Android Framework 音頻子系統（16）AudioRecord案例

該系列文章總綱連結：專題分綱目錄 Android Framework 音頻子系統

本章關鍵點總結 & 說明：

本章節主要關注➕ 以上思維導圖左上錄音部分即可。主要是說明了一些錄音的基礎知識，同時通過一個Native的測試程式對android的錄音流程有一個了解。

1 錄音基礎知識

@1 PCM與WAV的關系

PCM音頻資料是原始音頻資料，無法使用播放器播放，需要給它加上一個頭部，表明聲音有幾個通道，采樣率是多少等等。将

PCM音頻資料轉換為WAV格式，這樣其它播放器才能播放出來。

@2 錄音的3個關鍵參數

采樣率：每秒對聲波的采樣次數。常用的采樣率有8000、11025、44100。。。
采樣精度：目前Android系統上固定為16bit。
聲道數：雙聲道（Stereo立體聲，每個采樣點記錄左右聲道的值）還是單聲道（Mono）。

2 錄音測試程式

這裡一共涉及3個關鍵檔案：

AudioRecordTest.cpp：錄音程式，最終輸出pcm音頻格式資料。
pcm2wav.cpp：pcm格式添加頭資訊，轉換成WAV格式。
Android.mk：編譯配置。

@1 AudioRecordTest.cpp，用于錄制pcm資料，但是這裡沒有添加頭資訊。代碼如下：

#include <utils/Log.h>
#include <media/AudioRecord.h>
#include <stdlib.h>

using namespace android;
 
//==============================================
//  Audio Record Defination
//==============================================
#ifdef LOG_TAG
#undef LOG_TAG
#endif

#define LOG_TAG "AudioRecordTest"
 
static pthread_t    g_AudioRecordThread;
static pthread_t *  g_AudioRecordThreadPtr = NULL;
 
volatile bool       g_bQuitAudioRecordThread = false;
volatile int        g_iInSampleTime = 0;
int                 g_iNotificationPeriodInFrames = 8000/10; 
// g_iNotificationPeriodInFrames should be change when sample rate changes.

static void *   AudioRecordThread(int sample_rate, int channels, void *fileName)
{
    uint64_t                        inHostTime              = 0;
    void *                              inBuffer                    = NULL; 
    audio_source_t              inputSource             = AUDIO_SOURCE_MIC;
    audio_format_t              audioFormat             = AUDIO_FORMAT_PCM_16_BIT;  
    audio_channel_mask_t    channelConfig       = AUDIO_CHANNEL_IN_MONO;
    int                                     bufferSizeInBytes;
    int                                     sampleRateInHz      = sample_rate; //8000; //44100; 
    android::AudioRecord *  pAudioRecord        = NULL;
    FILE *                                  g_pAudioRecordFile      = NULL;
    char *                                      strAudioFile                = (char *)fileName;
 
    int iNbChannels         = channels; // 1 channel for mono, 2 channel for streo
    int iBytesPerSample = 2;    // 16bits pcm, 2Bytes
    int frameSize           = 0;    // frameSize = iNbChannels * iBytesPerSample
    size_t  minFrameCount   = 0;    // get from AudroRecord object
    int iWriteDataCount = 0;    // how many data are there write to file
    
    // log the thread id for debug info
    ALOGD("%s  Thread ID  = %d  \n", __FUNCTION__,  pthread_self());  
    g_iInSampleTime = 0;
    g_pAudioRecordFile = fopen(strAudioFile, "wb+");    
    
    //printf("sample_rate = %d, channels = %d, iNbChannels = %d, channelConfig = 0x%x\n", sample_rate, channels, iNbChannels, channelConfig);
    
    //iNbChannels = (channelConfig == AUDIO_CHANNEL_IN_STEREO) ? 2 : 1;
    if (iNbChannels == 2) {
        channelConfig = AUDIO_CHANNEL_IN_STEREO;
    }
    printf("sample_rate = %d, channels = %d, iNbChannels = %d, channelConfig = 0x%x\n", sample_rate, channels, iNbChannels, channelConfig);
    
    frameSize   = iNbChannels * iBytesPerSample;    
    
    android::status_t   status = android::AudioRecord::getMinFrameCount(
        &minFrameCount, sampleRateInHz, audioFormat, channelConfig);    
    
    if(status != android::NO_ERROR)
    {
        ALOGE("%s  AudioRecord.getMinFrameCount fail \n", __FUNCTION__);
        goto exit ;
    }
    
    ALOGE("sampleRateInHz = %d minFrameCount = %d iNbChannels = %d channelConfig = 0x%x frameSize = %d ", 
        sampleRateInHz, minFrameCount, iNbChannels, channelConfig, frameSize);  
    
    bufferSizeInBytes = minFrameCount * frameSize;
    //申請記憶體
    inBuffer = malloc(bufferSizeInBytes); 
    if(inBuffer == NULL)
    {       
        ALOGE("%s  alloc mem failed \n", __FUNCTION__);     
        goto exit ; 
    }
 
    g_iNotificationPeriodInFrames = sampleRateInHz/10;  
    //建立AudioRecord
    pAudioRecord  = new android::AudioRecord(); 
    if(NULL == pAudioRecord)
    {
        ALOGE(" create native AudioRecord failed! ");
        goto exit;
    }
    //set操作
    pAudioRecord->set( inputSource,
                                    sampleRateInHz,
                                    audioFormat,
                                    channelConfig,
                                    0,
                                    NULL, //AudioRecordCallback,
                                    NULL,
                                    0,
                                    true,
                                    0); 
 
    if(pAudioRecord->initCheck() != android::NO_ERROR)  
    {
        ALOGE("AudioTrack initCheck error!");
        goto exit;
    }
    //開始錄音
    if(pAudioRecord->start()!= android::NO_ERROR)
    {
        ALOGE("AudioTrack start error!");
        goto exit;
    }   
    
    while (!g_bQuitAudioRecordThread)
    {
        //從底層讀取音頻資料
        int readLen = pAudioRecord->read(inBuffer, bufferSizeInBytes);      
        int writeResult = -1;
        
        if(readLen > 0) 
        {
            iWriteDataCount += readLen;
            if(NULL != g_pAudioRecordFile)
            {
                //将音頻資料寫入指定檔案中
                writeResult = fwrite(inBuffer, 1, readLen, g_pAudioRecordFile);             
                if(writeResult < readLen)
                {
                    ALOGE("Write Audio Record Stream error");
                }
            }           
 
            //ALOGD("readLen = %d  writeResult = %d  iWriteDataCount = %d", readLen, writeResult, iWriteDataCount);         
        }
        else 
        {
            ALOGE("pAudioRecord->read  readLen = 0");
        }
    }
        
exit:
    if(NULL != g_pAudioRecordFile)
    {
        fflush(g_pAudioRecordFile);
        fclose(g_pAudioRecordFile);
        g_pAudioRecordFile = NULL;
    }
 
    if(pAudioRecord)
    {
        pAudioRecord->stop();
        //delete pAudioRecord;
        //pAudioRecord == NULL;
    }
 
    if(inBuffer)
    {
        free(inBuffer);
        inBuffer = NULL;
    }
    
    ALOGD("%s  Thread ID  = %d  quit\n", __FUNCTION__,  pthread_self());
    return NULL;
}

int main(int argc, char **argv)
{
    if (argc != 4)
    {
        printf("Usage:\n");
        printf("%s <sample_rate> <channels> <out_file>\n", argv[0]);
        return -1;
    }
    AudioRecordThread(strtol(argv[1], NULL, 0), strtol(argv[2], NULL, 0), argv[3]);
    return 0;
}

@2 pcm2wav.cpp

#include <stdio.h>
#include <string.h>
#include <stdlib.h>

/* 參考https://blog.csdn.net/u010011236/article/details/53026127 */

/**
 * Convert PCM16LE raw data to WAVE format
 * @param pcmpath       Input PCM file.
 * @param channels      Channel number of PCM file.
 * @param sample_rate   Sample rate of PCM file.
 * @param wavepath      Output WAVE file.
 */
int simplest_pcm16le_to_wave(const char *pcmpath, int sample_rate, int channels, const char *wavepath)
{
    typedef struct WAVE_HEADER{
        char    fccID[4];       //内容為""RIFF
        unsigned long dwSize;   //最後填寫，WAVE格式音頻的大小
        char    fccType[4];     //内容為"WAVE"
    }WAVE_HEADER;

    typedef struct WAVE_FMT{
        char    fccID[4];          //内容為"fmt "
        unsigned long  dwSize;     //内容為WAVE_FMT占的位元組數，為16
        unsigned short wFormatTag; //如果為PCM，改值為 1
        unsigned short wChannels;  //通道數，單通道=1，雙通道=2
        unsigned long  dwSamplesPerSec;//采用頻率
        unsigned long  dwAvgBytesPerSec;/* ==dwSamplesPerSec*wChannels*uiBitsPerSample/8 */
        unsigned short wBlockAlign;//==wChannels*uiBitsPerSample/8
        unsigned short uiBitsPerSample;//每個采樣點的bit數，8bits=8, 16bits=16
    }WAVE_FMT;

    typedef struct WAVE_DATA{
        char    fccID[4];       //内容為"data"
        unsigned long dwSize;   //==NumSamples*wChannels*uiBitsPerSample/8
    }WAVE_DATA;

    int bits = 16;

    WAVE_HEADER pcmHEADER;
    WAVE_FMT    pcmFMT;
    WAVE_DATA   pcmDATA;

    unsigned short m_pcmData;
    FILE *fp, *fpout;

    fp = fopen(pcmpath, "rb+");
    if(fp==NULL)
    {
        printf("Open pcm file error.\n");
        return -1;
    }
    fpout = fopen(wavepath, "wb+");
    if(fpout==NULL)
    {
        printf("Create wav file error.\n");
        return -1;
    }

    /* WAVE_HEADER */
    memcpy(pcmHEADER.fccID, "RIFF", strlen("RIFF"));
    memcpy(pcmHEADER.fccType, "WAVE", strlen("WAVE"));
    fseek(fpout, sizeof(WAVE_HEADER), 1);   //1=SEEK_CUR
    /* WAVE_FMT */
    memcpy(pcmFMT.fccID, "fmt ", strlen("fmt "));
    pcmFMT.dwSize = 16;
    pcmFMT.wFormatTag = 1;
    pcmFMT.wChannels = channels;
    pcmFMT.dwSamplesPerSec = sample_rate;
    pcmFMT.uiBitsPerSample = bits;
    /* ==dwSamplesPerSec*wChannels*uiBitsPerSample/8 */
    pcmFMT.dwAvgBytesPerSec = pcmFMT.dwSamplesPerSec*pcmFMT.wChannels*pcmFMT.uiBitsPerSample/8;
    /* ==wChannels*uiBitsPerSample/8 */
    pcmFMT.wBlockAlign = pcmFMT.wChannels*pcmFMT.uiBitsPerSample/8;


    fwrite(&pcmFMT, sizeof(WAVE_FMT), 1, fpout);

    /* WAVE_DATA */
    memcpy(pcmDATA.fccID, "data", strlen("data"));
    pcmDATA.dwSize = 0;
    fseek(fpout, sizeof(WAVE_DATA), SEEK_CUR);

    fread(&m_pcmData, sizeof(unsigned short), 1, fp);
    while(!feof(fp))
    {
        pcmDATA.dwSize += 2;
        fwrite(&m_pcmData, sizeof(unsigned short), 1, fpout);
        fread(&m_pcmData, sizeof(unsigned short), 1, fp);
    }

    /*pcmHEADER.dwSize = 44 + pcmDATA.dwSize;*/
    pcmHEADER.dwSize = 36 + pcmDATA.dwSize;

    rewind(fpout);
    fwrite(&pcmHEADER, sizeof(WAVE_HEADER), 1, fpout);
    fseek(fpout, sizeof(WAVE_FMT), SEEK_CUR);
    fwrite(&pcmDATA, sizeof(WAVE_DATA), 1, fpout);

    fclose(fp);
    fclose(fpout);

    return 0;
}

int main(int argc, char **argv)
{
	if (argc != 5)
	{
		printf("Usage:\n");
		printf("%s <input pcm file> <sample_rate> <channels>  <output wav file>\n", argv[0]);
		return -1;
	}
	
    simplest_pcm16le_to_wave(argv[1], strtol(argv[2], NULL, 0), strtol(argv[3], NULL, 0), argv[4]);

    return 0;
}

@3 Android.mk實作

LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)

LOCAL_SRC_FILES:= \
	AudioRecordTest.cpp

LOCAL_SHARED_LIBRARIES := \
	libcutils \
	libutils \
    libmedia

LOCAL_MODULE:= AudioRecordTest

LOCAL_MODULE_TAGS := tests

include $(BUILD_EXECUTABLE)

include $(CLEAR_VARS)

LOCAL_SRC_FILES:= \
	pcm2wav.cpp

LOCAL_SHARED_LIBRARIES := \
	libcutils \
	libutils \
    libmedia

LOCAL_MODULE:= pcm2wav

LOCAL_MODULE_TAGS := tests

include $(BUILD_EXECUTABLE)

3 測試程式執行步驟

@1 開始錄音：

 ./AudioRecordTest 44100 2 my.pcm

@2 檔案格式轉換（PCM轉WAV）：

./pcm2wav my.pcm 44100 2 my.wav

4 遇到問題以及分析

@1 tinyplay 不能播放單聲道聲音

如果一定要播放，那就隻能用其他播放器。

@2 為何錄音時用雙聲通，播放時隻有1個耳朵有聲音？而錄音時用單聲通，播放時2個耳朵都有聲音？

硬體上、驅動上是雙聲道的; 但是我們隻接了一個MIC，是以驅動程式錄音時得到的雙聲道資料中，其中一個聲道資料恒為0，而AudioRecordTest錄音時如果指定了雙聲道，那麼得到的PCM資料裡其中一個聲道恒為0，它播放時就會導緻隻有一個耳朵有聲音
AudioRecordTest錄音時如果指定了單聲道，那麼得到的PCM資料隻含有一個聲道資料,它是硬體左、右聲道的混合,這個混合是AudioFlinger系統實作的，在播放時單聲道資料時，AudioFlinger系統會把單聲道資料既發給硬體Left DAC(左聲道)、也發給硬體Right DAC(右聲道)，是以2個耳朵都可以聽到聲音

Android Framework 音頻子系統（16）AudioRecord案例

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