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Playing system recorded sound


How can i get 2 wavefilereaders to play sequencially?Continuous record and play audio in C# visual studioHow to record and playback with NAudio using AsioOutHow to record audio using naudio onto byte[] rather than fileNaudio Recording and playing audioPlay sound at an exact momentNAudio WasapiLoopbackCapture intermittent soundRecord sound from microphone using NAudio. Why not works correctly record the whole buffer from the list?Cant play created mp3 file, but can play existing fileNAudio Asio Record and Playback at the same time






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0















I am trying to record my system sound and stream it over tcp to then be played on another pc, but my problem is that then i play the sound i recieve it is playing super slow. I'm using WasapiLoopbackCapture to record system sound, but when i try to play the bytes i record it is playing slowly.
How do i fix this? If i write to a file and then play it, it sounds fine, but i cant directly play the bytes.


This is the code i use to play the sound.



bufferedWaveProvider = new BufferedWaveProvider(CaptureInstance.WaveFormat);
savingWaveProvider = new SavingWaveProvider(bufferedWaveProvider, outputFilePath);
soundPlayer = new WasapiOut();
soundPlayer.Init(bufferedWaveProvider);
soundPlayer.Play();


And later in the code when i recieve a buffer over TCP i add the sample.



bufferedWaveProvider.AddSamples(bytes, 0, bytes.Length);









share|improve this question




























    0















    I am trying to record my system sound and stream it over tcp to then be played on another pc, but my problem is that then i play the sound i recieve it is playing super slow. I'm using WasapiLoopbackCapture to record system sound, but when i try to play the bytes i record it is playing slowly.
    How do i fix this? If i write to a file and then play it, it sounds fine, but i cant directly play the bytes.


    This is the code i use to play the sound.



    bufferedWaveProvider = new BufferedWaveProvider(CaptureInstance.WaveFormat);
    savingWaveProvider = new SavingWaveProvider(bufferedWaveProvider, outputFilePath);
    soundPlayer = new WasapiOut();
    soundPlayer.Init(bufferedWaveProvider);
    soundPlayer.Play();


    And later in the code when i recieve a buffer over TCP i add the sample.



    bufferedWaveProvider.AddSamples(bytes, 0, bytes.Length);









    share|improve this question
























      0












      0








      0








      I am trying to record my system sound and stream it over tcp to then be played on another pc, but my problem is that then i play the sound i recieve it is playing super slow. I'm using WasapiLoopbackCapture to record system sound, but when i try to play the bytes i record it is playing slowly.
      How do i fix this? If i write to a file and then play it, it sounds fine, but i cant directly play the bytes.


      This is the code i use to play the sound.



      bufferedWaveProvider = new BufferedWaveProvider(CaptureInstance.WaveFormat);
      savingWaveProvider = new SavingWaveProvider(bufferedWaveProvider, outputFilePath);
      soundPlayer = new WasapiOut();
      soundPlayer.Init(bufferedWaveProvider);
      soundPlayer.Play();


      And later in the code when i recieve a buffer over TCP i add the sample.



      bufferedWaveProvider.AddSamples(bytes, 0, bytes.Length);









      share|improve this question














      I am trying to record my system sound and stream it over tcp to then be played on another pc, but my problem is that then i play the sound i recieve it is playing super slow. I'm using WasapiLoopbackCapture to record system sound, but when i try to play the bytes i record it is playing slowly.
      How do i fix this? If i write to a file and then play it, it sounds fine, but i cant directly play the bytes.


      This is the code i use to play the sound.



      bufferedWaveProvider = new BufferedWaveProvider(CaptureInstance.WaveFormat);
      savingWaveProvider = new SavingWaveProvider(bufferedWaveProvider, outputFilePath);
      soundPlayer = new WasapiOut();
      soundPlayer.Init(bufferedWaveProvider);
      soundPlayer.Play();


      And later in the code when i recieve a buffer over TCP i add the sample.



      bufferedWaveProvider.AddSamples(bytes, 0, bytes.Length);






      naudio






      share|improve this question













      share|improve this question











      share|improve this question




      share|improve this question










      asked Mar 24 at 11:41









      Mikkel PedersenMikkel Pedersen

      32




      32






















          1 Answer
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          0














          You need to make sure the wave formats match exactly. Typically WASAPI loopback capture will be capturing stereo IEEE floating point samples at either 44.1kHz or 48kHz. So the WaveFormat of the BufferedWaveProvider should be set to exactly the same format as audio is being captured in.



          Another issue to be aware of is that you should be using the BytesRecorded property of the DataAvailable event rather than the length of the buffer. The capture buffer may not always be completely full.



          Finally, raw stereo IEEE float samples at 44.kHz or higher is not a very efficient way of transmitting audio over a network. It may just be that your network can't keep up. Typically programs that transmit audio over a network will apply some kind of codec to reduce the bandwidth requirements.






          share|improve this answer























          • It does use the same waveformat for recording and playing, and im using the BytesRecorded property, but it still doesn't sound smooth when i play the audio. For the codec part im looking into it, but just wanted to get the basic record, steam and play to work first. What else could be wrong, the buffer fills up in no time and crashes.

            – Mikkel Pedersen
            Mar 26 at 8:15












          Your Answer






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          0














          You need to make sure the wave formats match exactly. Typically WASAPI loopback capture will be capturing stereo IEEE floating point samples at either 44.1kHz or 48kHz. So the WaveFormat of the BufferedWaveProvider should be set to exactly the same format as audio is being captured in.



          Another issue to be aware of is that you should be using the BytesRecorded property of the DataAvailable event rather than the length of the buffer. The capture buffer may not always be completely full.



          Finally, raw stereo IEEE float samples at 44.kHz or higher is not a very efficient way of transmitting audio over a network. It may just be that your network can't keep up. Typically programs that transmit audio over a network will apply some kind of codec to reduce the bandwidth requirements.






          share|improve this answer























          • It does use the same waveformat for recording and playing, and im using the BytesRecorded property, but it still doesn't sound smooth when i play the audio. For the codec part im looking into it, but just wanted to get the basic record, steam and play to work first. What else could be wrong, the buffer fills up in no time and crashes.

            – Mikkel Pedersen
            Mar 26 at 8:15
















          0














          You need to make sure the wave formats match exactly. Typically WASAPI loopback capture will be capturing stereo IEEE floating point samples at either 44.1kHz or 48kHz. So the WaveFormat of the BufferedWaveProvider should be set to exactly the same format as audio is being captured in.



          Another issue to be aware of is that you should be using the BytesRecorded property of the DataAvailable event rather than the length of the buffer. The capture buffer may not always be completely full.



          Finally, raw stereo IEEE float samples at 44.kHz or higher is not a very efficient way of transmitting audio over a network. It may just be that your network can't keep up. Typically programs that transmit audio over a network will apply some kind of codec to reduce the bandwidth requirements.






          share|improve this answer























          • It does use the same waveformat for recording and playing, and im using the BytesRecorded property, but it still doesn't sound smooth when i play the audio. For the codec part im looking into it, but just wanted to get the basic record, steam and play to work first. What else could be wrong, the buffer fills up in no time and crashes.

            – Mikkel Pedersen
            Mar 26 at 8:15














          0












          0








          0







          You need to make sure the wave formats match exactly. Typically WASAPI loopback capture will be capturing stereo IEEE floating point samples at either 44.1kHz or 48kHz. So the WaveFormat of the BufferedWaveProvider should be set to exactly the same format as audio is being captured in.



          Another issue to be aware of is that you should be using the BytesRecorded property of the DataAvailable event rather than the length of the buffer. The capture buffer may not always be completely full.



          Finally, raw stereo IEEE float samples at 44.kHz or higher is not a very efficient way of transmitting audio over a network. It may just be that your network can't keep up. Typically programs that transmit audio over a network will apply some kind of codec to reduce the bandwidth requirements.






          share|improve this answer













          You need to make sure the wave formats match exactly. Typically WASAPI loopback capture will be capturing stereo IEEE floating point samples at either 44.1kHz or 48kHz. So the WaveFormat of the BufferedWaveProvider should be set to exactly the same format as audio is being captured in.



          Another issue to be aware of is that you should be using the BytesRecorded property of the DataAvailable event rather than the length of the buffer. The capture buffer may not always be completely full.



          Finally, raw stereo IEEE float samples at 44.kHz or higher is not a very efficient way of transmitting audio over a network. It may just be that your network can't keep up. Typically programs that transmit audio over a network will apply some kind of codec to reduce the bandwidth requirements.







          share|improve this answer












          share|improve this answer



          share|improve this answer










          answered Mar 25 at 15:37









          Mark HeathMark Heath

          36.9k24110177




          36.9k24110177












          • It does use the same waveformat for recording and playing, and im using the BytesRecorded property, but it still doesn't sound smooth when i play the audio. For the codec part im looking into it, but just wanted to get the basic record, steam and play to work first. What else could be wrong, the buffer fills up in no time and crashes.

            – Mikkel Pedersen
            Mar 26 at 8:15


















          • It does use the same waveformat for recording and playing, and im using the BytesRecorded property, but it still doesn't sound smooth when i play the audio. For the codec part im looking into it, but just wanted to get the basic record, steam and play to work first. What else could be wrong, the buffer fills up in no time and crashes.

            – Mikkel Pedersen
            Mar 26 at 8:15

















          It does use the same waveformat for recording and playing, and im using the BytesRecorded property, but it still doesn't sound smooth when i play the audio. For the codec part im looking into it, but just wanted to get the basic record, steam and play to work first. What else could be wrong, the buffer fills up in no time and crashes.

          – Mikkel Pedersen
          Mar 26 at 8:15






          It does use the same waveformat for recording and playing, and im using the BytesRecorded property, but it still doesn't sound smooth when i play the audio. For the codec part im looking into it, but just wanted to get the basic record, steam and play to work first. What else could be wrong, the buffer fills up in no time and crashes.

          – Mikkel Pedersen
          Mar 26 at 8:15




















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