Basic Implementation of Audio Processing in C Language

Audio processing is an important component of computer science and digital signal processing. With the efficiency of the C language, we can implement some basic audio operations. This article will introduce how to perform simple audio reading, playback, and processing using C language.

1. Overview of Basic Knowledge

Audio files generally consist of two main parts: header information and data section. The header information usually contains details about the audio format, such as sample rate, number of channels, and bit depth. We will take the WAV format as an example, which is quite common and easy to understand.

WAV File Structure

The common structure of a WAV file is as follows:

RIFF Header      (12 bytes)
Format Chunk     (24 bytes)
Data Chunk       (Variable size)

• RIFF Header: Contains file identifier, file size, etc.
• Format Chunk: Contains information about the waveform format, such as sample rate, number of channels, and bit depth.
• Data Chunk: The actual sound data.

2. Preparation Work

Ensure you have a development environment, such as the GCC compiler, that can compile and run C programs from the command line. Additionally, you need to find a WAV format audio file as an example.

3. Reading WAV Files

Below is a simple demonstration of how to open and read a WAV file using C language:

#include <stdio.h>
#include <stdlib.h>

typedef struct {
    char riff[4];          // RIFF Header
    unsigned int fileSize;
    char wave[4];          // WAVE header
    char fmt[4];           // format chunk identifier
    unsigned int fmtSize;   // size of the format chunk
    short audioFormat;     // audio format type
    short numChannels;     // number of channels
    unsigned int sampleRate;       // sampling frequency in Hz
    unsigned int byteRate;         // SampleRate * NumChannels * BitsPerSample/8
    short blockAlign;      // NumChannels * BitsPerSample/8
    short bitsPerSample;   // bits per sample for PCM
} WavHeader;

void readWavFile(const char* filename) {
    FILE* file = fopen(filename, "rb");
    if (!file) {
        fprintf(stderr, "Could not open file %s\n", filename);
        return;
    }
    WavHeader header;
    fread(&header, sizeof(WavHeader), 1, file);
    printf("Audio Format: %d\n", header.audioFormat);
    printf("Number of Channels: %d\n", header.numChannels);
    printf("Sample Rate: %d Hz\n", header.sampleRate);
    printf("Bits Per Sample: %d\n", header.bitsPerSample);
    fclose(file);
}

int main() {
    readWavFile("example.wav");
    return 0;
}

Code Explanation

1. We defined the <span>WavHeader</span> structure to store the header information of the WAV file.
2. <span>readWavFile</span> function takes the filename as a parameter and opens the file for binary reading.
3. Using <span>fread</span> function, we read the header data from the file and print relevant information such as audio format, number of channels, and sample rate.

4. Playing WAV Audio (Linux System)

In a Linux system, we can use the ALSA library to play audio data that has been loaded into memory. If ALSA is not installed, you can install the corresponding library using a package manager (such as apt or yum).

The following code snippet demonstrates how to play an audio segment using ALSA:

#include <alsa/asoundlib.h>
// Code omitted for brevity

void playAudio(short* buffer, long dataSize) {
   snd_pcm_t *handle;
   snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
   snd_pcm_set_params(handle,
                      SND_PCM_FORMAT_S16_LE,
                      SND_PCM_ACCESS_RW_INTERLEAVED,
                      2,
                      44100,
                      1,
                      &latency);
   snd_pcm_writei(handle, buffer, dataSize / sizeof(short));
   snd_pcm_drain(handle);
   snd_pcm_close(handle);
}

int main() {
    // Code omitted for brevity
}

Notes:

• In practical applications, ensure that your <span>buffer</span> is filled with the correct PCM format data, and be mindful of the data size required by the audio player.

5. Conclusion and Outlook

This article only covers the basic audio reading and playback functions, but this is just a small part of digital signal processing. In practice, more features can be added, such as speed adjustment, pitch change, or adding effect filters, which will promote the development and expansion of related projects. For example, we can further explore how to modify waveform data to produce different effects and learn more complex topics such as streaming technology or cross-platform support.

I hope this article helps you understand how C language and audio processing work together at a fundamental level.