alsa学习--6.ffmpeg解码mp3后用alsa播放

一. 
1. 己经学会了用alsa播放pcm并且ffmpeg可以解码mp3为pcm,那么把这两者合并起来
就是ffmpeg解码mp3后用alsa播放
2. 代码

 #include "utils.h"
 #include <libavutil/avutil.h>
 #include <libavutil/attributes.h>
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavutil/imgutils.h>
 #include <libavutil/samplefmt.h>
 #include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 #include <libavcodec/avcodec.h>
 #include <libswscale/swscale.h>
 #include <libavutil/mathematics.h>
 #include <libswresample/swresample.h>
 #include <libavutil/channel_layout.h>
 #include <libavutil/common.h>
 #include <libavformat/avio.h>
 #include <libavutil/file.h>
 #include <libswresample/swresample.h>
 #include <alsa/asoundlib.h>
 #define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000

 typedef struct {
     int videoindex;
     int sndindex;
     AVFormatContext* pFormatCtx;
     AVCodecContext* sndCodecCtx;
     AVCodec* sndCodec;
     SwrContext *swr_ctx;
     snd_pcm_t *pcm;
     DECLARE_ALIGNED(16,uint8_t,audio_buf) [AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
 }AudioState;

 void ffmpeg_fmt_to_alsa_fmt(AVCodecContext *pCodecCtx, snd_pcm_t *pcm, snd_pcm_hw_params_t *params)
 {
     switch(pCodecCtx->sample_fmt) {
         case AV_SAMPLE_FMT_U8: //unsigned 8 bits
             snd_pcm_hw_params_set_format(pcm, params, SND_PCM_FORMAT_S8);
             break;
         case AV_SAMPLE_FMT_S16: //signed 16 bits
             dbmsg("AV_SAMPLE_FMT_S16");
             //SND_PCM_FORMAT_S16 is ok, not care SND_PCM_FORMAT_S16_LE or SND_PCM_FORMAT_S16_BE
             snd_pcm_hw_params_set_format(pcm, params, SND_PCM_FORMAT_S16); //SND_PCM_FORMAT_S16_LE
             break;
         case AV_SAMPLE_FMT_S16P: //signed 16 bits, planar
             dbmsg("AV_SAMPLE_FMT_S16P");
             //SND_PCM_FORMAT_S16 is ok, not care SND_PCM_FORMAT_S16_LE or SND_PCM_FORMAT_S16_BE
             snd_pcm_hw_params_set_format(pcm, params, SND_PCM_FORMAT_S16);
             break;
         default:
             break;
     }
 }

 int init_ffmpeg_alsa(AudioState* is, char* filepath)
 {
     int i=0;
     int ret;
     //alsa
     snd_pcm_hw_params_t *params;
     unsigned int val;
     int dir;
     snd_pcm_uframes_t buffer_size;

     is->sndindex = -1;
     if(NULL == filepath)
     {
         dbmsg("input file is NULL");
         return -1;
     }
     avcodec_register_all();
     avfilter_register_all();
     av_register_all();

     is->pFormatCtx = avformat_alloc_context();

     if(avformat_open_input(&is->pFormatCtx, filepath, NULL, NULL)!=0)
         return -1;

     if(avformat_find_stream_info(is->pFormatCtx, NULL)<0)
         return -1;
     av_dump_format(is->pFormatCtx,0, 0, 0);
     is->videoindex = av_find_best_stream(is->pFormatCtx, AVMEDIA_TYPE_VIDEO, is->videoindex, -1, NULL, 0);
     is->sndindex = av_find_best_stream(is->pFormatCtx, AVMEDIA_TYPE_AUDIO,is->sndindex, is->videoindex, NULL, 0);
     dbmsg("videoindex=%d, sndindex=%d", is->videoindex, is->sndindex);
     if(is->sndindex != -1)
     {
         is->sndCodecCtx = is->pFormatCtx->streams[is->sndindex]->codec;
         is->sndCodec = avcodec_find_decoder(is->sndCodecCtx->codec_id);
         if(is->sndCodec == NULL)
         {
             dbmsg("Codec not found");
             return -1;
         }
         if(avcodec_open2(is->sndCodecCtx, is->sndCodec, NULL) < 0)
             return -1;
         snd_pcm_open(&is->pcm, "default", SND_PCM_STREAM_PLAYBACK, 0);
         snd_pcm_hw_params_alloca(&params);
         snd_pcm_hw_params_any(is->pcm, params);
         snd_pcm_hw_params_set_access(is->pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED);
         ffmpeg_fmt_to_alsa_fmt(is->sndCodecCtx, is->pcm, params);
         snd_pcm_hw_params_set_channels(is->pcm, params, is->sndCodecCtx->channels);
         val = is->sndCodecCtx->sample_rate;
         dbmsg("is->sndCodecCtx->sample_rate=%d", is->sndCodecCtx->sample_rate);
         snd_pcm_hw_params_set_rate_near(is->pcm, params, &val, &dir);
         snd_pcm_hw_params(is->pcm, params);
     }
     return 0;
 }

 int main(int argc, char **argv)
 {
     int ret;
     FILE* fp;
     int file_data_size = 0;
     int len2,len3, data_size, got_frame;
     AVPacket *packet = av_mallocz(sizeof(AVPacket));
     AVFrame *frame = av_frame_alloc();
     AudioState* is = (AudioState*) av_mallocz(sizeof(AudioState));
     uint8_t *out[] = { is->audio_buf };
     if( (ret=init_ffmpeg_alsa(is, argv[1])) != 0)
     {
         dbmsg("init_ffmpeg error");
         return -1;
     }
     while( (av_read_frame(is->pFormatCtx, packet)>=0) )
     {
         if(packet->stream_index != is->sndindex)
             continue;
         if((ret=avcodec_decode_audio4(is->sndCodecCtx, frame, &got_frame, packet)) < 0) //decode data is store in frame
         {
             dbmsg("file eof");
             break;
         }

         if(got_frame <= 0) /* No data yet, get more frames */
             continue;
         if(NULL==is->swr_ctx)
         {
             if(is->swr_ctx != NULL)
                 swr_free(&is->swr_ctx);
             dbmsg("AV_CH_LAYOUT_STEREO=%d, AV_SAMPLE_FMT_S16=%d, freq=44100", AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16);
             dbmsg("frame: channnels=%d, default_layout=%d, format=%d, sample_rate=%d", frame->channels,av_get_default_channel_layout(frame->channels), frame->format, frame->sample_rate);
             is->swr_ctx = swr_alloc_set_opts(NULL, AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, is->sndCodecCtx->sample_rate, av_get_default_channel_layout(frame->channels), frame->format, frame->sample_rate, 0, NULL);
             if(is->swr_ctx == NULL)
             {
                 dbmsg("swr_ctx == NULL");
             }
             swr_init(is->swr_ctx);
         }
         //dbmsg("next swr_convert");
         len2 = swr_convert(is->swr_ctx, out, is->sndCodecCtx->sample_rate,(const uint8_t **)frame->extended_data, frame->nb_samples);
         data_size = av_samples_get_buffer_size(NULL, is->sndCodecCtx->channels, frame->nb_samples, is->sndCodecCtx->sample_fmt, 1);
         //dbmsg("data_size 111 = %d", data_size);
         //data_size = len2 * 2 * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);         -->通过av_sample_get_buffer_size的值与这儿计算的值是一样的
         //dbmsg("data_size 222 = %d", data_size);
         //dbmsg("next pcm write, data_size=%d", data_size);
         ret = snd_pcm_writei(is->pcm, is->audio_buf, data_size/4);                   -->最后面的4应该通过计算获得: 采样位数/8*声道数
         if (ret < 0)
             ret = snd_pcm_recover(is->pcm, ret, 0); //ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
         //dbmsg("pcm write over, data_size=%d", data_size/4);
     }
     av_free_packet(packet);
     av_free(frame);
     avcodec_close(is->sndCodecCtx);
     avformat_close_input(&is->pFormatCtx);
     snd_pcm_close(is->pcm);
     fclose(fp);
     return 0;
 }

关于snd_pcm_writei --> 最后的i代表是SND_PCM_ACCESS_RW_INTERLEAVED类型的接口
即:SND_PCM_ACCESS_RW_INTERLEAVED类型的读写接口是 snd_pcm_readi/snd_pcm_writei
snd_pcm_sframes_t snd_pcm_writei(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
它的最后一个参数是snd_pcm_uframes_t,不像系统read write都是字节数,两者是不一样的。
这个size=buffer_size/(采样位数/8*声道数)
3.运行结果

 cong@msi:/work/ffmpeg/test/alsa/testalsa/6ffmpeg_alsa$ make run
 export LD_LIBRARY_PATH=/work/ffmpeg/out/lib/ \
     && ./ffmpeg_alsa /work/ffmpeg/test/resource//yeye.mp3
 libavutil/crc.c:av_crc_init[313]:
 [mp3 @ 0xe9d3e0] Skipping 0 bytes of junk at 2169.
 Input #0, mp3, from '(null)':
   Metadata:
     album : 八度空间
     artist : 周杰伦
     title : 爷爷泡的茶
   Duration: 00:03:58.55, start: 0.025057, bitrate: 320 kb/s
     Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
     Metadata:
       encoder : LAME3.98r
     Side data:
       replaygain: track gain - -10.100000, track peak - unknown, album gain - unknown, album peak - unknown,
 ffmpeg_alsa.c:init_ffmpeg_alsa[84]: videoindex=-1381258232, sndindex=0
 ffmpeg_alsa.c:ffmpeg_fmt_to_alsa_fmt[45]: AV_SAMPLE_FMT_S16P
 ffmpeg_alsa.c:init_ffmpeg_alsa[103]: is->sndCodecCtx->sample_rate=44100
 ffmpeg_alsa.c:main[142]: AV_CH_LAYOUT_STEREO=3, AV_SAMPLE_FMT_S16=1, freq=44100
 ffmpeg_alsa.c:main[143]: frame: channnels=2, default_layout=3, format=6, sample_rate=44100
 ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred              --> 这就是为什么要加上snd_pcm_recover

4.代码打包 
6ffmpeg_alsa.rar (下载后改名为6ffmpeg_alsa.tar.gz)