SuperpoweredAndroidAudioIO和UDP套接字音频传输
SuperpoweredAndroidAudioIO和UDP套接字音频传输
提问于 2017-09-30 20:38:15
我正试图制作一个应用程序，将SuperpoweredAndroidAudioIO缓冲区从一个安卓设备传输到另一个安卓设备。通过下面的代码，我成功地从音频回调中发送和接收了短int缓冲区。但是在接收端，声音在回放过程中会受到很大的扭曲。
注意:为了简洁起见，我没有包含一些似乎与问题无关的代码，包括套接字初始化相关函数。如果需要的话，我可以添加代码。
发送端：
Mic.cpp
static bool SendBuffer(
int sd,
sockaddr_in address,
short int *buffer,
size_t bufferSize) {
// Send data buffer to the socket
ssize_t sentSize = sendto(sd,
buffer,
bufferSize,
0,
(struct sockaddr*)&address,
sizeof address);
// If send is failed
if (sentSize == -1){
__android_log_print(ANDROID_LOG_INFO, "Sent size ", "%i error %i",
sentSize , errno);
}
else if (sentSize > 0) {
__android_log_print(ANDROID_LOG_INFO, "DatagramSent : ", "%hi size: %hi",
buffer, sentSize);
}
return true;
}
// audio callback
static bool micProcessing(
void *clientdata,
short int *audioInputOutput,
int numberOfSamples,
int __unused samplerate) {
return SendBuffer(micSocket, dsocket, audioInputOutput, numberOfSamples);
}
// Constructor
Mic::Mic(JNIEnv *env,
unsigned int samplerate,
unsigned int buffersize,
unsigned int port){
micSocket = NewUdpSocket(env);
dsocket = initDestinationSocket(port); // where to send
__android_log_write(ANDROID_LOG_DEBUG, "Sockets", "initialised");
// init IO
microphone = new SuperpoweredAndroidAudioIO(samplerate,
buffersize,
true,
false,
micProcessing,
this,
-1,
SL_ANDROID_STREAM_MEDIA,
buffersize*2);
__android_log_write(ANDROID_LOG_DEBUG, "Mic", "initialised");
}接收端由两个部分组成：混频器、和信道。
Mixer.cpp
//audio callback
static bool mainprocess(
void *clientdata,
short int *audioInputOutput,
int numberOfSamples,
int __unused samplerate) {
return ((Mixer*)clientdata)->processMain(audioInputOutput, numberOfSamples);
}
// Setting up Mixer
Mixer::Mixer(JNIEnv *env,unsigned int samplerate, unsigned int buffersize) {
//Short int buffers for recieving
channel1 = new Channel(samplerate,buffersize);
//output buffer
outputFloat = ((float *)memalign(16, (buffersize + 16) * sizeof(float) * 2));
//volumes
outputLevel = 0.5f;
channel1level = 0.2f;
channel2level = 0.2f;
channel3level = 0.2f;
channel4level = 0.2f;
mainmixer = new SuperpoweredMonoMixer();
__android_log_print(ANDROID_LOG_INFO, "Mixer", " Created");
main = new SuperpoweredAndroidAudioIO(
samplerate,
buffersize,
false,
true,
mainprocess,
this,
-1,
SL_ANDROID_STREAM_MEDIA,
buffersize*2);
__android_log_write(ANDROID_LOG_INFO, "Main AudioIO created", " ");
main->stop();
SuperpoweredCPU::setSustainedPerformanceMode(true); // Prevents CPU drops
}
// processing.
bool Mixer::processMain(short int *outputbuffer, unsigned int numberOfSamples{
// a bit awkward
channel1->returnFloatBuffer();
inputs[0] = channel1->floatBuffer;
inputs[1] = NULL;
inputs[2] = NULL;
inputs[3] = NULL;
__android_log_print(ANDROID_LOG_INFO, "Channels are inside", " of mixer");
inputLevels[0] = channel1level;
inputLevels[1] = channel2level;
inputLevels[2] = channel3level;
inputLevels[3] = channel4level;
mainmixer->process(inputs,
outputFloat,
inputLevels,
outputLevel,
numberOfSamples);
SuperpoweredFloatToShortInt(outputFloat, outputbuffer, numberOfSamples);
return true;
}Channel.cpp
//receiving buffer
static bool ReceiveDatagramFromSocket(int sd, short int *buffer, size_t bufferSize) {
ssize_t recvSize = recvfrom(sd, buffer, bufferSize, 0, NULL, NULL);
if (-1 == recvSize){ // If failed
__android_log_print(ANDROID_LOG_INFO, "receive failed", " %i ", errno);
}
else {
// If data is received
if (recvSize > 0) {
__android_log_print(ANDROID_LOG_INFO, "Received"," %i bytes: %hi", recvSize, buffer);
}
}
return true;
}
// init channel
Channel::Channel(unsigned int samplerate, unsigned int buffersize){
socketIn = NewUdpSocket();
BindSocketToPort(socketIn);
samplerRate = samplerate;
bufferSize = buffersize;
shortIntBuffer = (short int *)malloc((buffersize + 16) * sizeof(short int)*4);
floatBuffer = (float *)memalign(16, (buffersize + 16) * sizeof(float) * 2);
bandEQ = new Superpowered3BandEQ(samplerate);
bandEQ->enable(true);
__android_log_print(ANDROID_LOG_INFO, "Channel ", "created");
}
// this function is called from Mixer.cpp
void Channel::returnFloatBuffer(){
ReceiveDatagramFromSocket(socketIn, shortIntBuffer, bufferSize);
Converting the 16-bit integer samples to 32-bit floating point.
SuperpoweredShortIntToFloat(shortIntBuffer, floatBuffer, bufferSize, 1);
bandEQ->process(floatBuffer, floatBuffer, bufferSize );
__android_log_print(ANDROID_LOG_INFO, "EQ", " processing");
}一开始我认为，因为AudioIO的两边都是用不同的缓冲区大小(不同的设备240和512)初始化的，但后来我尝试将512硬编码到这两个缓冲区中，但这没有帮助。
我还试图将sendto()中的缓冲区大小从()增加到4096，这样做听起来更容易识别，但仍然过于扭曲。
我还应该补充一点，我是C++的新手，我坚持“天真”和“任何可行”的方法，这使我走到了这一步。
我想了解我是否在正确的轨道上，我应该采取什么方法来传输音频而不失真。
回答 1
Stack Overflow用户
发布于 2017-10-02 15:17:12
您的方法有两个主要问题：
- 应该避免阻塞函数，如网络，从音频处理回调。您需要从不同的线程执行网络(两边)，并且需要在音频处理回调和网络线程之间进行一些缓冲以传递音频。
- 您需要“打包”传输，您需要处理双方的网络数据包。网络传输既不快也不可靠，所以您需要聪明的机制来处理这个问题。
一般来说，这种音频传输的实现对当前代码来说要复杂得多。
页面原文内容由Stack Overflow提供。腾讯云小微IT领域专用引擎提供翻译支持
原文链接：
https://stackoverflow.com/questions/46506848
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