mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-12 17:17:24 +01:00
5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
102 lines
3.7 KiB
C++
102 lines
3.7 KiB
C++
/*
|
|
* Copyright (c) 2018 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/agc2/fixed_gain_controller.h"
|
|
|
|
#include "api/array_view.h"
|
|
#include "common_audio/include/audio_util.h"
|
|
#include "modules/audio_processing/agc2/agc2_common.h"
|
|
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/logging.h"
|
|
#include "rtc_base/numerics/safe_minmax.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
// Returns true when the gain factor is so close to 1 that it would
|
|
// not affect int16 samples.
|
|
bool CloseToOne(float gain_factor) {
|
|
return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
|
|
gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
|
|
}
|
|
} // namespace
|
|
|
|
FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper)
|
|
: FixedGainController(apm_data_dumper, "Agc2") {}
|
|
|
|
FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper,
|
|
std::string histogram_name_prefix)
|
|
: apm_data_dumper_(apm_data_dumper),
|
|
limiter_(48000, apm_data_dumper_, histogram_name_prefix) {
|
|
// Do update histograms.xml when adding name prefixes.
|
|
RTC_DCHECK(histogram_name_prefix == "" || histogram_name_prefix == "Test" ||
|
|
histogram_name_prefix == "AudioMixer" ||
|
|
histogram_name_prefix == "Agc2");
|
|
}
|
|
|
|
void FixedGainController::SetGain(float gain_to_apply_db) {
|
|
// Changes in gain_to_apply_ cause discontinuities. We assume
|
|
// gain_to_apply_ is set in the beginning of the call. If it is
|
|
// frequently changed, we should add interpolation between the
|
|
// values.
|
|
// The gain
|
|
RTC_DCHECK_LE(-50.f, gain_to_apply_db);
|
|
RTC_DCHECK_LE(gain_to_apply_db, 50.f);
|
|
const float previous_applied_gained = gain_to_apply_;
|
|
gain_to_apply_ = DbToRatio(gain_to_apply_db);
|
|
RTC_DCHECK_LT(0.f, gain_to_apply_);
|
|
RTC_DLOG(LS_INFO) << "Gain to apply: " << gain_to_apply_db << " db.";
|
|
// Reset the gain curve applier to quickly react on abrupt level changes
|
|
// caused by large changes of the applied gain.
|
|
if (previous_applied_gained != gain_to_apply_) {
|
|
limiter_.Reset();
|
|
}
|
|
}
|
|
|
|
void FixedGainController::SetSampleRate(size_t sample_rate_hz) {
|
|
limiter_.SetSampleRate(sample_rate_hz);
|
|
}
|
|
|
|
void FixedGainController::Process(AudioFrameView<float> signal) {
|
|
// Apply fixed digital gain. One of the
|
|
// planned usages of the FGC is to only use the limiter. In that
|
|
// case, the gain would be 1.0. Not doing the multiplications speeds
|
|
// it up considerably. Hence the check.
|
|
if (!CloseToOne(gain_to_apply_)) {
|
|
for (size_t k = 0; k < signal.num_channels(); ++k) {
|
|
rtc::ArrayView<float> channel_view = signal.channel(k);
|
|
for (auto& sample : channel_view) {
|
|
sample *= gain_to_apply_;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Use the limiter.
|
|
limiter_.Process(signal);
|
|
|
|
// Dump data for debug.
|
|
const auto channel_view = signal.channel(0);
|
|
apm_data_dumper_->DumpRaw("agc2_fixed_digital_gain_curve_applier",
|
|
channel_view.size(), channel_view.data());
|
|
// Hard-clipping.
|
|
for (size_t k = 0; k < signal.num_channels(); ++k) {
|
|
rtc::ArrayView<float> channel_view = signal.channel(k);
|
|
for (auto& sample : channel_view) {
|
|
sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
|
|
}
|
|
}
|
|
}
|
|
|
|
float FixedGainController::LastAudioLevel() const {
|
|
return limiter_.LastAudioLevel();
|
|
}
|
|
} // namespace webrtc
|