ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
104 lines
2.8 KiB
C++
104 lines
2.8 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 "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
|
#include "webrtc/modules/include/module_common_types.h"
|
|
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
|
|
#include "webrtc/voice_engine/level_indicator.h"
|
|
|
|
namespace webrtc {
|
|
|
|
namespace voe {
|
|
|
|
// Number of bars on the indicator.
|
|
// Note that the number of elements is specified because we are indexing it
|
|
// in the range of 0-32
|
|
const int8_t permutation[33] =
|
|
{0,1,2,3,4,4,5,5,5,5,6,6,6,6,6,7,7,7,7,8,8,8,9,9,9,9,9,9,9,9,9,9,9};
|
|
|
|
|
|
AudioLevel::AudioLevel() :
|
|
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
|
|
_absMax(0),
|
|
_count(0),
|
|
_currentLevel(0),
|
|
_currentLevelFullRange(0) {
|
|
}
|
|
|
|
AudioLevel::~AudioLevel() {
|
|
delete &_critSect;
|
|
}
|
|
|
|
void AudioLevel::Clear()
|
|
{
|
|
CriticalSectionScoped cs(&_critSect);
|
|
_absMax = 0;
|
|
_count = 0;
|
|
_currentLevel = 0;
|
|
_currentLevelFullRange = 0;
|
|
}
|
|
|
|
void AudioLevel::ComputeLevel(const AudioFrame& audioFrame)
|
|
{
|
|
int16_t absValue(0);
|
|
|
|
// Check speech level (works for 2 channels as well)
|
|
absValue = 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.
|
|
CriticalSectionScoped cs(&_critSect);
|
|
|
|
if (absValue > _absMax)
|
|
_absMax = absValue;
|
|
|
|
// Update level approximately 10 times per second
|
|
if (_count++ == kUpdateFrequency)
|
|
{
|
|
_currentLevelFullRange = _absMax;
|
|
|
|
_count = 0;
|
|
|
|
// Highest value for a int16_t is 0x7fff = 32767
|
|
// Divide with 1000 to get in the range of 0-32 which is the range of
|
|
// the permutation vector
|
|
int32_t position = _absMax/1000;
|
|
|
|
// Make it less likely that the bar stays at position 0. I.e. only if
|
|
// its in the range 0-250 (instead of 0-1000)
|
|
if ((position == 0) && (_absMax > 250))
|
|
{
|
|
position = 1;
|
|
}
|
|
_currentLevel = permutation[position];
|
|
|
|
// Decay the absolute maximum (divide by 4)
|
|
_absMax >>= 2;
|
|
}
|
|
}
|
|
|
|
int8_t AudioLevel::Level() const
|
|
{
|
|
CriticalSectionScoped cs(&_critSect);
|
|
return _currentLevel;
|
|
}
|
|
|
|
int16_t AudioLevel::LevelFullRange() const
|
|
{
|
|
CriticalSectionScoped cs(&_critSect);
|
|
return _currentLevelFullRange;
|
|
}
|
|
|
|
} // namespace voe
|
|
|
|
} // namespace webrtc
|