Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:
for m in PLOG \
LOG_TAG \
LOG_GLEM \
LOG_GLE_EX \
LOG_GLE \
LAST_SYSTEM_ERROR \
LOG_ERRNO_EX \
LOG_ERRNO \
LOG_ERR_EX \
LOG_ERR \
LOG_V \
LOG_F \
LOG_T_F \
LOG_E \
LOG_T \
LOG_CHECK_LEVEL_V \
LOG_CHECK_LEVEL \
LOG
do
git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format
Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 3320d1b..361278f 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -455,9 +455,9 @@
codec_histogram_bins_log_(),
number_of_consecutive_empty_packets_(0) {
if (InitializeReceiverSafe() < 0) {
- LOG(LS_ERROR) << "Cannot initialize receiver";
+ RTC_LOG(LS_ERROR) << "Cannot initialize receiver";
}
- LOG(LS_INFO) << "Created";
+ RTC_LOG(LS_INFO) << "Created";
}
AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
@@ -629,7 +629,7 @@
rtc::CritScope lock(&acm_crit_sect_);
if (!encoder_stack_) {
- LOG(LS_ERROR) << "SendFrequency Failed, no codec is registered";
+ RTC_LOG(LS_ERROR) << "SendFrequency Failed, no codec is registered";
return -1;
}
@@ -665,26 +665,26 @@
InputData* input_data) {
if (audio_frame.samples_per_channel_ == 0) {
assert(false);
- LOG(LS_ERROR) << "Cannot Add 10 ms audio, payload length is zero";
+ RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, payload length is zero";
return -1;
}
if (audio_frame.sample_rate_hz_ > 48000) {
assert(false);
- LOG(LS_ERROR) << "Cannot Add 10 ms audio, input frequency not valid";
+ RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, input frequency not valid";
return -1;
}
// If the length and frequency matches. We currently just support raw PCM.
if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) !=
audio_frame.samples_per_channel_) {
- LOG(LS_ERROR)
+ RTC_LOG(LS_ERROR)
<< "Cannot Add 10 ms audio, input frequency and length doesn't match";
return -1;
}
if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) {
- LOG(LS_ERROR) << "Cannot Add 10 ms audio, invalid number of channels.";
+ RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, invalid number of channels.";
return -1;
}
@@ -757,8 +757,8 @@
expected_codec_ts_ = in_frame.timestamp_;
first_10ms_data_ = true;
} else if (in_frame.timestamp_ != expected_in_ts_) {
- LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_
- << ", expected: " << expected_in_ts_;
+ RTC_LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_
+ << ", expected: " << expected_in_ts_;
expected_codec_ts_ +=
(in_frame.timestamp_ - expected_in_ts_) *
static_cast<uint32_t>(
@@ -816,7 +816,7 @@
dest_ptr_audio);
if (samples_per_channel < 0) {
- LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed";
+ RTC_LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed";
return -1;
}
preprocess_frame_.samples_per_channel_ =
@@ -853,7 +853,7 @@
encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
return 0;
#else
- LOG(LS_WARNING) << " WEBRTC_CODEC_RED is undefined";
+ RTC_LOG(LS_WARNING) << " WEBRTC_CODEC_RED is undefined";
return -1;
#endif
}
@@ -971,8 +971,8 @@
RTC_DCHECK(receiver_initialized_);
if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
- LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
- << " for decoder.";
+ RTC_LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
+ << " for decoder.";
return false;
}
@@ -998,14 +998,15 @@
rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) {
RTC_DCHECK(receiver_initialized_);
if (codec.channels > 2) {
- LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
+ RTC_LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
return -1;
}
auto codec_id = acm2::RentACodec::CodecIdByParams(codec.plname, codec.plfreq,
codec.channels);
if (!codec_id) {
- LOG_F(LS_ERROR) << "Wrong codec params to be registered as receive codec";
+ RTC_LOG_F(LS_ERROR)
+ << "Wrong codec params to be registered as receive codec";
return -1;
}
auto codec_index = acm2::RentACodec::CodecIndexFromId(*codec_id);
@@ -1013,8 +1014,8 @@
// Check if the payload-type is valid.
if (!acm2::RentACodec::IsPayloadTypeValid(codec.pltype)) {
- LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for "
- << codec.plname;
+ RTC_LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for "
+ << codec.plname;
return -1;
}
@@ -1040,14 +1041,14 @@
rtc::CritScope lock(&acm_crit_sect_);
RTC_DCHECK(receiver_initialized_);
if (num_channels > 2 || num_channels < 0) {
- LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
+ RTC_LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
return -1;
}
// Check if the payload-type is valid.
if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
- LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
- << " for external decoder.";
+ RTC_LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
+ << " for external decoder.";
return -1;
}
@@ -1079,7 +1080,7 @@
// Minimum playout delay (Used for lip-sync).
int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
if ((time_ms < 0) || (time_ms > 10000)) {
- LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
+ RTC_LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
return -1;
}
return receiver_.SetMinimumDelay(time_ms);
@@ -1087,7 +1088,7 @@
int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
if ((time_ms < 0) || (time_ms > 10000)) {
- LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
+ RTC_LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
return -1;
}
return receiver_.SetMaximumDelay(time_ms);
@@ -1100,7 +1101,7 @@
bool* muted) {
// GetAudio always returns 10 ms, at the requested sample rate.
if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
- LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
+ RTC_LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
return -1;
}
return 0;
@@ -1126,7 +1127,7 @@
}
int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
- LOG(LS_VERBOSE) << "RegisterVADCallback()";
+ RTC_LOG(LS_VERBOSE) << "RegisterVADCallback()";
rtc::CritScope lock(&callback_crit_sect_);
vad_callback_ = vad_callback;
return 0;
@@ -1196,7 +1197,7 @@
bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
if (!encoder_stack_) {
- LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";
+ RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";
return false;
}
return true;
@@ -1331,7 +1332,7 @@
bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
bool valid = acm2::RentACodec::IsCodecValid(codec);
if (!valid)
- LOG(LS_ERROR) << "Invalid codec setting";
+ RTC_LOG(LS_ERROR) << "Invalid codec setting";
return valid;
}