简单mp3播放器：音频播放核心

简单mp3播放器：音频播放核心

tag：C，mp3解码，libMAD，PortAudio

/* Create by zyzx
* Created 2008-08-09
* Modified 2008-08-09
*/

一、准备
       1、依赖库PortAudio（ http://www.portaudio.com/）
             Win32平台编译见《 Win32环境PortAudio库编译简介：音频播放》

       2、依赖库libMAD ( http://www.underbit.com/products/mad/)
             简介：开源跨平台的mpeg-1，mpeg-2，mpeg-2.5，MP3解码C库。
             编译环境 Win32 + VS
             下载libMAD 打开 ..\msvc++\LibMad.sln 可以用VS直接编译。

二、编码试验
        如上一篇《 简单音频播放：音频播放核心实现》的代码实现，不同的是LibMad 提供给我们的解码方式稍微有些区别。

        如下源码参照 PortAudio库中示例patest_read_write_wire.c 和 LibMad库中示例minimad.c文件进行的修改，还有Audacity中的部分实现。。所以还有很多地方懒得去理会，代码难看点。。

        需要注意的地方：
        基本上同上一篇。
        1）PCM格式，分左右声道2个缓存区域
        2）PortAudio接受一个缓存区域，左右声道并列方式组织。

        3）当然如果要让你的音频播放模块能够支持更多种格式，可要做一番设计了。。要做到可以替换编解码模块，驱动声卡模块，跨平台，移植ARM那些系统，还是不容易的。。


如下是全部的源码：

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "portaudio.h"

#include <sys/types.h>

#include "mad.h"

#include "windows.h"

/* #define SAMPLE_RATE (17932) // Test failure to open with this value. */
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER ( 64 * 1024 )
#define NUM_CHANNELS    (2)
/* #define DITHER_FLAG     (paDitherOff) */
#define DITHER_FLAG     (0) /**/

/* Select sample format. */
#if 1
#define PA_SAMPLE_TYPE paFloat32
#define SAMPLE_SIZE (4)
#define SAMPLE_SILENCE (0.0f)
#define CLEAR(a) bzero( (a), FRAMES_PER_BUFFER * NUM_CHANNELS * SAMPLE_SIZE )
#define PRINTF_S_FORMAT "%.8f"
#elif 0
#define PA_SAMPLE_TYPE paInt16
#define SAMPLE_SIZE (2)
#define SAMPLE_SILENCE (0)
#define CLEAR(a) bzero( (a), FRAMES_PER_BUFFER * NUM_CHANNELS * SAMPLE_SIZE )
#define PRINTF_S_FORMAT "%d"
#elif 0
#define PA_SAMPLE_TYPE paInt24
#define SAMPLE_SIZE (3)
#define SAMPLE_SILENCE (0)
#define CLEAR(a) bzero( (a), FRAMES_PER_BUFFER * NUM_CHANNELS * SAMPLE_SIZE )
#define PRINTF_S_FORMAT "%d"
#elif 0
#define PA_SAMPLE_TYPE paInt8
#define SAMPLE_SIZE (1)
#define SAMPLE_SILENCE (0)
#define CLEAR(a) bzero( (a), FRAMES_PER_BUFFER * NUM_CHANNELS * SAMPLE_SIZE )
#define PRINTF_S_FORMAT "%d"
#else
#define PA_SAMPLE_TYPE paFloat32//paUInt8
#define SAMPLE_SIZE ( sizeof(float) )
#define SAMPLE_SILENCE (128)
#define CLEAR( a ) { \
    int i; \
    for( i=0; i<FRAMES_PER_BUFFER*NUM_CHANNELS; i++ ) \
    ((unsigned char *)a)[i] = (SAMPLE_SILENCE); \
}
#define PRINTF_S_FORMAT "%d"
#endif

#define WIN32_AUDIO_BUFF_LEN    32 * 1024 * 1024

#define WIN32_BUFF_MAD_LEN        FRAMES_PER_BUFFER

typedef struct
{
    float*        buffer;                               //* - 解码后的数据存储区域
    long        current, bufferlen ;

    unsigned char*    bufferMad;           //* - 由于LibMad是一段一段读写，这里是缓冲区域
    long    bfMadLen;

    unsigned int samplerate;        /* sampling frequency (Hz) */
    unsigned short channels;        /* number of channels */

    FILE* file;

} Win32_Data ;

struct MadBuffer {
    unsigned char const *start;
    unsigned long length;
};

static
enum mad_flow input(void *data,
struct mad_stream *stream)
{
    Win32_Data *buffer = (Win32_Data*)data;

    //if (!buffer->length)
    //    return MAD_FLOW_STOP;

    //mad_stream_buffer(stream, buffer->start, buffer->length);

    //buffer->length = 0;

    if( feof( buffer->file ) ) {
        return MAD_FLOW_STOP;
    }

    //* - 这段摘自Audacity( http://audacity.sourceforge.net/)
    //* - 由于mp3编码方式，可能某次读取的数据并不是完整的，我们只取完整的 frame段，
    //* - 将剩余的部分留给下段。

    unsigned int unconsumedBytes;
    if(stream->next_frame) {
        unconsumedBytes = buffer->bufferMad + WIN32_BUFF_MAD_LEN - stream->next_frame;
        memmove(buffer->bufferMad, stream->next_frame, unconsumedBytes);
    }
    else
        unconsumedBytes = 0;

    off_t read = fread(buffer->bufferMad + unconsumedBytes, sizeof( unsigned char ), WIN32_BUFF_MAD_LEN - unconsumedBytes, buffer->file);

    mad_stream_buffer(stream, buffer->bufferMad, read + unconsumedBytes);

    return MAD_FLOW_CONTINUE;
}

/*
* The following utility routine performs simple rounding, clipping, and
* scaling of MAD's high-resolution samples down to 16 bits. It does not
* perform any dithering or noise shaping, which would be recommended to
* obtain any exceptional audio quality. It is therefore not recommended to
* use this routine if high-quality output is desired.
*/

static inline
float scale(mad_fixed_t sample)
{
    /* round */
    //sample += (1L << (MAD_F_FRACBITS - 16));

    ///* clip */
    //if (sample >= MAD_F_ONE)
    //    sample = MAD_F_ONE - 1;
    //else if (sample < -MAD_F_ONE)
    //    sample = -MAD_F_ONE;

    ///* quantize */
    //return sample >> (MAD_F_FRACBITS + 1 - 16);

    //* 将libMad库内置的采样点格式转换为 32位float类型
    return (float) (sample / (float) (1L << MAD_F_FRACBITS));
}

/*
* This is the output callback function. It is called after each frame of
* MPEG audio data has been completely decoded. The purpose of this callback
* is to output (or play) the decoded PCM audio.
*/

static
enum mad_flow output(void *data,
struct mad_header const *header,
struct mad_pcm *pcm)
{
    unsigned int nsamples;
    mad_fixed_t const *left_ch, *right_ch;
    Win32_Data* buffer = ( Win32_Data* ) data;

    ///* pcm->samplerate contains the sampling frequency */

    buffer->channels = pcm->channels;
    buffer->samplerate = pcm->samplerate;
    nsamples = pcm->length;
    left_ch   = pcm->samples[0];
    right_ch = pcm->samples[1];

    while (nsamples--) {
        float sample;

        /* output sample(s) in 16-bit signed little-endian PCM */
        //* - PCM格式 的数据是分布在2个缓存区域中分左右声道，而PortAudio库是只接受一个区域
        //* - 左右声道数据相邻，所以转换下。
        if ( buffer->channels == 2) {
            sample = scale(*left_ch++);
            buffer->buffer[ buffer->current++ ] = sample;
            sample = scale(*right_ch++);
            buffer->buffer[ buffer->current++ ] = sample;
        }
    }

    return MAD_FLOW_CONTINUE;
}

/*
* This is the error callback function. It is called whenever a decoding
* error occurs. The error is indicated by stream->error; the list of
* possible MAD_ERROR_* errors can be found in the mad.h (or stream.h)
* header file.
*/

static
enum mad_flow error(void *data,
struct mad_stream *stream,
struct mad_frame *frame)
{
    //struct buffer *buffer = data;

    //fprintf(stderr, "decoding error 0x%04x (%s) at byte offset %u\n",
    //    stream->error, mad_stream_errorstr(stream),
    //    stream->this_frame - buffer->start);

    ///* return MAD_FLOW_BREAK here to stop decoding (and propagate an error) */

    return MAD_FLOW_CONTINUE;
}

/*
* This is the function called by main() above to perform all the decoding.
* It instantiates a decoder object and configures it with the input,
* output, and error callback functions above. A single call to
* mad_decoder_run() continues until a callback function returns
* MAD_FLOW_STOP (to stop decoding) or MAD_FLOW_BREAK (to stop decoding and
* signal an error).
*/

static
int decode( Win32_Data * data )
{
    //struct buffer buffer;
    struct mad_decoder decoder;
    int result;

    ///* initialize our private message structure */

    //buffer.start = start;
    //buffer.length = length;

    ///* configure input, output, and error functions */

    mad_decoder_init(&decoder, data,
                    input, 0 /* header */, 0 /* filter */, output,
                    error, 0 /* message */);

    ///* start decoding */

    //* - 此函数是同步函数，在“input”函数返回MAD_FLOW_STOP后才完成。
    //* - 出错在error函数中定义行为。
    result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);

    ///* release the decoder */

    mad_decoder_finish(&decoder);

    return result;
}

/*******************************************************************/
int main(int argc, char *argv [] );
int main(int argc, char *argv [] )
{

    Win32_Data audio_data;

    audio_data.buffer = new float[ WIN32_AUDIO_BUFF_LEN ] ;
    audio_data.bufferMad = new unsigned char[ WIN32_BUFF_MAD_LEN ];
    audio_data.current = 0;

    PaStreamParameters inputParameters, outputParameters;
    PaStream *stream = NULL;
    PaError err;
    char *sampleBlock;
    int i;
    int numBytes;


    printf("patest_read_write_wire.c\n"); fflush(stdout);

    //numBytes = FRAMES_PER_BUFFER * NUM_CHANNELS * SAMPLE_SIZE ;
    //sampleBlock = (char *) malloc( numBytes );
    //if( sampleBlock == NULL )
    //{
    //    printf("Could not allocate record array.\n");
    //    exit(1);
    //}

    audio_data.file = fopen ( argv[1] , "rb" );

    if (audio_data.file == NULL)
    {
        fputs ("File error",stderr);
        exit (1);
    }

    printf( "-- Start Decode %s\n" , argv[1] );

    decode( &audio_data );
    audio_data.bufferlen = audio_data.current;
    audio_data.current = 0;

    fclose( audio_data.file );
    printf( "-- End Decode\n" );

    printf( "-------------------------------\n" );

    printf( "Output samplerate %d\n", audio_data.samplerate );
    printf( "Output channels %d\n", audio_data.channels );

    err = Pa_Initialize();
    if( err != paNoError ) goto error;

    //printf( "-------------------------------\n" );

    //printf( "Output samplerate %d\n", audio_data.sfinfo.samplerate );
    //printf( "Output frames %d\n", audio_data.sfinfo.frames );
    //printf( "Output channels %d\n", audio_data.sfinfo.channels );
    //printf( "seekable %d\n ", audio_data.sfinfo.seekable );
    //printf( "format is %d\n ", audio_data.sfinfo.format );
    //printf( "Sections is %d\n ", audio_data.sfinfo.sections );

    //printf( "Read buffer len %d\n" , audio_data.bufferlen );

    //printf( "-------------------------------\n" );

    //inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
    //printf( "Input device # %d.\n", inputParameters.device );
    //printf( "Input LL: %g s\n", Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency );
    //printf( "Input HL: %g s\n", Pa_GetDeviceInfo( inputParameters.device )->defaultHighInputLatency );
    //inputParameters.channelCount = audio_data.sfinfo.channels;//NUM_CHANNELS;
    //inputParameters.sampleFormat = PA_SAMPLE_TYPE;
    //inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultHighInputLatency ;
    //inputParameters.hostApiSpecificStreamInfo = NULL;

    outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
    printf( "Output device # %d.\n", outputParameters.device );
    printf( "Output LL: %g s\n", Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency );
    printf( "Output HL: %g s\n", Pa_GetDeviceInfo( outputParameters.device )->defaultHighOutputLatency );
    outputParameters.channelCount = audio_data.channels;//NUM_CHANNELS;
    outputParameters.sampleFormat = PA_SAMPLE_TYPE;
    outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultHighOutputLatency;
    outputParameters.hostApiSpecificStreamInfo = NULL;

    /* -- setup -- */

    err = Pa_OpenStream(
        &stream,
        NULL,//&inputParameters,
        &outputParameters,
        SAMPLE_RATE,
        FRAMES_PER_BUFFER,
        paClipOff,      /* we won't output out of range samples so don't bother clipping them */
        NULL, /* no callback, use blocking API */
        NULL ); /* no callback, so no callback userData */
    if( err != paNoError ) goto error;

    //err = Pa_StartStream( stream );
    //if( err != paNoError ) goto error;
    //printf("Wire on. Will run one minute.\n"); fflush(stdout);

    //for( i=0; i<(60*SAMPLE_RATE)/FRAMES_PER_BUFFER; ++i )
    //{
    //    err = Pa_WriteStream( stream, sampleBlock, FRAMES_PER_BUFFER );
    //    if( err ) goto xrun;
    //    err = Pa_ReadStream( stream, sampleBlock, FRAMES_PER_BUFFER );
    //    if( err ) goto xrun;
    //}
    //err = Pa_StopStream( stream );
    //if( err != paNoError ) goto error;

    //CLEAR( sampleBlock );

    err = Pa_StartStream( stream );
    if( err != paNoError ) goto error;
    printf("Wire on. Interrupt to stop.\n"); fflush(stdout);

    while( 1 )
    {

        err = Pa_WriteStream( stream, audio_data.buffer + audio_data.current/*sampleBlock*/, FRAMES_PER_BUFFER );
        audio_data.current = audio_data.current + FRAMES_PER_BUFFER * NUM_CHANNELS /* SAMPLE_SIZE */;

        //if( audio_data.current + FRAMES_PER_BUFFER * NUM_CHANNELS * SAMPLE_SIZE > audio_data.bufferlen ) break;

        printf( "%d\n", audio_data.current);

        if( err ) goto xrun;
        //err = Pa_ReadStream( stream, sampleBlock, FRAMES_PER_BUFFER );
        //if( err ) goto xrun;
    }
    err = Pa_StopStream( stream );
    if( err != paNoError ) goto error;

    Pa_CloseStream( stream );

    free( sampleBlock );

    Pa_Terminate();
    return 0;

xrun:
    if( stream ) {
        Pa_AbortStream( stream );
        Pa_CloseStream( stream );
    }
    free( sampleBlock );
    Pa_Terminate();
    if( err & paInputOverflow )
        fprintf( stderr, "Input Overflow.\n" );
    if( err & paOutputUnderflow )
        fprintf( stderr, "Output Underflow.\n" );

    delete [] audio_data.buffer;
    delete [] audio_data.bufferMad;
    return -2;

error:
    if( stream ) {
        Pa_AbortStream( stream );
        Pa_CloseStream( stream );
    }
    free( sampleBlock );
    Pa_Terminate();
    fprintf( stderr, "An error occured while using the portaudio stream\n" );
    fprintf( stderr, "Error number: %d\n", err );
    fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );

    delete [] audio_data.buffer;
    delete [] audio_data.bufferMad;
    return -1;
}

三、某此播放mp3文件 截图 -- 留念