Declare LERROR deprecated and remove all usage in webrtc
Bug: webrtc:13362
Change-Id: I1c6c6eccd950d73be616b34f96db7832ff94377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238804
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35416}
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 6d9211c..aa98169 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -66,14 +66,14 @@
int AcmReceiver::SetMinimumDelay(int delay_ms) {
if (neteq_->SetMinimumDelay(delay_ms))
return 0;
- RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
+ RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
return -1;
}
int AcmReceiver::SetMaximumDelay(int delay_ms) {
if (neteq_->SetMaximumDelay(delay_ms))
return 0;
- RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
+ RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
return -1;
}
@@ -134,9 +134,9 @@
} // `mutex_` is released.
if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
- RTC_LOG(LERROR) << "AcmReceiver::InsertPacket "
- << static_cast<int>(rtp_header.payloadType)
- << " Failed to insert packet";
+ RTC_LOG(LS_ERROR) << "AcmReceiver::InsertPacket "
+ << static_cast<int>(rtp_header.payloadType)
+ << " Failed to insert packet";
return -1;
}
return 0;
@@ -150,7 +150,7 @@
int current_sample_rate_hz = 0;
if (neteq_->GetAudio(audio_frame, muted, ¤t_sample_rate_hz) !=
NetEq::kOK) {
- RTC_LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
+ RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
return -1;
}
@@ -170,8 +170,8 @@
audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
temp_output);
if (samples_per_channel_int < 0) {
- RTC_LOG(LERROR) << "AcmReceiver::GetAudio - "
- "Resampling last_audio_buffer_ failed.";
+ RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - "
+ "Resampling last_audio_buffer_ failed.";
return -1;
}
}
@@ -185,7 +185,7 @@
audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
audio_frame->mutable_data());
if (samples_per_channel_int < 0) {
- RTC_LOG(LERROR)
+ RTC_LOG(LS_ERROR)
<< "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
return -1;
}