Android音频(10)——多App同时录音实现
转自：https://www.cnblogs.com/hellokitty2/p/10947364.html
一、使用c++编写录音程序
-
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。 -
录音时要确定3个参数
(1)采样率：一秒钟对声波采样的次数。常用的采样率有8000,11025,22050,32000,44100.
高版本的Android应该可以支持更高的采样率。
(2)每个采样值使用多少bit表示
目前Android系统上固定为16bit
(3)声道数
Stereo：立体声，每个采样点记录左右声道的值
Mono: 单声道
- tinyplay工具只能播放双声道的音频数据。
4.测试程序
(1)AudioRecordTest.cpp，用于不做pcm数据
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#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> <out_file>\n", argv[0]);
return -1;
}
AudioRecordThread(strtol(argv[1], NULL, 0), strtol(argv[2], NULL, 0), argv[3]);
return 0;
}
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(2)pcm2wav.cpp，用于将pcm转换为wav格式
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#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]; //ÄÚÈÝΪ"“RIFF
unsigned long dwSize; //×îºóÌîд£¬WAVE¸ñʽÒôƵµÄ´óС
char fccType[4]; //ÄÚÈÝΪ"WAVE”
}WAVE_HEADER;typedef struct WAVE_FMT{
char fccID[4]; //ÄÚÈÝΪ"fmt "
unsigned long dwSize; //ÄÚÈÝΪWAVE_FMTÕ¼µÄ×Ö½ÚÊý£¬Îª16
unsigned short wFormatTag; //Èç¹ûΪPCM£¬¸ÄֵΪ 1
unsigned short wChannels; //ͨµÀÊý£¬µ¥Í¨µÀ=1£¬Ë«Í¨µÀ=2
unsigned long dwSamplesPerSec;//²ÉÓÃƵÂÊ
unsigned long dwAvgBytesPerSec;/* ==dwSamplesPerSecwChannelsuiBitsPerSample/8 /
unsigned short wBlockAlign;//==wChannelsuiBitsPerSample/8
unsigned short uiBitsPerSample;//ÿ¸ö²ÉÑùµãµÄbitÊý£¬8bits=8, 16bits=16
}WAVE_FMT;typedef struct WAVE_DATA{
char fccID[4]; //ÄÚÈÝΪ"data"
unsigned long dwSize; //==NumSampleswChannelsuiBitsPerSample/8
}WAVE_DATA;
#if 0
if(channels2 || sample_rate0)
{
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;*/
//ÐÞ¸Äʱ¼ä£º2018Äê1ÔÂ5ÈÕ
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 <sample_rate> \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;
}
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(3)Android.mk
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LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
LOCAL_SRC_FILES:=
AudioRecordTest.cpp
LOCAL_SHARED_LIBRARIES :=
libcutils
libutils
libmedia
LOCAL_MODULE:= audio_record_test
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)
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然后使用tinyplay播放产生的wav文件。
录音程序参考:
Android Native C++ 层中使用AudioRecord录制PCM音频: https://blog.csdn.net/romantic_energy/article/details/50521970
pcm转wav参考:
PCM、WAV格式介绍及用C语言实现PCM转WAV: https://blog.csdn.net/u010011236/article/details/53026127
- 耳机的只有一边播放有声音的原因
./AudioRecordTest 44100 2 my.pcm
./pcm2wav my.pcm 44100 2 my.wav
tinyplay my.wav
只有1个耳朵都听到声音
./AudioRecordTest 44100 1 my.pcm
./pcm2wav my.pcm 44100 1 my.wav
tinyplay 不能播放单声道声音, 用其他播放器来播放my.wav，2个耳朵都听到声音
为何录音时用双声通，播放时只有1个耳朵有声音?
反而录音时用单声通，播放时2个耳朵都有声音?
答案:
a. 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
b. 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
c. 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
二、录音框架及代码流程
-
playbackThread 就是MixerThread，多个App对应着一个线程。
-
原生的Android录音流程
根据App传入的声音来源找到对应的device
找到profile(audio_policy.conf产生的)
根据profile找到module，即对应一个声卡，然后加载对应声卡的HAL文件
调用HAL文件中的openInput()来打开一个输入通道。 -
录音时只要App执行了set(),就会创建一个RecordThread(),多个App可能导致并发访问声卡，导致竞争访
问声卡数据的问题。 -
录音框架及代码流程
a. APP创建、设置AudioRecord, 指定了声音来源: inputSource, 比如: AUDIO_SOURCE_MIC，还指定了采样率、通道数、格式等参数
b. AudioPolicyManager根据inputSource等参数确定录音设备: device
c. AudioFlinger创建一个RecordThread, 以后该线程将从上述device读取声音数据
d. 在RecordThread内部为APP的AudioRecord创建一个对应的RecordTrack，APP的AudioRecord 与 RecordThread内部的RecordTrack 通过共享内存传递数据
e. RecordThread从HAL中得到数据, 再通过内部的RecordTrack把数据传给APP的AudioRecord
注意:
在原生代码中,APP的一个AudioRecord会导致创建一个RecordThread,在一个device上有可能存在多个RecordThread,
任意时刻只能有一个RecordThread在运行,所以只能有一个APP在录音，不能多个APP同时录音
三、修改代码支持多APP同时录音
修改AudioPolicyManager.cpp，补丁如下：
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Subject: [PATCH] v2, support Multi AudioRecord at same time
AudioPolicyManager.cpp | 11 +++++++++++
1 file changed, 11 insertions(+)
diff --git a/AudioPolicyManager.cpp b/AudioPolicyManager.cpp
index 536987a…6c87508 100644
— a/AudioPolicyManager.cpp
+++ b/AudioPolicyManager.cpp
@@ -1356,6 +1356,17 @@ audio_io_handle_t AudioPolicyManager::getInput(audio_source_t inputSource,
config.channel_mask = channelMask;
config.format = format;
- /* check wether have an AudioInputDescriptor use the same profile */
- for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
-
sp<AudioInputDescriptor> desc; -
desc = mInputs.valueAt(input_index); -
if (desc->mProfile == profile) { -
desc->mOpenRefCount++; // ÒýÓüÆÊý¼Ó1 -
desc->mSessions.add(session); // session -
return desc->mIoHandle; -
} - }
- status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
&input,
&config,
–
1.9.1
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