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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
111 lines
3.6 KiB
C++
111 lines
3.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "voice_engine/audio_level.h"
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#include "common_audio/signal_processing/include/signal_processing_library.h"
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#include "modules/include/module_common_types.h"
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namespace webrtc {
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namespace voe {
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// Number of bars on the indicator.
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// Note that the number of elements is specified because we are indexing it
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// in the range of 0-32
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constexpr int8_t kPermutation[33] = {0, 1, 2, 3, 4, 4, 5, 5, 5, 5, 6,
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6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8,
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9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9};
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AudioLevel::AudioLevel()
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: abs_max_(0), count_(0), current_level_(0), current_level_full_range_(0) {
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WebRtcSpl_Init();
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}
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AudioLevel::~AudioLevel() {}
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int8_t AudioLevel::Level() const {
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rtc::CritScope cs(&crit_sect_);
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return current_level_;
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}
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int16_t AudioLevel::LevelFullRange() const {
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rtc::CritScope cs(&crit_sect_);
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return current_level_full_range_;
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}
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void AudioLevel::Clear() {
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rtc::CritScope cs(&crit_sect_);
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abs_max_ = 0;
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count_ = 0;
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current_level_ = 0;
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current_level_full_range_ = 0;
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}
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double AudioLevel::TotalEnergy() const {
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rtc::CritScope cs(&crit_sect_);
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return total_energy_;
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}
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double AudioLevel::TotalDuration() const {
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rtc::CritScope cs(&crit_sect_);
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return total_duration_;
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}
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void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
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// Check speech level (works for 2 channels as well)
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int16_t abs_value = audioFrame.muted() ? 0 :
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WebRtcSpl_MaxAbsValueW16(
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audioFrame.data(),
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audioFrame.samples_per_channel_ * audioFrame.num_channels_);
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// Protect member access using a lock since this method is called on a
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// dedicated audio thread in the RecordedDataIsAvailable() callback.
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rtc::CritScope cs(&crit_sect_);
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if (abs_value > abs_max_)
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abs_max_ = abs_value;
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// Update level approximately 10 times per second
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if (count_++ == kUpdateFrequency) {
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current_level_full_range_ = abs_max_;
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count_ = 0;
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// Highest value for a int16_t is 0x7fff = 32767
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// Divide with 1000 to get in the range of 0-32 which is the range of the
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// permutation vector
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int32_t position = abs_max_ / 1000;
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// Make it less likely that the bar stays at position 0. I.e. only if it's
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// in the range 0-250 (instead of 0-1000)
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if ((position == 0) && (abs_max_ > 250)) {
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position = 1;
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}
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current_level_ = kPermutation[position];
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// Decay the absolute maximum (divide by 4)
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abs_max_ >>= 2;
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}
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// See the description for "totalAudioEnergy" in the WebRTC stats spec
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// (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy)
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// for an explanation of these formulas. In short, we need a value that can
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// be used to compute RMS audio levels over different time intervals, by
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// taking the difference between the results from two getStats calls. To do
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// this, the value needs to be of units "squared sample value * time".
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double additional_energy =
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static_cast<double>(current_level_full_range_) / INT16_MAX;
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additional_energy *= additional_energy;
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total_energy_ += additional_energy * duration;
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total_duration_ += duration;
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}
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} // namespace voe
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} // namespace webrtc
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