在android5.0,6.0,7.0,8.0中，都沒有實作多個APP同時錄音的功能，假設有一個應用程式在錄音，其他的APP則不能再錄音。我們先寫一個C++的錄音程式。然後在講解錄音的架構，以及代碼流程。
使用C++編寫一個程式，他可以獲得原始的音頻資料，但是這些原始的資料是沒有辦法使用播放器直接播放的，需要給他加上一個頭部，表明聲音有幾個通道，采樣率是多少等等。是以需要把pcm格式轉化為Wav格式。這樣其他的播放器才能播放，
錄音時，我們需要知道
1.采樣率：一般android采樣率有8000,11025,22050,32000,44100.。
2.每個采樣值用多少位（fomat）來表示，android一般使用16位
3.通道，左右聲道，當然有的時候也隻采樣一個通道。
4.每個采樣點，記錄兩個值，分别為左右聲道，每個值用16位表示，是以每個采樣點為32位。
即在采樣的時候我們隻需要提供3個參數：采樣率，格式（format），以及通道數。
下面建立一個檔案夾，編寫如下代碼APP_0011_AudioRecordTest：
AudioRecordTest.cpp
#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;
inBuffer = malloc(bufferSizeInBytes);
if(inBuffer == NULL)
{
ALOGE("%s alloc mem failed \n", __FUNCTION__);
goto exit ;
}
g_iNotificationPeriodInFrames = sampleRateInHz/10;
pAudioRecord = new android::AudioRecord();
if(NULL == pAudioRecord)
{
ALOGE(" create native AudioRecord failed! ");
goto exit;
}
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;
}
if(pAudioRecord->start()!= android::NO_ERROR)
{
ALOGE("AudioTrack start error!");
goto exit;
}
while (!g_bQuitAudioRecordThread)
{
int readLen = pAudioRecord->read(inBuffer, bufferSizeInBytes);
int writeResult = -1;
if(readLen > 0)
{
iWriteDataCount += readLen;
if(NULL != g_pAudioRecordFile)
{
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;
}
pcm2wav.cpp
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
/* 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;
#if 0
if(channels==2 || sample_rate==0)
{
channels = 2;
sample_rate = 44100;
}
#endif
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;*/
//修改時間：2018年1月5日
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;
}
Android.mk
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)
編寫完成之後，把檔案夾拷貝到frameworks\testing下，使用mmm指令編譯，重新燒寫system.img鏡像。然後在開發闆上執行：
record audio: ./AudioRecordTest 44100 2 my.pcm
covert pcm to wav: ./pcm2wav my.pcm 44100 2 my.wav
play wav: tinyplay my.wav
其上的tinyplay隻能播放雙通道的資料。
當我們運作兩個錄音的時候，就會出現失敗的現象，根據列印資訊，我們可以知道，其是在if(pAudioRecord->start()!= android::NO_ERROR)出了問題。也就是說，在android源生的代碼，是不支援兩個程式同時錄音的功能的。
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