_WARNING) << "This method has no effect";
return initialized_ ? 0 : -1;
}
bool MicrophoneIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DLOG(LS_WARNING) << "This method has no effect";
return initialized_;
@@ -364,7 +364,7 @@
int32_t SpeakerVolumeIsAvailable(bool* available) override {
// TODO(henrika): improve support.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
*available = false;
return 0;
@@ -377,7 +377,7 @@
int32_t MicrophoneVolumeIsAvailable(bool* available) override {
// TODO(henrika): improve support.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
*available = false;
return 0;
@@ -398,7 +398,7 @@
int32_t StereoPlayoutIsAvailable(bool* available) const override {
// TODO(henrika): improve support.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
*available = true;
return 0;
@@ -406,14 +406,14 @@
int32_t SetStereoPlayout(bool enable) override {
// TODO(henrika): improve support.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return 0;
}
int32_t StereoPlayout(bool* enabled) const override {
// TODO(henrika): improve support.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
*enabled = true;
return 0;
@@ -421,7 +421,7 @@
int32_t StereoRecordingIsAvailable(bool* available) const override {
// TODO(henrika): improve support.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
*available = true;
return 0;
@@ -429,14 +429,14 @@
int32_t SetStereoRecording(bool enable) override {
// TODO(henrika): improve support.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return 0;
}
int32_t StereoRecording(bool* enabled) const override {
// TODO(henrika): improve support.
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
*enabled = true;
return 0;
@@ -453,33 +453,33 @@
int32_t EnableBuiltInNS(bool enable) override { return 0; }
int32_t AttachAudioBuffer() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
output_->AttachAudioBuffer(audio_device_buffer_.get());
input_->AttachAudioBuffer(audio_device_buffer_.get());
return 0;
}
int RestartPlayoutInternally() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
RETURN_IF_OUTPUT_RESTARTS(0);
return output_->RestartPlayout();
}
int RestartRecordingInternally() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return input_->RestartRecording();
}
int SetPlayoutSampleRate(uint32_t sample_rate) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return output_->SetSampleRate(sample_rate);
}
int SetRecordingSampleRate(uint32_t sample_rate) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return input_->SetSampleRate(sample_rate);
}
@@ -513,7 +513,7 @@
std::unique_ptr<AudioInput> audio_input,
std::unique_ptr<AudioOutput> audio_output,
TaskQueueFactory* task_queue_factory) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return rtc::make_ref_counted<WindowsAudioDeviceModule>(
std::move(audio_input), std::move(audio_output), task_queue_factory);
}
diff --git a/modules/audio_device/win/core_audio_base_win.cc b/modules/audio_device/win/core_audio_base_win.cc
index c42c091..40645d5 100644
--- a/modules/audio_device/win/core_audio_base_win.cc
+++ b/modules/audio_device/win/core_audio_base_win.cc
@@ -125,7 +125,7 @@
bool IsLowLatencySupported(IAudioClient3* client3,
const WAVEFORMATEXTENSIBLE* format,
uint32_t* min_period_in_frames) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get the range of periodicities supported by the engine for the specified
// stream format.
@@ -143,7 +143,7 @@
// default engine period.
// TODO(henrika): verify that this assumption is correct.
const bool low_latency = min_period < default_period;
- RTC_LOG(INFO) << "low_latency: " << low_latency;
+ RTC_LOG(LS_INFO) << "low_latency: " << low_latency;
*min_period_in_frames = low_latency ? min_period : 0;
return low_latency;
}
@@ -161,9 +161,10 @@
on_error_callback_(error_callback),
device_index_(kUndefined),
is_restarting_(false) {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction) << "]";
- RTC_DLOG(INFO) << "Automatic restart: " << automatic_restart;
- RTC_DLOG(INFO) << "Windows version: " << rtc::rtc_win::GetVersion();
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction)
+ << "]";
+ RTC_DLOG(LS_INFO) << "Automatic restart: " << automatic_restart;
+ RTC_DLOG(LS_INFO) << "Windows version: " << rtc::rtc_win::GetVersion();
// Create the event which the audio engine will signal each time a buffer
// becomes ready to be processed by the client.
@@ -181,7 +182,7 @@
}
CoreAudioBase::~CoreAudioBase() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_EQ(ref_count_, 1);
}
@@ -207,7 +208,7 @@
}
void CoreAudioBase::ReleaseCOMObjects() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// ComPtr::Reset() sets the ComPtr to nullptr releasing any previous
// reference.
if (audio_client_) {
@@ -288,15 +289,15 @@
}
int CoreAudioBase::SetDevice(int index) {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
- << "]: index=" << IndexToString(index);
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+ << "]: index=" << IndexToString(index);
if (initialized_) {
return -1;
}
std::string device_id = GetDeviceID(index);
- RTC_DLOG(INFO) << "index=" << IndexToString(index)
- << " => device_id: " << device_id;
+ RTC_DLOG(LS_INFO) << "index=" << IndexToString(index)
+ << " => device_id: " << device_id;
device_index_ = index;
device_id_ = device_id;
@@ -306,8 +307,8 @@
int CoreAudioBase::DeviceName(int index,
std::string* name,
std::string* guid) const {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
- << "]: index=" << IndexToString(index);
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+ << "]: index=" << IndexToString(index);
if (index > NumberOfEnumeratedDevices() - 1) {
RTC_LOG(LS_ERROR) << "Invalid device index";
return -1;
@@ -324,17 +325,17 @@
}
*name = device_names[index].device_name;
- RTC_DLOG(INFO) << "name: " << *name;
+ RTC_DLOG(LS_INFO) << "name: " << *name;
if (guid != nullptr) {
*guid = device_names[index].unique_id;
- RTC_DLOG(INFO) << "guid: " << *guid;
+ RTC_DLOG(LS_INFO) << "guid: " << *guid;
}
return 0;
}
bool CoreAudioBase::Init() {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
- << "]";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+ << "]";
RTC_DCHECK_GE(device_index_, 0);
RTC_DCHECK(!device_id_.empty());
RTC_DCHECK(audio_device_buffer_);
@@ -360,15 +361,15 @@
// an audio stream between an audio application and the audio engine.
ComPtr<IAudioClient> audio_client;
if (core_audio_utility::GetAudioClientVersion() == 3) {
- RTC_DLOG(INFO) << "Using IAudioClient3";
+ RTC_DLOG(LS_INFO) << "Using IAudioClient3";
audio_client =
core_audio_utility::CreateClient3(device_id, GetDataFlow(), role);
} else if (core_audio_utility::GetAudioClientVersion() == 2) {
- RTC_DLOG(INFO) << "Using IAudioClient2";
+ RTC_DLOG(LS_INFO) << "Using IAudioClient2";
audio_client =
core_audio_utility::CreateClient2(device_id, GetDataFlow(), role);
} else {
- RTC_DLOG(INFO) << "Using IAudioClient";
+ RTC_DLOG(LS_INFO) << "Using IAudioClient";
audio_client =
core_audio_utility::CreateClient(device_id, GetDataFlow(), role);
}
@@ -429,7 +430,7 @@
format_.dwChannelMask =
format->nChannels == 1 ? KSAUDIO_SPEAKER_MONO : KSAUDIO_SPEAKER_STEREO;
format_.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
- RTC_DLOG(INFO) << core_audio_utility::WaveFormatToString(&format_);
+ RTC_DLOG(LS_INFO) << core_audio_utility::WaveFormatToString(&format_);
// Verify that the format is supported but exclude the test if the default
// sample rate has been overridden. If so, the WASAPI audio engine will do
@@ -502,8 +503,8 @@
1000.0L;
const int preferred_frames_per_buffer =
static_cast<int>(params.sample_rate() * device_period_in_seconds + 0.5);
- RTC_DLOG(INFO) << "preferred_frames_per_buffer: "
- << preferred_frames_per_buffer;
+ 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 "
@@ -525,7 +526,7 @@
if (FAILED(audio_session_control->GetState(&state))) {
return false;
}
- RTC_DLOG(INFO) << "audio session state: " << SessionStateToString(state);
+ RTC_DLOG(LS_INFO) << "audio session state: " << SessionStateToString(state);
RTC_DCHECK_EQ(state, AudioSessionStateInactive);
// Register the client to receive notifications of session events, including
@@ -542,8 +543,8 @@
}
bool CoreAudioBase::Start() {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
- << "]";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+ << "]";
if (IsRestarting()) {
// Audio thread should be alive during internal restart since the restart
// callback is triggered on that thread and it also makes the restart
@@ -559,8 +560,8 @@
audio_thread_ = rtc::PlatformThread::SpawnJoinable(
[this] { ThreadRun(); }, name,
rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
- RTC_DLOG(INFO) << "Started thread with name: " << name
- << " and handle: " << *audio_thread_.GetHandle();
+ RTC_DLOG(LS_INFO) << "Started thread with name: " << name
+ << " and handle: " << *audio_thread_.GetHandle();
}
// Start streaming data between the endpoint buffer and the audio engine.
@@ -579,9 +580,9 @@
}
bool CoreAudioBase::Stop() {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
- << "]";
- RTC_DLOG(INFO) << "total activity time: " << TimeSinceStart();
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+ << "]";
+ RTC_DLOG(LS_INFO) << "total activity time: " << TimeSinceStart();
// Stop audio streaming.
_com_error error = audio_client_->Stop();
@@ -614,8 +615,8 @@
// Delete the previous registration by the client to receive notifications
// about audio session events.
- RTC_DLOG(INFO) << "audio session state: "
- << SessionStateToString(GetAudioSessionState());
+ RTC_DLOG(LS_INFO) << "audio session state: "
+ << SessionStateToString(GetAudioSessionState());
error = audio_session_control_->UnregisterAudioSessionNotification(this);
if (FAILED(error.Error())) {
RTC_LOG(LS_ERROR)
@@ -663,7 +664,7 @@
<< core_audio_utility::ErrorToString(error);
*available = false;
}
- RTC_DLOG(INFO) << "master volume for output audio session: " << volume;
+ RTC_DLOG(LS_INFO) << "master volume for output audio session: " << volume;
*available = true;
return false;
@@ -674,8 +675,8 @@
// device notifications. Hence, the emulated restart sequence covers most parts
// of a real sequence expect the actual device switch.
bool CoreAudioBase::Restart() {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
- << "]";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+ << "]";
if (!automatic_restart()) {
return false;
}
@@ -685,12 +686,12 @@
}
void CoreAudioBase::StopThread() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(!IsRestarting());
if (!audio_thread_.empty()) {
- RTC_DLOG(INFO) << "Sets stop_event...";
+ RTC_DLOG(LS_INFO) << "Sets stop_event...";
SetEvent(stop_event_.Get());
- RTC_DLOG(INFO) << "PlatformThread::Finalize...";
+ RTC_DLOG(LS_INFO) << "PlatformThread::Finalize...";
audio_thread_.Finalize();
// Ensure that we don't quit the main thread loop immediately next
@@ -701,8 +702,8 @@
}
bool CoreAudioBase::HandleRestartEvent() {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
- << "]";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+ << "]";
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
RTC_DCHECK(!audio_thread_.empty());
RTC_DCHECK(IsRestarting());
@@ -716,13 +717,13 @@
}
bool CoreAudioBase::SwitchDeviceIfNeeded() {
- RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
- << "]";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+ << "]";
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
RTC_DCHECK(IsRestarting());
- RTC_DLOG(INFO) << "device_index=" << device_index_
- << " => device_id: " << device_id_;
+ RTC_DLOG(LS_INFO) << "device_index=" << device_index_
+ << " => device_id: " << device_id_;
// Ensure that at least one device exists and can be utilized. The most
// probable cause for ending up here is that a device has been removed.
@@ -746,7 +747,7 @@
return false;
}
} else {
- RTC_LOG(INFO)
+ RTC_LOG(LS_INFO)
<< "Device configuration has not changed => keeping selected device";
}
return true;
@@ -766,14 +767,14 @@
// TODO(henrika): only used for debugging purposes currently.
ULONG CoreAudioBase::AddRef() {
ULONG new_ref = InterlockedIncrement(&ref_count_);
- // RTC_DLOG(INFO) << "__AddRef => " << new_ref;
+ // RTC_DLOG(LS_INFO) << "__AddRef => " << new_ref;
return new_ref;
}
// TODO(henrika): does not call delete this.
ULONG CoreAudioBase::Release() {
ULONG new_ref = InterlockedDecrement(&ref_count_);
- // RTC_DLOG(INFO) << "__Release => " << new_ref;
+ // RTC_DLOG(LS_INFO) << "__Release => " << new_ref;
return new_ref;
}
@@ -792,9 +793,9 @@
// IAudioSessionEvents::OnStateChanged.
HRESULT CoreAudioBase::OnStateChanged(AudioSessionState new_state) {
- RTC_DLOG(INFO) << "___" << __FUNCTION__ << "["
- << DirectionToString(direction())
- << "] new_state: " << SessionStateToString(new_state);
+ RTC_DLOG(LS_INFO) << "___" << __FUNCTION__ << "["
+ << DirectionToString(direction())
+ << "] new_state: " << SessionStateToString(new_state);
return S_OK;
}
@@ -806,9 +807,9 @@
// same event.
HRESULT CoreAudioBase::OnSessionDisconnected(
AudioSessionDisconnectReason disconnect_reason) {
- RTC_DLOG(INFO) << "___" << __FUNCTION__ << "["
- << DirectionToString(direction()) << "] reason: "
- << SessionDisconnectReasonToString(disconnect_reason);
+ RTC_DLOG(LS_INFO) << "___" << __FUNCTION__ << "["
+ << DirectionToString(direction()) << "] reason: "
+ << SessionDisconnectReasonToString(disconnect_reason);
// Ignore changes in the audio session (don't try to restart) if the user
// has explicitly asked for this type of ADM during construction.
if (!automatic_restart()) {
@@ -869,8 +870,8 @@
RTC_LOG(LS_ERROR) << "MMCSS is not supported";
return;
}
- RTC_DLOG(INFO) << "[" << DirectionToString(direction())
- << "] ThreadRun starts...";
+ RTC_DLOG(LS_INFO) << "[" << DirectionToString(direction())
+ << "] ThreadRun starts...";
// TODO(henrika): difference between "Pro Audio" and "Audio"?
ScopedMMCSSRegistration mmcss_registration(L"Pro Audio");
ScopedCOMInitializer com_initializer(ScopedCOMInitializer::kMTA);
@@ -938,8 +939,8 @@
// this stream should be destroyed instead of reused in the future.
}
- RTC_DLOG(INFO) << "[" << DirectionToString(direction())
- << "] ...ThreadRun stops";
+ RTC_DLOG(LS_INFO) << "[" << DirectionToString(direction())
+ << "] ...ThreadRun stops";
}
} // namespace webrtc_win
diff --git a/modules/audio_device/win/core_audio_input_win.cc b/modules/audio_device/win/core_audio_input_win.cc
index be4aec8..6ad5f46 100644
--- a/modules/audio_device/win/core_audio_input_win.cc
+++ b/modules/audio_device/win/core_audio_input_win.cc
@@ -33,24 +33,24 @@
automatic_restart,
[this](uint64_t freq) { return OnDataCallback(freq); },
[this](ErrorType err) { return OnErrorCallback(err); }) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
thread_checker_audio_.Detach();
}
CoreAudioInput::~CoreAudioInput() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
}
int CoreAudioInput::Init() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return 0;
}
int CoreAudioInput::Terminate() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
StopRecording();
return 0;
@@ -62,17 +62,17 @@
}
int CoreAudioInput::SetDevice(int index) {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
RTC_DCHECK_GE(index, 0);
RTC_DCHECK_RUN_ON(&thread_checker_);
return CoreAudioBase::SetDevice(index);
}
int CoreAudioInput::SetDevice(AudioDeviceModule::WindowsDeviceType device) {
- RTC_DLOG(INFO) << __FUNCTION__ << ": "
- << ((device == AudioDeviceModule::kDefaultDevice)
- ? "Default"
- : "DefaultCommunication");
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": "
+ << ((device == AudioDeviceModule::kDefaultDevice)
+ ? "Default"
+ : "DefaultCommunication");
RTC_DCHECK_RUN_ON(&thread_checker_);
return SetDevice((device == AudioDeviceModule::kDefaultDevice) ? 0 : 1);
}
@@ -80,26 +80,26 @@
int CoreAudioInput::DeviceName(int index,
std::string* name,
std::string* guid) {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(name);
return CoreAudioBase::DeviceName(index, name, guid);
}
void CoreAudioInput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
audio_device_buffer_ = audio_buffer;
}
bool CoreAudioInput::RecordingIsInitialized() const {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << initialized_;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
RTC_DCHECK_RUN_ON(&thread_checker_);
return initialized_;
}
int CoreAudioInput::InitRecording() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(!initialized_);
RTC_DCHECK(!Recording());
RTC_DCHECK(!audio_capture_client_);
@@ -155,7 +155,7 @@
}
int CoreAudioInput::StartRecording() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(!Recording());
RTC_DCHECK(fine_audio_buffer_);
RTC_DCHECK(audio_device_buffer_);
@@ -179,7 +179,7 @@
}
int CoreAudioInput::StopRecording() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_) {
return 0;
}
@@ -214,7 +214,7 @@
}
bool CoreAudioInput::Recording() {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << is_active_;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << is_active_;
return is_active_;
}
@@ -222,7 +222,7 @@
// are not compatible with the old ADM implementation since it allows accessing
// the volume control with any active audio output stream.
int CoreAudioInput::VolumeIsAvailable(bool* available) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return IsVolumeControlAvailable(available) ? 0 : -1;
}
@@ -230,7 +230,7 @@
// Triggers the restart sequence. Only used for testing purposes to emulate
// a real event where e.g. an active input device is removed.
int CoreAudioInput::RestartRecording() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!Recording()) {
return 0;
@@ -249,14 +249,14 @@
}
int CoreAudioInput::SetSampleRate(uint32_t sample_rate) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
sample_rate_ = sample_rate;
return 0;
}
void CoreAudioInput::ReleaseCOMObjects() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CoreAudioBase::ReleaseCOMObjects();
if (audio_capture_client_.Get()) {
audio_capture_client_.Reset();
@@ -273,7 +273,7 @@
return false;
}
if (num_data_callbacks_ == 0) {
- RTC_LOG(INFO) << "--- Input audio stream is alive ---";
+ RTC_LOG(LS_INFO) << "--- Input audio stream is alive ---";
}
UINT32 num_frames_in_next_packet = 0;
_com_error error =
@@ -328,7 +328,7 @@
}
}
if (num_data_callbacks_ % 500 == 0) {
- RTC_DLOG(INFO) << "latency: " << latency_ms_;
+ RTC_DLOG(LS_INFO) << "latency: " << latency_ms_;
}
// The data in the packet is not correlated with the previous packet's
@@ -382,7 +382,7 @@
}
bool CoreAudioInput::OnErrorCallback(ErrorType error) {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << as_integer(error);
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << as_integer(error);
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
HandleStreamDisconnected();
@@ -426,7 +426,7 @@
// safe.
// TODO(henrika): add more details.
bool CoreAudioInput::HandleStreamDisconnected() {
- RTC_DLOG(INFO) << "<<<--- " << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << "<<<--- " << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
RTC_DCHECK(automatic_restart());
@@ -445,7 +445,7 @@
return false;
}
- RTC_DLOG(INFO) << __FUNCTION__ << " --->>>";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << " --->>>";
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 bd4132a..7522922 100644
--- a/modules/audio_device/win/core_audio_output_win.cc
+++ b/modules/audio_device/win/core_audio_output_win.cc
@@ -29,25 +29,25 @@
automatic_restart,
[this](uint64_t freq) { return OnDataCallback(freq); },
[this](ErrorType err) { return OnErrorCallback(err); }) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
thread_checker_audio_.Detach();
}
CoreAudioOutput::~CoreAudioOutput() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
Terminate();
}
int CoreAudioOutput::Init() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return 0;
}
int CoreAudioOutput::Terminate() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
StopPlayout();
return 0;
@@ -59,17 +59,17 @@
}
int CoreAudioOutput::SetDevice(int index) {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
RTC_DCHECK_GE(index, 0);
RTC_DCHECK_RUN_ON(&thread_checker_);
return CoreAudioBase::SetDevice(index);
}
int CoreAudioOutput::SetDevice(AudioDeviceModule::WindowsDeviceType device) {
- RTC_DLOG(INFO) << __FUNCTION__ << ": "
- << ((device == AudioDeviceModule::kDefaultDevice)
- ? "Default"
- : "DefaultCommunication");
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": "
+ << ((device == AudioDeviceModule::kDefaultDevice)
+ ? "Default"
+ : "DefaultCommunication");
RTC_DCHECK_RUN_ON(&thread_checker_);
return SetDevice((device == AudioDeviceModule::kDefaultDevice) ? 0 : 1);
}
@@ -77,26 +77,26 @@
int CoreAudioOutput::DeviceName(int index,
std::string* name,
std::string* guid) {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(name);
return CoreAudioBase::DeviceName(index, name, guid);
}
void CoreAudioOutput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
audio_device_buffer_ = audio_buffer;
}
bool CoreAudioOutput::PlayoutIsInitialized() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return initialized_;
}
int CoreAudioOutput::InitPlayout() {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << IsRestarting();
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
RTC_DCHECK(!initialized_);
RTC_DCHECK(!Playing());
RTC_DCHECK(!audio_render_client_);
@@ -150,7 +150,7 @@
}
int CoreAudioOutput::StartPlayout() {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << IsRestarting();
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
RTC_DCHECK(!Playing());
RTC_DCHECK(fine_audio_buffer_);
RTC_DCHECK(audio_device_buffer_);
@@ -180,7 +180,7 @@
}
int CoreAudioOutput::StopPlayout() {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << IsRestarting();
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
if (!initialized_) {
return 0;
}
@@ -214,7 +214,7 @@
}
bool CoreAudioOutput::Playing() {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << is_active_;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << is_active_;
return is_active_;
}
@@ -222,7 +222,7 @@
// are not compatible with the old ADM implementation since it allows accessing
// the volume control with any active audio output stream.
int CoreAudioOutput::VolumeIsAvailable(bool* available) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return IsVolumeControlAvailable(available) ? 0 : -1;
}
@@ -230,7 +230,7 @@
// Triggers the restart sequence. Only used for testing purposes to emulate
// a real event where e.g. an active output device is removed.
int CoreAudioOutput::RestartPlayout() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
if (!Playing()) {
return 0;
@@ -243,20 +243,20 @@
}
bool CoreAudioOutput::Restarting() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
return IsRestarting();
}
int CoreAudioOutput::SetSampleRate(uint32_t sample_rate) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_);
sample_rate_ = sample_rate;
return 0;
}
void CoreAudioOutput::ReleaseCOMObjects() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CoreAudioBase::ReleaseCOMObjects();
if (audio_render_client_.Get()) {
audio_render_client_.Reset();
@@ -264,7 +264,7 @@
}
bool CoreAudioOutput::OnErrorCallback(ErrorType error) {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << as_integer(error);
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << as_integer(error);
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
if (!initialized_ || !Playing()) {
return true;
@@ -281,7 +281,7 @@
bool CoreAudioOutput::OnDataCallback(uint64_t device_frequency) {
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
if (num_data_callbacks_ == 0) {
- RTC_LOG(INFO) << "--- Output audio stream is alive ---";
+ RTC_LOG(LS_INFO) << "--- Output audio stream is alive ---";
}
// Get the padding value which indicates the amount of valid unread data that
// the endpoint buffer currently contains.
@@ -329,7 +329,7 @@
// TODO(henrika): note that FineAudioBuffer adds latency as well.
latency_ms_ = EstimateOutputLatencyMillis(device_frequency);
if (num_data_callbacks_ % 500 == 0) {
- RTC_DLOG(INFO) << "latency: " << latency_ms_;
+ RTC_DLOG(LS_INFO) << "latency: " << latency_ms_;
}
}
@@ -394,7 +394,7 @@
// safe.
// TODO(henrika): add more details.
bool CoreAudioOutput::HandleStreamDisconnected() {
- RTC_DLOG(INFO) << "<<<--- " << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << "<<<--- " << __FUNCTION__;
RTC_DCHECK_RUN_ON(&thread_checker_audio_);
RTC_DCHECK(automatic_restart());
@@ -413,7 +413,7 @@
return false;
}
- RTC_DLOG(INFO) << __FUNCTION__ << " --->>>";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << " --->>>";
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 5950c8d..976edc8 100644
--- a/modules/audio_device/win/core_audio_utility_win.cc
+++ b/modules/audio_device/win/core_audio_utility_win.cc
@@ -205,7 +205,7 @@
L"%WINDIR%\\system32\\audioses.dll";
wchar_t path[MAX_PATH] = {0};
ExpandEnvironmentStringsW(kAudiosesDLL, path, arraysize(path));
- RTC_DLOG(INFO) << rtc::ToUtf8(path);
+ RTC_DLOG(LS_INFO) << rtc::ToUtf8(path);
return (LoadLibraryExW(path, nullptr, LOAD_WITH_ALTERED_SEARCH_PATH) !=
nullptr);
}
@@ -214,7 +214,7 @@
static const wchar_t* const kAvrtDLL = L"%WINDIR%\\system32\\Avrt.dll";
wchar_t path[MAX_PATH] = {0};
ExpandEnvironmentStringsW(kAvrtDLL, path, arraysize(path));
- RTC_DLOG(INFO) << rtc::ToUtf8(path);
+ RTC_DLOG(LS_INFO) << rtc::ToUtf8(path);
return (LoadLibraryExW(path, nullptr, LOAD_WITH_ALTERED_SEARCH_PATH) !=
nullptr);
}
@@ -283,10 +283,10 @@
ComPtr<IMMDevice> CreateDeviceInternal(const std::string& device_id,
EDataFlow data_flow,
ERole role) {
- RTC_DLOG(INFO) << "CreateDeviceInternal: "
- "id="
- << device_id << ", flow=" << FlowToString(data_flow)
- << ", role=" << RoleToString(role);
+ RTC_DLOG(LS_INFO) << "CreateDeviceInternal: "
+ "id="
+ << device_id << ", flow=" << FlowToString(data_flow)
+ << ", role=" << RoleToString(role);
ComPtr<IMMDevice> audio_endpoint_device;
// Create the IMMDeviceEnumerator interface.
@@ -587,10 +587,10 @@
// [2] friendly name: Headset Microphone (2- Arctis 7 Chat)
// [2] unique id : {0.0.1.00000000}.{ff9eed76-196e-467a-b295-26986e69451c}
for (size_t i = 0; i < device_names->size(); ++i) {
- RTC_DLOG(INFO) << "[" << i
- << "] friendly name: " << (*device_names)[i].device_name;
- RTC_DLOG(INFO) << "[" << i
- << "] unique id : " << (*device_names)[i].unique_id;
+ RTC_DLOG(LS_INFO) << "[" << i
+ << "] friendly name: " << (*device_names)[i].device_name;
+ RTC_DLOG(LS_INFO) << "[" << i
+ << "] unique id : " << (*device_names)[i].unique_id;
}
return true;
@@ -614,8 +614,8 @@
// Override default sample rate if `fixed_sample_rate` is set and different
// from the default rate.
if (fixed_sample_rate > 0 && fixed_sample_rate != sample_rate) {
- RTC_DLOG(INFO) << "Using fixed sample rate instead of the preferred: "
- << sample_rate << " is replaced by " << fixed_sample_rate;
+ RTC_DLOG(LS_INFO) << "Using fixed sample rate instead of the preferred: "
+ << sample_rate << " is replaced by " << fixed_sample_rate;
sample_rate = fixed_sample_rate;
}
// TODO(henrika): utilize full mix_format.Format.wBitsPerSample.
@@ -634,7 +634,7 @@
AudioParameters audio_params(sample_rate, channels, frames_per_buffer);
*params = audio_params;
- RTC_DLOG(INFO) << audio_params.ToString();
+ RTC_DLOG(LS_INFO) << audio_params.ToString();
return hr;
}
@@ -669,13 +669,13 @@
}
bool IsSupported() {
- RTC_DLOG(INFO) << "IsSupported";
+ RTC_DLOG(LS_INFO) << "IsSupported";
static bool g_is_supported = IsSupportedInternal();
return g_is_supported;
}
bool IsMMCSSSupported() {
- RTC_DLOG(INFO) << "IsMMCSSSupported";
+ RTC_DLOG(LS_INFO) << "IsMMCSSSupported";
return LoadAvrtDll();
}
@@ -698,7 +698,7 @@
} else if (data_flow == eAll) {
str = "Total number of devices: ";
}
- RTC_DLOG(INFO) << str << number_of_active_devices;
+ RTC_DLOG(LS_INFO) << str << number_of_active_devices;
return static_cast<int>(number_of_active_devices);
}
@@ -713,33 +713,33 @@
}
ComPtr<IMMDeviceEnumerator> CreateDeviceEnumerator() {
- RTC_DLOG(INFO) << "CreateDeviceEnumerator";
+ RTC_DLOG(LS_INFO) << "CreateDeviceEnumerator";
return CreateDeviceEnumeratorInternal(true);
}
std::string GetDefaultInputDeviceID() {
- RTC_DLOG(INFO) << "GetDefaultInputDeviceID";
+ RTC_DLOG(LS_INFO) << "GetDefaultInputDeviceID";
ComPtr<IMMDevice> device(
CreateDevice(AudioDeviceName::kDefaultDeviceId, eCapture, eConsole));
return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
}
std::string GetDefaultOutputDeviceID() {
- RTC_DLOG(INFO) << "GetDefaultOutputDeviceID";
+ RTC_DLOG(LS_INFO) << "GetDefaultOutputDeviceID";
ComPtr<IMMDevice> device(
CreateDevice(AudioDeviceName::kDefaultDeviceId, eRender, eConsole));
return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
}
std::string GetCommunicationsInputDeviceID() {
- RTC_DLOG(INFO) << "GetCommunicationsInputDeviceID";
+ RTC_DLOG(LS_INFO) << "GetCommunicationsInputDeviceID";
ComPtr<IMMDevice> device(CreateDevice(AudioDeviceName::kDefaultDeviceId,
eCapture, eCommunications));
return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
}
std::string GetCommunicationsOutputDeviceID() {
- RTC_DLOG(INFO) << "GetCommunicationsOutputDeviceID";
+ RTC_DLOG(LS_INFO) << "GetCommunicationsOutputDeviceID";
ComPtr<IMMDevice> device(CreateDevice(AudioDeviceName::kDefaultDeviceId,
eRender, eCommunications));
return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
@@ -748,24 +748,24 @@
ComPtr<IMMDevice> CreateDevice(const std::string& device_id,
EDataFlow data_flow,
ERole role) {
- RTC_DLOG(INFO) << "CreateDevice";
+ RTC_DLOG(LS_INFO) << "CreateDevice";
return CreateDeviceInternal(device_id, data_flow, role);
}
AudioDeviceName GetDeviceName(IMMDevice* device) {
- RTC_DLOG(INFO) << "GetDeviceName";
+ RTC_DLOG(LS_INFO) << "GetDeviceName";
RTC_DCHECK(device);
AudioDeviceName device_name(GetDeviceFriendlyNameInternal(device),
GetDeviceIdInternal(device));
- RTC_DLOG(INFO) << "friendly name: " << device_name.device_name;
- RTC_DLOG(INFO) << "unique id : " << device_name.unique_id;
+ RTC_DLOG(LS_INFO) << "friendly name: " << device_name.device_name;
+ RTC_DLOG(LS_INFO) << "unique id : " << device_name.unique_id;
return device_name;
}
std::string GetFriendlyName(const std::string& device_id,
EDataFlow data_flow,
ERole role) {
- RTC_DLOG(INFO) << "GetFriendlyName";
+ RTC_DLOG(LS_INFO) << "GetFriendlyName";
ComPtr<IMMDevice> audio_device = CreateDevice(device_id, data_flow, role);
if (!audio_device.Get())
return std::string();
@@ -775,7 +775,7 @@
}
EDataFlow GetDataFlow(IMMDevice* device) {
- RTC_DLOG(INFO) << "GetDataFlow";
+ RTC_DLOG(LS_INFO) << "GetDataFlow";
RTC_DCHECK(device);
ComPtr<IMMEndpoint> endpoint;
_com_error error = device->QueryInterface(endpoint.GetAddressOf());
@@ -796,32 +796,32 @@
}
bool GetInputDeviceNames(webrtc::AudioDeviceNames* device_names) {
- RTC_DLOG(INFO) << "GetInputDeviceNames";
+ RTC_DLOG(LS_INFO) << "GetInputDeviceNames";
RTC_DCHECK(device_names);
RTC_DCHECK(device_names->empty());
return GetDeviceNamesInternal(eCapture, device_names);
}
bool GetOutputDeviceNames(webrtc::AudioDeviceNames* device_names) {
- RTC_DLOG(INFO) << "GetOutputDeviceNames";
+ RTC_DLOG(LS_INFO) << "GetOutputDeviceNames";
RTC_DCHECK(device_names);
RTC_DCHECK(device_names->empty());
return GetDeviceNamesInternal(eRender, device_names);
}
ComPtr<IAudioSessionManager2> CreateSessionManager2(IMMDevice* device) {
- RTC_DLOG(INFO) << "CreateSessionManager2";
+ RTC_DLOG(LS_INFO) << "CreateSessionManager2";
return CreateSessionManager2Internal(device);
}
Microsoft::WRL::ComPtr<IAudioSessionEnumerator> CreateSessionEnumerator(
IMMDevice* device) {
- RTC_DLOG(INFO) << "CreateSessionEnumerator";
+ RTC_DLOG(LS_INFO) << "CreateSessionEnumerator";
return CreateSessionEnumeratorInternal(device);
}
int NumberOfActiveSessions(IMMDevice* device) {
- RTC_DLOG(INFO) << "NumberOfActiveSessions";
+ RTC_DLOG(LS_INFO) << "NumberOfActiveSessions";
ComPtr<IAudioSessionEnumerator> session_enumerator =
CreateSessionEnumerator(device);
@@ -833,7 +833,7 @@
<< ErrorToString(error);
return 0;
}
- RTC_DLOG(INFO) << "Total number of audio sessions: " << session_count;
+ RTC_DLOG(LS_INFO) << "Total number of audio sessions: " << session_count;
int num_active = 0;
for (int session = 0; session < session_count; session++) {
@@ -849,8 +849,8 @@
// Log the display name of the audio session for debugging purposes.
LPWSTR display_name;
if (SUCCEEDED(session_control->GetDisplayName(&display_name))) {
- RTC_DLOG(INFO) << "display name: "
- << rtc::ToUtf8(display_name, wcslen(display_name));
+ RTC_DLOG(LS_INFO) << "display name: "
+ << rtc::ToUtf8(display_name, wcslen(display_name));
CoTaskMemFree(display_name);
}
@@ -867,14 +867,14 @@
}
}
- RTC_DLOG(INFO) << "Number of active audio sessions: " << num_active;
+ RTC_DLOG(LS_INFO) << "Number of active audio sessions: " << num_active;
return num_active;
}
ComPtr<IAudioClient> CreateClient(const std::string& device_id,
EDataFlow data_flow,
ERole role) {
- RTC_DLOG(INFO) << "CreateClient";
+ RTC_DLOG(LS_INFO) << "CreateClient";
ComPtr<IMMDevice> device(CreateDevice(device_id, data_flow, role));
return CreateClientInternal(device.Get());
}
@@ -882,7 +882,7 @@
ComPtr<IAudioClient2> CreateClient2(const std::string& device_id,
EDataFlow data_flow,
ERole role) {
- RTC_DLOG(INFO) << "CreateClient2";
+ RTC_DLOG(LS_INFO) << "CreateClient2";
ComPtr<IMMDevice> device(CreateDevice(device_id, data_flow, role));
return CreateClient2Internal(device.Get());
}
@@ -890,13 +890,13 @@
ComPtr<IAudioClient3> CreateClient3(const std::string& device_id,
EDataFlow data_flow,
ERole role) {
- RTC_DLOG(INFO) << "CreateClient3";
+ RTC_DLOG(LS_INFO) << "CreateClient3";
ComPtr<IMMDevice> device(CreateDevice(device_id, data_flow, role));
return CreateClient3Internal(device.Get());
}
HRESULT SetClientProperties(IAudioClient2* client) {
- RTC_DLOG(INFO) << "SetClientProperties";
+ RTC_DLOG(LS_INFO) << "SetClientProperties";
RTC_DCHECK(client);
if (GetAudioClientVersion() < 2) {
RTC_LOG(LS_WARNING) << "Requires IAudioClient2 or higher";
@@ -919,10 +919,10 @@
RTC_LOG(LS_ERROR) << "IAudioClient2::IsOffloadCapable failed: "
<< ErrorToString(error);
}
- RTC_DLOG(INFO) << "supports_offload: " << supports_offload;
+ RTC_DLOG(LS_INFO) << "supports_offload: " << supports_offload;
props.bIsOffload = false;
#if (NTDDI_VERSION < NTDDI_WINBLUE)
- RTC_DLOG(INFO) << "options: Not supported in this build";
+ RTC_DLOG(LS_INFO) << "options: Not supported in this build";
#else
// TODO(henrika): pros and cons compared with AUDCLNT_STREAMOPTIONS_NONE?
props.Options |= AUDCLNT_STREAMOPTIONS_NONE;
@@ -939,7 +939,7 @@
// an appropriate interface to use for communications scenarios.
// This interface is mainly meant for pro audio scenarios.
// props.Options |= AUDCLNT_STREAMOPTIONS_MATCH_FORMAT;
- RTC_DLOG(INFO) << "options: 0x" << rtc::ToHex(props.Options);
+ RTC_DLOG(LS_INFO) << "options: 0x" << rtc::ToHex(props.Options);
#endif
error = client->SetClientProperties(&props);
if (FAILED(error.Error())) {
@@ -953,7 +953,7 @@
const WAVEFORMATEXTENSIBLE* format,
REFERENCE_TIME* min_buffer_duration,
REFERENCE_TIME* max_buffer_duration) {
- RTC_DLOG(INFO) << "GetBufferSizeLimits";
+ RTC_DLOG(LS_INFO) << "GetBufferSizeLimits";
RTC_DCHECK(client);
if (GetAudioClientVersion() < 2) {
RTC_LOG(LS_WARNING) << "Requires IAudioClient2 or higher";
@@ -975,15 +975,15 @@
} else {
*min_buffer_duration = min_duration;
*max_buffer_duration = max_duration;
- RTC_DLOG(INFO) << "min_buffer_duration: " << min_buffer_duration;
- RTC_DLOG(INFO) << "max_buffer_duration: " << max_buffer_duration;
+ RTC_DLOG(LS_INFO) << "min_buffer_duration: " << min_buffer_duration;
+ RTC_DLOG(LS_INFO) << "max_buffer_duration: " << max_buffer_duration;
}
return error.Error();
}
HRESULT GetSharedModeMixFormat(IAudioClient* client,
WAVEFORMATEXTENSIBLE* format) {
- RTC_DLOG(INFO) << "GetSharedModeMixFormat";
+ RTC_DLOG(LS_INFO) << "GetSharedModeMixFormat";
RTC_DCHECK(client);
// The GetMixFormat method retrieves the stream format that the audio engine
@@ -1030,7 +1030,7 @@
// the returned structure is correctly extended or not.
RTC_CHECK_LE(wrapped_format.size(), sizeof(WAVEFORMATEXTENSIBLE));
memcpy(format, wrapped_format.get(), wrapped_format.size());
- RTC_DLOG(INFO) << WaveFormatToString(format);
+ RTC_DLOG(LS_INFO) << WaveFormatToString(format);
return error.Error();
}
@@ -1038,7 +1038,7 @@
bool IsFormatSupported(IAudioClient* client,
AUDCLNT_SHAREMODE share_mode,
const WAVEFORMATEXTENSIBLE* format) {
- RTC_DLOG(INFO) << "IsFormatSupported";
+ RTC_DLOG(LS_INFO) << "IsFormatSupported";
RTC_DCHECK(client);
ScopedCoMem<WAVEFORMATEX> closest_match;
// This method provides a way for a client to determine, before calling
@@ -1049,23 +1049,23 @@
_com_error error = client->IsFormatSupported(
share_mode, reinterpret_cast<const WAVEFORMATEX*>(format),
&closest_match);
- RTC_LOG(INFO) << WaveFormatToString(
+ RTC_LOG(LS_INFO) << WaveFormatToString(
const_cast<WAVEFORMATEXTENSIBLE*>(format));
if ((error.Error() == S_OK) && (closest_match == nullptr)) {
- RTC_DLOG(INFO)
+ RTC_DLOG(LS_INFO)
<< "The audio endpoint device supports the specified stream format";
} else if ((error.Error() == S_FALSE) && (closest_match != nullptr)) {
// Call succeeded with a closest match to the specified format. This log can
// only be triggered for shared mode.
RTC_LOG(LS_WARNING)
<< "Exact format is not supported, but a closest match exists";
- RTC_LOG(INFO) << WaveFormatToString(closest_match.Get());
+ RTC_LOG(LS_INFO) << WaveFormatToString(closest_match.Get());
} else if ((error.Error() == AUDCLNT_E_UNSUPPORTED_FORMAT) &&
(closest_match == nullptr)) {
// The audio engine does not support the caller-specified format or any
// similar format.
- RTC_DLOG(INFO) << "The audio endpoint device does not support the "
- "specified stream format";
+ RTC_DLOG(LS_INFO) << "The audio endpoint device does not support the "
+ "specified stream format";
} else {
RTC_LOG(LS_ERROR) << "IAudioClient::IsFormatSupported failed: "
<< ErrorToString(error);
@@ -1077,7 +1077,7 @@
HRESULT GetDevicePeriod(IAudioClient* client,
AUDCLNT_SHAREMODE share_mode,
REFERENCE_TIME* device_period) {
- RTC_DLOG(INFO) << "GetDevicePeriod";
+ RTC_DLOG(LS_INFO) << "GetDevicePeriod";
RTC_DCHECK(client);
// The `default_period` parameter specifies the default scheduling period
// for a shared-mode stream. The `minimum_period` parameter specifies the
@@ -1094,10 +1094,10 @@
*device_period = (share_mode == AUDCLNT_SHAREMODE_SHARED) ? default_period
: minimum_period;
- RTC_LOG(INFO) << "device_period: "
- << ReferenceTimeToTimeDelta(*device_period).ms() << " [ms]";
- RTC_LOG(INFO) << "minimum_period: "
- << ReferenceTimeToTimeDelta(minimum_period).ms() << " [ms]";
+ RTC_LOG(LS_INFO) << "device_period: "
+ << ReferenceTimeToTimeDelta(*device_period).ms() << " [ms]";
+ RTC_LOG(LS_INFO) << "minimum_period: "
+ << ReferenceTimeToTimeDelta(minimum_period).ms() << " [ms]";
return error.Error();
}
@@ -1107,7 +1107,7 @@
uint32_t* fundamental_period_in_frames,
uint32_t* min_period_in_frames,
uint32_t* max_period_in_frames) {
- RTC_DLOG(INFO) << "GetSharedModeEnginePeriod";
+ RTC_DLOG(LS_INFO) << "GetSharedModeEnginePeriod";
RTC_DCHECK(client3);
UINT32 default_period = 0;
@@ -1125,15 +1125,17 @@
WAVEFORMATEX format_ex = format->Format;
const WORD sample_rate = format_ex.nSamplesPerSec;
- RTC_LOG(INFO) << "default_period_in_frames: " << default_period << " ("
- << FramesToMilliseconds(default_period, sample_rate) << " ms)";
- RTC_LOG(INFO) << "fundamental_period_in_frames: " << fundamental_period
- << " (" << FramesToMilliseconds(fundamental_period, sample_rate)
- << " ms)";
- RTC_LOG(INFO) << "min_period_in_frames: " << min_period << " ("
- << FramesToMilliseconds(min_period, sample_rate) << " ms)";
- RTC_LOG(INFO) << "max_period_in_frames: " << max_period << " ("
- << FramesToMilliseconds(max_period, sample_rate) << " ms)";
+ RTC_LOG(LS_INFO) << "default_period_in_frames: " << default_period << " ("
+ << FramesToMilliseconds(default_period, sample_rate)
+ << " ms)";
+ RTC_LOG(LS_INFO) << "fundamental_period_in_frames: " << fundamental_period
+ << " ("
+ << FramesToMilliseconds(fundamental_period, sample_rate)
+ << " ms)";
+ RTC_LOG(LS_INFO) << "min_period_in_frames: " << min_period << " ("
+ << FramesToMilliseconds(min_period, sample_rate) << " ms)";
+ RTC_LOG(LS_INFO) << "max_period_in_frames: " << max_period << " ("
+ << FramesToMilliseconds(max_period, sample_rate) << " ms)";
*default_period_in_frames = default_period;
*fundamental_period_in_frames = fundamental_period;
*min_period_in_frames = min_period;
@@ -1143,7 +1145,7 @@
HRESULT GetPreferredAudioParameters(IAudioClient* client,
AudioParameters* params) {
- RTC_DLOG(INFO) << "GetPreferredAudioParameters";
+ RTC_DLOG(LS_INFO) << "GetPreferredAudioParameters";
RTC_DCHECK(client);
return GetPreferredAudioParametersInternal(client, params, -1);
}
@@ -1151,7 +1153,7 @@
HRESULT GetPreferredAudioParameters(IAudioClient* client,
webrtc::AudioParameters* params,
uint32_t sample_rate) {
- RTC_DLOG(INFO) << "GetPreferredAudioParameters: " << sample_rate;
+ RTC_DLOG(LS_INFO) << "GetPreferredAudioParameters: " << sample_rate;
RTC_DCHECK(client);
return GetPreferredAudioParametersInternal(client, params, sample_rate);
}
@@ -1162,8 +1164,9 @@
REFERENCE_TIME buffer_duration,
bool auto_convert_pcm,
uint32_t* endpoint_buffer_size) {
- RTC_DLOG(INFO) << "SharedModeInitialize: buffer_duration=" << buffer_duration
- << ", auto_convert_pcm=" << auto_convert_pcm;
+ RTC_DLOG(LS_INFO) << "SharedModeInitialize: buffer_duration="
+ << buffer_duration
+ << ", auto_convert_pcm=" << auto_convert_pcm;
RTC_DCHECK(client);
RTC_DCHECK_GE(buffer_duration, 0);
if (buffer_duration != 0) {
@@ -1188,7 +1191,7 @@
(event_handle != nullptr && event_handle != INVALID_HANDLE_VALUE);
if (use_event) {
stream_flags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
- RTC_DLOG(INFO) << "The stream is initialized to be event driven";
+ RTC_DLOG(LS_INFO) << "The stream is initialized to be event driven";
}
// Check if sample-rate conversion is requested.
@@ -1197,11 +1200,11 @@
// from our (the client's) format to the audio engine mix format.
// Currently only supported for testing, i.e., not possible to enable using
// public APIs.
- RTC_DLOG(INFO) << "The stream is initialized to support rate conversion";
+ RTC_DLOG(LS_INFO) << "The stream is initialized to support rate conversion";
stream_flags |= AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
stream_flags |= AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
}
- RTC_DLOG(INFO) << "stream_flags: 0x" << rtc::ToHex(stream_flags);
+ RTC_DLOG(LS_INFO) << "stream_flags: 0x" << rtc::ToHex(stream_flags);
// Initialize the shared mode client for minimal delay if `buffer_duration`
// is 0 or possibly a higher delay (more robust) if `buffer_duration` is
@@ -1243,23 +1246,23 @@
}
*endpoint_buffer_size = buffer_size_in_frames;
- RTC_DLOG(INFO) << "endpoint buffer size: " << buffer_size_in_frames
- << " [audio frames]";
+ RTC_DLOG(LS_INFO) << "endpoint buffer size: " << buffer_size_in_frames
+ << " [audio frames]";
const double size_in_ms = static_cast<double>(buffer_size_in_frames) /
(format->Format.nSamplesPerSec / 1000.0);
- RTC_DLOG(INFO) << "endpoint buffer size: "
- << static_cast<int>(size_in_ms + 0.5) << " [ms]";
- RTC_DLOG(INFO) << "bytes per audio frame: " << format->Format.nBlockAlign;
- RTC_DLOG(INFO) << "endpoint buffer size: "
- << buffer_size_in_frames * format->Format.nChannels *
- (format->Format.wBitsPerSample / 8)
- << " [bytes]";
+ RTC_DLOG(LS_INFO) << "endpoint buffer size: "
+ << static_cast<int>(size_in_ms + 0.5) << " [ms]";
+ RTC_DLOG(LS_INFO) << "bytes per audio frame: " << format->Format.nBlockAlign;
+ RTC_DLOG(LS_INFO) << "endpoint buffer size: "
+ << buffer_size_in_frames * format->Format.nChannels *
+ (format->Format.wBitsPerSample / 8)
+ << " [bytes]";
// TODO(henrika): utilize when delay measurements are added.
REFERENCE_TIME latency = 0;
error = client->GetStreamLatency(&latency);
- RTC_DLOG(INFO) << "stream latency: " << ReferenceTimeToTimeDelta(latency).ms()
- << " [ms]";
+ RTC_DLOG(LS_INFO) << "stream latency: "
+ << ReferenceTimeToTimeDelta(latency).ms() << " [ms]";
return error.Error();
}
@@ -1269,9 +1272,9 @@
uint32_t period_in_frames,
bool auto_convert_pcm,
uint32_t* endpoint_buffer_size) {
- RTC_DLOG(INFO) << "SharedModeInitializeLowLatency: period_in_frames="
- << period_in_frames
- << ", auto_convert_pcm=" << auto_convert_pcm;
+ RTC_DLOG(LS_INFO) << "SharedModeInitializeLowLatency: period_in_frames="
+ << period_in_frames
+ << ", auto_convert_pcm=" << auto_convert_pcm;
RTC_DCHECK(client);
RTC_DCHECK_GT(period_in_frames, 0);
if (auto_convert_pcm) {
@@ -1284,13 +1287,13 @@
(event_handle != nullptr && event_handle != INVALID_HANDLE_VALUE);
if (use_event) {
stream_flags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
- RTC_DLOG(INFO) << "The stream is initialized to be event driven";
+ RTC_DLOG(LS_INFO) << "The stream is initialized to be event driven";
}
if (auto_convert_pcm) {
stream_flags |= AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
stream_flags |= AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
}
- RTC_DLOG(INFO) << "stream_flags: 0x" << rtc::ToHex(stream_flags);
+ RTC_DLOG(LS_INFO) << "stream_flags: 0x" << rtc::ToHex(stream_flags);
// Initialize the shared mode client for lowest possible latency.
// It is assumed that GetSharedModeEnginePeriod() has been used to query the
@@ -1324,17 +1327,17 @@
}
*endpoint_buffer_size = buffer_size_in_frames;
- RTC_DLOG(INFO) << "endpoint buffer size: " << buffer_size_in_frames
- << " [audio frames]";
+ RTC_DLOG(LS_INFO) << "endpoint buffer size: " << buffer_size_in_frames
+ << " [audio frames]";
const double size_in_ms = static_cast<double>(buffer_size_in_frames) /
(format->Format.nSamplesPerSec / 1000.0);
- RTC_DLOG(INFO) << "endpoint buffer size: "
- << static_cast<int>(size_in_ms + 0.5) << " [ms]";
- RTC_DLOG(INFO) << "bytes per audio frame: " << format->Format.nBlockAlign;
- RTC_DLOG(INFO) << "endpoint buffer size: "
- << buffer_size_in_frames * format->Format.nChannels *
- (format->Format.wBitsPerSample / 8)
- << " [bytes]";
+ RTC_DLOG(LS_INFO) << "endpoint buffer size: "
+ << static_cast<int>(size_in_ms + 0.5) << " [ms]";
+ RTC_DLOG(LS_INFO) << "bytes per audio frame: " << format->Format.nBlockAlign;
+ RTC_DLOG(LS_INFO) << "endpoint buffer size: "
+ << buffer_size_in_frames * format->Format.nChannels *
+ (format->Format.wBitsPerSample / 8)
+ << " [bytes]";
// TODO(henrika): utilize when delay measurements are added.
REFERENCE_TIME latency = 0;
@@ -1343,14 +1346,14 @@
RTC_LOG(LS_WARNING) << "IAudioClient::GetStreamLatency failed: "
<< ErrorToString(error);
} else {
- RTC_DLOG(INFO) << "stream latency: "
- << ReferenceTimeToTimeDelta(latency).ms() << " [ms]";
+ RTC_DLOG(LS_INFO) << "stream latency: "
+ << ReferenceTimeToTimeDelta(latency).ms() << " [ms]";
}
return error.Error();
}
ComPtr<IAudioRenderClient> CreateRenderClient(IAudioClient* client) {
- RTC_DLOG(INFO) << "CreateRenderClient";
+ RTC_DLOG(LS_INFO) << "CreateRenderClient";
RTC_DCHECK(client);
// Get access to the IAudioRenderClient interface. This interface
// enables us to write output data to a rendering endpoint buffer.
@@ -1366,7 +1369,7 @@
}
ComPtr<IAudioCaptureClient> CreateCaptureClient(IAudioClient* client) {
- RTC_DLOG(INFO) << "CreateCaptureClient";
+ RTC_DLOG(LS_INFO) << "CreateCaptureClient";
RTC_DCHECK(client);
// Get access to the IAudioCaptureClient interface. This interface
// enables us to read input data from a capturing endpoint buffer.
@@ -1382,7 +1385,7 @@
}
ComPtr<IAudioClock> CreateAudioClock(IAudioClient* client) {
- RTC_DLOG(INFO) << "CreateAudioClock";
+ RTC_DLOG(LS_INFO) << "CreateAudioClock";
RTC_DCHECK(client);
// Get access to the IAudioClock interface. This interface enables us to
// monitor a stream's data rate and the current position in the stream.
@@ -1397,7 +1400,7 @@
}
ComPtr<IAudioSessionControl> CreateAudioSessionControl(IAudioClient* client) {
- RTC_DLOG(INFO) << "CreateAudioSessionControl";
+ RTC_DLOG(LS_INFO) << "CreateAudioSessionControl";
RTC_DCHECK(client);
ComPtr<IAudioSessionControl> audio_session_control;
_com_error error = client->GetService(IID_PPV_ARGS(&audio_session_control));
@@ -1410,7 +1413,7 @@
}
ComPtr<ISimpleAudioVolume> CreateSimpleAudioVolume(IAudioClient* client) {
- RTC_DLOG(INFO) << "CreateSimpleAudioVolume";
+ RTC_DLOG(LS_INFO) << "CreateSimpleAudioVolume";
RTC_DCHECK(client);
// Get access to the ISimpleAudioVolume interface. This interface enables a
// client to control the master volume level of an audio session.
@@ -1427,7 +1430,7 @@
bool FillRenderEndpointBufferWithSilence(IAudioClient* client,
IAudioRenderClient* render_client) {
- RTC_DLOG(INFO) << "FillRenderEndpointBufferWithSilence";
+ RTC_DLOG(LS_INFO) << "FillRenderEndpointBufferWithSilence";
RTC_DCHECK(client);
RTC_DCHECK(render_client);
UINT32 endpoint_buffer_size = 0;
@@ -1447,11 +1450,11 @@
<< ErrorToString(error);
return false;
}
- RTC_DLOG(INFO) << "num_queued_frames: " << num_queued_frames;
+ RTC_DLOG(LS_INFO) << "num_queued_frames: " << num_queued_frames;
BYTE* data = nullptr;
int num_frames_to_fill = endpoint_buffer_size - num_queued_frames;
- RTC_DLOG(INFO) << "num_frames_to_fill: " << num_frames_to_fill;
+ RTC_DLOG(LS_INFO) << "num_frames_to_fill: " << num_frames_to_fill;
error = render_client->GetBuffer(num_frames_to_fill, &data);
if (FAILED(error.Error())) {
RTC_LOG(LS_ERROR) << "IAudioRenderClient::GetBuffer failed: "
diff --git a/modules/audio_device/win/core_audio_utility_win.h b/modules/audio_device/win/core_audio_utility_win.h
index 95ed911..afadee5 100644
--- a/modules/audio_device/win/core_audio_utility_win.h
+++ b/modules/audio_device/win/core_audio_utility_win.h
@@ -83,7 +83,7 @@
}
explicit ScopedMMCSSRegistration(const wchar_t* task_name) {
- RTC_DLOG(INFO) << "ScopedMMCSSRegistration: " << rtc::ToUtf8(task_name);
+ RTC_DLOG(LS_INFO) << "ScopedMMCSSRegistration: " << rtc::ToUtf8(task_name);
// Register the calling thread with MMCSS for the supplied `task_name`.
DWORD mmcss_task_index = 0;
mmcss_handle_ = AvSetMmThreadCharacteristicsW(task_name, &mmcss_task_index);
@@ -93,18 +93,18 @@
} else {
const DWORD priority_class = GetPriorityClass(GetCurrentProcess());
const int priority = GetThreadPriority(GetCurrentThread());
- RTC_DLOG(INFO) << "priority class: "
- << PriorityClassToString(priority_class) << "("
- << priority_class << ")";
- RTC_DLOG(INFO) << "priority: " << PriorityToString(priority) << "("
- << priority << ")";
+ RTC_DLOG(LS_INFO) << "priority class: "
+ << PriorityClassToString(priority_class) << "("
+ << priority_class << ")";
+ RTC_DLOG(LS_INFO) << "priority: " << PriorityToString(priority) << "("
+ << priority << ")";
}
}
~ScopedMMCSSRegistration() {
if (Succeeded()) {
// Deregister with MMCSS.
- RTC_DLOG(INFO) << "~ScopedMMCSSRegistration";
+ RTC_DLOG(LS_INFO) << "~ScopedMMCSSRegistration";
AvRevertMmThreadCharacteristics(mmcss_handle_);
}
}
diff --git a/modules/utility/source/jvm_android.cc b/modules/utility/source/jvm_android.cc
index 8e24daa..ee9930b 100644
--- a/modules/utility/source/jvm_android.cc
+++ b/modules/utility/source/jvm_android.cc
@@ -38,10 +38,10 @@
// stack. Consequently, we only look up all classes once in native WebRTC.
// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
void LoadClasses(JNIEnv* jni) {
- RTC_LOG(INFO) << "LoadClasses:";
+ RTC_LOG(LS_INFO) << "LoadClasses:";
for (auto& c : loaded_classes) {
jclass localRef = FindClass(jni, c.name);
- RTC_LOG(INFO) << "name: " << c.name;
+ RTC_LOG(LS_INFO) << "name: " << c.name;
CHECK_EXCEPTION(jni) << "Error during FindClass: " << c.name;
RTC_CHECK(localRef) << c.name;
jclass globalRef = reinterpret_cast<jclass>(jni->NewGlobalRef(localRef));
@@ -69,12 +69,12 @@
// JvmThreadConnector implementation.
JvmThreadConnector::JvmThreadConnector() : attached_(false) {
- RTC_LOG(INFO) << "JvmThreadConnector::ctor";
+ RTC_LOG(LS_INFO) << "JvmThreadConnector::ctor";
JavaVM* jvm = JVM::GetInstance()->jvm();
RTC_CHECK(jvm);
JNIEnv* jni = GetEnv(jvm);
if (!jni) {
- RTC_LOG(INFO) << "Attaching thread to JVM";
+ RTC_LOG(LS_INFO) << "Attaching thread to JVM";
JNIEnv* env = nullptr;
jint ret = jvm->AttachCurrentThread(&env, nullptr);
attached_ = (ret == JNI_OK);
@@ -82,10 +82,10 @@
}
JvmThreadConnector::~JvmThreadConnector() {
- RTC_LOG(INFO) << "JvmThreadConnector::dtor";
+ RTC_LOG(LS_INFO) << "JvmThreadConnector::dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
if (attached_) {
- RTC_LOG(INFO) << "Detaching thread from JVM";
+ RTC_LOG(LS_INFO) << "Detaching thread from JVM";
jint res = JVM::GetInstance()->jvm()->DetachCurrentThread();
RTC_CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res;
}
@@ -94,11 +94,11 @@
// GlobalRef implementation.
GlobalRef::GlobalRef(JNIEnv* jni, jobject object)
: jni_(jni), j_object_(NewGlobalRef(jni, object)) {
- RTC_LOG(INFO) << "GlobalRef::ctor";
+ RTC_LOG(LS_INFO) << "GlobalRef::ctor";
}
GlobalRef::~GlobalRef() {
- RTC_LOG(INFO) << "GlobalRef::dtor";
+ RTC_LOG(LS_INFO) << "GlobalRef::dtor";
DeleteGlobalRef(jni_, j_object_);
}
@@ -131,11 +131,11 @@
// NativeRegistration implementation.
NativeRegistration::NativeRegistration(JNIEnv* jni, jclass clazz)
: JavaClass(jni, clazz), jni_(jni) {
- RTC_LOG(INFO) << "NativeRegistration::ctor";
+ RTC_LOG(LS_INFO) << "NativeRegistration::ctor";
}
NativeRegistration::~NativeRegistration() {
- RTC_LOG(INFO) << "NativeRegistration::dtor";
+ RTC_LOG(LS_INFO) << "NativeRegistration::dtor";
jni_->UnregisterNatives(j_class_);
CHECK_EXCEPTION(jni_) << "Error during UnregisterNatives";
}
@@ -143,7 +143,7 @@
std::unique_ptr<GlobalRef> NativeRegistration::NewObject(const char* name,
const char* signature,
...) {
- RTC_LOG(INFO) << "NativeRegistration::NewObject";
+ RTC_LOG(LS_INFO) << "NativeRegistration::NewObject";
va_list args;
va_start(args, signature);
jobject obj = jni_->NewObjectV(
@@ -181,11 +181,11 @@
// JNIEnvironment implementation.
JNIEnvironment::JNIEnvironment(JNIEnv* jni) : jni_(jni) {
- RTC_LOG(INFO) << "JNIEnvironment::ctor";
+ RTC_LOG(LS_INFO) << "JNIEnvironment::ctor";
}
JNIEnvironment::~JNIEnvironment() {
- RTC_LOG(INFO) << "JNIEnvironment::dtor";
+ RTC_LOG(LS_INFO) << "JNIEnvironment::dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
}
@@ -193,7 +193,7 @@
const char* name,
const JNINativeMethod* methods,
int num_methods) {
- RTC_LOG(INFO) << "JNIEnvironment::RegisterNatives: " << name;
+ RTC_LOG(LS_INFO) << "JNIEnvironment::RegisterNatives: " << name;
RTC_DCHECK(thread_checker_.IsCurrent());
jclass clazz = LookUpClass(name);
jni_->RegisterNatives(clazz, methods, num_methods);
@@ -216,7 +216,7 @@
// static
void JVM::Initialize(JavaVM* jvm) {
- RTC_LOG(INFO) << "JVM::Initialize";
+ RTC_LOG(LS_INFO) << "JVM::Initialize";
RTC_CHECK(!g_jvm);
g_jvm = new JVM(jvm);
}
@@ -234,7 +234,7 @@
// static
void JVM::Uninitialize() {
- RTC_LOG(INFO) << "JVM::Uninitialize";
+ RTC_LOG(LS_INFO) << "JVM::Uninitialize";
RTC_DCHECK(g_jvm);
delete g_jvm;
g_jvm = nullptr;
@@ -247,19 +247,19 @@
}
JVM::JVM(JavaVM* jvm) : jvm_(jvm) {
- RTC_LOG(INFO) << "JVM::JVM";
+ RTC_LOG(LS_INFO) << "JVM::JVM";
RTC_CHECK(jni()) << "AttachCurrentThread() must be called on this thread.";
LoadClasses(jni());
}
JVM::~JVM() {
- RTC_LOG(INFO) << "JVM::~JVM";
+ RTC_LOG(LS_INFO) << "JVM::~JVM";
RTC_DCHECK(thread_checker_.IsCurrent());
FreeClassReferences(jni());
}
std::unique_ptr<JNIEnvironment> JVM::environment() {
- RTC_LOG(INFO) << "JVM::environment";
+ RTC_LOG(LS_INFO) << "JVM::environment";
;
// The JNIEnv is used for thread-local storage. For this reason, we cannot
// share a JNIEnv between threads. If a piece of code has no other way to get
@@ -276,7 +276,7 @@
}
JavaClass JVM::GetClass(const char* name) {
- RTC_LOG(INFO) << "JVM::GetClass: " << name;
+ RTC_LOG(LS_INFO) << "JVM::GetClass: " << name;
RTC_DCHECK(thread_checker_.IsCurrent());
return JavaClass(jni(), LookUpClass(name));
}
diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
index 887aa58..4256926 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
@@ -586,8 +586,8 @@
// theoretically use all available reference buffers.
encoder_params.iNumRefFrame = encoder_params.iTemporalLayerNum - 1;
}
- RTC_LOG(INFO) << "OpenH264 version is " << OPENH264_MAJOR << "."
- << OPENH264_MINOR;
+ RTC_LOG(LS_INFO) << "OpenH264 version is " << OPENH264_MAJOR << "."
+ << OPENH264_MINOR;
switch (packetization_mode_) {
case H264PacketizationMode::SingleNalUnit:
// Limit the size of the packets produced.
@@ -596,8 +596,8 @@
SM_SIZELIMITED_SLICE;
encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceSizeConstraint =
static_cast<unsigned int>(max_payload_size_);
- RTC_LOG(INFO) << "Encoder is configured with NALU constraint: "
- << max_payload_size_ << " bytes";
+ RTC_LOG(LS_INFO) << "Encoder is configured with NALU constraint: "
+ << max_payload_size_ << " bytes";
break;
case H264PacketizationMode::NonInterleaved:
// When uiSliceMode = SM_FIXEDSLCNUM_SLICE, uiSliceNum = 0 means auto
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
index 88b840c..0d6fdfa 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
@@ -1293,8 +1293,8 @@
<< "Not all raw images had the right format!";
return;
}
- RTC_LOG(INFO) << "Updating vp8 encoder pixel format to "
- << (fmt == VPX_IMG_FMT_NV12 ? "NV12" : "I420");
+ RTC_LOG(LS_INFO) << "Updating vp8 encoder pixel format to "
+ << (fmt == VPX_IMG_FMT_NV12 ? "NV12" : "I420");
for (size_t i = 0; i < raw_images_.size(); ++i) {
vpx_image_t& img = raw_images_[i];
auto d_w = img.d_w;
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
index ffbaaaa..c32673d 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
@@ -1930,8 +1930,8 @@
raw_ = libvpx_->img_wrap(nullptr, fmt, codec_.width, codec_.height, 1,
nullptr);
} else if (raw_->fmt != fmt) {
- RTC_LOG(INFO) << "Switching VP9 encoder pixel format to "
- << (fmt == VPX_IMG_FMT_NV12 ? "NV12" : "I420");
+ RTC_LOG(LS_INFO) << "Switching VP9 encoder pixel format to "
+ << (fmt == VPX_IMG_FMT_NV12 ? "NV12" : "I420");
libvpx_->img_free(raw_);
raw_ = libvpx_->img_wrap(nullptr, fmt, codec_.width, codec_.height, 1,
nullptr);
diff --git a/modules/video_coding/utility/ivf_file_reader.cc b/modules/video_coding/utility/ivf_file_reader.cc
index f326c8c..6392570 100644
--- a/modules/video_coding/utility/ivf_file_reader.cc
+++ b/modules/video_coding/utility/ivf_file_reader.cc
@@ -104,10 +104,10 @@
has_error_ = false;
const char* codec_name = CodecTypeToPayloadString(codec_type_);
- RTC_LOG(INFO) << "Opened IVF file with codec data of type " << codec_name
- << " at resolution " << width_ << " x " << height_ << ", using "
- << (using_capture_timestamps_ ? "1" : "90")
- << "kHz clock resolution.";
+ RTC_LOG(LS_INFO) << "Opened IVF file with codec data of type " << codec_name
+ << " at resolution " << width_ << " x " << height_
+ << ", using " << (using_capture_timestamps_ ? "1" : "90")
+ << "kHz clock resolution.";
return true;
}
diff --git a/p2p/base/fake_ice_transport.h b/p2p/base/fake_ice_transport.h
index 8b52fe9..c053abd 100644
--- a/p2p/base/fake_ice_transport.h
+++ b/p2p/base/fake_ice_transport.h
@@ -364,7 +364,7 @@
if (writable_ == writable) {
return;
}
- RTC_LOG(INFO) << "Change writable_ to " << writable;
+ RTC_LOG(LS_INFO) << "Change writable_ to " << writable;
writable_ = writable;
if (writable_) {
SignalReadyToSend(this);
diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc
index 5587a84..5c6b32c 100644
--- a/p2p/base/p2p_transport_channel.cc
+++ b/p2p/base/p2p_transport_channel.cc
@@ -1476,11 +1476,11 @@
// It is not legal to try to change any of the parameters of an existing
// connection; however, the other side can send a duplicate candidate.
if (!remote_candidate.IsEquivalent(connection->remote_candidate())) {
- RTC_LOG(INFO) << "Attempt to change a remote candidate."
- " Existing remote candidate: "
- << connection->remote_candidate().ToSensitiveString()
- << "New remote candidate: "
- << remote_candidate.ToSensitiveString();
+ RTC_LOG(LS_INFO) << "Attempt to change a remote candidate."
+ " Existing remote candidate: "
+ << connection->remote_candidate().ToSensitiveString()
+ << "New remote candidate: "
+ << remote_candidate.ToSensitiveString();
}
return false;
}
@@ -1532,8 +1532,8 @@
size_t i = 0;
while (i < remote_candidates_.size()) {
if (remote_candidates_[i].generation() < remote_candidate.generation()) {
- RTC_LOG(INFO) << "Pruning candidate from old generation: "
- << remote_candidates_[i].address().ToSensitiveString();
+ RTC_LOG(LS_INFO) << "Pruning candidate from old generation: "
+ << remote_candidates_[i].address().ToSensitiveString();
remote_candidates_.erase(remote_candidates_.begin() + i);
} else {
i += 1;
@@ -1542,8 +1542,8 @@
// Make sure this candidate is not a duplicate.
if (IsDuplicateRemoteCandidate(remote_candidate)) {
- RTC_LOG(INFO) << "Duplicate candidate: "
- << remote_candidate.ToSensitiveString();
+ RTC_LOG(LS_INFO) << "Duplicate candidate: "
+ << remote_candidate.ToSensitiveString();
return;
}
@@ -2147,8 +2147,8 @@
pruned_ports_.erase(
std::remove(pruned_ports_.begin(), pruned_ports_.end(), port),
pruned_ports_.end());
- RTC_LOG(INFO) << "Removed port because it is destroyed: " << ports_.size()
- << " remaining";
+ RTC_LOG(LS_INFO) << "Removed port because it is destroyed: " << ports_.size()
+ << " remaining";
}
void P2PTransportChannel::OnPortsPruned(
@@ -2157,8 +2157,8 @@
RTC_DCHECK_RUN_ON(network_thread_);
for (PortInterface* port : ports) {
if (PrunePort(port)) {
- RTC_LOG(INFO) << "Removed port: " << port->ToString() << " "
- << ports_.size() << " remaining";
+ RTC_LOG(LS_INFO) << "Removed port: " << port->ToString() << " "
+ << ports_.size() << " remaining";
}
}
}
diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc
index 06dafe8..dc32463 100644
--- a/p2p/base/port_unittest.cc
+++ b/p2p/base/port_unittest.cc
@@ -350,7 +350,7 @@
void OnDestroyed(Connection* conn) {
ASSERT_EQ(conn_, conn);
- RTC_LOG(INFO) << "OnDestroy connection " << conn << " deleted";
+ RTC_LOG(LS_INFO) << "OnDestroy connection " << conn << " deleted";
conn_ = NULL;
// When the connection is destroyed, also clear these fields so future
// connections are possible.
diff --git a/p2p/client/basic_port_allocator.cc b/p2p/client/basic_port_allocator.cc
index 5a26934..15f59bd 100644
--- a/p2p/client/basic_port_allocator.cc
+++ b/p2p/client/basic_port_allocator.cc
@@ -103,9 +103,9 @@
if (start_to_remove == networks->end()) {
return;
}
- RTC_LOG(INFO) << "Filtered out " << filter.description << " networks:";
+ RTC_LOG(LS_INFO) << "Filtered out " << filter.description << " networks:";
for (auto it = start_to_remove; it != networks->end(); ++it) {
- RTC_LOG(INFO) << (*it)->ToString();
+ RTC_LOG(LS_INFO) << (*it)->ToString();
}
networks->erase(start_to_remove, networks->end());
}
diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc
index e844264..e40c7b5 100644
--- a/pc/jsep_transport.cc
+++ b/pc/jsep_transport.cc
@@ -365,7 +365,7 @@
void JsepTransport::SetActiveResetSrtpParams(bool active_reset_srtp_params) {
RTC_DCHECK_RUN_ON(network_thread_);
if (dtls_srtp_transport_) {
- RTC_LOG(INFO)
+ RTC_LOG(LS_INFO)
<< "Setting active_reset_srtp_params of DtlsSrtpTransport to: "
<< active_reset_srtp_params;
dtls_srtp_transport_->SetActiveResetSrtpParams(active_reset_srtp_params);
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 475389c..24154fd 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -366,7 +366,7 @@
return;
}
RTC_DCHECK_RUN_ON(network_thread_);
- RTC_LOG(INFO)
+ RTC_LOG(LS_INFO)
<< "Updating the active_reset_srtp_params for JsepTransportController: "
<< active_reset_srtp_params;
active_reset_srtp_params_ = active_reset_srtp_params;
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index 4fbbb5c..9bfef80 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -89,6 +89,8 @@
LS_WARNING,
LS_ERROR,
LS_NONE,
+ // Compatibility aliases, to be deleted.
+ // TODO(bugs.webrtc.org/13362): Remove usage and delete.
INFO = LS_INFO,
WARNING = LS_WARNING,
LERROR = LS_ERROR
diff --git a/rtc_base/network_unittest.cc b/rtc_base/network_unittest.cc
index 4d94e7c..87df0d3 100644
--- a/rtc_base/network_unittest.cc
+++ b/rtc_base/network_unittest.cc
@@ -128,11 +128,11 @@
bool SameNameAndPrefix(const rtc::Network& a, const rtc::Network& b) {
if (a.name() != b.name()) {
- RTC_LOG(INFO) << "Different interface names.";
+ RTC_LOG(LS_INFO) << "Different interface names.";
return false;
}
if (a.prefix_length() != b.prefix_length() || a.prefix() != b.prefix()) {
- RTC_LOG(INFO) << "Different IP prefixes.";
+ RTC_LOG(LS_INFO) << "Different IP prefixes.";
return false;
}
return true;
diff --git a/rtc_base/socket_adapters.cc b/rtc_base/socket_adapters.cc
index c6cd196..abe5e30 100644
--- a/rtc_base/socket_adapters.cc
+++ b/rtc_base/socket_adapters.cc
@@ -105,7 +105,7 @@
buffer_size_ - data_len_, nullptr);
if (len < 0) {
// TODO: Do something better like forwarding the error to the user.
- RTC_LOG_ERR(INFO) << "Recv";
+ RTC_LOG_ERR(LS_INFO) << "Recv";
return;
}
diff --git a/rtc_base/win/scoped_com_initializer.cc b/rtc_base/win/scoped_com_initializer.cc
index 81079fb..e791adc 100644
--- a/rtc_base/win/scoped_com_initializer.cc
+++ b/rtc_base/win/scoped_com_initializer.cc
@@ -16,13 +16,13 @@
namespace webrtc {
ScopedCOMInitializer::ScopedCOMInitializer() {
- RTC_DLOG(INFO) << "Single-Threaded Apartment (STA) COM thread";
+ RTC_DLOG(LS_INFO) << "Single-Threaded Apartment (STA) COM thread";
Initialize(COINIT_APARTMENTTHREADED);
}
// Constructor for MTA initialization.
ScopedCOMInitializer::ScopedCOMInitializer(SelectMTA mta) {
- RTC_DLOG(INFO) << "Multi-Threaded Apartment (MTA) COM thread";
+ RTC_DLOG(LS_INFO) << "Multi-Threaded Apartment (MTA) COM thread";
Initialize(COINIT_MULTITHREADED);
}
@@ -46,7 +46,7 @@
// successful call to CoInitializeEx, including any call that returns
// S_FALSE, must be balanced by a corresponding call to CoUninitialize.
if (hr_ == S_OK) {
- RTC_DLOG(INFO)
+ RTC_DLOG(LS_INFO)
<< "The COM library was initialized successfully on this thread";
} else if (hr_ == S_FALSE) {
RTC_DLOG(WARNING)
diff --git a/rtc_base/win/windows_version_unittest.cc b/rtc_base/win/windows_version_unittest.cc
index 9e582e5..e1cd920 100644
--- a/rtc_base/win/windows_version_unittest.cc
+++ b/rtc_base/win/windows_version_unittest.cc
@@ -18,11 +18,11 @@
namespace {
void MethodSupportedOnWin10AndLater() {
- RTC_DLOG(INFO) << "MethodSupportedOnWin10AndLater";
+ RTC_DLOG(LS_INFO) << "MethodSupportedOnWin10AndLater";
}
void MethodNotSupportedOnWin10AndLater() {
- RTC_DLOG(INFO) << "MethodNotSupportedOnWin10AndLater";
+ RTC_DLOG(LS_INFO) << "MethodNotSupportedOnWin10AndLater";
}
// Use global GetVersion() and use it in a way a user would typically use it
@@ -39,7 +39,7 @@
TEST(WindowsVersion, ProcessorModelName) {
std::string name = OSInfo::GetInstance()->processor_model_name();
EXPECT_FALSE(name.empty());
- RTC_DLOG(INFO) << "processor_model_name: " << name;
+ RTC_DLOG(LS_INFO) << "processor_model_name: " << name;
}
} // namespace
diff --git a/rtc_tools/converter/yuv_to_ivf_converter.cc b/rtc_tools/converter/yuv_to_ivf_converter.cc
index e4a1e12..d00457d 100644
--- a/rtc_tools/converter/yuv_to_ivf_converter.cc
+++ b/rtc_tools/converter/yuv_to_ivf_converter.cc
@@ -81,8 +81,8 @@
received_frames_count_++;
RTC_CHECK_LE(received_frames_count_, expected_frames_count_);
if (received_frames_count_ % kFrameLogInterval == 0) {
- RTC_LOG(INFO) << received_frames_count_ << " out of "
- << expected_frames_count_ << " frames written";
+ RTC_LOG(LS_INFO) << received_frames_count_ << " out of "
+ << expected_frames_count_ << " frames written";
}
next_frame_written_.Set();
return Result(Result::Error::OK);
@@ -231,11 +231,11 @@
encoder.WaitNextFrameWritten(kMaxFrameEncodeWaitTimeoutMs);
if ((i + 1) % kFrameLogInterval == 0) {
- RTC_LOG(INFO) << i + 1 << " out of " << frames_count
- << " frames are sent for encoding";
+ RTC_LOG(LS_INFO) << i + 1 << " out of " << frames_count
+ << " frames are sent for encoding";
}
}
- RTC_LOG(INFO) << "All " << frames_count << " frame are sent for encoding";
+ RTC_LOG(LS_INFO) << "All " << frames_count << " frame are sent for encoding";
}
} // namespace
diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.cc b/sdk/android/native_api/audio_device_module/audio_device_android.cc
index 8a57e4a..9a42361 100644
--- a/sdk/android/native_api/audio_device_module/audio_device_android.cc
+++ b/sdk/android/native_api/audio_device_module/audio_device_android.cc
@@ -57,7 +57,7 @@
rtc::scoped_refptr<AudioDeviceModule> CreateAAudioAudioDeviceModule(
JNIEnv* env,
jobject application_context) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
AudioParameters input_parameters;
AudioParameters output_parameters;
@@ -76,7 +76,7 @@
rtc::scoped_refptr<AudioDeviceModule> CreateJavaAudioDeviceModule(
JNIEnv* env,
jobject application_context) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
const JavaParamRef<jobject> j_context(application_context);
const ScopedJavaLocalRef<jobject> j_audio_manager =
@@ -104,7 +104,7 @@
rtc::scoped_refptr<AudioDeviceModule> CreateOpenSLESAudioDeviceModule(
JNIEnv* env,
jobject application_context) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
AudioParameters input_parameters;
AudioParameters output_parameters;
@@ -127,7 +127,7 @@
rtc::scoped_refptr<AudioDeviceModule>
CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env,
jobject application_context) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
// Get default audio input/output parameters.
const JavaParamRef<jobject> j_context(application_context);
const ScopedJavaLocalRef<jobject> j_audio_manager =
diff --git a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
index 4cd62bc..fd988a5 100644
--- a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
+++ b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
@@ -77,10 +77,10 @@
TEST(PeerConnectionFactoryTest, NativeToJavaPeerConnectionFactory) {
JNIEnv* jni = AttachCurrentThreadIfNeeded();
- RTC_LOG(INFO) << "Initializing java peer connection factory.";
+ RTC_LOG(LS_INFO) << "Initializing java peer connection factory.";
jni::Java_PeerConnectionFactoryInitializationHelper_initializeFactoryForTests(
jni);
- RTC_LOG(INFO) << "Java peer connection factory initialized.";
+ RTC_LOG(LS_INFO) << "Java peer connection factory initialized.";
auto socket_server = std::make_unique<rtc::PhysicalSocketServer>();
@@ -105,7 +105,7 @@
jni, factory, std::move(socket_server), std::move(network_thread),
std::move(worker_thread), std::move(signaling_thread));
- RTC_LOG(INFO) << java_factory;
+ RTC_LOG(LS_INFO) << java_factory;
EXPECT_NE(java_factory, nullptr);
}
diff --git a/sdk/android/src/jni/audio_device/aaudio_player.cc b/sdk/android/src/jni/audio_device/aaudio_player.cc
index da68b83..29bcfae 100644
--- a/sdk/android/src/jni/audio_device/aaudio_player.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_player.cc
@@ -28,19 +28,19 @@
AAudioPlayer::AAudioPlayer(const AudioParameters& audio_parameters)
: main_thread_(rtc::Thread::Current()),
aaudio_(audio_parameters, AAUDIO_DIRECTION_OUTPUT, this) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
thread_checker_aaudio_.Detach();
}
AAudioPlayer::~AAudioPlayer() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
Terminate();
- RTC_LOG(INFO) << "#detected underruns: " << underrun_count_;
+ RTC_LOG(LS_INFO) << "#detected underruns: " << underrun_count_;
}
int AAudioPlayer::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
if (aaudio_.audio_parameters().channels() == 2) {
RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
@@ -49,14 +49,14 @@
}
int AAudioPlayer::Terminate() {
- RTC_LOG(INFO) << "Terminate";
+ RTC_LOG(LS_INFO) << "Terminate";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
StopPlayout();
return 0;
}
int AAudioPlayer::InitPlayout() {
- RTC_LOG(INFO) << "InitPlayout";
+ RTC_LOG(LS_INFO) << "InitPlayout";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
RTC_DCHECK(!initialized_);
RTC_DCHECK(!playing_);
@@ -73,7 +73,7 @@
}
int AAudioPlayer::StartPlayout() {
- RTC_LOG(INFO) << "StartPlayout";
+ RTC_LOG(LS_INFO) << "StartPlayout";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
RTC_DCHECK(!playing_);
if (!initialized_) {
@@ -94,7 +94,7 @@
}
int AAudioPlayer::StopPlayout() {
- RTC_LOG(INFO) << "StopPlayout";
+ RTC_LOG(LS_INFO) << "StopPlayout";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
if (!initialized_ || !playing_) {
return 0;
@@ -115,7 +115,7 @@
}
void AAudioPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_DLOG(INFO) << "AttachAudioBuffer";
+ RTC_DLOG(LS_INFO) << "AttachAudioBuffer";
RTC_DCHECK_RUN_ON(&main_thread_checker_);
audio_device_buffer_ = audioBuffer;
const AudioParameters audio_parameters = aaudio_.audio_parameters();
@@ -173,9 +173,9 @@
// Log device id in first data callback to ensure that a valid device is
// utilized.
if (first_data_callback_) {
- RTC_LOG(INFO) << "--- First output data callback: "
- "device id="
- << aaudio_.device_id();
+ RTC_LOG(LS_INFO) << "--- First output data callback: "
+ "device id="
+ << aaudio_.device_id();
first_data_callback_ = false;
}
@@ -195,8 +195,8 @@
latency_millis_ = aaudio_.EstimateLatencyMillis();
// TODO(henrika): use for development only.
if (aaudio_.frames_written() % (1000 * aaudio_.frames_per_burst()) == 0) {
- RTC_DLOG(INFO) << "output latency: " << latency_millis_
- << ", num_frames: " << num_frames;
+ RTC_DLOG(LS_INFO) << "output latency: " << latency_millis_
+ << ", num_frames: " << num_frames;
}
// Read audio data from the WebRTC source using the FineAudioBuffer object
@@ -231,7 +231,7 @@
void AAudioPlayer::HandleStreamDisconnected() {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
- RTC_DLOG(INFO) << "HandleStreamDisconnected";
+ RTC_DLOG(LS_INFO) << "HandleStreamDisconnected";
if (!initialized_ || !playing_) {
return;
}
diff --git a/sdk/android/src/jni/audio_device/aaudio_recorder.cc b/sdk/android/src/jni/audio_device/aaudio_recorder.cc
index 8a4c353..8ab097d 100644
--- a/sdk/android/src/jni/audio_device/aaudio_recorder.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_recorder.cc
@@ -29,19 +29,19 @@
AAudioRecorder::AAudioRecorder(const AudioParameters& audio_parameters)
: main_thread_(rtc::Thread::Current()),
aaudio_(audio_parameters, AAUDIO_DIRECTION_INPUT, this) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
thread_checker_aaudio_.Detach();
}
AAudioRecorder::~AAudioRecorder() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
Terminate();
- RTC_LOG(INFO) << "detected owerflows: " << overflow_count_;
+ RTC_LOG(LS_INFO) << "detected owerflows: " << overflow_count_;
}
int AAudioRecorder::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK(thread_checker_.IsCurrent());
if (aaudio_.audio_parameters().channels() == 2) {
RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
@@ -50,14 +50,14 @@
}
int AAudioRecorder::Terminate() {
- RTC_LOG(INFO) << "Terminate";
+ RTC_LOG(LS_INFO) << "Terminate";
RTC_DCHECK(thread_checker_.IsCurrent());
StopRecording();
return 0;
}
int AAudioRecorder::InitRecording() {
- RTC_LOG(INFO) << "InitRecording";
+ RTC_LOG(LS_INFO) << "InitRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!initialized_);
RTC_DCHECK(!recording_);
@@ -73,7 +73,7 @@
}
int AAudioRecorder::StartRecording() {
- RTC_LOG(INFO) << "StartRecording";
+ RTC_LOG(LS_INFO) << "StartRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(initialized_);
RTC_DCHECK(!recording_);
@@ -90,7 +90,7 @@
}
int AAudioRecorder::StopRecording() {
- RTC_LOG(INFO) << "StopRecording";
+ RTC_LOG(LS_INFO) << "StopRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
if (!initialized_ || !recording_) {
return 0;
@@ -109,7 +109,7 @@
}
void AAudioRecorder::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(INFO) << "AttachAudioBuffer";
+ RTC_LOG(LS_INFO) << "AttachAudioBuffer";
RTC_DCHECK(thread_checker_.IsCurrent());
audio_device_buffer_ = audioBuffer;
const AudioParameters audio_parameters = aaudio_.audio_parameters();
@@ -131,13 +131,13 @@
}
int AAudioRecorder::EnableBuiltInAEC(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInAEC: " << enable;
+ RTC_LOG(LS_INFO) << "EnableBuiltInAEC: " << enable;
RTC_LOG(LS_ERROR) << "Not implemented";
return -1;
}
int AAudioRecorder::EnableBuiltInNS(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInNS: " << enable;
+ RTC_LOG(LS_INFO) << "EnableBuiltInNS: " << enable;
RTC_LOG(LS_ERROR) << "Not implemented";
return -1;
}
@@ -165,14 +165,14 @@
int32_t num_frames) {
// TODO(henrika): figure out why we sometimes hit this one.
// RTC_DCHECK(thread_checker_aaudio_.IsCurrent());
- // RTC_LOG(INFO) << "OnDataCallback: " << num_frames;
+ // RTC_LOG(LS_INFO) << "OnDataCallback: " << num_frames;
// Drain the input buffer at first callback to ensure that it does not
// contain any old data. Will also ensure that the lowest possible latency
// is obtained.
if (first_data_callback_) {
- RTC_LOG(INFO) << "--- First input data callback: "
- "device id="
- << aaudio_.device_id();
+ RTC_LOG(LS_INFO) << "--- First input data callback: "
+ "device id="
+ << aaudio_.device_id();
aaudio_.ClearInputStream(audio_data, num_frames);
first_data_callback_ = false;
}
@@ -188,8 +188,8 @@
latency_millis_ = aaudio_.EstimateLatencyMillis();
// TODO(henrika): use for development only.
if (aaudio_.frames_read() % (1000 * aaudio_.frames_per_burst()) == 0) {
- RTC_DLOG(INFO) << "input latency: " << latency_millis_
- << ", num_frames: " << num_frames;
+ RTC_DLOG(LS_INFO) << "input latency: " << latency_millis_
+ << ", num_frames: " << num_frames;
}
// Copy recorded audio in `audio_data` to the WebRTC sink using the
// FineAudioBuffer object.
@@ -215,7 +215,7 @@
void AAudioRecorder::HandleStreamDisconnected() {
RTC_DCHECK_RUN_ON(&thread_checker_);
- RTC_LOG(INFO) << "HandleStreamDisconnected";
+ RTC_LOG(LS_INFO) << "HandleStreamDisconnected";
if (!initialized_ || !recording_) {
return;
}
diff --git a/sdk/android/src/jni/audio_device/aaudio_wrapper.cc b/sdk/android/src/jni/audio_device/aaudio_wrapper.cc
index 8ce720f..8fc8e78 100644
--- a/sdk/android/src/jni/audio_device/aaudio_wrapper.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_wrapper.cc
@@ -137,20 +137,20 @@
: audio_parameters_(audio_parameters),
direction_(direction),
observer_(observer) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
RTC_DCHECK(observer_);
aaudio_thread_checker_.Detach();
- RTC_LOG(INFO) << audio_parameters_.ToString();
+ RTC_LOG(LS_INFO) << audio_parameters_.ToString();
}
AAudioWrapper::~AAudioWrapper() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!stream_);
}
bool AAudioWrapper::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
RTC_DCHECK(thread_checker_.IsCurrent());
// Creates a stream builder which can be used to open an audio stream.
ScopedStreamBuilder builder;
@@ -174,7 +174,7 @@
}
bool AAudioWrapper::Start() {
- RTC_LOG(INFO) << "Start";
+ RTC_LOG(LS_INFO) << "Start";
RTC_DCHECK(thread_checker_.IsCurrent());
// TODO(henrika): this state check might not be needed.
aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
@@ -190,7 +190,7 @@
}
bool AAudioWrapper::Stop() {
- RTC_LOG(INFO) << "Stop: " << DirectionToString(direction());
+ RTC_LOG(LS_INFO) << "Stop: " << DirectionToString(direction());
RTC_DCHECK(thread_checker_.IsCurrent());
// Asynchronous request for the stream to stop.
RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
@@ -240,7 +240,7 @@
// Returns new buffer size or a negative error value if buffer size could not
// be increased.
bool AAudioWrapper::IncreaseOutputBufferSize() {
- RTC_LOG(INFO) << "IncreaseBufferSize";
+ RTC_LOG(LS_INFO) << "IncreaseBufferSize";
RTC_DCHECK(stream_);
RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
@@ -255,20 +255,20 @@
<< ") is higher than max: " << max_buffer_size;
return false;
}
- RTC_LOG(INFO) << "Updating buffer size to: " << buffer_size
- << " (max=" << max_buffer_size << ")";
+ RTC_LOG(LS_INFO) << "Updating buffer size to: " << buffer_size
+ << " (max=" << max_buffer_size << ")";
buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
if (buffer_size < 0) {
RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
<< AAudio_convertResultToText(buffer_size);
return false;
}
- RTC_LOG(INFO) << "Buffer size changed to: " << buffer_size;
+ RTC_LOG(LS_INFO) << "Buffer size changed to: " << buffer_size;
return true;
}
void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
- RTC_LOG(INFO) << "ClearInputStream";
+ RTC_LOG(LS_INFO) << "ClearInputStream";
RTC_DCHECK(stream_);
RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
@@ -357,7 +357,7 @@
}
void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
- RTC_LOG(INFO) << "SetStreamConfiguration";
+ RTC_LOG(LS_INFO) << "SetStreamConfiguration";
RTC_DCHECK(builder);
RTC_DCHECK(thread_checker_.IsCurrent());
// Request usage of default primary output/input device.
@@ -390,7 +390,7 @@
}
bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
- RTC_LOG(INFO) << "OpenStream";
+ RTC_LOG(LS_INFO) << "OpenStream";
RTC_DCHECK(builder);
AAudioStream* stream = nullptr;
RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
@@ -400,7 +400,7 @@
}
void AAudioWrapper::CloseStream() {
- RTC_LOG(INFO) << "CloseStream";
+ RTC_LOG(LS_INFO) << "CloseStream";
RTC_DCHECK(stream_);
LOG_ON_ERROR(AAudioStream_close(stream_));
stream_ = nullptr;
@@ -419,16 +419,16 @@
ss << ", direction=" << DirectionToString(direction());
ss << ", device id=" << AAudioStream_getDeviceId(stream_);
ss << ", frames per callback=" << frames_per_callback();
- RTC_LOG(INFO) << ss.str();
+ RTC_LOG(LS_INFO) << ss.str();
}
void AAudioWrapper::LogStreamState() {
- RTC_LOG(INFO) << "AAudio stream state: "
- << AAudio_convertStreamStateToText(stream_state());
+ RTC_LOG(LS_INFO) << "AAudio stream state: "
+ << AAudio_convertStreamStateToText(stream_state());
}
bool AAudioWrapper::VerifyStreamConfiguration() {
- RTC_LOG(INFO) << "VerifyStreamConfiguration";
+ RTC_LOG(LS_INFO) << "VerifyStreamConfiguration";
RTC_DCHECK(stream_);
// TODO(henrika): should we verify device ID as well?
if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
@@ -466,16 +466,16 @@
}
bool AAudioWrapper::OptimizeBuffers() {
- RTC_LOG(INFO) << "OptimizeBuffers";
+ RTC_LOG(LS_INFO) << "OptimizeBuffers";
RTC_DCHECK(stream_);
// Maximum number of frames that can be filled without blocking.
- RTC_LOG(INFO) << "max buffer capacity in frames: "
- << buffer_capacity_in_frames();
+ RTC_LOG(LS_INFO) << "max buffer capacity in frames: "
+ << buffer_capacity_in_frames();
// Query the number of frames that the application should read or write at
// one time for optimal performance.
int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
- RTC_LOG(INFO) << "frames per burst for optimal performance: "
- << frames_per_burst;
+ RTC_LOG(LS_INFO) << "frames per burst for optimal performance: "
+ << frames_per_burst;
frames_per_burst_ = frames_per_burst;
if (direction() == AAUDIO_DIRECTION_INPUT) {
// There is no point in calling setBufferSizeInFrames() for input streams
@@ -492,7 +492,7 @@
return false;
}
// Maximum number of frames that can be filled without blocking.
- RTC_LOG(INFO) << "buffer burst size in frames: " << buffer_size;
+ RTC_LOG(LS_INFO) << "buffer burst size in frames: " << buffer_size;
return true;
}
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 4c9c36b..21c644f 100644
--- a/sdk/android/src/jni/audio_device/audio_device_module.cc
+++ b/sdk/android/src/jni/audio_device/audio_device_module.cc
@@ -70,26 +70,26 @@
initialized_(false) {
RTC_CHECK(input_);
RTC_CHECK(output_);
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
thread_checker_.Detach();
}
- ~AndroidAudioDeviceModule() override { RTC_DLOG(INFO) << __FUNCTION__; }
+ ~AndroidAudioDeviceModule() override { RTC_DLOG(LS_INFO) << __FUNCTION__; }
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer* audioLayer) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
*audioLayer = audio_layer_;
return 0;
}
int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return audio_device_buffer_->RegisterAudioCallback(audioCallback);
}
int32_t Init() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.IsCurrent());
audio_device_buffer_ =
std::make_unique<AudioDeviceBuffer>(task_queue_factory_.get());
@@ -118,7 +118,7 @@
}
int32_t Terminate() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return 0;
RTC_DCHECK(thread_checker_.IsCurrent());
@@ -132,19 +132,19 @@
}
bool Initialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__ << ":" << initialized_;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ":" << initialized_;
return initialized_;
}
int16_t PlayoutDevices() override {
- RTC_DLOG(INFO) << __FUNCTION__;
- RTC_LOG(INFO) << "output: " << 1;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << "output: " << 1;
return 1;
}
int16_t RecordingDevices() override {
- RTC_DLOG(INFO) << __FUNCTION__;
- RTC_DLOG(INFO) << "output: " << 1;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << "output: " << 1;
return 1;
}
@@ -163,7 +163,7 @@
int32_t SetPlayoutDevice(uint16_t index) override {
// OK to use but it has no effect currently since device selection is
// done using Andoid APIs instead.
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << index << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
return 0;
}
@@ -175,7 +175,7 @@
int32_t SetRecordingDevice(uint16_t index) override {
// OK to use but it has no effect currently since device selection is
// done using Andoid APIs instead.
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << index << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
return 0;
}
@@ -185,66 +185,66 @@
}
int32_t PlayoutIsAvailable(bool* available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
*available = true;
- RTC_DLOG(INFO) << "output: " << *available;
+ RTC_DLOG(LS_INFO) << "output: " << *available;
return 0;
}
int32_t InitPlayout() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
if (PlayoutIsInitialized()) {
return 0;
}
int32_t result = output_->InitPlayout();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
bool PlayoutIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_->PlayoutIsInitialized();
}
int32_t RecordingIsAvailable(bool* available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
*available = true;
- RTC_DLOG(INFO) << "output: " << *available;
+ RTC_DLOG(LS_INFO) << "output: " << *available;
return 0;
}
int32_t InitRecording() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
if (RecordingIsInitialized()) {
return 0;
}
int32_t result = input_->InitRecording();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool RecordingIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return input_->RecordingIsInitialized();
}
int32_t StartPlayout() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
if (Playing()) {
return 0;
}
int32_t result = output_->StartPlayout();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
static_cast<int>(result == 0));
if (result == 0) {
@@ -256,34 +256,34 @@
}
int32_t StopPlayout() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
if (!Playing())
return 0;
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
audio_device_buffer_->StopPlayout();
int32_t result = output_->StopPlayout();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
bool Playing() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return output_->Playing();
}
int32_t StartRecording() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
if (Recording()) {
return 0;
}
int32_t result = input_->StartRecording();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
static_cast<int>(result == 0));
if (result == 0) {
@@ -295,74 +295,74 @@
}
int32_t StopRecording() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
if (!Recording())
return 0;
audio_device_buffer_->StopRecording();
int32_t result = input_->StopRecording();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool Recording() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return input_->Recording();
}
int32_t InitSpeaker() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return initialized_ ? 0 : -1;
}
bool SpeakerIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return initialized_;
}
int32_t InitMicrophone() override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return initialized_ ? 0 : -1;
}
bool MicrophoneIsInitialized() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return initialized_;
}
int32_t SpeakerVolumeIsAvailable(bool* available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
*available = output_->SpeakerVolumeIsAvailable();
- RTC_DLOG(INFO) << "output: " << *available;
+ RTC_DLOG(LS_INFO) << "output: " << *available;
return 0;
}
int32_t SetSpeakerVolume(uint32_t volume) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
return output_->SetSpeakerVolume(volume);
}
int32_t SpeakerVolume(uint32_t* output_volume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
absl::optional<uint32_t> volume = output_->SpeakerVolume();
if (!volume)
return -1;
*output_volume = *volume;
- RTC_DLOG(INFO) << "output: " << *volume;
+ RTC_DLOG(LS_INFO) << "output: " << *volume;
return 0;
}
int32_t MaxSpeakerVolume(uint32_t* output_max_volume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
absl::optional<uint32_t> max_volume = output_->MaxSpeakerVolume();
@@ -373,7 +373,7 @@
}
int32_t MinSpeakerVolume(uint32_t* output_min_volume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return -1;
absl::optional<uint32_t> min_volume = output_->MinSpeakerVolume();
@@ -384,71 +384,71 @@
}
int32_t MicrophoneVolumeIsAvailable(bool* available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
*available = false;
- RTC_DLOG(INFO) << "output: " << *available;
+ RTC_DLOG(LS_INFO) << "output: " << *available;
return -1;
}
int32_t SetMicrophoneVolume(uint32_t volume) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << volume << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
RTC_CHECK_NOTREACHED();
}
int32_t MicrophoneVolume(uint32_t* volume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_CHECK_NOTREACHED();
}
int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_CHECK_NOTREACHED();
}
int32_t MinMicrophoneVolume(uint32_t* minVolume) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_CHECK_NOTREACHED();
}
int32_t SpeakerMuteIsAvailable(bool* available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_CHECK_NOTREACHED();
}
int32_t SetSpeakerMute(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK_NOTREACHED();
}
int32_t SpeakerMute(bool* enabled) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_CHECK_NOTREACHED();
}
int32_t MicrophoneMuteIsAvailable(bool* available) override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_CHECK_NOTREACHED();
}
int32_t SetMicrophoneMute(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK_NOTREACHED();
}
int32_t MicrophoneMute(bool* enabled) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
RTC_CHECK_NOTREACHED();
}
int32_t StereoPlayoutIsAvailable(bool* available) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
*available = is_stereo_playout_supported_;
- RTC_DLOG(INFO) << "output: " << *available;
+ RTC_DLOG(LS_INFO) << "output: " << *available;
return 0;
}
int32_t SetStereoPlayout(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
// Android does not support changes between mono and stero on the fly. The
// use of stereo or mono is determined by the audio layer. It is allowed
// to call this method if that same state is not modified.
@@ -461,21 +461,21 @@
}
int32_t StereoPlayout(bool* enabled) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
*enabled = is_stereo_playout_supported_;
- RTC_DLOG(INFO) << "output: " << *enabled;
+ RTC_DLOG(LS_INFO) << "output: " << *enabled;
return 0;
}
int32_t StereoRecordingIsAvailable(bool* available) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
*available = is_stereo_record_supported_;
- RTC_DLOG(INFO) << "output: " << *available;
+ RTC_DLOG(LS_INFO) << "output: " << *available;
return 0;
}
int32_t SetStereoRecording(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
// Android does not support changes between mono and stero on the fly. The
// use of stereo or mono is determined by the audio layer. It is allowed
// to call this method if that same state is not modified.
@@ -488,9 +488,9 @@
}
int32_t StereoRecording(bool* enabled) const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
*enabled = is_stereo_record_supported_;
- RTC_DLOG(INFO) << "output: " << *enabled;
+ RTC_DLOG(LS_INFO) << "output: " << *enabled;
return 0;
}
@@ -514,18 +514,18 @@
// a "Not Implemented" log will be filed. This non-perfect state will remain
// until I have added full support for audio effects based on OpenSL ES APIs.
bool BuiltInAECIsAvailable() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return false;
bool isAvailable = input_->IsAcousticEchoCancelerSupported();
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return isAvailable;
}
// Not implemented for any input device on Android.
bool BuiltInAGCIsAvailable() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
- RTC_DLOG(INFO) << "output: " << false;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << "output: " << false;
return false;
}
@@ -534,38 +534,38 @@
// TODO(henrika): add implementation for OpenSL ES based audio as well.
// In addition, see comments for BuiltInAECIsAvailable().
bool BuiltInNSIsAvailable() const override {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return false;
bool isAvailable = input_->IsNoiseSuppressorSupported();
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return isAvailable;
}
// TODO(henrika): add implementation for OpenSL ES based audio as well.
int32_t EnableBuiltInAEC(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
if (!initialized_)
return -1;
RTC_CHECK(BuiltInAECIsAvailable()) << "HW AEC is not available";
int32_t result = input_->EnableBuiltInAEC(enable);
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
return result;
}
int32_t EnableBuiltInAGC(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK_NOTREACHED();
}
// TODO(henrika): add implementation for OpenSL ES based audio as well.
int32_t EnableBuiltInNS(bool enable) override {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
if (!initialized_)
return -1;
RTC_CHECK(BuiltInNSIsAvailable()) << "HW NS is not available";
int32_t result = input_->EnableBuiltInNS(enable);
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
return result;
}
@@ -576,7 +576,7 @@
}
int32_t AttachAudioBuffer() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
output_->AttachAudioBuffer(audio_device_buffer_.get());
input_->AttachAudioBuffer(audio_device_buffer_.get());
return 0;
@@ -640,7 +640,7 @@
uint16_t playout_delay_ms,
std::unique_ptr<AudioInput> audio_input,
std::unique_ptr<AudioOutput> audio_output) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return rtc::make_ref_counted<AndroidAudioDeviceModule>(
audio_layer, is_stereo_playout_supported, is_stereo_record_supported,
playout_delay_ms, std::move(audio_input), std::move(audio_output));
diff --git a/sdk/android/src/jni/audio_device/audio_record_jni.cc b/sdk/android/src/jni/audio_device/audio_record_jni.cc
index 2739522..170c81a 100644
--- a/sdk/android/src/jni/audio_device/audio_record_jni.cc
+++ b/sdk/android/src/jni/audio_device/audio_record_jni.cc
@@ -38,7 +38,7 @@
~ScopedHistogramTimer() {
const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
- RTC_LOG(INFO) << histogram_name_ << ": " << life_time_ms;
+ RTC_LOG(LS_INFO) << histogram_name_ << ": " << life_time_ms;
}
private:
@@ -68,7 +68,7 @@
initialized_(false),
recording_(false),
audio_device_buffer_(nullptr) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
RTC_DCHECK(audio_parameters_.is_valid());
Java_WebRtcAudioRecord_setNativeAudioRecord(env, j_audio_record_,
jni::jlongFromPointer(this));
@@ -79,20 +79,20 @@
}
AudioRecordJni::~AudioRecordJni() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
Terminate();
}
int32_t AudioRecordJni::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
env_ = AttachCurrentThreadIfNeeded();
RTC_DCHECK(thread_checker_.IsCurrent());
return 0;
}
int32_t AudioRecordJni::Terminate() {
- RTC_LOG(INFO) << "Terminate";
+ RTC_LOG(LS_INFO) << "Terminate";
RTC_DCHECK(thread_checker_.IsCurrent());
StopRecording();
thread_checker_.Detach();
@@ -100,7 +100,7 @@
}
int32_t AudioRecordJni::InitRecording() {
- RTC_LOG(INFO) << "InitRecording";
+ RTC_LOG(LS_INFO) << "InitRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
if (initialized_) {
// Already initialized.
@@ -118,7 +118,7 @@
return -1;
}
frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
- RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_;
+ RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
frames_per_buffer_ * bytes_per_frame);
@@ -132,7 +132,7 @@
}
int32_t AudioRecordJni::StartRecording() {
- RTC_LOG(INFO) << "StartRecording";
+ RTC_LOG(LS_INFO) << "StartRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
if (recording_) {
// Already recording.
@@ -153,7 +153,7 @@
}
int32_t AudioRecordJni::StopRecording() {
- RTC_LOG(INFO) << "StopRecording";
+ RTC_LOG(LS_INFO) << "StopRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
if (!initialized_ || !recording_) {
return 0;
@@ -166,8 +166,9 @@
env_, j_audio_record_);
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.SourceMatchesRecordingSession",
session_was_ok);
- RTC_LOG(INFO) << "HISTOGRAM(WebRTC.Audio.SourceMatchesRecordingSession): "
- << session_was_ok;
+ RTC_LOG(LS_INFO)
+ << "HISTOGRAM(WebRTC.Audio.SourceMatchesRecordingSession): "
+ << session_was_ok;
}
if (!Java_WebRtcAudioRecord_stopRecording(env_, j_audio_record_)) {
RTC_LOG(LS_ERROR) << "StopRecording failed";
@@ -188,14 +189,14 @@
}
void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(INFO) << "AttachAudioBuffer";
+ RTC_LOG(LS_INFO) << "AttachAudioBuffer";
RTC_DCHECK(thread_checker_.IsCurrent());
audio_device_buffer_ = audioBuffer;
const int sample_rate_hz = audio_parameters_.sample_rate();
- RTC_LOG(INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
+ RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
const size_t channels = audio_parameters_.channels();
- RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")";
+ RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
audio_device_buffer_->SetRecordingChannels(channels);
}
@@ -212,7 +213,7 @@
}
int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInAEC(" << enable << ")";
+ RTC_LOG(LS_INFO) << "EnableBuiltInAEC(" << enable << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
return Java_WebRtcAudioRecord_enableBuiltInAEC(env_, j_audio_record_, enable)
? 0
@@ -220,7 +221,7 @@
}
int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
- RTC_LOG(INFO) << "EnableBuiltInNS(" << enable << ")";
+ RTC_LOG(LS_INFO) << "EnableBuiltInNS(" << enable << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
return Java_WebRtcAudioRecord_enableBuiltInNS(env_, j_audio_record_, enable)
? 0
@@ -231,12 +232,12 @@
JNIEnv* env,
const JavaParamRef<jobject>& j_caller,
const JavaParamRef<jobject>& byte_buffer) {
- RTC_LOG(INFO) << "OnCacheDirectBufferAddress";
+ RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!direct_buffer_address_);
direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer.obj());
jlong capacity = env->GetDirectBufferCapacity(byte_buffer.obj());
- RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
+ RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
}
@@ -257,7 +258,7 @@
// of `playDelayMs` and `recDelayMs`, hence the distributions does not matter.
audio_device_buffer_->SetVQEData(total_delay_ms_, 0);
if (audio_device_buffer_->DeliverRecordedData() == -1) {
- RTC_LOG(INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
+ RTC_LOG(LS_INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
}
}
diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.cc b/sdk/android/src/jni/audio_device/audio_track_jni.cc
index 85adee2..f2f22f9 100644
--- a/sdk/android/src/jni/audio_device/audio_track_jni.cc
+++ b/sdk/android/src/jni/audio_device/audio_track_jni.cc
@@ -44,7 +44,7 @@
initialized_(false),
playing_(false),
audio_device_buffer_(nullptr) {
- RTC_LOG(INFO) << "ctor";
+ RTC_LOG(LS_INFO) << "ctor";
RTC_DCHECK(audio_parameters_.is_valid());
Java_WebRtcAudioTrack_setNativeAudioTrack(env, j_audio_track_,
jni::jlongFromPointer(this));
@@ -55,20 +55,20 @@
}
AudioTrackJni::~AudioTrackJni() {
- RTC_LOG(INFO) << "dtor";
+ RTC_LOG(LS_INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
Terminate();
}
int32_t AudioTrackJni::Init() {
- RTC_LOG(INFO) << "Init";
+ RTC_LOG(LS_INFO) << "Init";
env_ = AttachCurrentThreadIfNeeded();
RTC_DCHECK(thread_checker_.IsCurrent());
return 0;
}
int32_t AudioTrackJni::Terminate() {
- RTC_LOG(INFO) << "Terminate";
+ RTC_LOG(LS_INFO) << "Terminate";
RTC_DCHECK(thread_checker_.IsCurrent());
StopPlayout();
thread_checker_.Detach();
@@ -76,7 +76,7 @@
}
int32_t AudioTrackJni::InitPlayout() {
- RTC_LOG(INFO) << "InitPlayout";
+ RTC_LOG(LS_INFO) << "InitPlayout";
RTC_DCHECK(thread_checker_.IsCurrent());
if (initialized_) {
// Already initialized.
@@ -126,7 +126,7 @@
}
int32_t AudioTrackJni::StartPlayout() {
- RTC_LOG(INFO) << "StartPlayout";
+ RTC_LOG(LS_INFO) << "StartPlayout";
RTC_DCHECK(thread_checker_.IsCurrent());
if (playing_) {
// Already playing.
@@ -146,7 +146,7 @@
}
int32_t AudioTrackJni::StopPlayout() {
- RTC_LOG(INFO) << "StopPlayout";
+ RTC_LOG(LS_INFO) << "StopPlayout";
RTC_DCHECK(thread_checker_.IsCurrent());
if (!initialized_ || !playing_) {
return 0;
@@ -185,7 +185,7 @@
}
int AudioTrackJni::SetSpeakerVolume(uint32_t volume) {
- RTC_LOG(INFO) << "SetSpeakerVolume(" << volume << ")";
+ RTC_LOG(LS_INFO) << "SetSpeakerVolume(" << volume << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
return Java_WebRtcAudioTrack_setStreamVolume(env_, j_audio_track_,
static_cast<int>(volume))
@@ -207,7 +207,7 @@
RTC_DCHECK(thread_checker_.IsCurrent());
const uint32_t volume =
Java_WebRtcAudioTrack_getStreamVolume(env_, j_audio_track_);
- RTC_LOG(INFO) << "SpeakerVolume: " << volume;
+ RTC_LOG(LS_INFO) << "SpeakerVolume: " << volume;
return volume;
}
@@ -217,30 +217,30 @@
// TODO(henrika): possibly add stereo support.
void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
- RTC_LOG(INFO) << "AttachAudioBuffer";
+ RTC_LOG(LS_INFO) << "AttachAudioBuffer";
RTC_DCHECK(thread_checker_.IsCurrent());
audio_device_buffer_ = audioBuffer;
const int sample_rate_hz = audio_parameters_.sample_rate();
- RTC_LOG(INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")";
+ RTC_LOG(LS_INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")";
audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
const size_t channels = audio_parameters_.channels();
- RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")";
+ RTC_LOG(LS_INFO) << "SetPlayoutChannels(" << channels << ")";
audio_device_buffer_->SetPlayoutChannels(channels);
}
void AudioTrackJni::CacheDirectBufferAddress(
JNIEnv* env,
const JavaParamRef<jobject>& byte_buffer) {
- RTC_LOG(INFO) << "OnCacheDirectBufferAddress";
+ RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!direct_buffer_address_);
direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer.obj());
jlong capacity = env->GetDirectBufferCapacity(byte_buffer.obj());
- RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
+ RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / bytes_per_frame;
- RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_;
+ RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
}
// This method is called on a high-priority thread from Java. The name of
diff --git a/sdk/android/src/jni/audio_device/opensles_common.cc b/sdk/android/src/jni/audio_device/opensles_common.cc
index 0f35b27..abc415d 100644
--- a/sdk/android/src/jni/audio_device/opensles_common.cc
+++ b/sdk/android/src/jni/audio_device/opensles_common.cc
@@ -107,7 +107,7 @@
}
SLObjectItf OpenSLEngineManager::GetOpenSLEngine() {
- RTC_LOG(INFO) << "GetOpenSLEngine";
+ RTC_LOG(LS_INFO) << "GetOpenSLEngine";
RTC_DCHECK(thread_checker_.IsCurrent());
// OpenSL ES for Android only supports a single engine per application.
// If one already has been created, return existing object instead of
diff --git a/sdk/objc/native/api/audio_device_module.mm b/sdk/objc/native/api/audio_device_module.mm
index dd95775..3c2790e 100644
--- a/sdk/objc/native/api/audio_device_module.mm
+++ b/sdk/objc/native/api/audio_device_module.mm
@@ -18,12 +18,11 @@
namespace webrtc {
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(bool bypass_voice_processing) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
#if defined(WEBRTC_IOS)
return new rtc::RefCountedObject<ios_adm::AudioDeviceModuleIOS>(bypass_voice_processing);
#else
- RTC_LOG(LERROR)
- << "current platform is not supported => this module will self destruct!";
+ RTC_LOG(LS_ERROR) << "current platform is not supported => this module will self destruct!";
return nullptr;
#endif
}
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 859442d..db4ed21 100644
--- a/sdk/objc/native/src/audio/audio_device_module_ios.mm
+++ b/sdk/objc/native/src/audio/audio_device_module_ios.mm
@@ -43,22 +43,22 @@
AudioDeviceModuleIOS::AudioDeviceModuleIOS(bool bypass_voice_processing)
: bypass_voice_processing_(bypass_voice_processing),
task_queue_factory_(CreateDefaultTaskQueueFactory()) {
- RTC_LOG(INFO) << "current platform is IOS";
- RTC_LOG(INFO) << "iPhone Audio APIs will be utilized.";
+ RTC_LOG(LS_INFO) << "current platform is IOS";
+ RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
}
int32_t AudioDeviceModuleIOS::AttachAudioBuffer() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
audio_device_->AttachAudioBuffer(audio_device_buffer_.get());
return 0;
}
AudioDeviceModuleIOS::~AudioDeviceModuleIOS() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
}
int32_t AudioDeviceModuleIOS::ActiveAudioLayer(AudioLayer* audioLayer) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
AudioLayer activeAudio;
if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
return -1;
@@ -68,7 +68,7 @@
}
int32_t AudioDeviceModuleIOS::Init() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (initialized_)
return 0;
@@ -91,7 +91,7 @@
}
int32_t AudioDeviceModuleIOS::Terminate() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
if (!initialized_)
return 0;
if (audio_device_->Terminate() == -1) {
@@ -102,65 +102,65 @@
}
bool AudioDeviceModuleIOS::Initialized() const {
- RTC_DLOG(INFO) << __FUNCTION__ << ": " << initialized_;
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
return initialized_;
}
int32_t AudioDeviceModuleIOS::InitSpeaker() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->InitSpeaker();
}
int32_t AudioDeviceModuleIOS::InitMicrophone() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->InitMicrophone();
}
int32_t AudioDeviceModuleIOS::SpeakerVolumeIsAvailable(bool* available) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetSpeakerVolume(uint32_t volume) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << volume << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
CHECKinitialized_();
return audio_device_->SetSpeakerVolume(volume);
}
int32_t AudioDeviceModuleIOS::SpeakerVolume(uint32_t* volume) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
uint32_t level = 0;
if (audio_device_->SpeakerVolume(level) == -1) {
return -1;
}
*volume = level;
- RTC_DLOG(INFO) << "output: " << *volume;
+ RTC_DLOG(LS_INFO) << "output: " << *volume;
return 0;
}
bool AudioDeviceModuleIOS::SpeakerIsInitialized() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isInitialized = audio_device_->SpeakerIsInitialized();
- RTC_DLOG(INFO) << "output: " << isInitialized;
+ RTC_DLOG(LS_INFO) << "output: " << isInitialized;
return isInitialized;
}
bool AudioDeviceModuleIOS::MicrophoneIsInitialized() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isInitialized = audio_device_->MicrophoneIsInitialized();
- RTC_DLOG(INFO) << "output: " << isInitialized;
+ RTC_DLOG(LS_INFO) << "output: " << isInitialized;
return isInitialized;
}
@@ -185,110 +185,110 @@
}
int32_t AudioDeviceModuleIOS::SpeakerMuteIsAvailable(bool* available) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetSpeakerMute(bool enable) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return audio_device_->SetSpeakerMute(enable);
}
int32_t AudioDeviceModuleIOS::SpeakerMute(bool* enabled) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool muted = false;
if (audio_device_->SpeakerMute(muted) == -1) {
return -1;
}
*enabled = muted;
- RTC_DLOG(INFO) << "output: " << muted;
+ RTC_DLOG(LS_INFO) << "output: " << muted;
return 0;
}
int32_t AudioDeviceModuleIOS::MicrophoneMuteIsAvailable(bool* available) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetMicrophoneMute(bool enable) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return (audio_device_->SetMicrophoneMute(enable));
}
int32_t AudioDeviceModuleIOS::MicrophoneMute(bool* enabled) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool muted = false;
if (audio_device_->MicrophoneMute(muted) == -1) {
return -1;
}
*enabled = muted;
- RTC_DLOG(INFO) << "output: " << muted;
+ RTC_DLOG(LS_INFO) << "output: " << muted;
return 0;
}
int32_t AudioDeviceModuleIOS::MicrophoneVolumeIsAvailable(bool* available) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetMicrophoneVolume(uint32_t volume) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << volume << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
CHECKinitialized_();
return (audio_device_->SetMicrophoneVolume(volume));
}
int32_t AudioDeviceModuleIOS::MicrophoneVolume(uint32_t* volume) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
uint32_t level = 0;
if (audio_device_->MicrophoneVolume(level) == -1) {
return -1;
}
*volume = level;
- RTC_DLOG(INFO) << "output: " << *volume;
+ RTC_DLOG(LS_INFO) << "output: " << *volume;
return 0;
}
int32_t AudioDeviceModuleIOS::StereoRecordingIsAvailable(
bool* available) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetStereoRecording(bool enable) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (enable) {
RTC_LOG(WARNING) << "recording in stereo is not supported";
@@ -297,31 +297,31 @@
}
int32_t AudioDeviceModuleIOS::StereoRecording(bool* enabled) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool stereo = false;
if (audio_device_->StereoRecording(stereo) == -1) {
return -1;
}
*enabled = stereo;
- RTC_DLOG(INFO) << "output: " << stereo;
+ RTC_DLOG(LS_INFO) << "output: " << stereo;
return 0;
}
int32_t AudioDeviceModuleIOS::StereoPlayoutIsAvailable(bool* available) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::SetStereoPlayout(bool enable) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->PlayoutIsInitialized()) {
RTC_LOG(LERROR)
@@ -341,38 +341,38 @@
}
int32_t AudioDeviceModuleIOS::StereoPlayout(bool* enabled) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool stereo = false;
if (audio_device_->StereoPlayout(stereo) == -1) {
return -1;
}
*enabled = stereo;
- RTC_DLOG(INFO) << "output: " << stereo;
+ RTC_DLOG(LS_INFO) << "output: " << stereo;
return 0;
}
int32_t AudioDeviceModuleIOS::PlayoutIsAvailable(bool* available) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleIOS::RecordingIsAvailable(bool* available) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return 0;
}
@@ -397,21 +397,21 @@
}
int16_t AudioDeviceModuleIOS::PlayoutDevices() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
- RTC_DLOG(INFO) << "output: " << nPlayoutDevices;
+ RTC_DLOG(LS_INFO) << "output: " << nPlayoutDevices;
return (int16_t)(nPlayoutDevices);
}
int32_t AudioDeviceModuleIOS::SetPlayoutDevice(uint16_t index) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << index << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
CHECKinitialized_();
return audio_device_->SetPlayoutDevice(index);
}
int32_t AudioDeviceModuleIOS::SetPlayoutDevice(WindowsDeviceType device) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->SetPlayoutDevice(device);
}
@@ -420,7 +420,7 @@
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
CHECKinitialized_();
if (name == NULL) {
return -1;
@@ -429,10 +429,10 @@
return -1;
}
if (name != NULL) {
- RTC_DLOG(INFO) << "output: name = " << name;
+ RTC_DLOG(LS_INFO) << "output: name = " << name;
}
if (guid != NULL) {
- RTC_DLOG(INFO) << "output: guid = " << guid;
+ RTC_DLOG(LS_INFO) << "output: guid = " << guid;
}
return 0;
}
@@ -441,7 +441,7 @@
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
CHECKinitialized_();
if (name == NULL) {
return -1;
@@ -450,137 +450,137 @@
return -1;
}
if (name != NULL) {
- RTC_DLOG(INFO) << "output: name = " << name;
+ RTC_DLOG(LS_INFO) << "output: name = " << name;
}
if (guid != NULL) {
- RTC_DLOG(INFO) << "output: guid = " << guid;
+ RTC_DLOG(LS_INFO) << "output: guid = " << guid;
}
return 0;
}
int16_t AudioDeviceModuleIOS::RecordingDevices() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
uint16_t nRecordingDevices = audio_device_->RecordingDevices();
- RTC_DLOG(INFO) << "output: " << nRecordingDevices;
+ RTC_DLOG(LS_INFO) << "output: " << nRecordingDevices;
return (int16_t)nRecordingDevices;
}
int32_t AudioDeviceModuleIOS::SetRecordingDevice(uint16_t index) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << index << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
CHECKinitialized_();
return audio_device_->SetRecordingDevice(index);
}
int32_t AudioDeviceModuleIOS::SetRecordingDevice(WindowsDeviceType device) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->SetRecordingDevice(device);
}
int32_t AudioDeviceModuleIOS::InitPlayout() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
if (PlayoutIsInitialized()) {
return 0;
}
int32_t result = audio_device_->InitPlayout();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleIOS::InitRecording() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
if (RecordingIsInitialized()) {
return 0;
}
int32_t result = audio_device_->InitRecording();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleIOS::PlayoutIsInitialized() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->PlayoutIsInitialized();
}
bool AudioDeviceModuleIOS::RecordingIsInitialized() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->RecordingIsInitialized();
}
int32_t AudioDeviceModuleIOS::StartPlayout() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
if (Playing()) {
return 0;
}
audio_device_buffer_.get()->StartPlayout();
int32_t result = audio_device_->StartPlayout();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleIOS::StopPlayout() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopPlayout();
audio_device_buffer_.get()->StopPlayout();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleIOS::Playing() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->Playing();
}
int32_t AudioDeviceModuleIOS::StartRecording() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
if (Recording()) {
return 0;
}
audio_device_buffer_.get()->StartRecording();
int32_t result = audio_device_->StartRecording();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleIOS::StopRecording() {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopRecording();
audio_device_buffer_.get()->StopRecording();
- RTC_DLOG(INFO) << "output: " << result;
+ RTC_DLOG(LS_INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleIOS::Recording() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->Recording();
}
int32_t AudioDeviceModuleIOS::RegisterAudioCallback(
AudioTransport* audioCallback) {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
return audio_device_buffer_.get()->RegisterAudioCallback(audioCallback);
}
@@ -596,50 +596,50 @@
}
bool AudioDeviceModuleIOS::BuiltInAECIsAvailable() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInAECIsAvailable();
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleIOS::EnableBuiltInAEC(bool enable) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInAEC(enable);
- RTC_DLOG(INFO) << "output: " << ok;
+ RTC_DLOG(LS_INFO) << "output: " << ok;
return ok;
}
bool AudioDeviceModuleIOS::BuiltInAGCIsAvailable() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleIOS::EnableBuiltInAGC(bool enable) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInAGC(enable);
- RTC_DLOG(INFO) << "output: " << ok;
+ RTC_DLOG(LS_INFO) << "output: " << ok;
return ok;
}
bool AudioDeviceModuleIOS::BuiltInNSIsAvailable() const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInNSIsAvailable();
- RTC_DLOG(INFO) << "output: " << isAvailable;
+ RTC_DLOG(LS_INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleIOS::EnableBuiltInNS(bool enable) {
- RTC_DLOG(INFO) << __FUNCTION__ << "(" << enable << ")";
+ RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInNS(enable);
- RTC_DLOG(INFO) << "output: " << ok;
+ RTC_DLOG(LS_INFO) << "output: " << ok;
return ok;
}
@@ -653,17 +653,17 @@
#if defined(WEBRTC_IOS)
int AudioDeviceModuleIOS::GetPlayoutAudioParameters(
AudioParameters* params) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
int r = audio_device_->GetPlayoutAudioParameters(params);
- RTC_DLOG(INFO) << "output: " << r;
+ RTC_DLOG(LS_INFO) << "output: " << r;
return r;
}
int AudioDeviceModuleIOS::GetRecordAudioParameters(
AudioParameters* params) const {
- RTC_DLOG(INFO) << __FUNCTION__;
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
int r = audio_device_->GetRecordAudioParameters(params);
- RTC_DLOG(INFO) << "output: " << r;
+ RTC_DLOG(LS_INFO) << "output: " << r;
return r;
}
#endif // WEBRTC_IOS
diff --git a/test/network/cross_traffic_unittest.cc b/test/network/cross_traffic_unittest.cc
index 2744a90..0c1bb46 100644
--- a/test/network/cross_traffic_unittest.cc
+++ b/test/network/cross_traffic_unittest.cc
@@ -87,8 +87,8 @@
fixture.clock.AdvanceTimeMilliseconds(1);
}
- RTC_LOG(INFO) << fixture.counter.packets_count_ << " packets; "
- << fixture.counter.total_packets_size_ << " bytes";
+ RTC_LOG(LS_INFO) << fixture.counter.packets_count_ << " packets; "
+ << fixture.counter.total_packets_size_ << " bytes";
// Using 50% duty cycle.
const auto kExpectedDataSent = kRunTime * config.peak_rate * 0.5;
EXPECT_NEAR(fixture.counter.total_packets_size_, kExpectedDataSent.bytes(),
@@ -115,8 +115,8 @@
fixture.clock.AdvanceTimeMilliseconds(1);
}
- RTC_LOG(INFO) << fixture.counter.packets_count_ << " packets; "
- << fixture.counter.total_packets_size_ << " bytes";
+ RTC_LOG(LS_INFO) << fixture.counter.packets_count_ << " packets; "
+ << fixture.counter.total_packets_size_ << " bytes";
// Sending at peak rate since bias = 1.
const auto kExpectedDataSent = kRunTime * config.peak_rate;
EXPECT_NEAR(fixture.counter.total_packets_size_, kExpectedDataSent.bytes(),
diff --git a/test/network/fake_network_socket_server.cc b/test/network/fake_network_socket_server.cc
index 9597edb..28c0473 100644
--- a/test/network/fake_network_socket_server.cc
+++ b/test/network/fake_network_socket_server.cc
@@ -132,7 +132,7 @@
endpoint_ = socket_server_->GetEndpointNode(local_addr_.ipaddr());
if (!endpoint_) {
local_addr_.Clear();
- RTC_LOG(INFO) << "No endpoint for address: " << ToString(addr);
+ RTC_LOG(LS_INFO) << "No endpoint for address: " << ToString(addr);
error_ = EADDRNOTAVAIL;
return 2;
}
@@ -140,7 +140,7 @@
endpoint_->BindReceiver(local_addr_.port(), this);
if (!port) {
local_addr_.Clear();
- RTC_LOG(INFO) << "Cannot bind to in-use address: " << ToString(addr);
+ RTC_LOG(LS_INFO) << "Cannot bind to in-use address: " << ToString(addr);
error_ = EADDRINUSE;
return 1;
}
diff --git a/test/network/network_emulation.cc b/test/network/network_emulation.cc
index ada9ab5..4cd2fda 100644
--- a/test/network/network_emulation.cc
+++ b/test/network/network_emulation.cc
@@ -477,8 +477,8 @@
network_->AddIP(options_.ip);
enabled_state_checker_.Detach();
- RTC_LOG(INFO) << "Created emulated endpoint " << options_.log_name
- << "; id=" << options_.id;
+ RTC_LOG(LS_INFO) << "Created emulated endpoint " << options_.log_name
+ << "; id=" << options_.id;
}
EmulatedEndpointImpl::~EmulatedEndpointImpl() = default;
@@ -546,13 +546,13 @@
bool result =
port_to_receiver_.insert({port, {receiver, is_one_shot}}).second;
if (!result) {
- RTC_LOG(INFO) << "Can't bind receiver to used port " << desired_port
- << " in endpoint " << options_.log_name
- << "; id=" << options_.id;
+ RTC_LOG(LS_INFO) << "Can't bind receiver to used port " << desired_port
+ << " in endpoint " << options_.log_name
+ << "; id=" << options_.id;
return absl::nullopt;
}
- RTC_LOG(INFO) << "New receiver is binded to endpoint " << options_.log_name
- << "; id=" << options_.id << " on port " << port;
+ RTC_LOG(LS_INFO) << "New receiver is binded to endpoint " << options_.log_name
+ << "; id=" << options_.id << " on port " << port;
return port;
}
@@ -568,8 +568,9 @@
void EmulatedEndpointImpl::UnbindReceiver(uint16_t port) {
MutexLock lock(&receiver_lock_);
- RTC_LOG(INFO) << "Receiver is removed on port " << port << " from endpoint "
- << options_.log_name << "; id=" << options_.id;
+ RTC_LOG(LS_INFO) << "Receiver is removed on port " << port
+ << " from endpoint " << options_.log_name
+ << "; id=" << options_.id;
port_to_receiver_.erase(port);
}
@@ -579,15 +580,15 @@
RTC_CHECK(!default_receiver_.has_value())
<< "Endpoint " << options_.log_name << "; id=" << options_.id
<< " already has default receiver";
- RTC_LOG(INFO) << "Default receiver is binded to endpoint "
- << options_.log_name << "; id=" << options_.id;
+ RTC_LOG(LS_INFO) << "Default receiver is binded to endpoint "
+ << options_.log_name << "; id=" << options_.id;
default_receiver_ = receiver;
}
void EmulatedEndpointImpl::UnbindDefaultReceiver() {
MutexLock lock(&receiver_lock_);
- RTC_LOG(INFO) << "Default receiver is removed from endpoint "
- << options_.log_name << "; id=" << options_.id;
+ RTC_LOG(LS_INFO) << "Default receiver is removed from endpoint "
+ << options_.log_name << "; id=" << options_.id;
default_receiver_ = absl::nullopt;
}
@@ -616,9 +617,9 @@
// It can happen, that remote peer closed connection, but there still some
// packets, that are going to it. It can happen during peer connection close
// process: one peer closed connection, second still sending data.
- RTC_LOG(INFO) << "Drop packet: no receiver registered in "
- << options_.log_name << "; id=" << options_.id << " on port "
- << packet.to.port();
+ RTC_LOG(LS_INFO) << "Drop packet: no receiver registered in "
+ << options_.log_name << "; id=" << options_.id
+ << " on port " << packet.to.port();
stats_builder_.OnPacketDropped(packet.from.ipaddr(),
DataSize::Bytes(packet.ip_packet_size()),
options_.stats_gathering_mode);
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 e7b7f3f..9af65e2 100644
--- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -39,13 +39,13 @@
"considered dropped";
void LogFrameCounters(const std::string& name, const FrameCounters& counters) {
- RTC_LOG(INFO) << "[" << name << "] Captured : " << counters.captured;
- RTC_LOG(INFO) << "[" << name << "] Pre encoded : " << counters.pre_encoded;
- RTC_LOG(INFO) << "[" << name << "] Encoded : " << counters.encoded;
- RTC_LOG(INFO) << "[" << name << "] Received : " << counters.received;
- RTC_LOG(INFO) << "[" << name << "] Decoded : " << counters.decoded;
- RTC_LOG(INFO) << "[" << name << "] Rendered : " << counters.rendered;
- RTC_LOG(INFO) << "[" << name << "] Dropped : " << counters.dropped;
+ RTC_LOG(LS_INFO) << "[" << name << "] Captured : " << counters.captured;
+ RTC_LOG(LS_INFO) << "[" << name << "] Pre encoded : " << counters.pre_encoded;
+ RTC_LOG(LS_INFO) << "[" << name << "] Encoded : " << counters.encoded;
+ RTC_LOG(LS_INFO) << "[" << name << "] Received : " << counters.received;
+ RTC_LOG(LS_INFO) << "[" << name << "] Decoded : " << counters.decoded;
+ RTC_LOG(LS_INFO) << "[" << name << "] Rendered : " << counters.rendered;
+ RTC_LOG(LS_INFO) << "[" << name << "] Dropped : " << counters.dropped;
}
absl::string_view ToString(FrameDropPhase phase) {
@@ -67,8 +67,8 @@
const StreamStats& stats,
Timestamp start_time) {
for (const auto& entry : stats.dropped_by_phase) {
- RTC_LOG(INFO) << "[" << name << "] Dropped at " << ToString(entry.first)
- << ": " << entry.second;
+ RTC_LOG(LS_INFO) << "[" << name << "] Dropped at " << ToString(entry.first)
+ << ": " << entry.second;
}
Timestamp first_encoded_frame_time = Timestamp::PlusInfinity();
for (const StreamCodecInfo& encoder : stats.encoders) {
@@ -77,32 +77,32 @@
if (first_encoded_frame_time.IsInfinite()) {
first_encoded_frame_time = encoder.switched_on_at;
}
- RTC_LOG(INFO) << "[" << name << "] Used encoder: \"" << encoder.codec_name
- << "\" used from (frame_id=" << encoder.first_frame_id
- << "; from_stream_start="
- << (encoder.switched_on_at - stats.stream_started_time).ms()
- << "ms, from_call_start="
- << (encoder.switched_on_at - start_time).ms()
- << "ms) until (frame_id=" << encoder.last_frame_id
- << "; from_stream_start="
- << (encoder.switched_from_at - stats.stream_started_time).ms()
- << "ms, from_call_start="
- << (encoder.switched_from_at - start_time).ms() << "ms)";
+ RTC_LOG(LS_INFO)
+ << "[" << name << "] Used encoder: \"" << encoder.codec_name
+ << "\" used from (frame_id=" << encoder.first_frame_id
+ << "; from_stream_start="
+ << (encoder.switched_on_at - stats.stream_started_time).ms()
+ << "ms, from_call_start=" << (encoder.switched_on_at - start_time).ms()
+ << "ms) until (frame_id=" << encoder.last_frame_id
+ << "; from_stream_start="
+ << (encoder.switched_from_at - stats.stream_started_time).ms()
+ << "ms, from_call_start="
+ << (encoder.switched_from_at - start_time).ms() << "ms)";
}
for (const StreamCodecInfo& decoder : stats.decoders) {
RTC_DCHECK(decoder.switched_on_at.IsFinite());
RTC_DCHECK(decoder.switched_from_at.IsFinite());
- RTC_LOG(INFO) << "[" << name << "] Used decoder: \"" << decoder.codec_name
- << "\" used from (frame_id=" << decoder.first_frame_id
- << "; from_stream_start="
- << (decoder.switched_on_at - stats.stream_started_time).ms()
- << "ms, from_call_start="
- << (decoder.switched_on_at - start_time).ms()
- << "ms) until (frame_id=" << decoder.last_frame_id
- << "; from_stream_start="
- << (decoder.switched_from_at - stats.stream_started_time).ms()
- << "ms, from_call_start="
- << (decoder.switched_from_at - start_time).ms() << "ms)";
+ RTC_LOG(LS_INFO)
+ << "[" << name << "] Used decoder: \"" << decoder.codec_name
+ << "\" used from (frame_id=" << decoder.first_frame_id
+ << "; from_stream_start="
+ << (decoder.switched_on_at - stats.stream_started_time).ms()
+ << "ms, from_call_start=" << (decoder.switched_on_at - start_time).ms()
+ << "ms) until (frame_id=" << decoder.last_frame_id
+ << "; from_stream_start="
+ << (decoder.switched_from_at - stats.stream_started_time).ms()
+ << "ms, from_call_start="
+ << (decoder.switched_from_at - start_time).ms() << "ms)";
}
}
@@ -668,8 +668,8 @@
MutexLock lock(&mutex_);
std::set<StatsKey> out;
for (auto& item : frames_comparator_.stream_stats()) {
- RTC_LOG(INFO) << item.first.ToString() << " ==> "
- << ToStatsKey(item.first).ToString();
+ RTC_LOG(LS_INFO) << item.first.ToString() << " ==> "
+ << ToStatsKey(item.first).ToString();
out.insert(ToStatsKey(item.first));
}
return out;
@@ -722,17 +722,19 @@
start_time_);
}
if (!analyzer_stats_.comparisons_queue_size.IsEmpty()) {
- RTC_LOG(INFO) << "comparisons_queue_size min="
- << analyzer_stats_.comparisons_queue_size.GetMin()
- << "; max=" << analyzer_stats_.comparisons_queue_size.GetMax()
- << "; 99%="
- << analyzer_stats_.comparisons_queue_size.GetPercentile(0.99);
+ RTC_LOG(LS_INFO) << "comparisons_queue_size min="
+ << analyzer_stats_.comparisons_queue_size.GetMin()
+ << "; max="
+ << analyzer_stats_.comparisons_queue_size.GetMax()
+ << "; 99%="
+ << analyzer_stats_.comparisons_queue_size.GetPercentile(
+ 0.99);
}
- RTC_LOG(INFO) << "comparisons_done=" << analyzer_stats_.comparisons_done;
- RTC_LOG(INFO) << "cpu_overloaded_comparisons_done="
- << analyzer_stats_.cpu_overloaded_comparisons_done;
- RTC_LOG(INFO) << "memory_overloaded_comparisons_done="
- << analyzer_stats_.memory_overloaded_comparisons_done;
+ RTC_LOG(LS_INFO) << "comparisons_done=" << analyzer_stats_.comparisons_done;
+ RTC_LOG(LS_INFO) << "cpu_overloaded_comparisons_done="
+ << analyzer_stats_.cpu_overloaded_comparisons_done;
+ RTC_LOG(LS_INFO) << "memory_overloaded_comparisons_done="
+ << analyzer_stats_.memory_overloaded_comparisons_done;
}
void DefaultVideoQualityAnalyzer::ReportResults(
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 4808cf4..669fcfe 100644
--- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
@@ -66,7 +66,7 @@
void ExampleVideoQualityAnalyzer::OnFrameDropped(
absl::string_view peer_name,
webrtc::EncodedImageCallback::DropReason reason) {
- RTC_LOG(INFO) << "Frame dropped by encoder";
+ RTC_LOG(LS_INFO) << "Frame dropped by encoder";
MutexLock lock(&lock_);
++frames_dropped_;
}
@@ -112,8 +112,8 @@
void ExampleVideoQualityAnalyzer::Stop() {
MutexLock lock(&lock_);
- RTC_LOG(INFO) << "There are " << frames_in_flight_.size()
- << " frames in flight, assuming all of them are dropped";
+ RTC_LOG(LS_INFO) << "There are " << frames_in_flight_.size()
+ << " frames in flight, assuming all of them are dropped";
frames_dropped_ += frames_in_flight_.size();
}
diff --git a/test/pc/e2e/echo/echo_emulation.cc b/test/pc/e2e/echo/echo_emulation.cc
index 230e8e3..f2b4be9 100644
--- a/test/pc/e2e/echo/echo_emulation.cc
+++ b/test/pc/e2e/echo/echo_emulation.cc
@@ -57,7 +57,7 @@
}
queue_input_.assign(data.begin(), data.end());
if (!renderer_queue_.Insert(&queue_input_)) {
- RTC_LOG(WARNING) << "Echo queue is full";
+ RTC_LOG(LS_WARNING) << "Echo queue is full";
}
}
diff --git a/test/pc/e2e/media/media_helper.cc b/test/pc/e2e/media/media_helper.cc
index 6b1996a..c90b2e5 100644
--- a/test/pc/e2e/media/media_helper.cc
+++ b/test/pc/e2e/media/media_helper.cc
@@ -67,8 +67,8 @@
rtc::make_ref_counted<TestVideoCapturerVideoTrackSource>(
std::move(capturer), is_screencast);
out.push_back(source);
- RTC_LOG(INFO) << "Adding video with video_config.stream_label="
- << video_config.stream_label.value();
+ RTC_LOG(LS_INFO) << "Adding video with video_config.stream_label="
+ << video_config.stream_label.value();
rtc::scoped_refptr<VideoTrackInterface> track =
peer->pc_factory()->CreateVideoTrack(video_config.stream_label.value(),
source);
diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc
index f9a1294..4e9368f 100644
--- a/test/pc/e2e/peer_connection_quality_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test.cc
@@ -202,17 +202,18 @@
test::ScopedFieldTrials field_trials(GetFieldTrials(run_params));
// Print test summary
- RTC_LOG(INFO) << "Media quality test: " << *alice_configurer->params()->name
- << " will make a call to " << *bob_configurer->params()->name
- << " with media video="
- << !alice_configurer->params()->video_configs.empty()
- << "; audio="
- << alice_configurer->params()->audio_config.has_value() << ". "
- << *bob_configurer->params()->name
- << " will respond with media video="
- << !bob_configurer->params()->video_configs.empty()
- << "; audio="
- << bob_configurer->params()->audio_config.has_value();
+ RTC_LOG(LS_INFO) << "Media quality test: "
+ << *alice_configurer->params()->name
+ << " will make a call to " << *bob_configurer->params()->name
+ << " with media video="
+ << !alice_configurer->params()->video_configs.empty()
+ << "; audio="
+ << alice_configurer->params()->audio_config.has_value()
+ << ". " << *bob_configurer->params()->name
+ << " will respond with media video="
+ << !bob_configurer->params()->video_configs.empty()
+ << "; audio="
+ << bob_configurer->params()->audio_config.has_value();
const std::unique_ptr<rtc::Thread> signaling_thread =
time_controller_.CreateThread(kSignalThreadName);
@@ -276,7 +277,7 @@
}
video_analyzer_threads =
std::min(video_analyzer_threads, kMaxVideoAnalyzerThreads);
- RTC_LOG(INFO) << "video_analyzer_threads=" << video_analyzer_threads;
+ RTC_LOG(LS_INFO) << "video_analyzer_threads=" << video_analyzer_threads;
quality_metrics_reporters_.push_back(
std::make_unique<VideoQualityMetricsReporter>(
time_controller_.GetClock()));
@@ -315,8 +316,8 @@
return kAliveMessageLogInterval;
});
- RTC_LOG(INFO) << "Configuration is done. Now " << *alice_->params()->name
- << " is calling to " << *bob_->params()->name << "...";
+ RTC_LOG(LS_INFO) << "Configuration is done. Now " << *alice_->params()->name
+ << " is calling to " << *bob_->params()->name << "...";
// Setup stats poller.
std::vector<StatsObserverInterface*> observers = {
@@ -359,7 +360,7 @@
time_controller_.AdvanceTime(run_params.run_duration);
}
- RTC_LOG(INFO) << "Test is done, initiating disconnect sequence.";
+ RTC_LOG(LS_INFO) << "Test is done, initiating disconnect sequence.";
// Stop all client started tasks to prevent their access to any call related
// objects after these objects will be destroyed during call tear down.
@@ -382,7 +383,7 @@
[this] { TearDownCallOnSignalingThread(); });
Timestamp end_time = Now();
- RTC_LOG(INFO) << "All peers are disconnected.";
+ RTC_LOG(LS_INFO) << "All peers are disconnected.";
{
MutexLock lock(&lock_);
real_test_duration_ = end_time - start_time;
@@ -619,13 +620,13 @@
auto offer = alice_->CreateOffer();
RTC_CHECK(offer);
offer->ToString(&log_output);
- RTC_LOG(INFO) << "Original offer: " << log_output;
+ RTC_LOG(LS_INFO) << "Original offer: " << log_output;
LocalAndRemoteSdp patch_result = signaling_interceptor->PatchOffer(
std::move(offer), alice_->params()->video_codecs[0]);
patch_result.local_sdp->ToString(&log_output);
- RTC_LOG(INFO) << "Offer to set as local description: " << log_output;
+ RTC_LOG(LS_INFO) << "Offer to set as local description: " << log_output;
patch_result.remote_sdp->ToString(&log_output);
- RTC_LOG(INFO) << "Offer to set as remote description: " << log_output;
+ RTC_LOG(LS_INFO) << "Offer to set as remote description: " << log_output;
bool set_local_offer =
alice_->SetLocalDescription(std::move(patch_result.local_sdp));
@@ -636,13 +637,13 @@
auto answer = bob_->CreateAnswer();
RTC_CHECK(answer);
answer->ToString(&log_output);
- RTC_LOG(INFO) << "Original answer: " << log_output;
+ RTC_LOG(LS_INFO) << "Original answer: " << log_output;
patch_result = signaling_interceptor->PatchAnswer(
std::move(answer), bob_->params()->video_codecs[0]);
patch_result.local_sdp->ToString(&log_output);
- RTC_LOG(INFO) << "Answer to set as local description: " << log_output;
+ RTC_LOG(LS_INFO) << "Answer to set as local description: " << log_output;
patch_result.remote_sdp->ToString(&log_output);
- RTC_LOG(INFO) << "Answer to set as remote description: " << log_output;
+ RTC_LOG(LS_INFO) << "Answer to set as remote description: " << log_output;
bool set_local_answer =
bob_->SetLocalDescription(std::move(patch_result.local_sdp));
@@ -661,9 +662,9 @@
for (auto& candidate : alice_candidates) {
std::string candidate_str;
RTC_CHECK(candidate->ToString(&candidate_str));
- RTC_LOG(INFO) << *alice_->params()->name
- << " ICE candidate(mid= " << candidate->sdp_mid()
- << "): " << candidate_str;
+ RTC_LOG(LS_INFO) << *alice_->params()->name
+ << " ICE candidate(mid= " << candidate->sdp_mid()
+ << "): " << candidate_str;
}
ASSERT_TRUE(bob_->AddIceCandidates(std::move(alice_candidates)));
std::vector<std::unique_ptr<IceCandidateInterface>> bob_candidates =
@@ -672,9 +673,9 @@
for (auto& candidate : bob_candidates) {
std::string candidate_str;
RTC_CHECK(candidate->ToString(&candidate_str));
- RTC_LOG(INFO) << *bob_->params()->name
- << " ICE candidate(mid= " << candidate->sdp_mid()
- << "): " << candidate_str;
+ RTC_LOG(LS_INFO) << *bob_->params()->name
+ << " ICE candidate(mid= " << candidate->sdp_mid()
+ << "): " << candidate_str;
}
ASSERT_TRUE(alice_->AddIceCandidates(std::move(bob_candidates)));
}
diff --git a/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc b/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
index 1ba31a3..94a46fb 100644
--- a/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
+++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
@@ -378,7 +378,7 @@
}
}
- RTC_LOG(INFO) << log.str();
+ RTC_LOG(LS_INFO) << log.str();
}
} // namespace webrtc_pc_e2e
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index b77a475..ac565cc 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -1342,7 +1342,7 @@
rtc::scoped_refptr<AudioDeviceModule> VideoQualityTest::CreateAudioDevice() {
#ifdef WEBRTC_WIN
- RTC_LOG(INFO) << "Using latest version of ADM on Windows";
+ RTC_LOG(LS_INFO) << "Using latest version of ADM on Windows";
// We must initialize the COM library on a thread before we calling any of
// the library functions. All COM functions in the ADM will return
// CO_E_NOTINITIALIZED otherwise. The legacy ADM for Windows used internal
@@ -1433,7 +1433,7 @@
}
void VideoQualityTest::RunWithRenderers(const Params& params) {
- RTC_LOG(INFO) << __FUNCTION__;
+ RTC_LOG(LS_INFO) << __FUNCTION__;
num_video_streams_ = params.call.dual_video ? 2 : 1;
std::unique_ptr<test::LayerFilteringTransport> send_transport;
std::unique_ptr<test::DirectTransport> recv_transport;
diff --git a/video/video_source_sink_controller.cc b/video/video_source_sink_controller.cc
index 6955e3b..810a4ff 100644
--- a/video/video_source_sink_controller.cc
+++ b/video/video_source_sink_controller.cc
@@ -80,7 +80,8 @@
if (!source_)
return;
rtc::VideoSinkWants wants = CurrentSettingsToSinkWants();
- RTC_LOG(INFO) << "Pushing SourceSink restrictions: " << WantsToString(wants);
+ RTC_LOG(LS_INFO) << "Pushing SourceSink restrictions: "
+ << WantsToString(wants);
source_->AddOrUpdateSink(sink_, wants);
}
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index 1916a59..7e0e7fc 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -2173,9 +2173,9 @@
rtc::scoped_refptr<Resource> reason,
const VideoSourceRestrictions& unfiltered_restrictions) {
RTC_DCHECK_RUN_ON(&encoder_queue_);
- RTC_LOG(INFO) << "Updating sink restrictions from "
- << (reason ? reason->Name() : std::string("<null>")) << " to "
- << restrictions.ToString();
+ RTC_LOG(LS_INFO) << "Updating sink restrictions from "
+ << (reason ? reason->Name() : std::string("<null>"))
+ << " to " << restrictions.ToString();
worker_queue_->PostTask(ToQueuedTask(
task_safety_, [this, restrictions = std::move(restrictions)]() {
RTC_DCHECK_RUN_ON(worker_queue_);
diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc
index ca1a527..b41b4b0 100644
--- a/video/video_stream_encoder_unittest.cc
+++ b/video/video_stream_encoder_unittest.cc
@@ -510,9 +510,9 @@
int out_width = 0;
int out_height = 0;
if (adaption_enabled()) {
- RTC_DLOG(INFO) << "IncomingCapturedFrame: AdaptFrameResolution()"
- << "w=" << video_frame.width()
- << "h=" << video_frame.height();
+ RTC_DLOG(LS_INFO) << "IncomingCapturedFrame: AdaptFrameResolution()"
+ << "w=" << video_frame.width()
+ << "h=" << video_frame.height();
if (adapter_.AdaptFrameResolution(
video_frame.width(), video_frame.height(),
video_frame.timestamp_us() * 1000, &cropped_width,
@@ -540,7 +540,7 @@
last_height_ = absl::nullopt;
}
} else {
- RTC_DLOG(INFO) << "IncomingCapturedFrame: adaptation not enabled";
+ RTC_DLOG(LS_INFO) << "IncomingCapturedFrame: adaptation not enabled";
test::FrameForwarder::IncomingCapturedFrame(video_frame);
last_width_.emplace(video_frame.width());
last_height_.emplace(video_frame.height());
@@ -1352,7 +1352,7 @@
}
log << "]";
}
- RTC_DLOG(INFO) << "OnVideoLayersAllocationUpdated " << log.str();
+ RTC_DLOG(LS_INFO) << "OnVideoLayersAllocationUpdated " << log.str();
}
TimeController* const time_controller_;
