generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/

Browse Source 
include/ is renamed to legacy/ and analog_agc.* and digital_agc.* moved into the directory.

BUG=
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7909 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2014-12-16 10:38:10 +00:00
parent 55360ae402
commit b395a5ea65
8 changed files with 28 additions and 27 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
webrtc/modules/audio_processing/BUILD.gn
View File

@ -43,19 +43,19 @@ source_set("audio_processing") {
"agc/agc_audio_proc_internal.h",
"agc/agc_manager_direct.cc",
"agc/agc_manager_direct.h",
"agc/analog_agc.c",
"agc/analog_agc.h",
"agc/circular_buffer.cc",
"agc/circular_buffer.h",
"agc/common.h",
"agc/digital_agc.c",
"agc/digital_agc.h",
"agc/gain_map_internal.h",
"agc/gmm.cc",
"agc/gmm.h",
"agc/histogram.cc",
"agc/histogram.h",
"agc/include/gain_control.h",
"agc/legacy/analog_agc.c",
"agc/legacy/analog_agc.h",
"agc/legacy/digital_agc.c",
"agc/legacy/digital_agc.h",
"agc/legacy/gain_control.h",
"agc/noise_gmm_tables.h",
"agc/pitch_based_vad.cc",
"agc/pitch_based_vad.h",

3
webrtc/modules/audio_processing/agc/analog_agc.c → webrtc/modules/audio_processing/agc/legacy/analog_agc.c
View File

@ -17,12 +17,13 @@
*
*/
#include "webrtc/modules/audio_processing/agc/legacy/analog_agc.h"
#include <assert.h>
#include <stdlib.h>
#ifdef WEBRTC_AGC_DEBUG_DUMP
#include <stdio.h>
#endif
#include "webrtc/modules/audio_processing/agc/analog_agc.h"
/* The slope of in Q13*/
static const int16_t kSlope1[8] = {21793, 12517, 7189, 4129, 2372, 1362, 472, 78};

14
webrtc/modules/audio_processing/agc/analog_agc.h → webrtc/modules/audio_processing/agc/legacy/analog_agc.h
View File

@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
#include "webrtc/modules/audio_processing/agc/digital_agc.h"
#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
#include "webrtc/typedefs.h"
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
//#define MIC_LEVEL_FEEDBACK
#ifdef WEBRTC_AGC_DEBUG_DUMP
#include <stdio.h>
#endif
#include "webrtc/modules/audio_processing/agc/legacy/digital_agc.h"
#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
#include "webrtc/typedefs.h"
/* Analog Automatic Gain Control variables:
* Constant declarations (inner limits inside which no changes are done)
* In the beginning the range is narrower to widen as soon as the measure
@ -130,4 +130,4 @@ typedef struct
int16_t lowLevelSignal;
} Agc_t;
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_

4
webrtc/modules/audio_processing/agc/digital_agc.c → webrtc/modules/audio_processing/agc/legacy/digital_agc.c
View File

@ -12,7 +12,7 @@
*
*/
#include "webrtc/modules/audio_processing/agc/digital_agc.h"
#include "webrtc/modules/audio_processing/agc/legacy/digital_agc.h"
#include <assert.h>
#include <string.h>
@ -20,7 +20,7 @@
#include <stdio.h>
#endif
#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
// To generate the gaintable, copy&paste the following lines to a Matlab window:
// MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;

6
webrtc/modules/audio_processing/agc/digital_agc.h → webrtc/modules/audio_processing/agc/legacy/digital_agc.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
#ifdef WEBRTC_AGC_DEBUG_DUMP
#include <stdio.h>
@ -75,4 +75,4 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
uint8_t limiterEnable,
int16_t analogTarget);
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_

6
webrtc/modules/audio_processing/agc/include/gain_control.h → webrtc/modules/audio_processing/agc/legacy/gain_control.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#include "webrtc/typedefs.h"
@ -245,4 +245,4 @@ int WebRtcAgc_Init(void *agcInst,
}
#endif
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_

10
webrtc/modules/audio_processing/audio_processing.gypi
View File

@ -55,19 +55,19 @@
'agc/agc_audio_proc_internal.h',
'agc/agc_manager_direct.cc',
'agc/agc_manager_direct.h',
'agc/analog_agc.c',
'agc/analog_agc.h',
'agc/circular_buffer.cc',
'agc/circular_buffer.h',
'agc/common.h',
'agc/digital_agc.c',
'agc/digital_agc.h',
'agc/gain_map_internal.h',
'agc/gmm.cc',
'agc/gmm.h',
'agc/histogram.cc',
'agc/histogram.h',
'agc/include/gain_control.h',
'agc/legacy/analog_agc.c',
'agc/legacy/analog_agc.h',
'agc/legacy/digital_agc.c',
'agc/legacy/digital_agc.h',
'agc/legacy/gain_control.h',
'agc/noise_gmm_tables.h',
'agc/pitch_based_vad.cc',
'agc/pitch_based_vad.h',

2
webrtc/modules/audio_processing/gain_control_impl.cc
View File

@ -13,7 +13,7 @@
#include <assert.h>
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
namespace webrtc {