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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/agc/agc_manager_direct.h"
#include <cmath>
#ifdef WEBRTC_AGC_DEBUG_DUMP
#include <cstdio>
#endif
#include "modules/audio_processing/agc/gain_map_internal.h"
#include "modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h"
#include "modules/audio_processing/include/gain_control.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
int AgcManagerDirect::instance_counter_ = 0;
namespace {
// Amount the microphone level is lowered with every clipping event.
const int kClippedLevelStep = 15;
// Proportion of clipped samples required to declare a clipping event.
const float kClippedRatioThreshold = 0.1f;
// Time in frames to wait after a clipping event before checking again.
const int kClippedWaitFrames = 300;
// Amount of error we tolerate in the microphone level (presumably due to OS
// quantization) before we assume the user has manually adjusted the microphone.
const int kLevelQuantizationSlack = 25;
const int kDefaultCompressionGain = 7;
const int kMaxCompressionGain = 12;
const int kMinCompressionGain = 2;
// Controls the rate of compression changes towards the target.
const float kCompressionGainStep = 0.05f;
const int kMaxMicLevel = 255;
static_assert(kGainMapSize > kMaxMicLevel, "gain map too small");
const int kMinMicLevel = 12;
// Prevent very large microphone level changes.
const int kMaxResidualGainChange = 15;
// Maximum additional gain allowed to compensate for microphone level
// restrictions from clipping events.
const int kSurplusCompressionGain = 6;
int ClampLevel(int mic_level) {
return rtc::SafeClamp(mic_level, kMinMicLevel, kMaxMicLevel);
}
int LevelFromGainError(int gain_error, int level) {
RTC_DCHECK_GE(level, 0);
RTC_DCHECK_LE(level, kMaxMicLevel);
if (gain_error == 0) {
return level;
}
// TODO(ajm): Could be made more efficient with a binary search.
int new_level = level;
if (gain_error > 0) {
while (kGainMap[new_level] - kGainMap[level] < gain_error &&
new_level < kMaxMicLevel) {
++new_level;
}
} else {
while (kGainMap[new_level] - kGainMap[level] > gain_error &&
new_level > kMinMicLevel) {
--new_level;
}
}
return new_level;
}
int InitializeGainControl(GainControl* gain_control,
bool disable_digital_adaptive) {
if (gain_control->set_mode(GainControl::kFixedDigital) != 0) {
RTC_LOG(LS_ERROR) << "set_mode(GainControl::kFixedDigital) failed.";
return -1;
}
const int target_level_dbfs = disable_digital_adaptive ? 0 : 2;
if (gain_control->set_target_level_dbfs(target_level_dbfs) != 0) {
RTC_LOG(LS_ERROR) << "set_target_level_dbfs() failed.";
return -1;
}
const int compression_gain_db =
disable_digital_adaptive ? 0 : kDefaultCompressionGain;
if (gain_control->set_compression_gain_db(compression_gain_db) != 0) {
RTC_LOG(LS_ERROR) << "set_compression_gain_db() failed.";
return -1;
}
const bool enable_limiter = !disable_digital_adaptive;
if (gain_control->enable_limiter(enable_limiter) != 0) {
RTC_LOG(LS_ERROR) << "enable_limiter() failed.";
return -1;
}
return 0;
}
} // namespace
// Facility for dumping debug audio files. All methods are no-ops in the
// default case where WEBRTC_AGC_DEBUG_DUMP is undefined.
class DebugFile {
#ifdef WEBRTC_AGC_DEBUG_DUMP
public:
explicit DebugFile(const char* filename) : file_(fopen(filename, "wb")) {
RTC_DCHECK(file_);
}
~DebugFile() { fclose(file_); }
void Write(const int16_t* data, size_t length_samples) {
fwrite(data, 1, length_samples * sizeof(int16_t), file_);
}
private:
FILE* file_;
#else
public:
explicit DebugFile(const char* filename) {}
~DebugFile() {}
void Write(const int16_t* data, size_t length_samples) {}
#endif // WEBRTC_AGC_DEBUG_DUMP
};
AgcManagerDirect::AgcManagerDirect(GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level,
int clipped_level_min,
bool use_agc2_level_estimation,
bool disable_digital_adaptive)
: AgcManagerDirect(use_agc2_level_estimation ? nullptr : new Agc(),
gctrl,
volume_callbacks,
startup_min_level,
clipped_level_min,
use_agc2_level_estimation,
disable_digital_adaptive) {
RTC_DCHECK(agc_);
}
AgcManagerDirect::AgcManagerDirect(Agc* agc,
GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level,
int clipped_level_min)
: AgcManagerDirect(agc,
gctrl,
volume_callbacks,
startup_min_level,
clipped_level_min,
false,
false) {
RTC_DCHECK(agc_);
}
AgcManagerDirect::AgcManagerDirect(Agc* agc,
GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level,
int clipped_level_min,
bool use_agc2_level_estimation,
bool disable_digital_adaptive)
: data_dumper_(new ApmDataDumper(instance_counter_)),
agc_(agc),
gctrl_(gctrl),
volume_callbacks_(volume_callbacks),
frames_since_clipped_(kClippedWaitFrames),
level_(0),
max_level_(kMaxMicLevel),
max_compression_gain_(kMaxCompressionGain),
target_compression_(kDefaultCompressionGain),
compression_(target_compression_),
compression_accumulator_(compression_),
capture_muted_(false),
check_volume_on_next_process_(true), // Check at startup.
startup_(true),
use_agc2_level_estimation_(use_agc2_level_estimation),
disable_digital_adaptive_(disable_digital_adaptive),
startup_min_level_(ClampLevel(startup_min_level)),
clipped_level_min_(clipped_level_min),
file_preproc_(new DebugFile("agc_preproc.pcm")),
file_postproc_(new DebugFile("agc_postproc.pcm")) {
instance_counter_++;
if (use_agc2_level_estimation_) {
RTC_DCHECK(!agc);
agc_.reset(new AdaptiveModeLevelEstimatorAgc(data_dumper_.get()));
} else {
RTC_DCHECK(agc);
}
}
AgcManagerDirect::~AgcManagerDirect() {}
int AgcManagerDirect::Initialize() {
max_level_ = kMaxMicLevel;
max_compression_gain_ = kMaxCompressionGain;
target_compression_ = disable_digital_adaptive_ ? 0 : kDefaultCompressionGain;
compression_ = disable_digital_adaptive_ ? 0 : target_compression_;
compression_accumulator_ = compression_;
capture_muted_ = false;
check_volume_on_next_process_ = true;
// TODO(bjornv): Investigate if we need to reset |startup_| as well. For
// example, what happens when we change devices.
data_dumper_->InitiateNewSetOfRecordings();
return InitializeGainControl(gctrl_, disable_digital_adaptive_);
}
void AgcManagerDirect::AnalyzePreProcess(int16_t* audio,
int num_channels,
size_t samples_per_channel) {
size_t length = num_channels * samples_per_channel;
if (capture_muted_) {
return;
}
file_preproc_->Write(audio, length);
if (frames_since_clipped_ < kClippedWaitFrames) {
++frames_since_clipped_;
return;
}
// Check for clipped samples, as the AGC has difficulty detecting pitch
// under clipping distortion. We do this in the preprocessing phase in order
// to catch clipped echo as well.
//
// If we find a sufficiently clipped frame, drop the current microphone level
// and enforce a new maximum level, dropped the same amount from the current
// maximum. This harsh treatment is an effort to avoid repeated clipped echo
// events. As compensation for this restriction, the maximum compression
// gain is increased, through SetMaxLevel().
float clipped_ratio = agc_->AnalyzePreproc(audio, length);
if (clipped_ratio > kClippedRatioThreshold) {
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));
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.AgcClippingAdjustmentAllowed",
level_ - kClippedLevelStep >= clipped_level_min_);
if (level_ > clipped_level_min_) {
// Don't try to adjust the level if we're already below the limit. As
// a consequence, if the user has brought the level above the limit, we
// will still not react until the postproc updates the level.
SetLevel(std::max(clipped_level_min_, level_ - kClippedLevelStep));
// Reset the AGC since the level has changed.
agc_->Reset();
}
frames_since_clipped_ = 0;
}
}
void AgcManagerDirect::Process(const int16_t* audio,
size_t length,
int sample_rate_hz) {
if (capture_muted_) {
return;
}
if (check_volume_on_next_process_) {
check_volume_on_next_process_ = false;
// We have to wait until the first process call to check the volume,
// because Chromium doesn't guarantee it to be valid any earlier.
CheckVolumeAndReset();
}
agc_->Process(audio, length, sample_rate_hz);
UpdateGain();
if (!disable_digital_adaptive_) {
UpdateCompressor();
}
file_postproc_->Write(audio, length);
data_dumper_->DumpRaw("experimental_gain_control_compression_gain_db", 1,
&compression_);
}
void AgcManagerDirect::SetLevel(int new_level) {
int voe_level = volume_callbacks_->GetMicVolume();
if (voe_level == 0) {
RTC_DLOG(LS_INFO)
<< "[agc] VolumeCallbacks returned level=0, taking no action.";
return;
}
if (voe_level < 0 || voe_level > kMaxMicLevel) {
RTC_LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level="
<< voe_level;
return;
}
if (voe_level > level_ + kLevelQuantizationSlack ||
voe_level < level_ - kLevelQuantizationSlack) {
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_) {
SetMaxLevel(level_);
}
// Take no action in this case, since we can't be sure when the volume
// was manually adjusted. The compressor will still provide some of the
// desired gain change.
agc_->Reset();
return;
}
new_level = std::min(new_level, max_level_);
if (new_level == level_) {
return;
}
volume_callbacks_->SetMicVolume(new_level);
RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
<< "level_=" << level_ << ", "
<< "new_level=" << new_level;
level_ = new_level;
}
void AgcManagerDirect::SetMaxLevel(int level) {
RTC_DCHECK_GE(level, clipped_level_min_);
max_level_ = level;
// Scale the |kSurplusCompressionGain| linearly across the restricted
// level range.
max_compression_gain_ =
kMaxCompressionGain + std::floor((1.f * kMaxMicLevel - max_level_) /
(kMaxMicLevel - clipped_level_min_) *
kSurplusCompressionGain +
0.5f);
RTC_DLOG(LS_INFO) << "[agc] max_level_=" << max_level_
<< ", max_compression_gain_=" << max_compression_gain_;
}
void AgcManagerDirect::SetCaptureMuted(bool muted) {
if (capture_muted_ == muted) {
return;
}
capture_muted_ = muted;
if (!muted) {
// When we unmute, we should reset things to be safe.
check_volume_on_next_process_ = true;
}
}
float AgcManagerDirect::voice_probability() {
return agc_->voice_probability();
}
int AgcManagerDirect::CheckVolumeAndReset() {
int level = volume_callbacks_->GetMicVolume();
// 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_DLOG(LS_INFO)
<< "[agc] VolumeCallbacks returned level=0, taking no action.";
return 0;
}
if (level < 0 || level > kMaxMicLevel) {
RTC_LOG(LS_ERROR) << "[agc] VolumeCallbacks returned an invalid level="
<< level;
return -1;
}
RTC_DLOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
int minLevel = startup_ ? startup_min_level_ : kMinMicLevel;
if (level < minLevel) {
level = minLevel;
RTC_DLOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
volume_callbacks_->SetMicVolume(level);
}
agc_->Reset();
level_ = level;
startup_ = false;
return 0;
}
// Requests the RMS error from AGC and distributes the required gain change
// between the digital compression stage and volume slider. We use the
// compressor first, providing a slack region around the current slider
// position to reduce movement.
//
// If the slider needs to be moved, we check first if the user has adjusted
// it, in which case we take no action and cache the updated level.
void AgcManagerDirect::UpdateGain() {
int rms_error = 0;
if (!agc_->GetRmsErrorDb(&rms_error)) {
// No error update ready.
return;
}
// The compressor will always add at least kMinCompressionGain. In effect,
// this adjusts our target gain upward by the same amount and rms_error
// needs to reflect that.
rms_error += kMinCompressionGain;
// Handle as much error as possible with the compressor first.
int raw_compression =
rtc::SafeClamp(rms_error, kMinCompressionGain, max_compression_gain_);
// Deemphasize the compression gain error. Move halfway between the current
// target and the newly received target. This serves to soften perceptible
// intra-talkspurt adjustments, at the cost of some adaptation speed.
if ((raw_compression == max_compression_gain_ &&
target_compression_ == max_compression_gain_ - 1) ||
(raw_compression == kMinCompressionGain &&
target_compression_ == kMinCompressionGain + 1)) {
// Special case to allow the target to reach the endpoints of the
// compression range. The deemphasis would otherwise halt it at 1 dB shy.
target_compression_ = raw_compression;
} else {
target_compression_ =
(raw_compression - target_compression_) / 2 + target_compression_;
}
// Residual error will be handled by adjusting the volume slider. Use the
// raw rather than deemphasized compression here as we would otherwise
// shrink the amount of slack the compressor provides.
const int residual_gain =
rtc::SafeClamp(rms_error - raw_compression, -kMaxResidualGainChange,
kMaxResidualGainChange);
RTC_DLOG(LS_INFO) << "[agc] rms_error=" << rms_error
<< ", target_compression=" << target_compression_
<< ", residual_gain=" << residual_gain;
if (residual_gain == 0)
return;
int old_level = level_;
SetLevel(LevelFromGainError(residual_gain, level_));
if (old_level != level_) {
// level_ was updated by SetLevel; log the new value.
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.AgcSetLevel", level_, 1,
kMaxMicLevel, 50);
// Reset the AGC since the level has changed.
agc_->Reset();
}
}
void AgcManagerDirect::UpdateCompressor() {
calls_since_last_gain_log_++;
if (calls_since_last_gain_log_ == 100) {
calls_since_last_gain_log_ = 0;
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainApplied",
compression_, 0, kMaxCompressionGain,
kMaxCompressionGain + 1);
}
if (compression_ == target_compression_) {
return;
}
// Adapt the compression gain slowly towards the target, in order to avoid
// highly perceptible changes.
if (target_compression_ > compression_) {
compression_accumulator_ += kCompressionGainStep;
} else {
compression_accumulator_ -= kCompressionGainStep;
}
// The compressor accepts integer gains in dB. Adjust the gain when
// we've come within half a stepsize of the nearest integer. (We don't
// check for equality due to potential floating point imprecision).
int new_compression = compression_;
int nearest_neighbor = std::floor(compression_accumulator_ + 0.5);
if (std::fabs(compression_accumulator_ - nearest_neighbor) <
kCompressionGainStep / 2) {
new_compression = nearest_neighbor;
}
// Set the new compression gain.
if (new_compression != compression_) {
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainUpdated",
new_compression, 0, kMaxCompressionGain,
kMaxCompressionGain + 1);
compression_ = new_compression;
compression_accumulator_ = new_compression;
if (gctrl_->set_compression_gain_db(compression_) != 0) {
RTC_LOG(LS_ERROR) << "set_compression_gain_db(" << compression_
<< ") failed.";
}
}
}
} // namespace webrtc