Android錄音並進行本地轉碼爲MP3

Android錄音並進行本地轉碼

**
通過安卓手機進行錄音，
錄音後，使用lame進行轉碼操作

開發中需要使用這個功能，只是一個簡單的進行轉碼的工具，具體的代碼信息如下
項目的基本結構圖

1、下載工具包
下載lame工具包
使用的最新版本
安裝NDK環境

2、lame內容複製
首先在 src/main/目錄下新建一個 cpp文件夾，Lame源碼中 libmp3lame拷貝到 cpp文件夾下，可以重命名，例如我命名爲 lamemp3
將Lame源碼中的 include文件夾下的 lame.h複製到 lamemp3文件夾中
剔除 lamemp3中不必要的文件和目錄，.rc/am/in文件都可以刪除，Android這裏用不到，只保留 .c和 .h文件
修改 util.h的源碼。在570行找到 ieee754_float32_t數據類型，將其修改爲 float類型，因爲 ieee754_float32_t是Linux或者是Unix下支持的數據類型，在Android下並不支持。
set_get.h中24行將 include

defaultConfig {
        applicationId "com.example.gz.testmp3"
        minSdkVersion 23
        targetSdkVersion 27
        versionCode 1
        versionName "1.0"
        testInstrumentationRunner "android.support.test.runner.AndroidJUnitRunner"
        externalNativeBuild{
            cmake{
                cppFlags "-frtti -fexceptions" //設置cpp配置參數，c文件請使用CFlags
                abiFilters 'armeabi-v7a', "armeabi"
            }
        }
        ndk{
            abiFilters 'armeabi-v7a',"armeabi"
        }
    }
    sourceSets{
        main{
            jniLibs.srcDirs = ['libs']
        }
    }

如果下載的是最新的NDK與cmake，報出了一下的錯誤，

abis [mips64, armeabi, mips] are not supported for platform. supported
abis are [armeabi-v7a, arm64-v8a, x86, x86_64].

解決辦法將gradle中cmake abiFilters配置改爲：abiFilters ‘x86’, ‘x86_64’,’arm64-v8a’, ‘armeabi-v7a’
Google NDK r18解釋： Support for ARMv5 (armeabi), MIPS, and MIPS64 has been removed. Attempting to build any of these ABIs will result in an error
我這裏的abiFilters 只寫了兩個

4、編寫Java native方法 MP3Converter

public class MP3Converter {
    static {
        System.loadLibrary("lame-mp3-utils");//這裏一定要寫對名稱
    }
    //測試lame環境是否安裝成功
    public static native String getLameVersion();
    /**
     * init lame
     * @param inSampleRate
     *              input sample rate in Hz
     * @param channel
     *              number of channels
     * @param mode
     *              0 = CBR, 1 = VBR, 2 = ABR.  default = 0
     * @param outSampleRate
     *              output sample rate in Hz
     * @param outBitRate
     *              rate compression ratio in KHz
     * @param quality
     *              quality=0..9. 0=best (very slow). 9=worst.<br />
     *              recommended:<br />
     *              2 near-best quality, not too slow<br />
     *              5 good quality, fast<br />
     *              7 ok quality, really fast
     */
    public native static void init(int inSampleRate, int channel, int mode,
                                   int outSampleRate, int outBitRate, int quality);
    /**
     * file convert to mp3
     * it may cost a lot of time and better put it in a thread
     * @param inputPath
     *          file path to be converted
     * @param mp3Path
     *          mp3 output file path
     */
    public native  static void convertMp3(String inputPath, String mp3Path);
    /**
     * get converted bytes in inputBuffer
     * @return
     *          converted bytes in inputBuffer
     *          to ignore the deviation of the file size,when return to -1 represents convert complete
     */
    public native static long getConvertBytes();
}```

**5、編寫調用C/C++的cpp**
這個直接使用的是百度上的，僅限於能看懂，對於c++不懂

extern “C”
JNIEXPORT void JNICALL
Java_jaygoo_library_converter_MP3Converter_convertMp3(JNIEnv * env, jobject obj, jstring jInputPath, jstring jMp3Path) {
const char* cInput = env->GetStringUTFChars(jInputPath, 0);
const char* cMp3 = env->GetStringUTFChars(jMp3Path, 0);
//open input file and output file
FILE* fInput = fopen(cInput,”rb”);
FILE* fMp3 = fopen(cMp3,”wb”);
short int inputBuffer[BUFFER_SIZE * 2];
unsigned char mp3Buffer[BUFFER_SIZE];//You must specified at least 7200
int read = 0; // number of bytes in inputBuffer, if in the end return 0
int write = 0;// number of bytes output in mp3buffer. can be 0
long total = 0; // the bytes of reading input file
nowConvertBytes = 0;
//if you don’t init lame, it will init lame use the default value
if(lame == NULL){
lameInit(44100, 2, 0, 44100, 96, 7);
}

//convert to mp3
do{
    read = static_cast<int>(fread(inputBuffer, sizeof(short int) * 2, BUFFER_SIZE, fInput));
    total +=  read * sizeof(short int)*2;
    nowConvertBytes = total;
    if(read != 0){
        write = lame_encode_buffer_interleaved(lame, inputBuffer, read, mp3Buffer, BUFFER_SIZE);
        //write the converted buffer to the file
        fwrite(mp3Buffer, sizeof(unsigned char), static_cast<size_t>(write), fMp3);
    }
    //if in the end flush
    if(read == 0){
        lame_encode_flush(lame,mp3Buffer, BUFFER_SIZE);
    }
}while(read != 0);

//release resources
resetLame();
fclose(fInput);
fclose(fMp3);
env->ReleaseStringUTFChars(jInputPath, cInput);
env->ReleaseStringUTFChars(jMp3Path, cMp3);
nowConvertBytes = -1;

}


**6、剩下就是調用了**

/**
* 開始錄音
* @param mFlag
*/
public void recode(int mFlag){
if(mState != -1){
Log.e(mState+”1”, “mState: “+mState );
Message msg = new Message();
Bundle b = new Bundle();// 存放數據
b.putInt(“cmd”,CMD_RECORDFAIL);
b.putInt(“msg”, ErrorCode.E_STATE_RECODING);
msg.setData(b);
uiHandler.sendMessage(msg); // 向Handler發送消息,更新UI
return;
}
int mResult = -1;
AudioRecorderWav mRecord_1 = AudioRecorderWav.getInstance();
mResult = mRecord_1.startRecordAndFile(MainActivity.this);
if(mResult == ErrorCode.SUCCESS){
mState = mFlag;
uiThread = new UIThread();
new Thread(uiThread).start();
}else{
Message msg = new Message();
Bundle b = new Bundle();// 存放數據
b.putInt(“cmd”,CMD_RECORDFAIL);
b.putInt(“msg”, mResult);
msg.setData(b);
uiHandler.sendMessage(msg); // 向Handler發送消息,更新UI
}
}
“`
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