FFmpeg 音乐播放器类，自己写的~可以反复使用

根据网上到处收集的代码并自己修改写的一个FFmpegPlayer音频类如下：

.h头文件

#ifndef FFMPEGPLAYER_H
#define FFMPEGPLAYER_H

#define MAX_AUDIO_FRAME_SIZE  192000
#define SDL_AUDIO_BUFFER_SIZE  1024
#define MAX_AUDIO_SIZE (25 * 16 * 1024)
#define MAX_VIDEO_SIZE (25 * 256 * 1024)
#define FLUSH_DATA "FLUSH"

extern "C"
{
    # include <libavcodec\avcodec.h>
    # include <libavformat\avformat.h>
    # include <libswscale\swscale.h>
    # include <libswresample\swresample.h>
    # include <include/SDL2/SDL.h>
    # include <include/SDL2/SDL_thread.h>
}

#include<QThread>

typedef struct PacketQueue {
    AVPacketList *first_pkt, *last_pkt;
    int nb_packets;
    int size;
    SDL_mutex *mutex;
    SDL_cond *cond;
} PacketQueue;

typedef struct{
    SwrContext* swr_ctx ;//
    AVFrame *wanted_frame;//
    uint8_t* audio_pkt_data;
    int audio_pkt_size; //
    AVFrame *frame; //
    AVFormatContext* afct; //
    AVCodecContext *acct;//

    unsigned int audio_buf_size; //
    unsigned int audio_buf_index; //

    PacketQueue audioq; //
    AVPacket pkt; //
}mediaState;

enum playerStatus{
     playing,
     pausing,
     buffering,
     stopping
};



class FFmpegPlayer : public QThread
{
    Q_OBJECT
public:
    explicit FFmpegPlayer(QObject *parent = 0);
    void setMedia(const QString);
    void stop();
    void pause();
    void play();
    playerStatus getPlayerStatus();

    void FreeAllocSpace();
protected:
    virtual void run();
signals:

public slots:
private:
    QString m_url;
    mediaState m_MS;
};

#endif // FFMPEGPLAYER_H

.cpp

#include "FFmpegPlayer.h"
#include<QDebug>
#include<iostream>
#include<windows.h>
using namespace std;

bool isquit=false; //

// 包队列初始化
void packet_queue_init(PacketQueue* q)
{
    memset(q, 0, sizeof(PacketQueue));
    q->last_pkt = NULL;
    q->first_pkt = NULL;
    q->mutex = SDL_CreateMutex();
    q->cond = SDL_CreateCond();
}

// 放入packet到队列中，不带头指针的队列
int packet_queue_put(PacketQueue*q, AVPacket *pkt)
{
    AVPacketList *pktl;
    if (av_dup_packet(pkt) < 0)
        return -1;

    pktl = (AVPacketList*)av_malloc(sizeof(AVPacketList));
    if (!pktl)
        return -1;

    pktl->pkt = *pkt;
    pktl->next = nullptr;

    SDL_LockMutex(q->mutex);

    if (!q->last_pkt) // 队列为空，新插入元素为第一个元素
        q->first_pkt = pktl;
    else // 插入队尾
        q->last_pkt->next = pktl;

    q->last_pkt = pktl;

    q->nb_packets++;
    q->size += pkt->size;

    SDL_CondSignal(q->cond);
    SDL_UnlockMutex(q->mutex);
    return 0;
}

// 从队列中取出packet
static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block) {
    AVPacketList *pkt1;
    int ret;

    SDL_LockMutex(q->mutex);

    for (;;)
    {
        if (isquit)
         return -1;
        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;
            *pkt = pkt1->pkt;
            av_free(pkt1);
            ret = 1;
            break;
        } else if (!block) {
            ret = 0;
            break;
        } else {
            SDL_CondWait(q->cond, q->mutex);
        }
    }
    SDL_UnlockMutex(q->mutex);
    return ret;
}

static void packet_queue_flush(PacketQueue *q)
{
    SDL_LockMutex(q->mutex);
    AVPacketList *pkt, *pkt1;
    for(pkt = q->first_pkt; pkt != NULL; pkt = pkt1)
    {
        pkt1 = pkt->next;
        if(pkt1->pkt.data != (uint8_t *)"FLUSH")
        {

        }
        av_free_packet(&pkt->pkt);
        av_freep(&pkt);

    }
    q->last_pkt = NULL;
    q->first_pkt = NULL;
    q->nb_packets = 0;
    q->size = 0;
    SDL_UnlockMutex(q->mutex);
}
//
// 解码音频数据
int audio_decode_frame(mediaState* MS, uint8_t* audio_buf, int buf_size)
{
    int len1;
    int data_size = 0;
    if (isquit)
        return -1;

    while (true)
    {
        while (MS->audio_pkt_size > 0)
        {
            if (isquit)
                return -1;

            int got_frame = 0;
            len1 = avcodec_decode_audio4(MS->acct, MS->frame, &got_frame, &MS->pkt);
            if (len1 < 0) // 出错，跳过
            {
                MS->audio_pkt_size = 0;
                break;
            }

            MS->audio_pkt_data += len1;
            MS->audio_pkt_size -= len1;
            data_size = 0;
            if (got_frame)
            {
                data_size = av_samples_get_buffer_size(nullptr, MS->acct->channels,MS-> frame->nb_samples, MS->acct->sample_fmt, 1);
            }

            if (MS->frame->channels > 0 && MS->frame->channel_layout == 0)
                MS->frame->channel_layout = av_get_default_channel_layout(MS->frame->channels);
            else if (MS->frame->channels == 0 && MS->frame->channel_layout > 0)
                MS->frame->channels = av_get_channel_layout_nb_channels(MS->frame->channel_layout);

            if (MS->swr_ctx)
            {
                swr_free(&MS->swr_ctx);
                MS->swr_ctx = nullptr;
            }
            MS->swr_ctx = swr_alloc_set_opts(nullptr, MS->wanted_frame->channel_layout,
                                         (AVSampleFormat)MS->wanted_frame->format,
                                         MS->wanted_frame->sample_rate,
                                         MS->frame->channel_layout,
                                         (AVSampleFormat)MS->frame->format,
                                         MS->frame->sample_rate, 0, nullptr);

            if (!MS->swr_ctx || swr_init(MS->swr_ctx) < 0)
            {
                cout << "swr_init failed:" << endl;
                break;
            }
            int dst_nb_samples = av_rescale_rnd(swr_get_delay(MS->swr_ctx, MS->frame->sample_rate) + MS->frame->nb_samples, MS->frame->sample_rate, MS->frame->sample_rate, AVRounding(1));
            int len2 = swr_convert(MS->swr_ctx, &audio_buf, dst_nb_samples,(const uint8_t**)MS->frame->data, MS->frame->nb_samples);
            if (len2 < 0)
            {
                cout << "swr_convert failed\n";
                break;
            }
            return MS->wanted_frame->channels * len2 * av_get_bytes_per_sample((AVSampleFormat)MS->wanted_frame->format);

            if (data_size <= 0)
                continue; // No data yet,get more frames

            return data_size; // we have data,return it and come back for more later
        }
        if (MS->pkt.data)
            av_free_packet(&MS->pkt);

        if (packet_queue_get(&MS->audioq, &MS->pkt, true) < 0)
            return -1;

        MS->audio_pkt_data =MS->pkt.data;
        MS->audio_pkt_size = MS->pkt.size;
    }
}

// 解码后的回调函数
void audio_callback(void* userdata, Uint8* stream, int len)
{
    mediaState* MS = (mediaState*)userdata;
    int len1, audio_size;
    uint8_t audio_buff[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
    SDL_memset(stream, 0, len);
    if (isquit)
        return;
    while (len > 0)
    {
        if (MS->audio_buf_index >= MS->audio_buf_size)
        {
            audio_size = audio_decode_frame(MS, audio_buff, sizeof(audio_buff));
            if (isquit)
                return;
            if (audio_size < 0)
            {
                MS->audio_buf_size = 1024;
                memset(audio_buff, 0, MS->audio_buf_size);
            }
            else
                MS->audio_buf_size = audio_size;

            MS->audio_buf_index = 0;
        }
        len1 = MS->audio_buf_size - MS->audio_buf_index;
        if (len1 > len)
            len1 = len;

        SDL_MixAudio(stream, audio_buff + MS->audio_buf_index, len, SDL_MIX_MAXVOLUME);

        len -= len1;
        stream += len1;
        MS->audio_buf_index += len1;
    }
}

void FFmpegPlayer::FreeAllocSpace() //存在内在
{
    pause();
    SDL_CloseAudio();//Close SDL
    SDL_Quit();

    if(m_MS.wanted_frame)
    av_frame_free(&m_MS.wanted_frame);
    if(m_MS.frame)
    av_frame_free(&m_MS.frame);
    if(m_MS.acct)
    avcodec_close(m_MS.acct);
    if(m_MS.afct)
    avformat_close_input(&m_MS.afct);
    if(m_MS.swr_ctx)
    swr_free(&m_MS.swr_ctx);
    if(m_MS.audio_pkt_data)
    av_freep(m_MS.audio_pkt_data);
    if(m_MS.audioq.first_pkt)
    packet_queue_flush(&m_MS.audioq);
    m_MS={0};//自动对能初始化为0的都初始化了
    //使用memset(m_MS,0,sizeof(m_MS));会出现bug！！
}


FFmpegPlayer::FFmpegPlayer(QObject *parent) : QThread(parent)
{
    av_register_all();
    avformat_network_init();
    m_MS={0};//自动对能初始化为0的都初始化了
    //使用memset(m_MS,0,sizeof(m_MS));会出现bug！！
}

void FFmpegPlayer::setMedia(const QString url)
{
    stop();
    m_url=url;
    start();
    setPriority(QThread::HighestPriority);
}

void FFmpegPlayer::stop()
{
    isquit=1;
    Sleep(100);
    m_url="";
}

void FFmpegPlayer::pause()
{
    SDL_PauseAudio(1);
}

void FFmpegPlayer::play()
{
    SDL_PauseAudio(0);
}

playerStatus FFmpegPlayer::getPlayerStatus()
{
    if(SDL_AUDIO_PAUSED==SDL_GetAudioStatus())
        return pausing;
    if(SDL_AUDIO_PLAYING==SDL_GetAudioStatus())
        return playing;
        return stopping;
}


void FFmpegPlayer::run()
{
    isquit=0;
    SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER);

    // 读取文件头，将格式相关信息存放在AVFormatContext结构体中
    if (avformat_open_input(&m_MS.afct, m_url.toUtf8().data(), nullptr, nullptr) != 0)
    {
        FreeAllocSpace();
        return; // 打开失败
    }
    // 检测文件的流信息
    if (avformat_find_stream_info(m_MS.afct, nullptr) < 0)
    {
        FreeAllocSpace();
        return; // 没有检测到流信息 stream infomation
    }
    //查找第一个视频流 video stream
    int audioStream = -1;
    for (unsigned int i = 0; i < m_MS.afct->nb_streams; i++)
    {
        if (m_MS.afct->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
        {
            audioStream = i;
            break;
        }
    }

    // 3. 根据读取到的流信息查找相应的解码器并打开
    if (audioStream == -1)
    {
        FreeAllocSpace();
        return; // 没有检测到流信息 stream infomation
    }
    m_MS.acct = m_MS.afct->streams[audioStream]->codec; // codec context
    AVCodec* pCodec = avcodec_find_decoder(m_MS.acct->codec_id);

    if (!pCodec)
    {
        cout << "Unsupported codec!" << endl;
        FreeAllocSpace();
        return;
    }
    // Set audio settings from codec info
    SDL_AudioSpec wanted_spec, spec;
    wanted_spec.freq = m_MS.acct->sample_rate;
    wanted_spec.format = AUDIO_S16SYS;
    wanted_spec.channels = m_MS.acct->channels;
    wanted_spec.silence = 0;
    wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
    wanted_spec.callback = audio_callback;
    wanted_spec.userdata = &m_MS;
   // CoInitializeEx(NULL, COINIT_MULTITHREADED);
    if (SDL_OpenAudio(&wanted_spec, &spec) < 0)
    {
        cout << "Open audio failed:" << SDL_GetError() << endl;
        FreeAllocSpace();
        return ;
    }
    m_MS.wanted_frame=av_frame_alloc();
    m_MS.frame=av_frame_alloc();

    m_MS.wanted_frame->format = AV_SAMPLE_FMT_S16;
    m_MS.wanted_frame->sample_rate = spec.freq;
    m_MS.wanted_frame->channel_layout = av_get_default_channel_layout(spec.channels);
    m_MS.wanted_frame->channels = spec.channels;

    avcodec_open2(m_MS.acct, pCodec, nullptr);
    packet_queue_init(&m_MS.audioq);
    SDL_PauseAudio(0);

    AVPacket packet;
    while (true) //这里有一个顺序！先判断退出,再 读 再写入
    {
        SDL_Delay(10);
        if (isquit)
            break;
        if (m_MS.audioq.size > MAX_AUDIO_SIZE)
        {
            continue; //这个才是关键！
        }
        av_read_frame(m_MS.afct, &packet); 
        if (packet.stream_index == audioStream)
            packet_queue_put(&m_MS.audioq, &packet);
        else
            av_free_packet(&packet);
    }
    FreeAllocSpace();
}

然而我想写一个可以播放音频也可以播放视频~不管怎么说，我至少上路了

文件下载：https://github.com/what951006/FFmpeg-Player
更多文章：http://blog.csdn.net/what951006?viewmode=list
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