该系列文章总纲链接：专题分纲目录 Android Framework 音频子系统
本章关键点总结 & 说明：
本章节主要关注➕ 以上思维导图左上 录音部分 即可。主要是说明了一些录音的基础知识，同时通过一个Native的测试程序对android的录音流程有一个了解。
1 录音基础知识
@1 PCM与WAV的关系
PCM音频数据是原始音频数据，无法使用播放器播放，需要给它加上一个头部，表明声音有几个通道，采样率是多少等等。将
PCM音频数据转换为WAV格式，这样其它播放器才能播放出来。
@2 录音的3个关键参数
- 采样率：每秒对声波的采样次数。常用的采样率有8000、11025、44100。。。
- 采样精度：目前Android系统上固定为16bit。
- 声道数：双声道（Stereo立体声，每个采样点记录左右声道的值）还是单声道（Mono）。
2 录音测试程序
这里一共涉及3个关键文件：
- AudioRecordTest.cpp：录音程序，最终输出pcm音频格式数据。
- pcm2wav.cpp：pcm格式添加头信息，转换成WAV格式。
- Android.mk：编译配置。
@1 AudioRecordTest.cpp，用于录制pcm数据，但是这里没有添加头信息。代码如下：
#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;
//申请内存
inBuffer = malloc(bufferSizeInBytes);
if(inBuffer == NULL)
{
ALOGE("%s alloc mem failed \n", __FUNCTION__);
goto exit ;
}
g_iNotificationPeriodInFrames = sampleRateInHz/10;
//创建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;
}
//开始录音
if(pAudioRecord->start()!= android::NO_ERROR)
{
ALOGE("AudioTrack start error!");
goto exit;
}
while (!g_bQuitAudioRecordThread)
{
//从底层读取音频数据
int readLen = pAudioRecord->read(inBuffer, bufferSizeInBytes);
int writeResult = -1;
if(readLen > 0)
{
iWriteDataCount += readLen;
if(NULL != g_pAudioRecordFile)
{
//将音频数据写入指定文件中
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>
/* 参考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]; //内容为""RIFF
unsigned long dwSize; //最后填写，WAVE格式音频的大小
char fccType[4]; //内容为"WAVE"
}WAVE_HEADER;
typedef struct WAVE_FMT{
char fccID[4]; //内容为"fmt "
unsigned long dwSize; //内容为WAVE_FMT占的字节数，为16
unsigned short wFormatTag; //如果为PCM，改值为 1
unsigned short wChannels; //通道数，单通道=1，双通道=2
unsigned long dwSamplesPerSec;//采用频率
unsigned long dwAvgBytesPerSec;/* ==dwSamplesPerSec*wChannels*uiBitsPerSample/8 */
unsigned short wBlockAlign;//==wChannels*uiBitsPerSample/8
unsigned short uiBitsPerSample;//每个采样点的bit数，8bits=8, 16bits=16
}WAVE_FMT;
typedef struct WAVE_DATA{
char fccID[4]; //内容为"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实现
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 测试程序执行步骤
@1 开始录音：
./AudioRecordTest 44100 2 my.pcm@2 文件格式转换（PCM转WAV）：
./pcm2wav my.pcm 44100 2 my.wav4 遇到问题以及分析
@1 tinyplay 不能播放单声道声音
如果一定要播放，那就只能用其他播放器。
@2 为何录音时用双声通，播放时只有1个耳朵有声音？而录音时用单声通，播放时2个耳朵都有声音？
- 硬件上、驱动上是双声道的; 但是我们只接了一个MIC，所以驱动程序录音时得到的双声道数据中，其中一个声道数据恒为0，而AudioRecordTest录音时如果指定了双声道，那么得到的PCM数据里其中一个声道恒为0，它播放时就会导致只有一个耳朵有声音
- AudioRecordTest录音时如果指定了单声道，那么得到的PCM数据只含有一个声道数据,它是硬件左、右声道的混合,这个混合是AudioFlinger系统实现的，在播放时单声道数据时，AudioFlinger系统会把单声道数据既发给硬件Left DAC(左声道)、 也发给硬件Right DAC(右声道)，所以2个耳朵都可以听到声音
