Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 103 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc
new file mode 100644
index 0000000000..f9e276d3a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc
@@ -0,0 +1,103 @@
+/*
+ *  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/gain_applier.h"
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/agc2_common.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 GainCloseToOne(float gain_factor) {
+  return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
+         gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
+}
+
+void ClipSignal(AudioFrameView<float> signal) {
+  for (int 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);
+    }
+  }
+}
+
+void ApplyGainWithRamping(float last_gain_linear,
+                          float gain_at_end_of_frame_linear,
+                          float inverse_samples_per_channel,
+                          AudioFrameView<float> float_frame) {
+  // Do not modify the signal.
+  if (last_gain_linear == gain_at_end_of_frame_linear &&
+      GainCloseToOne(gain_at_end_of_frame_linear)) {
+    return;
+  }
+
+  // Gain is constant and different from 1.
+  if (last_gain_linear == gain_at_end_of_frame_linear) {
+    for (int k = 0; k < float_frame.num_channels(); ++k) {
+      rtc::ArrayView<float> channel_view = float_frame.channel(k);
+      for (auto& sample : channel_view) {
+        sample *= gain_at_end_of_frame_linear;
+      }
+    }
+    return;
+  }
+
+  // The gain changes. We have to change slowly to avoid discontinuities.
+  const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
+                          inverse_samples_per_channel;
+  float gain = last_gain_linear;
+  for (int i = 0; i < float_frame.samples_per_channel(); ++i) {
+    for (int ch = 0; ch < float_frame.num_channels(); ++ch) {
+      float_frame.channel(ch)[i] *= gain;
+    }
+    gain += increment;
+  }
+}
+
+}  // namespace
+
+GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
+    : hard_clip_samples_(hard_clip_samples),
+      last_gain_factor_(initial_gain_factor),
+      current_gain_factor_(initial_gain_factor) {}
+
+void GainApplier::ApplyGain(AudioFrameView<float> signal) {
+  if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
+    Initialize(signal.samples_per_channel());
+  }
+
+  ApplyGainWithRamping(last_gain_factor_, current_gain_factor_,
+                       inverse_samples_per_channel_, signal);
+
+  last_gain_factor_ = current_gain_factor_;
+
+  if (hard_clip_samples_) {
+    ClipSignal(signal);
+  }
+}
+
+// TODO(bugs.webrtc.org/7494): Remove once switched to gains in dB.
+void GainApplier::SetGainFactor(float gain_factor) {
+  RTC_DCHECK_GT(gain_factor, 0.f);
+  current_gain_factor_ = gain_factor;
+}
+
+void GainApplier::Initialize(int samples_per_channel) {
+  RTC_DCHECK_GT(samples_per_channel, 0);
+  samples_per_channel_ = static_cast<int>(samples_per_channel);
+  inverse_samples_per_channel_ = 1.f / samples_per_channel_;
+}
+
+}  // namespace webrtc