iOS Voip应用程序\ AudioQueue \ AVSession类别

Question

在我的iOS应用程序中，我使用AudioQueue进行音频录制和播放，基本上我已经在iOS上运行并移植了OSX版本。

我意识到在iOS中我需要配置/设置AV会话，直到现在我已经完成了，

-(void)initAudioSession{

    //get your app's audioSession singleton object

    AVAudioSession* session = [AVAudioSession sharedInstance];

    //error handling
    BOOL success;
    NSError* error;

    //set the audioSession category.
    //Needs to be Record or PlayAndRecord to use audioRouteOverride:

    success = [session setCategory:AVAudioSessionCategoryPlayAndRecord
                             error:&error];

    if (!success)  NSLog(@"AVAudioSession error setting category:%@",error);

    //set the audioSession override
    success = [session overrideOutputAudioPort:AVAudioSessionPortOverrideSpeaker
                                         error:&error];
    if (!success)  NSLog(@"AVAudioSession error overrideOutputAudioPort:%@",error);

    //activate the audio session
    success = [session setActive:YES error:&error];
    if (!success) NSLog(@"AVAudioSession error activating: %@",error);
    else NSLog(@"audioSession active");
}

现在所发生的是，议长AudioQueue的回调永远不会被称为，我检查了许多答案，评论在so，谷歌等.看起来是正确的，我所做的是

• 为输入和输出创建AudioQueue :配置线性PCM，16000采样率
• 分配缓冲器
• 设置具有有效回调的队列，
• 开始排队，

这似乎很好，我可以听到另一端的输出(即输入AudioQueue正在工作)，但是输出AudioQueue (即AudioQueueOutputCallback永远不会被调用)。

我怀疑我需要设置适当的AVSessionCatogery，我正在尝试所有可能的选择，但在演讲者中没有听到任何声音，

我将我的实现与在主线程上运行斯派克的苹果示例AudioQueue进行比较。

即使我不开始输入AudioQueue (麦克风)，我也一样的行为。而且很难有说话的行为，即停止记录和演奏。

谢谢你看它，期待你的评论/帮助。将能够共享代码片段。

Answer 1

谢谢你看，我意识到了问题，这是我的回调，

void AudioStream::AQBufferCallback(void *                   inUserData,
                                   AudioQueueRef            inAQ,
                                   AudioQueueBufferRef      inCompleteAQBuffer)
{
    AudioStream *THIS = (AudioStream *)inUserData;
    if (THIS->mIsDone) {
        return;
    }

    if ( !THIS->IsRunning()){
        NSLog(@" AudioQueue is not running");
        **return;** // Error part 

    }


    int bytes = THIS->bufferByteSize;       
    if ( !THIS->pSingleBuffer){
        THIS->pSingleBuffer = new unsigned char[bytes];
    }
    unsigned char *buffer = THIS->pSingleBuffer;



    if ((THIS->mNumPacketsToRead) > 0) {
        /* lets read only firt packet */
        memset(buffer,0x00,bytes);


        float volume = THIS->volume();

        if (THIS->volumeChange){
            SInt16 *editBuffer = (SInt16 *)buffer;

            // loop over every packet

            for (int nb = 0; nb < (sizeof(buffer) / 2); nb++) {

                // we check if the gain has been modified to save resoures
                if (volume != 0) {
                    // we need more accuracy in our calculation so we calculate with doubles
                    double gainSample = ((double)editBuffer[nb]) / 32767.0;

                    /*
                     at this point we multiply with our gain factor
                     we dont make a addition to prevent generation of sound where no sound is.

                     no noise
                     0*10=0

                     noise if zero
                     0+10=10
                     */
                    gainSample *= volume;

                    /**
                     our signal range cant be higher or lesser -1.0/1.0
                     we prevent that the signal got outside our range
                     */
                    gainSample = (gainSample < -1.0) ? -1.0 : (gainSample > 1.0) ? 1.0 : gainSample;

                    /*
                     This thing here is a little helper to shape our incoming wave.
                     The sound gets pretty warm and better and the noise is reduced a lot.
                     Feel free to outcomment this line and here again.

                     You can see here what happens here http://silentmatt.com/javascript-function-plotter/
                     Copy this to the command line and hit enter: plot y=(1.5*x)-0.5*x*x*x
                     */

                    gainSample = (1.5 * gainSample) - 0.5 * gainSample * gainSample * gainSample;

                    // multiply the new signal back to short
                    gainSample = gainSample * 32767.0;

                    // write calculate sample back to the buffer
                    editBuffer[nb] = (SInt16)gainSample;
                }
            }
        }
        else{
            //            NSLog(@" No change in the volume");
        }


        memcpy(inCompleteAQBuffer->mAudioData, buffer, 640);

        inCompleteAQBuffer->mAudioDataByteSize = 640;
        inCompleteAQBuffer->mPacketDescriptionCount = 320;
        show_err(AudioQueueEnqueueBuffer(inAQ, inCompleteAQBuffer, 0, NULL));
    }
}

由于在分配它时我没有排队，而且我认为它必须在启动之前对几个缓冲区进行排队，因此删除返回部分解决了我的问题。