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Stop using LOG macros in favor of RTC_ prefixed macros.

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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}
This commit is contained in:
Mirko Bonadei
2017-11-09 11:09:25 +01:00
committed by Commit Bot
parent 34fa309129
commit 675513b96a
407 changed files with 5753 additions and 5371 deletions
Download Patch File Download Diff File Expand all files Collapse all files

53
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -455,9 +455,9 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
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 @@ int AudioCodingModuleImpl::SendFrequency() const {
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 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
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 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
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 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
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 @@ int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
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 @@ bool AudioCodingModuleImpl::RegisterReceiveCodec(
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 @@ int AudioCodingModuleImpl::RegisterReceiveCodecUnlocked(
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 @@ int AudioCodingModuleImpl::RegisterReceiveCodecUnlocked(
// 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 @@ int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
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 @@ int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
// 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::SetMinimumPlayoutDelay(int time_ms) {
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 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
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::GetNetworkStatistics(NetworkStatistics* statistics) {
}
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 @@ int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
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 @@ int AudioCodingModule::Codec(const char* payload_name,
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;
}