Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 93 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_unittest.cc
new file mode 100644
index 0000000000..3296345e62
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_unittest.cc
@@ -0,0 +1,93 @@
+/*
+ *  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 <math.h>
+
+#include <algorithm>
+#include <limits>
+
+#include "modules/audio_processing/agc2/vector_float_frame.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+TEST(AutomaticGainController2GainApplier, InitialGainIsRespected) {
+  constexpr float initial_signal_level = 123.f;
+  constexpr float gain_factor = 10.f;
+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
+  GainApplier gain_applier(true, gain_factor);
+
+  gain_applier.ApplyGain(fake_audio.float_frame_view());
+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
+              initial_signal_level * gain_factor, 0.1f);
+}
+
+TEST(AutomaticGainController2GainApplier, ClippingIsDone) {
+  constexpr float initial_signal_level = 30000.f;
+  constexpr float gain_factor = 10.f;
+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
+  GainApplier gain_applier(true, gain_factor);
+
+  gain_applier.ApplyGain(fake_audio.float_frame_view());
+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
+              std::numeric_limits<int16_t>::max(), 0.1f);
+}
+
+TEST(AutomaticGainController2GainApplier, ClippingIsNotDone) {
+  constexpr float initial_signal_level = 30000.f;
+  constexpr float gain_factor = 10.f;
+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
+  GainApplier gain_applier(false, gain_factor);
+
+  gain_applier.ApplyGain(fake_audio.float_frame_view());
+
+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
+              initial_signal_level * gain_factor, 0.1f);
+}
+
+TEST(AutomaticGainController2GainApplier, RampingIsDone) {
+  constexpr float initial_signal_level = 30000.f;
+  constexpr float initial_gain_factor = 1.f;
+  constexpr float target_gain_factor = 0.5f;
+  constexpr int num_channels = 3;
+  constexpr int samples_per_channel = 4;
+  VectorFloatFrame fake_audio(num_channels, samples_per_channel,
+                              initial_signal_level);
+  GainApplier gain_applier(false, initial_gain_factor);
+
+  gain_applier.SetGainFactor(target_gain_factor);
+  gain_applier.ApplyGain(fake_audio.float_frame_view());
+
+  // The maximal gain change should be close to that in linear interpolation.
+  for (size_t channel = 0; channel < num_channels; ++channel) {
+    float max_signal_change = 0.f;
+    float last_signal_level = initial_signal_level;
+    for (const auto sample : fake_audio.float_frame_view().channel(channel)) {
+      const float current_change = fabs(last_signal_level - sample);
+      max_signal_change = std::max(max_signal_change, current_change);
+      last_signal_level = sample;
+    }
+    const float total_gain_change =
+        fabs((initial_gain_factor - target_gain_factor) * initial_signal_level);
+    EXPECT_NEAR(max_signal_change, total_gain_change / samples_per_channel,
+                0.1f);
+  }
+
+  // Next frame should have the desired level.
+  VectorFloatFrame next_fake_audio_frame(num_channels, samples_per_channel,
+                                         initial_signal_level);
+  gain_applier.ApplyGain(next_fake_audio_frame.float_frame_view());
+
+  // The last sample should have the new gain.
+  EXPECT_NEAR(next_fake_audio_frame.float_frame_view().channel(0)[0],
+              initial_signal_level * target_gain_factor, 0.1f);
+}
+}  // namespace webrtc