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gerrit.openfyde.cn / webrtc.googlesource.com / src / 557e92515d56b2e5d8f354921e8b037349518ac2 / . / webrtc / modules / audio_processing / agc / analog_agc.h
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niklase@google.com470e71d2011-07-07 08:21:25 +0000[diff] [blame]1/*
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
12#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
13
14#include "typedefs.h"
15#include "gain_control.h"
16#include "digital_agc.h"
17
18//#define AGC_DEBUG
19//#define MIC_LEVEL_FEEDBACK
20#ifdef AGC_DEBUG
21#include <stdio.h>
22#endif
23
24/* Analog Automatic Gain Control variables:
25 * Constant declarations (inner limits inside which no changes are done)
26 * In the beginning the range is narrower to widen as soon as the measure
27 * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0
28 * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal
29 * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm
30 * The limits are created by running the AGC with a file having the desired
31 * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined
32 * by out=10*log10(in/260537279.7); Set the target level to the average level
33 * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in
34 * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) )
35 */
36#define RXX_BUFFER_LEN 10
37
38static const WebRtc_Word16 kMsecSpeechInner = 520;
39static const WebRtc_Word16 kMsecSpeechOuter = 340;
40
41static const WebRtc_Word16 kNormalVadThreshold = 400;
42
43static const WebRtc_Word16 kAlphaShortTerm = 6; // 1 >> 6 = 0.0156
44static const WebRtc_Word16 kAlphaLongTerm = 10; // 1 >> 10 = 0.000977
45
46typedef struct
47{
48 // Configurable parameters/variables
49 WebRtc_UWord32 fs; // Sampling frequency
50 WebRtc_Word16 compressionGaindB; // Fixed gain level in dB
51 WebRtc_Word16 targetLevelDbfs; // Target level in -dBfs of envelope (default -3)
52 WebRtc_Word16 agcMode; // Hard coded mode (adaptAna/adaptDig/fixedDig)
53 WebRtc_UWord8 limiterEnable; // Enabling limiter (on/off (default off))
54 WebRtcAgc_config_t defaultConfig;
55 WebRtcAgc_config_t usedConfig;
56
57 // General variables
58 WebRtc_Word16 initFlag;
59 WebRtc_Word16 lastError;
60
61 // Target level parameters
62 // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7)
63 WebRtc_Word32 analogTargetLevel; // = RXX_BUFFER_LEN * 846805; -22 dBfs
64 WebRtc_Word32 startUpperLimit; // = RXX_BUFFER_LEN * 1066064; -21 dBfs
65 WebRtc_Word32 startLowerLimit; // = RXX_BUFFER_LEN * 672641; -23 dBfs
66 WebRtc_Word32 upperPrimaryLimit; // = RXX_BUFFER_LEN * 1342095; -20 dBfs
67 WebRtc_Word32 lowerPrimaryLimit; // = RXX_BUFFER_LEN * 534298; -24 dBfs
68 WebRtc_Word32 upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832; -17 dBfs
69 WebRtc_Word32 lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783; -27 dBfs
70 WebRtc_UWord16 targetIdx; // Table index for corresponding target level
71#ifdef MIC_LEVEL_FEEDBACK
72 WebRtc_UWord16 targetIdxOffset; // Table index offset for level compensation
73#endif
74 WebRtc_Word16 analogTarget; // Digital reference level in ENV scale
75
76 // Analog AGC specific variables
77 WebRtc_Word32 filterState[8]; // For downsampling wb to nb
78 WebRtc_Word32 upperLimit; // Upper limit for mic energy
79 WebRtc_Word32 lowerLimit; // Lower limit for mic energy
80 WebRtc_Word32 Rxx160w32; // Average energy for one frame
81 WebRtc_Word32 Rxx16_LPw32; // Low pass filtered subframe energies
82 WebRtc_Word32 Rxx160_LPw32; // Low pass filtered frame energies
83 WebRtc_Word32 Rxx16_LPw32Max; // Keeps track of largest energy subframe
84 WebRtc_Word32 Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies
85 WebRtc_Word32 Rxx16w32_array[2][5];// Energy values of microphone signal
86 WebRtc_Word32 env[2][10]; // Envelope values of subframes
87
88 WebRtc_Word16 Rxx16pos; // Current position in the Rxx16_vectorw32
89 WebRtc_Word16 envSum; // Filtered scaled envelope in subframes
90 WebRtc_Word16 vadThreshold; // Threshold for VAD decision
91 WebRtc_Word16 inActive; // Inactive time in milliseconds
92 WebRtc_Word16 msTooLow; // Milliseconds of speech at a too low level
93 WebRtc_Word16 msTooHigh; // Milliseconds of speech at a too high level
94 WebRtc_Word16 changeToSlowMode; // Change to slow mode after some time at target
95 WebRtc_Word16 firstCall; // First call to the process-function
96 WebRtc_Word16 msZero; // Milliseconds of zero input
97 WebRtc_Word16 msecSpeechOuterChange;// Min ms of speech between volume changes
98 WebRtc_Word16 msecSpeechInnerChange;// Min ms of speech between volume changes
99 WebRtc_Word16 activeSpeech; // Milliseconds of active speech
100 WebRtc_Word16 muteGuardMs; // Counter to prevent mute action
101 WebRtc_Word16 inQueue; // 10 ms batch indicator
102
103 // Microphone level variables
104 WebRtc_Word32 micRef; // Remember ref. mic level for virtual mic
105 WebRtc_UWord16 gainTableIdx; // Current position in virtual gain table
106 WebRtc_Word32 micGainIdx; // Gain index of mic level to increase slowly
107 WebRtc_Word32 micVol; // Remember volume between frames
108 WebRtc_Word32 maxLevel; // Max possible vol level, incl dig gain
109 WebRtc_Word32 maxAnalog; // Maximum possible analog volume level
110 WebRtc_Word32 maxInit; // Initial value of "max"
111 WebRtc_Word32 minLevel; // Minimum possible volume level
112 WebRtc_Word32 minOutput; // Minimum output volume level
113 WebRtc_Word32 zeroCtrlMax; // Remember max gain => don't amp low input
114
115 WebRtc_Word16 scale; // Scale factor for internal volume levels
116#ifdef MIC_LEVEL_FEEDBACK
117 WebRtc_Word16 numBlocksMicLvlSat;
118 WebRtc_UWord8 micLvlSat;
119#endif
120 // Structs for VAD and digital_agc
121 AgcVad_t vadMic;
122 DigitalAgc_t digitalAgc;
123
124#ifdef AGC_DEBUG
125 FILE* fpt;
126 FILE* agcLog;
127 WebRtc_Word32 fcount;
128#endif
129
130 WebRtc_Word16 lowLevelSignal;
131} Agc_t;
132
133#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_