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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/audio/audio_level.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
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.cc | 98 |
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 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 |