User Tools

Site Tools


inputting_20real-time_20audio

Differences

This shows you the differences between two versions of the page.

Link to this comparison view

inputting_20real-time_20audio [2018/03/31 13:19]
127.0.0.1 external edit
inputting_20real-time_20audio [2018/04/15 11:56] (current)
richardrussell Added syntax highlighting
Line 1: Line 1:
 =====Inputting real-time audio===== =====Inputting real-time audio=====
  
-//by Richard Russell, November 2006//\\ \\  Most PCs provide the capability of inputting audio, typically either from a microphone or from a line-level input. This article describes how a BBC BASIC for Windows program can read and process audio data from this source in real-time (i.e. without it having to be recorded to a file first).\\ \\ +//by Richard Russell, November 2006//\\ \\  Most PCs provide the capability of inputting audio, typically either from a microphone or from a line-level input. This article describes how a BBC BASIC for Windows program can read and process audio data from this source in real-time (i.e. without it having to be recorded to a file first). 
 + 
 ==== Preliminaries ==== ==== Preliminaries ====
-\\  ​As is usual for programs accessing the Windows API, it is important to trap errors, and closing the window, so that the necessary '​cleanup'​ operations can take place:\\ \\ + 
 +As is usual for programs accessing the Windows API, it is important to trap errors, and closing the window, so that the necessary '​cleanup'​ operations can take place: 
 + 
 +<code bb4w>
         ON ERROR PROCcleanup : SYS "​MessageBox",​ @hwnd%, REPORT$, 0, 48 : QUIT         ON ERROR PROCcleanup : SYS "​MessageBox",​ @hwnd%, REPORT$, 0, 48 : QUIT
         ON CLOSE PROCcleanup : QUIT         ON CLOSE PROCcleanup : QUIT
-The **PROCcleanup** routine is listed later. You may want to change the error reporting to a different method.\\ \\ +</​code>​ 
 + 
 +The **PROCcleanup** routine is listed later. You may want to change the error reporting to a different method. 
 ==== Selecting the audio format ==== ==== Selecting the audio format ====
-\\  ​The first step is to decide the audio //format// you will use: the principal choices being of sampling rate (the main ones being 11025 Hz, 22050 Hz and 44100 Hz) and number of channels (mono, 1 channel, or stereo, 2 channels). The higher the sampling rate the higher the audio frequency that can be received, but the more work your software needs to do. Normally you should choose the lowest sampling rate suitable for your application,​ remembering that it needs to be at least **double** the highest audio frequency in which you are interested (according to the [[http://​en.wikipedia.org/​wiki/​Nyquist_rate|Nyquist criterion]]).\\ \\  You set up the required audio format and open the wave input device as follows:\\ \\ + 
 +The first step is to decide the audio //format// you will use: the principal choices being of sampling rate (the main ones being 11025 Hz, 22050 Hz and 44100 Hz) and number of channels (mono, 1 channel, or stereo, 2 channels). The higher the sampling rate the higher the audio frequency that can be received, but the more work your software needs to do. Normally you should choose the lowest sampling rate suitable for your application,​ remembering that it needs to be at least **double** the highest audio frequency in which you are interested (according to the [[http://​en.wikipedia.org/​wiki/​Nyquist_rate|Nyquist criterion]]).\\ \\  You set up the required audio format and open the wave input device as follows: 
 + 
 +<code bb4w>
         DIM Format{wFormatTag{l&,​h&​},​ nChannels{l&,​h&​},​ nSamplesPerSec%,​ \         DIM Format{wFormatTag{l&,​h&​},​ nChannels{l&,​h&​},​ nSamplesPerSec%,​ \
         \          nAvgBytesPerSec%,​ nBlockAlign{l&,​h&​},​ wBitsPerSample{l&,​h&​},​ \         \          nAvgBytesPerSec%,​ nBlockAlign{l&,​h&​},​ wBitsPerSample{l&,​h&​},​ \
Line 23: Line 33:
         SYS "​waveInOpen",​ ^WaveIn%, _WAVE_MAPPER,​ Format{}, 0, 0, 0 TO ret%         SYS "​waveInOpen",​ ^WaveIn%, _WAVE_MAPPER,​ Format{}, 0, 0, 0 TO ret%
         IF ret% ERROR 100, "​waveInOpen failed: "​+STR$~ret%         IF ret% ERROR 100, "​waveInOpen failed: "​+STR$~ret%
-In this example a sampling rate of 44100 Hz has been selected. Note that you cannot be sure that the incoming audio is actually being sampled at the specified rate. On some PCs the sampling rate may be as low as 11025 Hz even if a higher frequency has been selected in your program.\\ \\ +</​code>​ 
 + 
 +In this example a sampling rate of 44100 Hz has been selected. Note that you cannot be sure that the incoming audio is actually being sampled at the specified rate. On some PCs the sampling rate may be as low as 11025 Hz even if a higher frequency has been selected in your program. 
 ==== Creating and initialising the buffers ==== ==== Creating and initialising the buffers ====
-\\  ​The next step is to decide how many audio buffers you need and how large they should be. To some extent this is an arbitrary decision, but it will depend on things like //latency// (how much time elapses between the audio arriving and it being processed by your program) and the amount of work needed to process the received audio data.\\ \\  Normally you should have at least three buffers: one inputting the sampled sound, one being processed by your program, and one spare (the buffers are reused cyclically). It is vitally important that your program can process the audio data quickly enough, otherwise data will be lost, with undesirable results. If there is any variability in the rate at which you can process the data (for example it depends on disk or network accesses) then you may need to use more and/or larger buffers to 'iron out' the fluctuation. Using more buffers is generally preferable to using larger buffers, to minimise any increase in latency.\\ \\  In the example below the number of buffers is three and the length of each buffer is 1024 samples; at 44100 Hz that implies a latency of at least 24 milliseconds. The code for creating and initialising the buffers is as follows:\\ \\ + 
 +The next step is to decide how many audio buffers you need and how large they should be. To some extent this is an arbitrary decision, but it will depend on things like //latency// (how much time elapses between the audio arriving and it being processed by your program) and the amount of work needed to process the received audio data.\\ \\  Normally you should have at least three buffers: one inputting the sampled sound, one being processed by your program, and one spare (the buffers are reused cyclically). It is vitally important that your program can process the audio data quickly enough, otherwise data will be lost, with undesirable results. If there is any variability in the rate at which you can process the data (for example it depends on disk or network accesses) then you may need to use more and/or larger buffers to 'iron out' the fluctuation. Using more buffers is generally preferable to using larger buffers, to minimise any increase in latency.\\ \\  In the example below the number of buffers is three and the length of each buffer is 1024 samples; at 44100 Hz that implies a latency of at least 24 milliseconds. The code for creating and initialising the buffers is as follows: 
 + 
 +<code bb4w>
         nBuffers% = 3         nBuffers% = 3
         SamplesPerBuffer% = 1024         SamplesPerBuffer% = 1024
Line 47: Line 63:
           IF ret% ERROR 100, "​waveInAddBuffer failed: "​+STR$~ret%           IF ret% ERROR 100, "​waveInAddBuffer failed: "​+STR$~ret%
         NEXT         NEXT
-Note that in this case the audio buffers are allocated from BASIC'​s heap; you could alternatively use the Windows API to allocate the memory.\\ \\ +</​code>​ 
 + 
 +Note that in this case the audio buffers are allocated from BASIC'​s heap; you could alternatively use the Windows API to allocate the memory. 
 ==== Starting audio capture ==== ==== Starting audio capture ====
-\\  ​Once you have prepared the wave input system using the above code, you can start the real-time capture as follows:\\ \\ + 
 +Once you have prepared the wave input system using the above code, you can start the real-time capture as follows: 
 + 
 +<code bb4w>
         SYS "​waveInStart",​ WaveIn% TO ret%         SYS "​waveInStart",​ WaveIn% TO ret%
         IF ret% ERROR 100, "​waveInStart failed: "​+STR$~ret%         IF ret% ERROR 100, "​waveInStart failed: "​+STR$~ret%
-\\ +</​code>​ 
 ==== Inputting in real-time ==== ==== Inputting in real-time ====
-\\  ​Once the above code has been executed you need to process the received audio buffers fast enough to keep up with the incoming data. The following code constantly checks whether any of the buffers needs processing and if so calls the **PROCprocessbuffer** routine:\\ \\ + 
 +Once the above code has been executed you need to process the received audio buffers fast enough to keep up with the incoming data. The following code constantly checks whether any of the buffers needs processing and if so calls the **PROCprocessbuffer** routine: 
 + 
 +<code bb4w>
         _WHDR_DONE = 1         _WHDR_DONE = 1
         REPEAT         REPEAT
Line 66: Line 92:
           SYS "​Sleep",​ 1           SYS "​Sleep",​ 1
         UNTIL FALSE         UNTIL FALSE
-In this example the audio processing continues indefinitely,​ but you can terminate the program prematurely if you wish. If you do, don't forget to execute **PROCcleanup** before exiting the program.\\ \\ +</​code>​ 
 + 
 +In this example the audio processing continues indefinitely,​ but you can terminate the program prematurely if you wish. If you do, don't forget to execute **PROCcleanup** before exiting the program. 
 ==== Processing the audio data ==== ==== Processing the audio data ====
-\\  ​Obviously it's only possible to describe this aspect in general terms, because precisely what audio processing takes place will depend on what the program is designed to do. The code below simply calculates the RMS (Root Mean Square) value of the incoming audio:\\ \\ + 
 +Obviously it's only possible to describe this aspect in general terms, because precisely what audio processing takes place will depend on what the program is designed to do. The code below simply calculates the RMS (Root Mean Square) value of the incoming audio: 
 + 
 +<code bb4w>
         DEF PROCprocessbuffer(B%,​N%)         DEF PROCprocessbuffer(B%,​N%)
         LOCAL I%, V%, sumsq         LOCAL I%, V%, sumsq
Line 77: Line 109:
         RMS = SQR(sumsq / N%)         RMS = SQR(sumsq / N%)
         ENDPROC         ENDPROC
-This code is appropriate for monaural input (one channel) where each audio sample consists of a signed 16-bit value in the range -32768 to +32767.\\ \\ +</​code>​ 
 + 
 +This code is appropriate for monaural input (one channel) where each audio sample consists of a signed 16-bit value in the range -32768 to +32767. 
 ==== Cleaning up ==== ==== Cleaning up ====
-\\  ​When you stop the sound capture, or exit the program, you need to shut down the audio input in a controlled fashion:\\ \\ + 
 +When you stop the sound capture, or exit the program, you need to shut down the audio input in a controlled fashion: 
 + 
 +<code bb4w>
         DEF PROCcleanup         DEF PROCcleanup
         WaveIn% += 0 : IF WaveIn% THEN         WaveIn% += 0 : IF WaveIn% THEN
Line 88: Line 126:
         ENDIF         ENDIF
         ENDPROC         ENDPROC
 +</​code>​
 +
 This code might form part of a larger routine, if there are other things that need to be shut down. This code might form part of a larger routine, if there are other things that need to be shut down.
inputting_20real-time_20audio.txt · Last modified: 2018/04/15 11:56 by richardrussell