注意：本文抽取的AGC源码基于webRtc源码2020年1月7日的提交
本文只提取源码中 legacy 版本的AGC模块
本文代码最终使用在Android设备上(其它设备根据情况自行调整)
抽取NS&NSX(降噪)模块文章链接：https://www.jianshu.com/p/72ae0ca2c0a7
本文简述步骤：下载源码 → 抽取文件 → 修改源码文件 → 编写jni文件 → 编写CMakeLists.txt文件 → 正常使用→ 特别注意→ GitHub链接
(提示： 非代码编写顺序 )
1.webRtc源码
使用Git工具Clone源码地址：https://webrtc.googlesource.com/src （需要翻墙）
2.AGC模块抽取
主函数入口为 gain_control.h
以下为主要源码文件：
modules/audio_processing/agc/legacy/
analog_agc.c
analog_agc.h
digital_agc.c
digital_agc.h
gain_control.h
以下为依赖的源码文件：
rtc_base/
numerics/
safe_conversions_impl.h
safe_conversions.h
checks.cc
checks.h
compile_assert_c.h
sanitizer.h
common_audio/
signal_processing/
include/
signal_processing_library.h
spl_inl.h
copy_set_operations.c
division_operations.c
dot_product_with_scale.cc
dot_product_with_scale.h
resample_by_2.c
spl_sqrt.c
third_party/spl_sqrt_floor/
spl_sqrt_floor.c
spl_sqrt_floor.h
3.修改源码文件
主要修改check.h与check.cc文件，因为这两个文件引用了absl库
主要修改是移除log打印，检查数据则修改使用基础库文件assert.h代替，效果无差别
下面贴出主要代码，详情查看GitHub中源码(因为全贴出来代码太长了)
check.h修改如下
//zhonghua code------------------------------------
#include <assert.h>
#define RTC_CHECK(condition) assert(condition)
#define RTC_CHECK_EQ(val1, val2) assert(val1 == val2)
#define RTC_CHECK_NE(val1, val2) assert(val1 != val2)
#define RTC_CHECK_LE(val1, val2) assert(val1 <= val2)
#define RTC_CHECK_LT(val1, val2) assert(val1 < val2)
#define RTC_CHECK_GE(val1, val2) assert(val1 >= val2)
#define RTC_CHECK_GT(val1, val2) assert(val1 > val2)
#define RTC_DCHECK(condition) RTC_CHECK(condition)
#define RTC_DCHECK_EQ(v1, v2) RTC_CHECK_EQ(v1, v2)
#define RTC_DCHECK_NE(v1, v2) RTC_CHECK_NE(v1, v2)
#define RTC_DCHECK_LE(v1, v2) RTC_CHECK_LE(v1, v2)
#define RTC_DCHECK_LT(v1, v2) RTC_CHECK_LT(v1, v2)
#define RTC_DCHECK_GE(v1, v2) RTC_CHECK_GE(v1, v2)
#define RTC_DCHECK_GT(v1, v2) RTC_CHECK_GT(v1, v2)
#define FATAL()
//zhonghua code------------------------------------
//为了缩减lib库大小，这样处理后可缩减200K大小
#ifdef USE_WEBRTC_CODE
#ifdef __cplusplus
// C++ version.
···
···[此处省略N行]
···
#endif // __cplusplus
#endif
check.cc文件主要是去掉执行代码，理论上是可以去掉这个文件
//zhonghua code---------------------------------------
//为了缩减lib库大小，这样处理后可缩减200K大小
#ifdef USE_WEBRTC_CODE
namespace {
#if defined(__GNUC__)
__attribute__((__format__(__printf__, 2, 3)))
#endif
void AppendFormat(std::string* s, const char* fmt, ...) {
···
···[此处省略N行]
···
#endif
//zhonghua code---------------------------------------
4.编写jni文件
这里的JNI文件指的是我们调用webRtc模块的C++文件
#include <jni.h>
#include <string>
#include <cstdlib>
#include "modules/audio_processing/agc/legacy/gain_control.h"
#if defined(__cplusplus)
extern "C" {
#endif
JNIEXPORT jlong JNICALL
Java_vip_inode_demo_webrtc_AutomaticGainControlUtils_agcCreate(JNIEnv *env, jobject obj) {
return (long) WebRtcAgc_Create();
}
JNIEXPORT jint JNICALL
Java_vip_inode_demo_webrtc_AutomaticGainControlUtils_agcFree(JNIEnv *env, jobject obj,
jlong agcInst) {
void *_agcInst = (void *) agcInst;
if (_agcInst == nullptr)
return -3;
WebRtcAgc_Free(_agcInst);
return 0;
}
JNIEXPORT jint JNICALL
Java_vip_inode_demo_webrtc_AutomaticGainControlUtils_agcInit(JNIEnv *env,
jobject obj, jlong agcInst,
jint minLevel, jint maxLevel,
jint agcMode, jint fs) {
void *_agcInst = (void *) agcInst;
if (_agcInst == nullptr)
return -3;
return WebRtcAgc_Init(_agcInst, minLevel, maxLevel, agcMode, fs);
}
JNIEXPORT jint JNICALL
Java_vip_inode_demo_webrtc_AutomaticGainControlUtils_agcSetConfig(JNIEnv *env, jobject obj,
jlong agcInst,
jshort targetLevelDbfs,
jshort compressionGaindB,
jboolean limiterEnable
) {
void *_agcInst = (void *) agcInst;
if (_agcInst == nullptr)
return -3;
WebRtcAgcConfig setConfig;
setConfig.targetLevelDbfs = targetLevelDbfs;
setConfig.compressionGaindB = compressionGaindB;
setConfig.limiterEnable = limiterEnable;
return WebRtcAgc_set_config(_agcInst, setConfig);
}
JNIEXPORT jint JNICALL
Java_vip_inode_demo_webrtc_AutomaticGainControlUtils_agcProcess(JNIEnv *env, jobject obj,
jlong agcInst,
jshortArray inNear,
jint num_bands,
jint samples, jshortArray out,
jint inMicLevel,
jint outMicLevel,
jint echo,
jboolean saturationWarning) {
void *_agcInst = (void *) agcInst;
if (_agcInst == nullptr)
return -3;
jshort *cinNear = env->GetShortArrayElements(inNear, nullptr);
jshort *cout = env->GetShortArrayElements(out, nullptr);
int32_t gains[11] = {};
jint ret = WebRtcAgc_Analyze(_agcInst, &cinNear, num_bands, samples, inMicLevel, &outMicLevel,
echo, &saturationWarning, gains);
if (ret == 0)
ret = WebRtcAgc_Process(_agcInst, gains, &cinNear, num_bands, &cout);
env->ReleaseShortArrayElements(inNear, cinNear, 0);
env->ReleaseShortArrayElements(out, cout, 0);
return ret;
}
JNIEXPORT jint JNICALL
Java_vip_inode_demo_webrtc_AutomaticGainControlUtils_agcAddFarend(JNIEnv *env, jobject obj,
jlong agcInst,
jshortArray inFar,
jint samples) {
void *_agcInst = (void *) agcInst;
if (_agcInst == nullptr)
return -3;
short *cinFar = env->GetShortArrayElements(inFar, nullptr);
jint ret = WebRtcAgc_AddFarend(_agcInst, cinFar, samples);
env->ReleaseShortArrayElements(inFar, cinFar, 0);
return ret;
}
JNIEXPORT jint JNICALL
Java_vip_inode_demo_webrtc_AutomaticGainControlUtils_agcAddMic(JNIEnv *env, jobject obj,
jlong agcInst,
jshortArray inMic,
jint num_bands, jint samples
) {
void *_agcInst = (void *) agcInst;
if (_agcInst == nullptr)
return -3;
short *cinMic = env->GetShortArrayElements(inMic, nullptr);
jint ret = WebRtcAgc_AddMic(_agcInst, &cinMic, num_bands, samples);
env->ReleaseShortArrayElements(inMic, cinMic, 0);
return ret;
}
JNIEXPORT jint JNICALL
Java_vip_inode_demo_webrtc_AutomaticGainControlUtils_agcVirtualMic(JNIEnv *env, jobject obj,
jlong agcInst,
jshortArray inMic,
jint num_bands,
jint samples,
jint micLevelIn,
jint micLevelOut
) {
void *_agcInst = (void *) agcInst;
if (_agcInst == nullptr)
return -3;
jshort *cinMic = env->GetShortArrayElements(inMic, nullptr);
jint ret = WebRtcAgc_VirtualMic(_agcInst, &cinMic, num_bands, samples, micLevelIn,
&micLevelOut);
env->ReleaseShortArrayElements(inMic, cinMic, 0);
return ret;
}
#if defined(__cplusplus)
}
#endif
5.编写CMakeLists.txt文件
Android中使用需要定义宏WEBRTC_ANDROID和WEBRTC_POSIX
CMakeLists.txt内容如下：
cmake_minimum_required(VERSION 3.4.1)
file(GLOB SRC_FILES
*/*.cc
*/*/*.c
*/*/*.cc
*/*/*/*/*.c
agc-lib.cpp
)
add_library(legacy_agc-lib SHARED ${SRC_FILES})
include_directories(./)
add_definitions(
-DWEBRTC_ANDROID
-DWEBRTC_POSIX
)
find_library(log-lib log)
target_link_libraries(legacy_agc-lib ${log-lib})
6.正常使用
在Java中使用需要先初始化并配置音频参数
主要使用代码：
val agcUtils = AutomaticGainControlUtils()
val agcId = agcUtils.agcCreate()
val agcInitResult = agcUtils.agcInit(agcId, 0, 255, 3, 16000)
val agcSetConfigResult = agcUtils.agcSetConfig(agcId, 9, 9, true)
Log.e(tag, "agcId : $agcId agcInit: $agcInitResult agcSetConfig: $agcSetConfigResult")
···
val inputData = ShortArray(160)
val outAgcData = ShortArray(160)
agcUtils.agcProcess(agcId, inputData, 1, 160, outAgcData, 0, 0, 0, false)
···
agcUtils.agcFree(agcId)
特别注意：每次处理的音频数据需始终为 10ms （否则会失败）
以下为官方注解：
/*·
* This function processes a 10 ms frame by applying precomputed digital gains.
*
* Input:
* - agcInst : AGC instance
* - gains : Vector of gains to apply for digital normalization
* - in_near : Near-end input speech vector for each band
* - num_bands : Number of bands in input/output vector
*
* Output:
* - out : Gain-adjusted near-end speech vector
* : May be the same vector as the input.
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_Process(const void *agcInst,
const int32_t gains[11],
const int16_t *const *in_near,
size_t num_bands,
int16_t *const *out);
GitHub链接：https://github.com/inodevip/WebRtcNsAgcModel
抽取NS&NSX(降噪)模块文章链接：https://www.jianshu.com/p/72ae0ca2c0a7
