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Declare LERROR deprecated and remove all usage in webrtc

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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}
This commit is contained in:
Harald Alvestrand
2021-11-24 10:01:32 +00:00
committed by WebRTC LUCI CQ
parent 4e8e36c745
commit 5f34130f26
10 changed files with 28 additions and 30 deletions
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18
modules/audio_coding/acm2/acm_receiver.cc
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@ -66,14 +66,14 @@ AcmReceiver::~AcmReceiver() = default;
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 @@ int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
} // `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 AcmReceiver::GetAudio(int desired_freq_hz,
int current_sample_rate_hz = 0;
if (neteq_->GetAudio(audio_frame, muted, &current_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 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
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 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
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;
}