36b1a5fcec
const int16_t* data() const; int16_t* mutable_data(); - data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames. - mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_. These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation. This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later. BUG=webrtc:7343 TBR=henrika Review-Url: https://codereview.webrtc.org/2750783004 Cr-Commit-Position: refs/heads/master@{#18543}
90 lines
2.7 KiB
C++
90 lines
2.7 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 "webrtc/voice_engine/audio_level.h"
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/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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void AudioLevel::ComputeLevel(const AudioFrame& audioFrame) {
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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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}
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} // namespace voe
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} // namespace webrtc
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