From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Fri, 19 Apr 2024 02:47:55 +0200
Subject: Adding upstream version 124.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 third_party/libwebrtc/audio/audio_level.cc | 98 ++++++++++++++++++++++++++++++
 1 file changed, 98 insertions(+)
 create mode 100644 third_party/libwebrtc/audio/audio_level.cc

(limited to 'third_party/libwebrtc/audio/audio_level.cc')

diff --git a/third_party/libwebrtc/audio/audio_level.cc b/third_party/libwebrtc/audio/audio_level.cc
new file mode 100644
index 0000000000..7874b73f1c
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_level.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2012 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 "audio/audio_level.h"
+
+#include "api/audio/audio_frame.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+namespace voe {
+
+AudioLevel::AudioLevel()
+    : abs_max_(0), count_(0), current_level_full_range_(0) {}
+
+AudioLevel::~AudioLevel() {}
+
+void AudioLevel::Reset() {
+  MutexLock lock(&mutex_);
+  abs_max_ = 0;
+  count_ = 0;
+  current_level_full_range_ = 0;
+  total_energy_ = 0.0;
+  total_duration_ = 0.0;
+}
+
+int16_t AudioLevel::LevelFullRange() const {
+  MutexLock lock(&mutex_);
+  return current_level_full_range_;
+}
+
+void AudioLevel::ResetLevelFullRange() {
+  MutexLock lock(&mutex_);
+  abs_max_ = 0;
+  count_ = 0;
+  current_level_full_range_ = 0;
+}
+
+double AudioLevel::TotalEnergy() const {
+  MutexLock lock(&mutex_);
+  return total_energy_;
+}
+
+double AudioLevel::TotalDuration() const {
+  MutexLock lock(&mutex_);
+  return total_duration_;
+}
+
+void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
+  // Check speech level (works for 2 channels as well)
+  int16_t abs_value =
+      audioFrame.muted()
+          ? 0
+          : WebRtcSpl_MaxAbsValueW16(
+                audioFrame.data(),
+                audioFrame.samples_per_channel_ * audioFrame.num_channels_);
+
+  // Protect member access using a lock since this method is called on a
+  // dedicated audio thread in the RecordedDataIsAvailable() callback.
+  MutexLock lock(&mutex_);
+
+  if (abs_value > abs_max_)
+    abs_max_ = abs_value;
+
+  // Update level approximately 9 times per second, assuming audio frame
+  // duration is approximately 10 ms. (The update frequency is every
+  // 11th (= |kUpdateFrequency+1|) call: 1000/(11*10)=9.09..., we should
+  // probably change this behavior, see https://crbug.com/webrtc/10784).
+  if (count_++ == kUpdateFrequency) {
+    current_level_full_range_ = abs_max_;
+
+    count_ = 0;
+
+    // Decay the absolute maximum (divide by 4)
+    abs_max_ >>= 2;
+  }
+
+  // See the description for "totalAudioEnergy" in the WebRTC stats spec
+  // (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy)
+  // for an explanation of these formulas. In short, we need a value that can
+  // be used to compute RMS audio levels over different time intervals, by
+  // taking the difference between the results from two getStats calls. To do
+  // this, the value needs to be of units "squared sample value * time".
+  double additional_energy =
+      static_cast<double>(current_level_full_range_) / INT16_MAX;
+  additional_energy *= additional_energy;
+  total_energy_ += additional_energy * duration;
+  total_duration_ += duration;
+}
+
+}  // namespace voe
+}  // namespace webrtc
-- 
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