通过套接字连接在iOS上播放音频

Question

希望你能帮我解决这个问题，我见过很多与此有关的问题，但它们都没有真正帮助我找出我在这里做错了什么。

因此，在Android上，我有一个AudioRecord，它记录音频，并通过套接字连接将音频作为字节数组发送到客户端。这部分在Android上非常容易，而且工作非常完美。

当我开始和iOS合作的时候，我发现要做到这一点并不容易，所以经过两天的研究，我已经得到了这个。仍然不播放任何音频。它在启动时会产生噪音，但是没有一个正在通过套接字传输的音频被播放。我通过记录缓冲区数组中的每个元素来确认套接字正在接收数据。

这里是我使用的所有代码，很多代码都是从一堆站点中重用的，不能记住所有的链接。(顺便说一句，使用AudioUnits)

首先，音频处理器:播放回调

static OSStatus playbackCallback(void *inRefCon,
                                 AudioUnitRenderActionFlags *ioActionFlags,
                                 const AudioTimeStamp *inTimeStamp,
                                 UInt32 inBusNumber,
                                 UInt32 inNumberFrames,
                                 AudioBufferList *ioData) {

    /**
     This is the reference to the object who owns the callback.
     */
    AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon;

    // iterate over incoming stream an copy to output stream
    for (int i=0; i < ioData->mNumberBuffers; i++) {
        AudioBuffer buffer = ioData->mBuffers[i];

        // find minimum size
        UInt32 size = min(buffer.mDataByteSize, [audioProcessor audioBuffer].mDataByteSize);

        // copy buffer to audio buffer which gets played after function return
        memcpy(buffer.mData, [audioProcessor audioBuffer].mData, size);

        // set data size
        buffer.mDataByteSize = size;
    }
    return noErr;
}

音频处理器初始化

-(void)initializeAudio
{
    OSStatus status;

    // We define the audio component
    AudioComponentDescription desc;
    desc.componentType = kAudioUnitType_Output; // we want to ouput
    desc.componentSubType = kAudioUnitSubType_RemoteIO; // we want in and ouput
    desc.componentFlags = 0; // must be zero
    desc.componentFlagsMask = 0; // must be zero
    desc.componentManufacturer = kAudioUnitManufacturer_Apple; // select provider

    // find the AU component by description
    AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);

    // create audio unit by component
    status = AudioComponentInstanceNew(inputComponent, &audioUnit);

    [self hasError:status:__FILE__:__LINE__];

    // define that we want record io on the input bus
    UInt32 flag = 1;


    // define that we want play on io on the output bus
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioOutputUnitProperty_EnableIO, // use io
                                  kAudioUnitScope_Output, // scope to output
                                  kOutputBus, // select output bus (0)
                                  &flag, // set flag
                                  sizeof(flag));
    [self hasError:status:__FILE__:__LINE__];

    /*
     We need to specifie our format on which we want to work.
     We use Linear PCM cause its uncompressed and we work on raw data.
     for more informations check.

     We want 16 bits, 2 bytes per packet/frames at 44khz
     */
    AudioStreamBasicDescription audioFormat;
    audioFormat.mSampleRate         = SAMPLE_RATE;
    audioFormat.mFormatID           = kAudioFormatLinearPCM;
    audioFormat.mFormatFlags        = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
    audioFormat.mFramesPerPacket    = 1;
    audioFormat.mChannelsPerFrame   = 1;
    audioFormat.mBitsPerChannel     = 16;
    audioFormat.mBytesPerPacket     = 2;
    audioFormat.mBytesPerFrame      = 2;

    // set the format on the output stream
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_StreamFormat,
                                  kAudioUnitScope_Output,
                                  kInputBus,
                                  &audioFormat,
                                  sizeof(audioFormat));

    [self hasError:status:__FILE__:__LINE__];



    /**
     We need to define a callback structure which holds
     a pointer to the recordingCallback and a reference to
     the audio processor object
     */
    AURenderCallbackStruct callbackStruct;

    /*
     We do the same on the output stream to hear what is coming
     from the input stream
     */
    callbackStruct.inputProc = playbackCallback;
    callbackStruct.inputProcRefCon = (__bridge void *)(self);

    // set playbackCallback as callback on our renderer for the output bus
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_SetRenderCallback,
                                  kAudioUnitScope_Global,
                                  kOutputBus,
                                  &callbackStruct,
                                  sizeof(callbackStruct));

    [self hasError:status:__FILE__:__LINE__];

    // reset flag to 0
    flag = 0;

    /*
     we need to tell the audio unit to allocate the render buffer,
     that we can directly write into it.
     */
    status = AudioUnitSetProperty(audioUnit,
                                  kAudioUnitProperty_ShouldAllocateBuffer,
                                  kAudioUnitScope_Output,
                                  kInputBus,
                                  &flag,
                                  sizeof(flag));

    /*
     we set the number of channels to mono and allocate our block size to
     1024 bytes.
     */
    audioBuffer.mNumberChannels = 1;
    audioBuffer.mDataByteSize = 512 * 2;
    audioBuffer.mData = malloc( 512 * 2 );

    // Initialize the Audio Unit and cross fingers =)
    status = AudioUnitInitialize(audioUnit);
    [self hasError:status:__FILE__:__LINE__];

    NSLog(@"Started");

}

开始演奏

-(void)start;
{
    // start the audio unit. You should hear something, hopefully <img src="http://www.stefanpopp.de/wp-includes/images/smilies/icon_smile.gif" alt=":)" class="wp-smiley">
    OSStatus status = AudioOutputUnitStart(audioUnit);
    [self hasError:status:__FILE__:__LINE__];
}

向缓冲区添加数据

-(void)processBuffer: (AudioBufferList*) audioBufferList
{
    AudioBuffer sourceBuffer = audioBufferList->mBuffers[0];

    // we check here if the input data byte size has changed
    if (audioBuffer.mDataByteSize != sourceBuffer.mDataByteSize) {
        // clear old buffer
        free(audioBuffer.mData);
        // assing new byte size and allocate them on mData
        audioBuffer.mDataByteSize = sourceBuffer.mDataByteSize;
        audioBuffer.mData = malloc(sourceBuffer.mDataByteSize);
    }
    // loop over every packet
    // copy incoming audio data to the audio buffer
    memcpy(audioBuffer.mData, audioBufferList->mBuffers[0].mData, audioBufferList->mBuffers[0].mDataByteSize);
}

流连接回调(套接字)

-(void)stream:(NSStream *)aStream handleEvent:(NSStreamEvent)eventCode
{
    if(eventCode == NSStreamEventHasBytesAvailable)
    {
        if(aStream == inputStream) {
            uint8_t buffer[1024];
            UInt32 len;
            while ([inputStream hasBytesAvailable]) {
                len = (UInt32)[inputStream read:buffer maxLength:sizeof(buffer)];
                if(len > 0)
                {
                    AudioBuffer abuffer;

                    abuffer.mDataByteSize = len; // sample size
                    abuffer.mNumberChannels = 1; // one channel
                    abuffer.mData = buffer;

                    int16_t audioBuffer[len];

                    for(int i = 0; i <= len; i++)
                    {
                        audioBuffer[i] = MuLaw_Decode(buffer[i]);
                    }

                    AudioBufferList bufferList;
                    bufferList.mNumberBuffers = 1;
                    bufferList.mBuffers[0] = abuffer;

                    NSLog(@"%", bufferList.mBuffers[0]);

                    [audioProcessor processBuffer:&bufferList];
                }
            }
        }
    }
}

The Decode

#define MULAW_BIAS 33
int16_t MuLaw_Decode(uint8_t number)
{
    uint8_t sign = 0, position = 0;
    int16_t decoded = 0;
    number =~ number;
    if(number&0x80)
    {
        number&=~(1<<7);
        sign = -1;
    }
    position= ((number & 0xF0) >> 4) + 5;
    decoded = ((1<<position) | ((number&0x0F) << (position - 4)) |(1<<(position-5))) - MULAW_BIAS;
    return (sign == 0) ? decoded : (-(decoded));
}

以及打开连接并初始化音频处理器的代码。

CFReadStreamRef readStream;
CFWriteStreamRef writeStream;



CFStreamCreatePairWithSocketToHost(NULL, (CFStringRef)@"10.0.0.14", 6000, &readStream, &writeStream);


inputStream = (__bridge_transfer NSInputStream *)readStream;
outputStream = (__bridge_transfer NSOutputStream *)writeStream;

[inputStream setDelegate:self];
[outputStream setDelegate:self];

[inputStream scheduleInRunLoop:[NSRunLoop currentRunLoop] forMode:NSDefaultRunLoopMode];
[outputStream scheduleInRunLoop:[NSRunLoop currentRunLoop] forMode:NSDefaultRunLoopMode];
[inputStream open];
[outputStream open];


audioProcessor = [[AudioProcessor alloc] init];
[audioProcessor start];
[audioProcessor setGain:1];

我相信我的代码中的问题是套接字连接回调，我没有对数据做正确的事情。

Answer 1

我最终解决了这个问题，见我的答案here

我打算把代码放在这里，但那将是大量的复制粘贴。