Linux平台的C语言实现ffmpeg框架+alsa框架的音乐播放器(wav、MP3格式转pcm)
1.ffpeg和alsa环境
(csdn教程一堆)
2.程序源码
#include <alsa/asoundlib.h>
#include <libavutil/time.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libavdevice/avdevice.h>
#include <libswresample/swresample.h>
#include <libswscale/swscale.h>
static char *device = "hw:1,0"; /* playback device */
static snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format */
static unsigned int rate = 44100; /* stream rate */
static unsigned int channels = 2; /* count of channels */
static unsigned int buffer_time = 500000; /* ring buffer length in us */
static unsigned int period_time = 100000; /* period time in us */
static int resample = 1; /* enable alsa-lib resampling */
static snd_pcm_sframes_t buffer_size;
static snd_pcm_sframes_t period_size;
snd_pcm_access_t mode = SND_PCM_ACCESS_RW_INTERLEAVED;
static snd_output_t *output = NULL;
/*配置参数*/
static int set_hwparams(snd_pcm_t *handle,snd_pcm_hw_params_t *params,snd_pcm_access_t access)
{
unsigned int rrate;
snd_pcm_uframes_t size;
int err, dir;
/* choose all parameters */
err = snd_pcm_hw_params_any(handle, params);
if (err < 0) {
printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
return err;
}
/* set hardware resampling */
err = snd_pcm_hw_params_set_rate_resample(handle, params, resample);
if (err < 0) {
printf("Resampling setup failed for playback: %s\n", snd_strerror(err));
return err;
}
/* set the interleaved read/write format */
/*访问格式*/
err = snd_pcm_hw_params_set_access(handle, params, mode);
if (err < 0) {
printf("Access type not available for playback: %s\n", snd_strerror(err));
return err;
}
/* set the sample format */
/*采样格式*/
err = snd_pcm_hw_params_set_format(handle, params, format);
if (err < 0) {
printf("Sample format not available for playback: %s\n", snd_strerror(err));
return err;
}
/* set the count of channels */
/*音频声道*/
err = snd_pcm_hw_params_set_channels(handle, params, channels);
if (err < 0) {
printf("Channels count (%u) not available for playbacks: %s\n", channels, snd_strerror(err));
return err;
}
/* set the stream rate */
/*采样率*/
rrate = rate;
err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0);
if (err < 0) {
printf("Rate %uHz not available for playback: %s\n", rate, snd_strerror(err));
return err;
}
if (rrate != rate) {
printf("Rate doesn't match (requested %uHz, get %iHz)\n", rate, err);
return -EINVAL;
}
/* set the buffer time */
/*底层buffer区间,以时间为单位,500000=0.5s*/
err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir);
if (err < 0) {
printf("Unable to set buffer time %u for playback: %s\n", buffer_time, snd_strerror(err));
return err;
}
err = snd_pcm_hw_params_get_buffer_size(params, &size);
if (err < 0) {
printf("Unable to get buffer size for playback: %s\n", snd_strerror(err));
return err;
}
buffer_size = size;
printf("buffer_size=%ld\n",buffer_size);
/* set the period time */
/*底层period区间,以时间为单位,100000=0.1s*/
err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir);
if (err < 0) {
printf("Unable to set period time %u for playback: %s\n", period_time, snd_strerror(err));
return err;
}
/*底层period区间,以字节为单位,44100*0.1=4410*/
err = snd_pcm_hw_params_get_period_size(params, &size, &dir);
if (err < 0) {
printf("Unable to get period size for playback: %s\n", snd_strerror(err));
return err;
}
period_size = size;
printf("period_size=%ld\n",period_size);
/* write the parameters to device */
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
printf("Unable to set hw params for playback: %s\n", snd_strerror(err));
return err;
}
return 0;
}
int main(int argc, char *argv[])
{
int rc;
int size;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
printf("Playback device is %s\n", device);
printf("Stream parameters are %uHz, %s, %u channels\n", rate, snd_pcm_format_name(format), channels);
int err;
err = snd_output_stdio_attach(&output, stdout, 0);
if (err < 0) {
printf("Output failed: %s\n", snd_strerror(err));
return 0;
}
/*设置播放模式*/
err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0)
{
printf("Playback open error: %s\n", snd_strerror(err));
return 0;
}
/*设置参数*/
err = set_hwparams(handle, hwparams, mode);
if (err < 0) {
printf("Setting of hwparams failed: %s\n", snd_strerror(err));
return 0;
}
//period_size大概是采样点数/帧——4410点/帧
//s16位代表两个字节,再加上双声道
//size公式=period_size*channels*16/8
size = (period_size * channels * snd_pcm_format_physical_width(format)) / 8; /* 2 bytes/sample, 1 channels */
printf("size:%d\n",size);
char *buffer;
buffer = (char *) malloc(size);
memset(buffer,0,size);
char *in_name="鄧紫棋 - 睡公主.wav";
int ret;
AVFormatContext* infmt_ctx = NULL;
//创建输入封装器
ret=avformat_open_input(&infmt_ctx, in_name, NULL, NULL);
if (ret != 0)
{
printf("failed alloc output context\n");
return -1;
}
//读取一部分视音频流并且获得一些相关的信息
avformat_find_stream_info(infmt_ctx, NULL);
//视频流和音频流的标志
int audioindex=-1;
//查找视频||音频流
for(int i=0; i<infmt_ctx->nb_streams; i++)
{
if(infmt_ctx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO)
{
audioindex=i;
break;
}
}
if (audioindex == -1)
{
printf("input video stream not exist\n");
return -1;
}
AVCodec* decodec = NULL;
AVCodecContext* decodec_ctx = NULL;
decodec_ctx=infmt_ctx->streams[audioindex]->codec;
//找到解码器
decodec = avcodec_find_decoder(decodec_ctx->codec_id);
if (!decodec)
{
printf("not find decoder\n");
avformat_close_input(&infmt_ctx);
return -1;
}
//打开解码器
ret = avcodec_open2(decodec_ctx, decodec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open codec: %s\n", av_err2str(ret));
return -1;
}
//查看输入封装内容
av_dump_format(infmt_ctx, 0, in_name,0);
AVPacket *dec_pkt;
dec_pkt = (AVPacket *)av_malloc(sizeof(AVPacket));
AVFrame *pframePCM;
pframePCM = av_frame_alloc();
pframePCM->format = AV_SAMPLE_FMT_S16;
pframePCM->channel_layout = AV_CH_LAYOUT_STEREO;
pframePCM->sample_rate = rate;
pframePCM->nb_samples = period_size;
pframePCM->channels = channels;
av_frame_get_buffer(pframePCM, 0);
AVFrame *pframeSRC;
pframeSRC = av_frame_alloc();
struct SwrContext *pcm_convert_ctx = swr_alloc();
if (!pcm_convert_ctx)
{
fprintf(stderr, "Could not allocate resampler context\n");
return -1;
}
swr_alloc_set_opts(pcm_convert_ctx,
AV_CH_LAYOUT_STEREO,
AV_SAMPLE_FMT_S16,
pframePCM->sample_rate,
av_get_default_channel_layout(decodec_ctx->channels),
decodec_ctx->sample_fmt,
decodec_ctx->sample_rate,
0,
NULL);
ret = swr_init(pcm_convert_ctx);
if (ret<0)
{
fprintf(stderr, "Failed to initialize the resampling context\n");
return -1;
}
int got_picture;
int nb_data;
while (1)
{
ret=av_read_frame(infmt_ctx,dec_pkt);
if (ret != 0)
{
printf("fail to read_frame\n");
break;
}
//解码获取初始音频
ret = avcodec_decode_audio4(decodec_ctx, pframeSRC, &got_picture, dec_pkt);
if(!got_picture)
{
printf("456\n");
continue;
}
//MP3->PCM,
ret=swr_convert(pcm_convert_ctx,pframePCM->data, pframePCM->nb_samples,(const uint8_t **)pframeSRC->data, pframeSRC->nb_samples);
if (ret <= 0)
{
printf("123\n");
continue;
}
nb_data=ret;
//向硬件写入音频数据
rc = snd_pcm_writei(handle, pframePCM->data[0], nb_data);
if (rc == -EPIPE) {
fprintf(stderr, "underrun occurred\n");
err=snd_pcm_prepare(handle);
if(err<0)
{
fprintf(stderr, "can not recover from underrun: %s\n",snd_strerror(err));
}
}
else if (rc < 0) {
fprintf(stderr,"error from writei: %s\n",snd_strerror(rc));
}
else if (rc != (int)nb_data) {
fprintf(stderr,"short write, write %d frames\n", rc);
}
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);
return 0;
}
3.编译
gcc alsa_ffmpeg_play.c -o alsa_ffmpeg_play -lavformat -lavcodec -lswscale -lswresample -lavutil -lasound
4.总结
大家可以尝试其他音频格式。