如何使用c/c++实时录制和播放我的声音

how can i record and play my voice in real time using c/c++

本文关键字:播放 我的声音 实时 何使用 c++      更新时间:2023-10-16

我正在用麦克风录音。为此,我使用了waveIn()和waveOut()函数。当我开始使用waveInStart()函数录音时,它首先在指定的时间内完整地记录我的声音,并将其存储在()缓冲区中,然后从该缓冲区中播放我的声音。在录制和播放期间,它不做任何事情。我想同时录音和演奏。为此,我想在记录期间访问缓冲区。这怎么可能呢?或者其他建议会有帮助的。

Here is the code :

#include <iostream>
#include <Windows.h>
using namespace std;
#pragma comment(lib, "winmm.lib")
 short int waveIn[8000 * 3];
void PlayRecord();
void writedataTofile(LPSTR lpData,DWORD dwBufferLength);
void StartRecord()
{
const int NUMPTS = 8000 * 3;   // 3 seconds
int sampleRate = 8000;  
// 'short int' is a 16-bit type; I request 16-bit samples below
                         // for 8-bit capture, you'd use 'unsigned char' or 'BYTE' 8-bit     types
 HWAVEIN      hWaveIn;
 MMRESULT result;
 WAVEFORMATEX pFormat;
 pFormat.wFormatTag=WAVE_FORMAT_PCM;     // simple, uncompressed format
 pFormat.nChannels=1;                    //  1=mono, 2=stereo
 pFormat.nSamplesPerSec=sampleRate;      // 8.0 kHz, 11.025 kHz, 22.05 kHz, and 44.1 kHz
 pFormat.nAvgBytesPerSec=sampleRate*2;   // =  nSamplesPerSec × nBlockAlign
 pFormat.nBlockAlign=2;                  // = (nChannels × wBitsPerSample) / 8
 pFormat.wBitsPerSample=16;              //  16 for high quality, 8 for telephone-grade
 pFormat.cbSize=0;
 // Specify recording parameters
 result = waveInOpen(&hWaveIn, WAVE_MAPPER,&pFormat,
        0L, 0L, WAVE_FORMAT_DIRECT);
  WAVEHDR      WaveInHdr;
 // Set up and prepare header for input
  WaveInHdr.lpData = (LPSTR)waveIn;
  WaveInHdr.dwBufferLength = NUMPTS*2;
  WaveInHdr.dwBytesRecorded=0;
  WaveInHdr.dwUser = 0L;
  WaveInHdr.dwFlags = 0L;
  WaveInHdr.dwLoops = 0L;
  waveInPrepareHeader(hWaveIn, &WaveInHdr, sizeof(WAVEHDR));
 // Insert a wave input buffer
  result = waveInAddBuffer(hWaveIn, &WaveInHdr, sizeof(WAVEHDR));

 // Commence sampling input
  result = waveInStart(hWaveIn);

 cout << "recording..." << endl;
  Sleep(3 * 1000);
 // Wait until finished recording
 waveInClose(hWaveIn);
PlayRecord();
}
void PlayRecord()
{
const int NUMPTS = 8000 * 3;   // 3 seconds
int sampleRate = 8000;  
// 'short int' is a 16-bit type; I request 16-bit samples below
                            // for 8-bit capture, you'd    use 'unsigned char' or 'BYTE' 8-bit types
HWAVEIN  hWaveIn;
WAVEFORMATEX pFormat;
pFormat.wFormatTag=WAVE_FORMAT_PCM;     // simple, uncompressed format
pFormat.nChannels=1;                    //  1=mono, 2=stereo
pFormat.nSamplesPerSec=sampleRate;      // 44100
pFormat.nAvgBytesPerSec=sampleRate*2;   // = nSamplesPerSec * n.Channels * wBitsPerSample/8
pFormat.nBlockAlign=2;                  // = n.Channels * wBitsPerSample/8
pFormat.wBitsPerSample=16;              //  16 for high quality, 8 for telephone-grade
pFormat.cbSize=0;
// Specify recording parameters
waveInOpen(&hWaveIn, WAVE_MAPPER,&pFormat, 0L, 0L, WAVE_FORMAT_DIRECT);
WAVEHDR      WaveInHdr;
// Set up and prepare header for input
WaveInHdr.lpData = (LPSTR)waveIn;
WaveInHdr.dwBufferLength = NUMPTS*2;
WaveInHdr.dwBytesRecorded=0;
WaveInHdr.dwUser = 0L;
WaveInHdr.dwFlags = 0L;
WaveInHdr.dwLoops = 0L;
waveInPrepareHeader(hWaveIn, &WaveInHdr, sizeof(WAVEHDR));
HWAVEOUT hWaveOut;
cout << "playing..." << endl;
waveOutOpen(&hWaveOut, WAVE_MAPPER, &pFormat, 0, 0, WAVE_FORMAT_DIRECT);
waveOutWrite(hWaveOut, &WaveInHdr, sizeof(WaveInHdr)); // Playing the data
Sleep(3 * 1000); //Sleep for as long as there was recorded

waveInClose(hWaveIn);
waveOutClose(hWaveOut);
}
int main()
{
 StartRecord();
    return 0;
}  

从技术上讲,如果您分配缓冲区,您可以将输入和输出分配到同一个缓冲区,并运行一个线程来执行播放,一个线程来执行录制。然而,我预计你需要的会比这多得多。

问题是缓冲区内容将通过某种机制从内存加载到硬件中,并且这将"预读取"数据以便在小块中播放,以及"缓冲"记录端。驱动程序和硬件都有这种"缓存"机制。这意味着回放将在数据从记录中存储到内存之前读取数据,这当然不会正常工作。

大多数音频处理系统的工作方式是稍微延迟输出,所以你输入一点,处理它,输出它。当然,这会导致一个小的延迟,这是相当恼人的。

在调用waveInStart之前,您可以准备并添加几个缓冲区到音频驱动程序。驱动程序将对它们进行排队,并在不丢失任何数据的情况下对缓冲区进行排序。你必须在waveInOpen中使用fdwOpen标志,这样每次缓冲区被填满时你都会得到一个通知。

waveOut具有相同的排队能力:您可以在播放前一个缓冲区时输出缓冲区,并且它将在缓冲区之间平滑地排序。

所以比起一个3秒的大缓冲区,你可以使用10个0.3秒的缓冲区,并编写代码来在这些缓冲区被填满时从记录到播放。结果将是没有暂停的音频,但有0.3秒的延迟。