Files
platform-external-webrtc/webrtc/modules/audio_processing/rms_level.cc
andrew@webrtc.org 21299d4e00 Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.

Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc

Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.

BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 19:00:59 +00:00

62 lines
1.4 KiB
C++

/*
* Copyright (c) 2014 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 "webrtc/modules/audio_processing/rms_level.h"
#include <assert.h>
#include <math.h>
namespace webrtc {
static const float kMaxSquaredLevel = 32768 * 32768;
RMSLevel::RMSLevel()
: sum_square_(0),
sample_count_(0) {}
RMSLevel::~RMSLevel() {}
void RMSLevel::Reset() {
sum_square_ = 0;
sample_count_ = 0;
}
void RMSLevel::Process(const int16_t* data, int length) {
for (int i = 0; i < length; ++i) {
sum_square_ += data[i] * data[i];
}
sample_count_ += length;
}
void RMSLevel::ProcessMuted(int length) {
sample_count_ += length;
}
int RMSLevel::RMS() {
if (sample_count_ == 0 || sum_square_ == 0) {
Reset();
return kMinLevel;
}
// Normalize by the max level.
float rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
// 20log_10(x^0.5) = 10log_10(x)
rms = 10 * log10(rms);
assert(rms <= 0);
if (rms < -kMinLevel)
rms = -kMinLevel;
rms = -rms;
Reset();
return static_cast<int>(rms + 0.5);
}
} // namespace webrtc