ffmpeg 分离音频 保存

<h1 class="postTitle" style="margin: 0px 0px 10px; padding: 0px 15px; font-size: 12px; font-weight: normal; background-image: url(http://www.cnblogs.com/skins/iMetro_HD/images/logo.png); font-family: Verdana, Arial, Helvetica, sans-serif; line-height: 18px; background-position: -10px 0px; background-repeat: no-repeat repeat;"><a target=_blank id="cb_post_title_url" class="postTitle2" href="http://www.cnblogs.com/xuanyuanchen/p/3161203.html" style="margin: 0px; padding: 0px; font-size: 24px; color: rgb(0, 0, 0); text-decoration: none;">Windwos平台上ffmpeg解码音频并且保存到wav文件中</a></h1>

#include <stdio.h>
#include <math.h>
#include "libavutil/avstring.h"
//修改colorspace.h中的inline为__inline
#include "libavutil/colorspace.h"
#include "libavutil/pixdesc.h"
#include "libavutil/imgutils.h"
#include "libavutil/dict.h"
#include "libavutil/parseutils.h"
#include "libavutil/samplefmt.h"
#include "libavutil/avassert.h"
#include "libavformat/avformat.h"
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
#include "libavcodec/audioconvert.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "cmdutils.h"
#include "pthread.h"

static AVPacket flush_pkt;//暂时不知道flush_pkt有什么作用，暂时先放这里。


//#define DEBUG_SYNC

#define MAX_QUEUE_SIZE (15 * 1024 * 1024)
#define MIN_AUDIOQ_SIZE (20 * 16 * 1024)
#define MIN_FRAMES 5

/* SDL audio buffer size, in samples. Should be small to have precise
   A/V sync as SDL does not have hardware buffer fullness info. */
#define SDL_AUDIO_BUFFER_SIZE 1024

/* no AV sync correction is done if below the AV sync threshold */
#define AV_SYNC_THRESHOLD 0.01
/* no AV correction is done if too big error */
#define AV_NOSYNC_THRESHOLD 10.0

#define FRAME_SKIP_FACTOR 0.05

/* maximum audio speed change to get correct sync */
#define SAMPLE_CORRECTION_PERCENT_MAX 10

/* we use about AUDIO_DIFF_AVG_NB A-V differences to make the average */
#define AUDIO_DIFF_AVG_NB   20

/* NOTE: the size must be big enough to compensate the hardware audio buffersize size */
#define SAMPLE_ARRAY_SIZE (2*65536)

typedef struct PacketQueue {
    AVPacketList *first_pkt, *last_pkt;
    int nb_packets;
    int size;
    int abort_request;
    pthread_mutex_t *mutex;//互斥锁
    pthread_cond_t *cond;//条件变量
} PacketQueue;

#define VIDEO_PICTURE_QUEUE_SIZE 2
#define SUBPICTURE_QUEUE_SIZE 4

typedef struct VideoPicture {
    double pts;                                  ///<presentation time stamp for this picture
    double target_clock;                         ///<av_gettime() time at which this should be displayed ideally
    int64_t pos;                                 ///<byte position in file
//    SDL_Overlay *bmp;
    int width, height; /* source height & width */
    int allocated;
    enum PixelFormat pix_fmt;

#if CONFIG_AVFILTER
    AVFilterBufferRef *picref;
#endif
} VideoPicture;

typedef struct SubPicture {
    double pts; /* presentation time stamp for this picture */
    AVSubtitle sub;
} SubPicture;

enum {
    AV_SYNC_AUDIO_MASTER, /* default choice */
    AV_SYNC_VIDEO_MASTER,
    AV_SYNC_EXTERNAL_CLOCK, /* synchronize to an external clock */
};

typedef struct VideoState {
    pthread_t *parse_tid;
    //SDL_Thread *parse_tid;
    pthread_t *video_tid;
    //SDL_Thread *video_tid;
    pthread_t *refresh_tid;
    //SDL_Thread *refresh_tid;
    AVInputFormat *iformat;
    int no_background;
    int abort_request;
    int paused;
    int last_paused;
    int seek_req;
    int seek_flags;
    int64_t seek_pos;
    int64_t seek_rel;
    int read_pause_return;
    AVFormatContext *ic;
    int dtg_active_format;

    int audio_stream;

    int av_sync_type;
    double external_clock; /* external clock base */
    int64_t external_clock_time;

    double audio_clock;
    double audio_diff_cum; /* used for AV difference average computation */
    double audio_diff_avg_coef;
    double audio_diff_threshold;
    int audio_diff_avg_count;
    AVStream *audio_st;
    PacketQueue audioq;
    int audio_hw_buf_size;
    /* samples output by the codec. we reserve more space for avsync
       compensation */
    DECLARE_ALIGNED(16,uint8_t,audio_buf1)[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
    DECLARE_ALIGNED(16,uint8_t,audio_buf2)[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
    uint8_t *audio_buf;
    unsigned int audio_buf_size; /* in bytes */
    int audio_buf_index; /* in bytes */
    AVPacket audio_pkt_temp;
    AVPacket audio_pkt;
    enum AVSampleFormat audio_src_fmt;
    AVAudioConvert *reformat_ctx;

    enum ShowMode {
        SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
    } show_mode;
    int16_t sample_array[SAMPLE_ARRAY_SIZE];
    int sample_array_index;
    int last_i_start;
    RDFTContext *rdft;
    int rdft_bits;
    FFTSample *rdft_data;
    int xpos;

    pthread_t *subtitle_tid;
    //SDL_Thread *subtitle_tid;
    int subtitle_stream;
    int subtitle_stream_changed;
    AVStream *subtitle_st;
    PacketQueue subtitleq;
    SubPicture subpq[SUBPICTURE_QUEUE_SIZE];
    int subpq_size, subpq_rindex, subpq_windex;

    pthread_mutex_t *subpq_mutex;
    pthread_cond_t *subpq_cond;
    //SDL_mutex *subpq_mutex;
    //SDL_cond *subpq_cond;

    double frame_timer;
    double frame_last_pts;
    double frame_last_delay;
    double video_clock;                          ///<pts of last decoded frame / predicted pts of next decoded frame
    int video_stream;
    AVStream *video_st;
    PacketQueue videoq;
    double video_current_pts;                    ///<current displayed pts (different from video_clock if frame fifos are used)
    double video_current_pts_drift;              ///<video_current_pts - time (av_gettime) at which we updated video_current_pts - used to have running video pts
    int64_t video_current_pos;                   ///<current displayed file pos
    VideoPicture pictq[VIDEO_PICTURE_QUEUE_SIZE];
    int pictq_size, pictq_rindex, pictq_windex;
    pthread_mutex_t *pictq_mutex;
    //SDL_mutex *pictq_mutex;
    pthread_cond_t *pictq_cond;
    //SDL_cond *pictq_cond;

    struct SwsContext *img_convert_ctx;

    //    QETimer *video_timer;
    char filename[1024];
    int width, height, xleft, ytop;

    //PtsCorrectionContext pts_ctx;

    float skip_frames;
    float skip_frames_index;
    int refresh;
} VideoState;


static int opt_help(const char *opt, const char *arg);

/* options specified by the user */
static AVInputFormat *file_iformat;
static const char *input_filename;
static const char *window_title;
static int fs_screen_width;
static int fs_screen_height;
static int screen_width = 0;
static int screen_height = 0;
static int frame_width = 0;
static int frame_height = 0;
static enum PixelFormat frame_pix_fmt = PIX_FMT_NONE;
static int audio_disable;
static int video_disable;
/*
static int wanted_stream[AVMEDIA_TYPE_NB]={
    [AVMEDIA_TYPE_AUDIO]=-1,
    [AVMEDIA_TYPE_VIDEO]=-1,
    [AVMEDIA_TYPE_SUBTITLE]=-1,
};
*/
static int wanted_stream[AVMEDIA_TYPE_NB]={-1,-1,0,-1,0};
static int seek_by_bytes=-1;
static int display_disable;
static int show_status = 1;
static int av_sync_type = AV_SYNC_AUDIO_MASTER;
static int64_t start_time = AV_NOPTS_VALUE;
static int64_t duration = AV_NOPTS_VALUE;
static int step = 0;
static int thread_count = 1;
static int workaround_bugs = 1;
static int fast = 0;
static int genpts = 0;
static int lowres = 0;
static int idct = FF_IDCT_AUTO;
static enum AVDiscard skip_frame= AVDISCARD_DEFAULT;
static enum AVDiscard skip_idct= AVDISCARD_DEFAULT;
static enum AVDiscard skip_loop_filter= AVDISCARD_DEFAULT;
static int error_recognition = FF_ER_CAREFUL;
static int error_concealment = 3;
static int decoder_reorder_pts= -1;
static int autoexit;
static int exit_on_keydown;
static int exit_on_mousedown;
static int loop=1;
static int framedrop=-1;
static enum ShowMode show_mode = SHOW_MODE_NONE;

static int rdftspeed=20;
#if CONFIG_AVFILTER
static char *vfilters = NULL;
#endif

/* current context */
static int is_full_screen;
static VideoState *cur_stream;
static int64_t audio_callback_time;
static AVPacket flush_pkt;//暂时不知道flush_pkt有什么作用，暂时先放这里。

static int packet_queue_put(PacketQueue *q, AVPacket *pkt);

/* packet queue handling */
//初始化队列
static void packet_queue_init(PacketQueue *q)
{
    memset(q, 0, sizeof(PacketQueue));
    pthread_mutex_init(q->mutex,NULL);
    pthread_cond_init(q->cond,NULL);
    //q->mutex = SDL_CreateMutex();
    //q->cond = SDL_CreateCond();
    packet_queue_put(q, &flush_pkt);
}

//清空队列
static void packet_queue_flush(PacketQueue *q)
{
    AVPacketList *pkt, *pkt1;

    pthread_mutex_lock(q->mutex);
    //SDL_LockMutex(q->mutex);
    for(pkt = q->first_pkt; pkt != NULL; pkt = pkt1) {
        pkt1 = pkt->next;
        av_free_packet(&pkt->pkt);
        av_freep(&pkt);
    }
    q->last_pkt = NULL;
    q->first_pkt = NULL;
    q->nb_packets = 0;
    q->size = 0;
    pthread_mutex_unlock(q->mutex);
    //SDL_UnlockMutex(q->mutex);
}

static void packet_queue_end(PacketQueue *q)
{
    packet_queue_flush(q);
    pthread_mutex_destroy(q->mutex);
    pthread_cond_destroy(q->cond);
    //SDL_DestroyMutex(q->mutex);
    //SDL_DestroyCond(q->cond);
}

static int packet_queue_put(PacketQueue *q, AVPacket *pkt)
{
    AVPacketList *pkt1;

    /* duplicate the packet */
    if (pkt!=&flush_pkt && av_dup_packet(pkt) < 0)
        return -1;

    pkt1 = av_malloc(sizeof(AVPacketList));
    if (!pkt1)
        return -1;
    pkt1->pkt = *pkt;
    pkt1->next = NULL;

    pthread_mutex_lock(q->mutex);
//    SDL_LockMutex(q->mutex);

    if (!q->last_pkt)

        q->first_pkt = pkt1;
    else
        q->last_pkt->next = pkt1;
    q->last_pkt = pkt1;
    q->nb_packets++;
    q->size += pkt1->pkt.size + sizeof(*pkt1);
    /* XXX: should duplicate packet data in DV case */
    pthread_cond_signal(q->cond);
//    SDL_CondSignal(q->cond);

//    SDL_UnlockMutex(q->mutex);
    pthread_mutex_unlock(q->mutex);
    return 0;
}


static void packet_queue_abort(PacketQueue *q)
{
    pthread_mutex_lock(q->mutex);
    //SDL_LockMutex(q->mutex);

    q->abort_request = 1;

    pthread_cond_signal(q->cond);
    //SDL_CondSignal(q->cond);

    pthread_mutex_unlock(q->mutex);
    //SDL_UnlockMutex(q->mutex);
}


//packet_queue_get 函数被调用的地方是audio_decode_frame,subtitle_thread,get_video_frame中，
//作用是从队列q中读取block(一般为)个packet，留待下一次进行解码
//avcodec_decode_audio3,avcodec_decode_video2
/* return < 0 if aborted, 0 if no packet and > 0 if packet.  */
static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block)
{
    AVPacketList *pkt1;
    int ret;

    pthread_mutex_lock(q->mutex);
    //SDL_LockMutex(q->mutex);

    for(;;) {
        if (q->abort_request) {
            ret = -1;
            break;
        }

        pkt1 = q->first_pkt;
        if (pkt1) {
            q->first_pkt = pkt1->next;
            if (!q->first_pkt)
                q->last_pkt = NULL;
            q->nb_packets--;
            q->size -= pkt1->pkt.size + sizeof(*pkt1);
            *pkt = pkt1->pkt;
            av_free(pkt1);
            ret = 1;
            break;
        } else if (!block) {
            ret = 0;
            break;
        } else {
            pthread_cond_wait(q->cond,q->mutex);
            //SDL_CondWait(q->cond, q->mutex);
        }
    }
    pthread_mutex_unlock(q->mutex);
    //SDL_UnlockMutex(q->mutex);
    return ret;
}



//声明了一个内联函数，写wav头
static __inline  void writeWavHeader(AVCodecContext *pCodecCtx,AVFormatContext *pFormatCtx,FILE *audioFile) {
    //wav文件有44字节的wav头，所以要写44字节的wav头
    int8_t *data;
    int32_t long_temp;
    int16_t short_temp;
    int16_t BlockAlign;
    int bits=16;
    int32_t fileSize;
    int32_t audioDataSize;

    switch(pCodecCtx->sample_fmt) {
        case AV_SAMPLE_FMT_S16:
            bits=16;
            break;
        case AV_SAMPLE_FMT_S32:
            bits=32;
            break;
        case AV_SAMPLE_FMT_U8:
            bits=8;
            break;
        default:
            bits=16;
            break;
    }
    audioDataSize=(pFormatCtx->duration)*(bits/8)*(pCodecCtx->sample_rate)*(pCodecCtx->channels);
    fileSize=audioDataSize+36;
    data="RIFF";
    fwrite(data,sizeof(char),4,audioFile);
    fwrite(&fileSize,sizeof(int32_t),1,audioFile);

    //"WAVE"
    data="WAVE";
    fwrite(data,sizeof(char),4,audioFile);
    data="fmt ";
    fwrite(data,sizeof(char),4,audioFile);
    long_temp=16;
    fwrite(&long_temp,sizeof(int32_t),1,audioFile);
    short_temp=0x01;
    fwrite(&short_temp,sizeof(int16_t),1,audioFile);
    short_temp=(pCodecCtx->channels);
    fwrite(&short_temp,sizeof(int16_t),1,audioFile);
    long_temp=(pCodecCtx->sample_rate);
    fwrite(&long_temp,sizeof(int32_t),1,audioFile);
    long_temp=(bits/8)*(pCodecCtx->channels)*(pCodecCtx->sample_rate);
    fwrite(&long_temp,sizeof(int32_t),1,audioFile);
    BlockAlign=(bits/8)*(pCodecCtx->channels);
    fwrite(&BlockAlign,sizeof(int16_t),1,audioFile);
    short_temp=(bits);
    fwrite(&short_temp,sizeof(int16_t),1,audioFile);
    data="data";
    fwrite(data,sizeof(char),4,audioFile);
    fwrite(&audioDataSize,sizeof(int32_t),1,audioFile);

    fseek(audioFile,44,SEEK_SET);

}

int main()
{
//    char *filename="rtsp://192.168.20.112/Love_You.mp4";
    //char *filename="E:\\flv\\3d.mp3";
    char *filename="E:\\flv\\MY.aac";
//    char *filename="mms://mms.cnr.cn/cnr003";
//    char *filename="mms://mms.cnr.cn/cnr001";
//    char *filename="rtsp://livewm.orange.fr/live-multicanaux";
//    char *filename="mms://211.167.102.66/ch-01";
    AVFormatContext *pFormatCtx;
    int audioStream=-1;
    int i;
    int iFrame=0;
    AVCodecContext *pCodecCtx;
    AVCodec *pCodec=NULL;
    static AVPacket packet;
    uint8_t *pktData=NULL;
    int pktSize;
    int outSize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
//    FILE *wavfile=NULL;

    //这里必须使用av_malloc
    uint8_t *inbuf=(uint8_t *)av_malloc(outSize);

    FILE *wavFile=NULL;
    int32_t audioFileSize=0;

    //注册所有的编解码器
    av_register_all();

    //打开文件
    if(av_open_input_file(&pFormatCtx,filename,NULL,0,NULL)!=0)
    {
        printf("Could not open input file %s\n",filename);
        return 0;
    }
    if(av_find_stream_info(pFormatCtx)<0)
    {
        printf("Could not find stream information\n");
    }

    //输出文件的音视频流信息
    av_dump_format(pFormatCtx,0,filename,0);

    //找到音频流
    for(i=0;i<pFormatCtx->nb_streams;i++) {
        if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO) {
            audioStream=i;
            break;
        }
    }

    //找到解码器
    pCodecCtx=pFormatCtx->streams[audioStream]->codec;
    pCodec=avcodec_find_decoder(pCodecCtx->codec_id);


    //打开解码器
    if(avcodec_open(pCodecCtx,pCodec)<0) {
        printf("Error avcodec_open failed.\n");
        return 1;
    }

    printf("\tbit_rate=%d\n \
    bytes_per_secondes=%d\n \
    sample_rate=%d\n \
    channels=%d\n \
    codec_name=%s\n",pCodecCtx->bit_rate,(pCodecCtx->codec_id==CODEC_ID_PCM_U8)?8:16,
    pCodecCtx->sample_rate,pCodecCtx->channels,pCodecCtx->codec->name);

    //wavFile=fopen("E:\\flv\\saveWav.wav","wb");
    wavFile=fopen("E:\\flv\\MY.wav","wb");
    //wavFile=fopen("E:\\flv\\test.wav","wb");
    if (wavFile==NULL)
    {
        printf("open error\n");
        return 1;
    }

    //写入wav文件头
    writeWavHeader(pCodecCtx,pFormatCtx,wavFile);

    //开始解码音频流
    av_free_packet(&packet);
    while(av_read_frame(pFormatCtx,&packet)>=0) {
        if(packet.stream_index==audioStream) {
            int len=0;
            if((iFrame++)>=4000)
                break;
            pktData=packet.data;
            pktSize=packet.size;
            while(pktSize>0) {
                outSize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
                len=avcodec_decode_audio3(pCodecCtx,(short *)inbuf,&outSize,&packet);
                if(len<0){
                    printf("Error while decoding\n");
                    break;
                }
                if(outSize>0) {
                    audioFileSize+=outSize;
                    fwrite(inbuf,1,outSize,wavFile);
                    fflush(wavFile);
                }
                pktSize-=len;
                pktData+=len;
            }
        }
        av_free_packet(&packet);
    }

    //wav文件的第40个字节开始的4个字节存放的是wav文件的有效数据长度
    fseek(wavFile,40,SEEK_SET);
    fwrite(&audioFileSize,1,sizeof(int32_t),wavFile);
    //wav文件的第4个字节开始的4个字节存放的是wav文件的文件长度(audioFileSize+44-8),44表示44个字节的头，8表示"RIFF"和"WAVE"
    audioFileSize+=36;
    fseek(wavFile,4,SEEK_SET);
    fwrite(&audioFileSize,1,sizeof(int32_t),wavFile);

    //关闭文件
    fclose(wavFile);

    //释放内存
    av_free(inbuf);
    if(pCodecCtx!=NULL){
        avcodec_close(pCodecCtx);
    }
    av_close_input_file(pFormatCtx);
    return 0;
}