blob: a565613d26b989ef2b14a48a9b2e7c2cb7ea3998 [file] [log] [blame]
Alex Loikoe36e8bb2018-02-16 11:54:07 +01001/*
2 * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include "modules/audio_processing/agc2/fixed_gain_controller.h"
12
13#include <algorithm>
14#include <cmath>
15
16#include "api/array_view.h"
17#include "common_audio/include/audio_util.h"
18#include "modules/audio_processing/agc2/agc2_common.h"
Alex Loikoa05ee822018-02-20 15:58:36 +010019#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
Alex Loikoe36e8bb2018-02-16 11:54:07 +010020#include "modules/audio_processing/logging/apm_data_dumper.h"
21#include "rtc_base/checks.h"
22#include "rtc_base/logging.h"
23#include "rtc_base/numerics/safe_minmax.h"
24
25namespace webrtc {
26namespace {
27
28// Returns true when the gain factor is so close to 1 that it would
29// not affect int16 samples.
30bool CloseToOne(float gain_factor) {
Alex Loikoa05ee822018-02-20 15:58:36 +010031 return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
32 gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
Alex Loikoe36e8bb2018-02-16 11:54:07 +010033}
34} // namespace
35
36FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper)
Alex Loikoa05ee822018-02-20 15:58:36 +010037 : apm_data_dumper_(apm_data_dumper),
38 gain_curve_applier_(48000, apm_data_dumper_) {}
Alex Loikoe36e8bb2018-02-16 11:54:07 +010039
40void FixedGainController::SetGain(float gain_to_apply_db) {
41 // Changes in gain_to_apply_ cause discontinuities. We assume
42 // gain_to_apply_ is set in the beginning of the call. If it is
43 // frequently changed, we should add interpolation between the
44 // values.
Alex Loikoa05ee822018-02-20 15:58:36 +010045 // The gain
46 RTC_DCHECK_LE(-50.f, gain_to_apply_db);
47 RTC_DCHECK_LE(gain_to_apply_db, 50.f);
Alex Loikoe36e8bb2018-02-16 11:54:07 +010048 gain_to_apply_ = DbToRatio(gain_to_apply_db);
Alex Loikoa05ee822018-02-20 15:58:36 +010049 RTC_DCHECK_LT(0.f, gain_to_apply_);
50 RTC_DLOG(LS_INFO) << "Gain to apply: " << gain_to_apply_db << " db.";
Alex Loikoe36e8bb2018-02-16 11:54:07 +010051}
52
53void FixedGainController::SetSampleRate(size_t sample_rate_hz) {
Alex Loikoa05ee822018-02-20 15:58:36 +010054 gain_curve_applier_.SetSampleRate(sample_rate_hz);
Alex Loikoe36e8bb2018-02-16 11:54:07 +010055}
56
57void FixedGainController::EnableLimiter(bool enable_limiter) {
58 enable_limiter_ = enable_limiter;
59}
60
61void FixedGainController::Process(AudioFrameView<float> signal) {
62 // Apply fixed digital gain; interpolate if necessary. One of the
63 // planned usages of the FGC is to only use the limiter. In that
64 // case, the gain would be 1.0. Not doing the multiplications speeds
65 // it up considerably. Hence the check.
66 if (!CloseToOne(gain_to_apply_)) {
67 for (size_t k = 0; k < signal.num_channels(); ++k) {
68 rtc::ArrayView<float> channel_view = signal.channel(k);
69 for (auto& sample : channel_view) {
70 sample *= gain_to_apply_;
71 }
72 }
73 }
74
75 // Use the limiter (if configured to).
76 if (enable_limiter_) {
Alex Loikoa05ee822018-02-20 15:58:36 +010077 gain_curve_applier_.Process(signal);
Alex Loikoe36e8bb2018-02-16 11:54:07 +010078
79 // Dump data for debug.
80 const auto channel_view = signal.channel(0);
81 apm_data_dumper_->DumpRaw("agc2_fixed_digital_gain_curve_applier",
82 channel_view.size(), channel_view.data());
83 }
84
85 // Hard-clipping.
86 for (size_t k = 0; k < signal.num_channels(); ++k) {
87 rtc::ArrayView<float> channel_view = signal.channel(k);
88 for (auto& sample : channel_view) {
Alex Loikoa05ee822018-02-20 15:58:36 +010089 sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
Alex Loikoe36e8bb2018-02-16 11:54:07 +010090 }
91 }
92}
93} // namespace webrtc