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Remove WEBRTC_TRACE from webrtc/modules/audio_coding

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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}
This commit is contained in:
Alex Loiko
2017-05-30 17:23:28 +02:00
committed by Commit Bot
parent 1d29c86cbf
commit 300ec8c8db
3 changed files with 38 additions and 67 deletions
Download Patch File Download Diff File Expand all files Collapse all files

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

@ -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 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
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 @@ rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
// 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 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
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 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
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 @@ int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
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 @@ int AudioCodingModuleImpl::ReceiveFrequency() const {
// 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 @@ 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)) {
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::SetMinimumPlayoutDelay(int time_ms) {
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 @@ 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) {
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::GetNetworkStatistics(NetworkStatistics* statistics) {
}
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 @@ int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
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 @@ int AudioCodingModule::Codec(const char* payload_name,
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;
}

47
webrtc/modules/audio_coding/acm2/codec_manager.cc
View File

@ -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 @@ namespace {
// 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 @@ bool CodecManager::RegisterEncoder(const CodecInst& send_codec) {
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 @@ bool CodecManager::RegisterEncoder(const CodecInst& 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 CNG"
" registration");
LOG(LS_ERROR) << "RegisterSendCodec() failed, invalid frequency for CNG"
" registration";
return false;
case RentACodec::RegistrationResult::kSkip:
break;
@ -135,15 +127,14 @@ CodecInst CodecManager::ForgeCodecInst(
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 @@ bool CodecManager::SetVAD(bool enable, ACMVADMode mode) {
? (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::SetVAD(bool enable, ACMVADMode mode) {
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;
}

5
webrtc/modules/audio_coding/test/TestAllCodecs.cc
View File

@ -14,12 +14,12 @@
#include <limits>
#include <string>
#include "webrtc/base/logging.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
@ -127,8 +127,7 @@ void TestAllCodecs::Perform() {
infile_a_.Open(file_name, 32000, "rb");
if (test_mode_ == 0) {
WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
"---------- TestAllCodecs ----------");
LOG(LS_INFO) << "---------- TestAllCodecs ----------";
}
acm_a_->InitializeReceiver();