Remove WEBRTC_TRACE from webrtc/modules/audio_coding
We'd like to remove all occurrences of WEBRTC_TRACE and delete the
macro! One logging mechanism is enough.
NOTRY=True
Bug: webrtc:5118
Change-Id: Ic226318e0aebe3a71785fcb4ce07371872ab7128
Reviewed-on: https://chromium-review.googlesource.com/518133
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18712}diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index 2fcbecf..d231a84 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -19,7 +19,6 @@
#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {
@@ -466,10 +465,9 @@
codec_histogram_bins_log_(),
number_of_consecutive_empty_packets_(0) {
if (InitializeReceiverSafe() < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Cannot initialize receiver");
+ LOG(LS_ERROR) << "Cannot initialize receiver";
}
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
+ LOG(LS_INFO) << "Created";
}
AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
@@ -638,13 +636,11 @@
// Get current send frequency.
int AudioCodingModuleImpl::SendFrequency() const {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
- "SendFrequency()");
+ LOG(LS_VERBOSE) << "SendFrequency()";
rtc::CritScope lock(&acm_crit_sect_);
if (!encoder_stack_) {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
- "SendFrequency Failed, no codec is registered");
+ LOG(LS_VERBOSE) << "SendFrequency Failed, no codec is registered";
return -1;
}
@@ -680,30 +676,26 @@
InputData* input_data) {
if (audio_frame.samples_per_channel_ == 0) {
assert(false);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Cannot Add 10 ms audio, payload length is zero");
+ LOG(LS_ERROR) << "Cannot Add 10 ms audio, payload length is zero";
return -1;
}
if (audio_frame.sample_rate_hz_ > 48000) {
assert(false);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Cannot Add 10 ms audio, input frequency not valid");
+ 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_) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Cannot Add 10 ms audio, input frequency and length doesn't"
- " match");
+ 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) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Cannot Add 10 ms audio, invalid number of channels.");
+ LOG(LS_ERROR) << "Cannot Add 10 ms audio, invalid number of channels.";
return -1;
}
@@ -835,8 +827,7 @@
dest_ptr_audio);
if (samples_per_channel < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Cannot add 10 ms audio, resampling failed");
+ LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed";
return -1;
}
preprocess_frame_.samples_per_channel_ =
@@ -873,8 +864,7 @@
encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
return 0;
#else
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
- " WEBRTC_CODEC_RED is undefined");
+ LOG(LS_WARNING) << " WEBRTC_CODEC_RED is undefined";
return -1;
#endif
}
@@ -976,8 +966,7 @@
// Get current playout frequency.
int AudioCodingModuleImpl::PlayoutFrequency() const {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
- "PlayoutFrequency()");
+ LOG(LS_VERBOSE) << "PlayoutFrequency()";
return receiver_.last_output_sample_rate_hz();
}
@@ -1102,8 +1091,7 @@
// Minimum playout delay (Used for lip-sync).
int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
if ((time_ms < 0) || (time_ms > 10000)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Delay must be in the range of 0-1000 milliseconds.");
+ LOG(LS_ERROR) << "Delay must be in the range of 0-1000 milliseconds.";
return -1;
}
return receiver_.SetMinimumDelay(time_ms);
@@ -1111,8 +1099,7 @@
int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
if ((time_ms < 0) || (time_ms > 10000)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "Delay must be in the range of 0-1000 milliseconds.");
+ LOG(LS_ERROR) << "Delay must be in the range of 0-1000 milliseconds.";
return -1;
}
return receiver_.SetMaximumDelay(time_ms);
@@ -1125,8 +1112,7 @@
bool* muted) {
// GetAudio always returns 10 ms, at the requested sample rate.
if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "PlayoutData failed, RecOut Failed");
+ LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
return -1;
}
audio_frame->id_ = id_;
@@ -1153,8 +1139,7 @@
}
int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
- "RegisterVADCallback()");
+ LOG(LS_VERBOSE) << "RegisterVADCallback()";
rtc::CritScope lock(&callback_crit_sect_);
vad_callback_ = vad_callback;
return 0;
@@ -1253,8 +1238,7 @@
bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
if (!encoder_stack_) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
- "%s failed: No send codec is registered.", caller_name);
+ LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";
return false;
}
return true;
@@ -1378,8 +1362,7 @@
bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
bool valid = acm2::RentACodec::IsCodecValid(codec);
if (!valid)
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
- "Invalid codec setting");
+ LOG(LS_ERROR) << "Invalid codec setting";
return valid;
}
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.cc b/webrtc/modules/audio_coding/acm2/codec_manager.cc
index 734358a..e218080 100644
--- a/webrtc/modules/audio_coding/acm2/codec_manager.cc
+++ b/webrtc/modules/audio_coding/acm2/codec_manager.cc
@@ -11,9 +11,9 @@
#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
#include "webrtc/base/checks.h"
-#include "webrtc/base/format_macros.h"
+//#include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -23,34 +23,29 @@
// Check if the given codec is a valid to be registered as send codec.
int IsValidSendCodec(const CodecInst& send_codec) {
- int dummy_id = 0;
if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
- "Wrong number of channels (%" PRIuS ", only mono and stereo "
- "are supported)",
- send_codec.channels);
+ LOG(LS_ERROR) << "Wrong number of channels (" << send_codec.channels
+ << "), only mono and stereo are supported)";
return -1;
}
auto maybe_codec_id = RentACodec::CodecIdByInst(send_codec);
if (!maybe_codec_id) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
- "Invalid codec setting for the send codec.");
+ LOG(LS_ERROR) << "Invalid codec setting for the send codec.";
return -1;
}
// Telephone-event cannot be a send codec.
if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
- "telephone-event cannot be a send codec");
+ LOG(LS_ERROR) << "telephone-event cannot be a send codec";
return -1;
}
if (!RentACodec::IsSupportedNumChannels(*maybe_codec_id, send_codec.channels)
.value_or(false)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
- "%" PRIuS " number of channels not supportedn for %s.",
- send_codec.channels, send_codec.plname);
+ LOG(LS_ERROR) << send_codec.channels
+ << " number of channels not supported for "
+ << send_codec.plname << ".";
return -1;
}
return RentACodec::CodecIndexFromId(*maybe_codec_id).value_or(-1);
@@ -81,15 +76,13 @@
return false;
}
- int dummy_id = 0;
switch (RentACodec::RegisterRedPayloadType(
&codec_stack_params_.red_payload_types, send_codec)) {
case RentACodec::RegistrationResult::kOk:
return true;
case RentACodec::RegistrationResult::kBadFreq:
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
- "RegisterSendCodec() failed, invalid frequency for RED"
- " registration");
+ LOG(LS_ERROR) << "RegisterSendCodec() failed, invalid frequency for RED"
+ " registration";
return false;
case RentACodec::RegistrationResult::kSkip:
break;
@@ -99,9 +92,8 @@
case RentACodec::RegistrationResult::kOk:
return true;
case RentACodec::RegistrationResult::kBadFreq:
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
- "RegisterSendCodec() failed, invalid frequency for CNG"
- " registration");
+ LOG(LS_ERROR) << "RegisterSendCodec() failed, invalid frequency for CNG"
+ " registration";
return false;
case RentACodec::RegistrationResult::kSkip:
break;
@@ -135,15 +127,14 @@
bool CodecManager::SetCopyRed(bool enable) {
if (enable && codec_stack_params_.use_codec_fec) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
- "Codec internal FEC and RED cannot be co-enabled.");
+ LOG(LS_WARNING) << "Codec internal FEC and RED cannot be co-enabled.";
return false;
}
if (enable && send_codec_inst_ &&
codec_stack_params_.red_payload_types.count(send_codec_inst_->plfreq) <
1) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
- "Cannot enable RED at %i Hz.", send_codec_inst_->plfreq);
+ LOG(LS_WARNING) << "Cannot enable RED at " << send_codec_inst_->plfreq
+ << " Hz.";
return false;
}
codec_stack_params_.use_red = enable;
@@ -162,8 +153,7 @@
? (codec_stack_params_.speech_encoder->NumChannels() != 1)
: false;
if (enable && stereo_send) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
- "VAD/DTX not supported for stereo sending");
+ LOG(LS_ERROR) << "VAD/DTX not supported for stereo sending";
return false;
}
@@ -181,8 +171,7 @@
bool CodecManager::SetCodecFEC(bool enable_codec_fec) {
if (enable_codec_fec && codec_stack_params_.use_red) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
- "Codec internal FEC and RED cannot be co-enabled.");
+ LOG(LS_WARNING) << "Codec internal FEC and RED cannot be co-enabled.";
return false;
}