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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/audio/audio_level.cc
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/audio_level.cc')
-rw-r--r--third_party/libwebrtc/audio/audio_level.cc98
1 files changed, 98 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/audio_level.cc b/third_party/libwebrtc/audio/audio_level.cc
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+/*
+ * 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