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Streamline error handling and logging in the audio processing module

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Bug: webrtc:8529
Change-Id: I40817d578c2c4106892e564df1bc734efcef5503
Reviewed-on: https://webrtc-review.googlesource.com/52540
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22034}
This commit is contained in:
Jonas Olsson 2018-02-15 15:16:27 +01:00 committed by Commit Bot
parent 2bde85046a
commit 645b027dc4
11 changed files with 71 additions and 83 deletions
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2
modules/audio_processing/aec3/adaptive_fir_filter.cc
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@ -466,9 +466,9 @@ void AdaptiveFirFilter::SetSizePartitions(size_t size) {
RTC_DCHECK_EQ(GetTimeDomainLength(max_size_partitions_), h_.capacity());
RTC_DCHECK_EQ(H_.size(), H2_.size());
RTC_DCHECK_EQ(h_.size(), GetTimeDomainLength(H_.size()));
RTC_DCHECK_LE(size, max_size_partitions_);
if (size > max_size_partitions_) {
RTC_LOG(LS_ERROR) << "Too large adaptive filter size specificed: " << size;
size = max_size_partitions_;
}

3
modules/audio_processing/agc/agc.cc
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@ -48,7 +48,7 @@ float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
return 1.0f * num_clipped / length;
}
int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
void Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
vad_.ProcessChunk(audio, length, sample_rate_hz);
const std::vector<double>& rms = vad_.chunkwise_rms();
const std::vector<double>& probabilities =
@ -57,7 +57,6 @@ int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
for (size_t i = 0; i < rms.size(); ++i) {
histogram_->Update(rms[i], probabilities[i]);
}
return 0;
}
bool Agc::GetRmsErrorDb(int* error) {

2
modules/audio_processing/agc/agc.h
View File

@ -31,7 +31,7 @@ class Agc {
virtual float AnalyzePreproc(const int16_t* audio, size_t length);
// |audio| must be mono; in a multi-channel stream, provide the first (usually
// left) channel.
virtual int Process(const int16_t* audio, size_t length, int sample_rate_hz);
virtual void Process(const int16_t* audio, size_t length, int sample_rate_hz);
// Retrieves the difference between the target RMS level and the current
// signal RMS level in dB. Returns true if an update is available and false

50
modules/audio_processing/agc/agc_manager_direct.cc
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@ -216,8 +216,8 @@ void AgcManagerDirect::AnalyzePreProcess(int16_t* audio,
// gain is increased, through SetMaxLevel().
float clipped_ratio = agc_->AnalyzePreproc(audio, length);
if (clipped_ratio > kClippedRatioThreshold) {
RTC_LOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
<< clipped_ratio;
RTC_DLOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
<< clipped_ratio;
// Always decrease the maximum level, even if the current level is below
// threshold.
SetMaxLevel(std::max(clipped_level_min_, max_level_ - kClippedLevelStep));
@ -249,10 +249,7 @@ void AgcManagerDirect::Process(const int16_t* audio,
CheckVolumeAndReset();
}
if (agc_->Process(audio, length, sample_rate_hz) != 0) {
RTC_LOG(LS_ERROR) << "Agc::Process failed";
RTC_NOTREACHED();
}
agc_->Process(audio, length, sample_rate_hz);
UpdateGain();
UpdateCompressor();
@ -262,15 +259,12 @@ void AgcManagerDirect::Process(const int16_t* audio,
void AgcManagerDirect::SetLevel(int new_level) {
int voe_level = volume_callbacks_->GetMicVolume();
if (voe_level < 0) {
return;
}
if (voe_level == 0) {
RTC_LOG(LS_INFO)
RTC_DLOG(LS_INFO)
<< "[agc] VolumeCallbacks returned level=0, taking no action.";
return;
}
if (voe_level > kMaxMicLevel) {
if (voe_level < 0 || voe_level > kMaxMicLevel) {
RTC_LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level="
<< voe_level;
return;
@ -278,8 +272,8 @@ void AgcManagerDirect::SetLevel(int new_level) {
if (voe_level > level_ + kLevelQuantizationSlack ||
voe_level < level_ - kLevelQuantizationSlack) {
RTC_LOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating "
<< "stored level from " << level_ << " to " << voe_level;
RTC_DLOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating "
"stored level from " << level_ << " to " << voe_level;
level_ = voe_level;
// Always allow the user to increase the volume.
if (level_ > max_level_) {
@ -298,9 +292,9 @@ void AgcManagerDirect::SetLevel(int new_level) {
}
volume_callbacks_->SetMicVolume(new_level);
RTC_LOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
<< "level_=" << level_ << ", "
<< "new_level=" << new_level;
RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
<< "level_=" << level_ << ", "
<< "new_level=" << new_level;
level_ = new_level;
}
@ -314,8 +308,8 @@ void AgcManagerDirect::SetMaxLevel(int level) {
(kMaxMicLevel - clipped_level_min_) *
kSurplusCompressionGain +
0.5f);
RTC_LOG(LS_INFO) << "[agc] max_level_=" << max_level_
<< ", max_compression_gain_=" << max_compression_gain_;
RTC_DLOG(LS_INFO) << "[agc] max_level_=" << max_level_
<< ", max_compression_gain_=" << max_compression_gain_;
}
void AgcManagerDirect::SetCaptureMuted(bool muted) {
@ -336,28 +330,26 @@ float AgcManagerDirect::voice_probability() {
int AgcManagerDirect::CheckVolumeAndReset() {
int level = volume_callbacks_->GetMicVolume();
if (level < 0) {
return -1;
}
// Reasons for taking action at startup:
// 1) A person starting a call is expected to be heard.
// 2) Independent of interpretation of |level| == 0 we should raise it so the
// AGC can do its job properly.
if (level == 0 && !startup_) {
RTC_LOG(LS_INFO)
RTC_DLOG(LS_INFO)
<< "[agc] VolumeCallbacks returned level=0, taking no action.";
return 0;
}
if (level > kMaxMicLevel) {
RTC_LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level=" << level;
if (level < 0 || level > kMaxMicLevel) {
RTC_LOG(LS_ERROR) << "[agc] VolumeCallbacks returned an invalid level="
<< level;
return -1;
}
RTC_LOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
RTC_DLOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
int minLevel = startup_ ? startup_min_level_ : kMinMicLevel;
if (level < minLevel) {
level = minLevel;
RTC_LOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
RTC_DLOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
volume_callbacks_->SetMicVolume(level);
}
agc_->Reset();
@ -409,9 +401,9 @@ void AgcManagerDirect::UpdateGain() {
const int residual_gain =
rtc::SafeClamp(rms_error - raw_compression, -kMaxResidualGainChange,
kMaxResidualGainChange);
RTC_LOG(LS_INFO) << "[agc] rms_error=" << rms_error << ", "
<< "target_compression=" << target_compression_ << ", "
<< "residual_gain=" << residual_gain;
RTC_DLOG(LS_INFO) << "[agc] rms_error=" << rms_error
<< ", target_compression=" << target_compression_
<< ", residual_gain=" << residual_gain;
if (residual_gain == 0)
return;

2
modules/audio_processing/agc/agc_manager_direct_unittest.cc
View File

@ -77,7 +77,7 @@ class AgcManagerDirectTest : public ::testing::Test {
void CallProcess(int num_calls) {
for (int i = 0; i < num_calls; ++i) {
EXPECT_CALL(*agc_, Process(_, _, _)).WillOnce(Return(0));
EXPECT_CALL(*agc_, Process(_, _, _)).WillOnce(Return());
manager_.Process(nullptr, kSamplesPerChannel, kSampleRateHz);
}
}

4
modules/audio_processing/agc/mock_agc.h
View File

@ -21,8 +21,8 @@ namespace webrtc {
class MockAgc : public Agc {
public:
MOCK_METHOD2(AnalyzePreproc, float(const int16_t* audio, size_t length));
MOCK_METHOD3(Process, int(const int16_t* audio, size_t length,
int sample_rate_hz));
MOCK_METHOD3(Process, void(const int16_t* audio, size_t length,
int sample_rate_hz));
MOCK_METHOD1(GetRmsErrorDb, bool(int* error));
MOCK_METHOD0(Reset, void());
MOCK_METHOD1(set_target_level_dbfs, int(int level));

19
modules/audio_processing/audio_processing_impl.cc
View File

@ -449,9 +449,8 @@ AudioProcessingImpl::AudioProcessingImpl(
private_submodules_->gain_controller2.reset(new GainController2());
RTC_LOG(LS_INFO) << "Capture post processor activated: "
<< !!private_submodules_->capture_post_processor;
RTC_LOG(LS_INFO) << "Render pre processor activated: "
<< !!private_submodules_->capture_post_processor
<< "\nRender pre processor activated: "
<< !!private_submodules_->render_pre_processor;
}
@ -709,11 +708,10 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
bool config_ok = LevelController::Validate(config_.level_controller);
if (!config_ok) {
RTC_LOG(LS_ERROR) << "AudioProcessing module config error" << std::endl
<< "level_controller: "
RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n"
"level_controller: "
<< LevelController::ToString(config_.level_controller)
<< std::endl
<< "Reverting to default parameter set";
<< "\nReverting to default parameter set";
config_.level_controller = AudioProcessing::Config::LevelController();
}
@ -743,11 +741,10 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
config_ok = GainController2::Validate(config_.gain_controller2);
if (!config_ok) {
RTC_LOG(LS_ERROR) << "AudioProcessing module config error" << std::endl
<< "Gain Controller 2: "
RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n"
"Gain Controller 2: "
<< GainController2::ToString(config_.gain_controller2)
<< std::endl
<< "Reverting to default parameter set";
<< "\nReverting to default parameter set";
config_.gain_controller2 = AudioProcessing::Config::GainController2();
}
InitializeGainController2();

6
modules/audio_processing/echo_control_mobile_impl.cc
View File

@ -353,9 +353,9 @@ void EchoControlMobileImpl::Initialize(int sample_rate_hz,
return;
}
if (stream_properties_->sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
RTC_LOG(LS_ERROR) << "AECM only supports 16 kHz or lower sample rates";
}
// AECM only supports 16 kHz or lower sample rates.
RTC_DCHECK_LE(stream_properties_->sample_rate_hz,
AudioProcessing::kSampleRate16kHz);
cancellers_.resize(
NumCancellersRequired(stream_properties_->num_output_channels,

32
modules/audio_processing/level_controller/level_controller.cc
View File

@ -156,21 +156,23 @@ void LevelController::Metrics::Update(float long_term_peak_level,
const int frame_peak_level_dbfs = static_cast<int>(
10 * log10(frame_peak_level * frame_peak_level + 1e-10f) - kdBFSOffset);
RTC_LOG(LS_INFO) << "Level Controller metrics: {"
<< "Max noise power: " << max_noise_power_dbfs << " dBFS, "
<< "Average noise power: " << average_noise_power_dbfs
<< " dBFS, "
<< "Max long term peak level: " << max_peak_level_dbfs
<< " dBFS, "
<< "Average long term peak level: "
<< average_peak_level_dbfs << " dBFS, "
<< "Max gain: " << max_gain_db << " dB, "
<< "Average gain: " << average_gain_db << " dB, "
<< "Long term peak level: " << long_term_peak_level_dbfs
<< " dBFS, "
<< "Last frame peak level: " << frame_peak_level_dbfs
<< " dBFS"
<< "}";
RTC_LOG(LS_INFO) << "Level Controller metrics: {Max noise power: "
<< max_noise_power_dbfs
<< " dBFS, Average noise power: "
<< average_noise_power_dbfs
<< " dBFS, Max long term peak level: "
<< max_peak_level_dbfs
<< " dBFS, Average long term peak level: "
<< average_peak_level_dbfs
<< " dBFS, Max gain: "
<< max_gain_db
<< " dB, Average gain: "
<< average_gain_db
<< " dB, Long term peak level: "
<< long_term_peak_level_dbfs
<< " dBFS, Last frame peak level: "
<< frame_peak_level_dbfs
<< " dBFS}";
Reset();
}

26
modules/audio_processing/residual_echo_detector.cc
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@ -141,19 +141,19 @@ void ResidualEchoDetector::AnalyzeCaptureAudio(
read_index -= kLookbackFrames;
}
RTC_DCHECK_LT(read_index, render_power_.size());
RTC_LOG_F(LS_ERROR) << "Echo detector internal state: {"
<< "Echo likelihood: " << echo_likelihood_
<< ", Best Delay: " << best_delay << ", Covariance: "
<< covariances_[best_delay].covariance()
<< ", Last capture power: " << capture_power
<< ", Capture mean: " << capture_mean
<< ", Capture_standard deviation: "
<< capture_std_deviation << ", Last render power: "
<< render_power_[read_index]
<< ", Render mean: " << render_power_mean_[read_index]
<< ", Render standard deviation: "
<< render_power_std_dev_[read_index]
<< ", Reliability: " << reliability_ << "}";
RTC_LOG_F(LS_ERROR)
<< "Echo detector internal state: {"
"Echo likelihood: " << echo_likelihood_
<< ", Best Delay: " << best_delay
<< ", Covariance: " << covariances_[best_delay].covariance()
<< ", Last capture power: " << capture_power
<< ", Capture mean: " << capture_mean
<< ", Capture_standard deviation: " << capture_std_deviation
<< ", Last render power: " << render_power_[read_index]
<< ", Render mean: " << render_power_mean_[read_index]
<< ", Render standard deviation: "
<< render_power_std_dev_[read_index]
<< ", Reliability: " << reliability_ << "}";
log_counter_++;
}
}

8
modules/audio_processing/transient/transient_suppression_test.cc
View File

@ -194,11 +194,9 @@ void void_main() {
detection_buffer.get(),
reference_file,
reference_buffer.get())) {
ASSERT_EQ(0,
agc.Process(audio_buffer_i.get(),
static_cast<int>(audio_buffer_size),
FLAG_sample_rate_hz))
<< "The AGC could not process the frame";
agc.Process(audio_buffer_i.get(),
static_cast<int>(audio_buffer_size),
FLAG_sample_rate_hz);
for (size_t i = 0; i < FLAG_num_channels * audio_buffer_size; ++i) {
audio_buffer_f[i] = audio_buffer_i[i];