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03ad9b892c
The FixedGainController is used in two places. One is the AudioMixer. There it's used to limit the audio level after adding streams. The other is GainController2, where it's placed after steps that could boost the audio level outside the allowed range. We log metrics from the FGC. To avoid confusion, this CL makes the two use cases log to different histograms. Chromium histogram CL is https://chromium-review.googlesource.com/c/chromium/src/+/1170833 Bug: webrtc:7494 Change-Id: I1abe60fd8e96556f144d2ee576254b15beca1174 Reviewed-on: https://webrtc-review.googlesource.com/93464 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24284}
95 lines
3.5 KiB
C++
95 lines
3.5 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/agc2/fixed_gain_controller.h"
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#include <algorithm>
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#include <cmath>
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#include "api/array_view.h"
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#include "common_audio/include/audio_util.h"
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_minmax.h"
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namespace webrtc {
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namespace {
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// Returns true when the gain factor is so close to 1 that it would
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// not affect int16 samples.
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bool CloseToOne(float gain_factor) {
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return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
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gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
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}
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} // namespace
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FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper)
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: FixedGainController(apm_data_dumper, "Agc2") {}
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FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper,
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std::string histogram_name_prefix)
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: apm_data_dumper_(apm_data_dumper),
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gain_curve_applier_(48000, apm_data_dumper_, histogram_name_prefix) {
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// Do update histograms.xml when adding name prefixes.
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RTC_DCHECK(histogram_name_prefix == "" || histogram_name_prefix == "Test" ||
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histogram_name_prefix == "AudioMixer" ||
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histogram_name_prefix == "Agc2");
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}
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void FixedGainController::SetGain(float gain_to_apply_db) {
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// Changes in gain_to_apply_ cause discontinuities. We assume
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// gain_to_apply_ is set in the beginning of the call. If it is
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// frequently changed, we should add interpolation between the
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// values.
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// The gain
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RTC_DCHECK_LE(-50.f, gain_to_apply_db);
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RTC_DCHECK_LE(gain_to_apply_db, 50.f);
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gain_to_apply_ = DbToRatio(gain_to_apply_db);
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RTC_DCHECK_LT(0.f, gain_to_apply_);
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RTC_DLOG(LS_INFO) << "Gain to apply: " << gain_to_apply_db << " db.";
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}
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void FixedGainController::SetSampleRate(size_t sample_rate_hz) {
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gain_curve_applier_.SetSampleRate(sample_rate_hz);
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}
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void FixedGainController::Process(AudioFrameView<float> signal) {
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// Apply fixed digital gain. One of the
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// planned usages of the FGC is to only use the limiter. In that
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// case, the gain would be 1.0. Not doing the multiplications speeds
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// it up considerably. Hence the check.
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if (!CloseToOne(gain_to_apply_)) {
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for (size_t k = 0; k < signal.num_channels(); ++k) {
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rtc::ArrayView<float> channel_view = signal.channel(k);
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for (auto& sample : channel_view) {
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sample *= gain_to_apply_;
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}
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}
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}
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// Use the limiter.
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gain_curve_applier_.Process(signal);
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// Dump data for debug.
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const auto channel_view = signal.channel(0);
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apm_data_dumper_->DumpRaw("agc2_fixed_digital_gain_curve_applier",
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channel_view.size(), channel_view.data());
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// Hard-clipping.
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for (size_t k = 0; k < signal.num_channels(); ++k) {
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rtc::ArrayView<float> channel_view = signal.channel(k);
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for (auto& sample : channel_view) {
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sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
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}
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}
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}
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} // namespace webrtc
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