webrtc/modules/audio_processing/level_estimator_impl.cc
Per Åhgren 928146f546 Removing all external access to the integer sample data in AudioBuffer
This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.

The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.

Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
2019-08-20 08:36:47 +00:00

70 lines
1.7 KiB
C++

/*
* 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 "modules/audio_processing/level_estimator_impl.h"
#include <stddef.h>
#include <stdint.h>
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/rms_level.h"
#include "rtc_base/checks.h"
namespace webrtc {
LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit)
: crit_(crit), rms_(new RmsLevel()) {
RTC_DCHECK(crit);
}
LevelEstimatorImpl::~LevelEstimatorImpl() {}
void LevelEstimatorImpl::Initialize() {
rtc::CritScope cs(crit_);
rms_->Reset();
}
void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
RTC_DCHECK(audio);
rtc::CritScope cs(crit_);
if (!enabled_) {
return;
}
for (size_t i = 0; i < audio->num_channels(); i++) {
rms_->Analyze(rtc::ArrayView<const float>(audio->channels_const_f()[i],
audio->num_frames()));
}
}
int LevelEstimatorImpl::Enable(bool enable) {
rtc::CritScope cs(crit_);
if (enable && !enabled_) {
rms_->Reset();
}
enabled_ = enable;
return AudioProcessing::kNoError;
}
bool LevelEstimatorImpl::is_enabled() const {
rtc::CritScope cs(crit_);
return enabled_;
}
int LevelEstimatorImpl::RMS() {
rtc::CritScope cs(crit_);
if (!enabled_) {
return AudioProcessing::kNotEnabledError;
}
return rms_->Average();
}
} // namespace webrtc