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