zlmediaKit中MediaPlayer解读

在源码中Player中有Player的基类，作者采用了模板委托，设计模式我第一次遇到。
MediaPlayer是干什么的？
我们以RtmpMediaPlay为例，它其实是一个rtmp客户端，

class RtmpPlayer : public PlayerBase, public TcpClient, public RtmpProtocol{}

从它的继承我们可以看到有PlayerBase，TcpClient，RtmpProtocol，直白点就是一个tcp客户端，上层握手协议为rtmp,连接rtmp后可以从play()处获取到demuxer的h264数据。

 ctx->player = mk_player_create();
 //回复play的时候触发
  mk_player_set_on_result(ctx->player, on_mk_play_event_func, ctx);
  mk_player_set_on_shutdown(ctx->player, on_mk_play_event_func, ctx);
  mk_player_set_on_data(ctx->player, on_mk_play_data_func, ctx);
  mk_player_play(ctx->player, url_pull);

下面分析一下mk_player_create()：

//首先
API_EXPORT mk_player API_CALL mk_player_create() {
    MediaPlayerForC::Ptr *obj = new MediaPlayerForC::Ptr(new MediaPlayerForC());
    (*obj)->setup();
    return obj;
}
//这里创建了一个MediaPlayerForC对象，里面有一个成员变量MediaPlayer::Ptr _player;
//然后执行setup()
    void setup(){
        weak_ptr<MediaPlayerForC> weak_self = shared_from_this();
        _player->setOnPlayResult([weak_self](const SockException &ex){
            auto strong_self = weak_self.lock();
            if(strong_self){
                strong_self->onEvent(false,ex);
            }
        });

        _player->setOnShutdown([weak_self](const SockException &ex){
            auto strong_self = weak_self.lock();
            if(strong_self){
                strong_self->onEvent(true,ex);
            }
        });
    }
//这里设置两个回调函数
 void onEvent(bool is_shutdown, const SockException &ex){
        lock_guard<recursive_mutex> lck(_mtx);
        if(is_shutdown){
            //播放中断
            if(_on_shutdown){
                _on_shutdown(_on_shutdown_data,ex.getErrCode(),ex.what());
            }
            return;
        }

        //播放结果
        if(_on_play){
        //mk_player_set_on_event设置的回调，创建的时候调一次
            _on_play(_on_play_data,ex.getErrCode(),ex.what());
        }

        if(ex){
            //播放失败
            return;
        }

        //播放成功,添加事件回调
        weak_ptr<MediaPlayerForC> weak_self = shared_from_this();
        auto delegate = std::make_shared<FrameWriterInterfaceHelper>([weak_self](const Frame::Ptr &frame) {
            auto strong_self = weak_self.lock();
            if (strong_self) {
                strong_self->onData(frame);
            }
        });
        //这里有track通道，可以看通道的抽象,这里为false说明通道还没有准备好
        for (auto &track : _player->getTracks(false)) {
            track->addDelegate(delegate);
        }
    }
    void onData(const Frame::Ptr &frame){
        lock_guard<recursive_mutex> lck(_mtx);
        if(_on_data){
                _on_data(_on_data_data,frame->getTrackType(),frame->getCodecId(),frame->data(),frame->size(),frame->dts(),frame->pts())；
               
        }
    }
    //在这里强调一下就是关于track是否准备好，什么是准备好呢，我们拿h264track来举例，它包含的主要成员变量是：
     bool _is_idr = false;
    int _width = 0;
    int _height = 0;
    float _fps = 0;
    string _sps;
    string _pps;
    再来看ready()
    bool H264Track::ready() {
    return !_sps.empty() && !_pps.empty();
}
//看到没，就是sps和pps都不为空了，说明这个通道就准备好了，一般我们创建一个通道的时候有两种方式，一种是
    /**
     * 不指定sps pps构造h264类型的媒体
     * 在随后的inputFrame中获取sps pps
     */
    H264Track() = default;
    另外一种是：
    /**
     * 构造h264类型的媒体
     * @param sps sps帧数据
     * @param pps pps帧数据
     * @param sps_prefix_len 264头长度，可以为3个或4个字节，一般为0x00 00 00 01
     * @param pps_prefix_len 264头长度，可以为3个或4个字节，一般为0x00 00 00 01
     */
    H264Track(const string &sps,const string &pps,int sps_prefix_len = 4,int pps_prefix_len = 4);

    /**
     * 构造h264类型的媒体
     * @param sps sps帧
     * @param pps pps帧
     */
    H264Track(const Frame::Ptr &sps,const Frame::Ptr &pps);
    但是我们常用的还是第一种，通过从后面获取到sps和pps直接为tracm赋值。
    那么player在说明时候创建trackn呢，通过代码找到：
    class Demuxer : public PlayerBase{
public:
    class Listener{
    public:
        Listener() = default;
        virtual ~Listener() = default;
        virtual void onAddTrack(const Track::Ptr &track) = 0;
    };

    Demuxer(){};
    virtual ~Demuxer(){};

    /**
     * 返回是否完成初始化完毕
     * 在构造RtspDemuxer对象时有些rtsp的sdp不包含sps pps信息
     * 所以要等待接收到到sps的rtp包后才能完成
     *
     * 在构造RtmpDemuxer对象时是无法获取sps pps aac_cfg等这些信息，
     * 所以要调用inputRtmp后才会获取到这些信息，这时才初始化成功
     * @param analysisMs 数据流最大分析时间 单位毫秒
     * @return
     */
    bool isInited(int analysisMs) override;

    /**
     * 获取所有Track
     * @return 所有Track
     */
    vector<Track::Ptr> getTracks(bool trackReady = true) const override;

    /**
     * 获取节目总时长
     * @return 节目总时长,单位秒
     */
    float getDuration() const override;

    /**
     * 设置track监听器
     */
    void setTrackListener(Listener *listener);
protected:
    void onAddTrack(const Track::Ptr &track);
protected:
    Listener *_listener = nullptr;
    AudioTrack::Ptr _audioTrack;//最原始的在这里track
    VideoTrack::Ptr _videoTrack;
    Ticker _ticker;
    float _fDuration = 0;
};
class PlayerBase : public DemuxerBase, public mINI
继续：
class RtmpPlayerImp: public PlayerImp<RtmpPlayer,RtmpDemuxer> 
在下面函数中找到创建track的代码
void RtmpDemuxer::inputRtmp(const RtmpPacket::Ptr &pkt) {
    switch (pkt->type_id) {
        case MSG_VIDEO: {
            if (!_try_get_video_track) {
                _try_get_video_track = true;
                auto codec = AMFValue(pkt->getMediaType());
                makeVideoTrack(codec, 0);//这里创建track
            }
            if (_video_rtmp_decoder) {
                _video_rtmp_decoder->inputRtmp(pkt);
            }
            break;
        }

        case MSG_AUDIO: {
            if (!_try_get_audio_track) {
                _try_get_audio_track = true;
                auto codec = AMFValue(pkt->getMediaType());
                makeAudioTrack(codec, pkt->getAudioSampleRate(), pkt->getAudioChannel(), pkt->getAudioSampleBit(), 0);
            }
            if (_audio_rtmp_decoder) {
                _audio_rtmp_decoder->inputRtmp(pkt);
            }
            break;
        }
        default : break;
    }
}
    
    
//其实真正创建rtmpPlayer在这个地方
API_EXPORT void API_CALL mk_player_play(mk_player ctx, const char *url) {
    assert(ctx && url);
    MediaPlayerForC &obj = **((MediaPlayerForC::Ptr *)ctx);
    auto player = obj.getPlayer();
    string url_str(url);
    player->getPoller()->async([url_str,player](){
        //切换线程后再操作
        player->play(url_str);
    });
}
void MediaPlayer::play(const string &url) {
//根据url创建相应的player
    _delegate = PlayerBase::createPlayer(_poller, url);
    assert(_delegate);
    setOnCreateSocket_l(_delegate, _on_create_socket);
    _delegate->setOnShutdown(_shutdownCB);
    _delegate->setOnPlayResult(_playResultCB);
    _delegate->setOnResume(_resumeCB);
    _delegate->setMediaSource(_pMediaSrc);
    _delegate->mINI::operator=(*this);
    //最终执行player在这里
    _delegate->play(url);
}

//下面是rtmpplayer的player
void RtmpPlayer::play(const string &strUrl)  {
    teardown();
    string host_url = FindField(strUrl.data(), "://", "/");
    _app = FindField(strUrl.data(), (host_url + "/").data(), "/");
    _stream_id = FindField(strUrl.data(), (host_url + "/" + _app + "/").data(), NULL);
    _tc_url = string("rtmp://") + host_url + "/" + _app;

    if (!_app.size() || !_stream_id.size()) {
        onPlayResult_l(SockException(Err_other, "rtmp url非法"), false);
        return;
    }
    DebugL << host_url << " " << _app << " " << _stream_id;

    auto iPort = atoi(FindField(host_url.data(), ":", NULL).data());
    if (iPort <= 0) {
        //rtmp 默认端口1935
        iPort = 1935;
    } else {
        //服务器域名
        host_url = FindField(host_url.data(), NULL, ":");
    }
    if (!(*this)[kNetAdapter].empty()) {
        setNetAdapter((*this)[kNetAdapter]);
    }

    weak_ptr<RtmpPlayer> weak_self = dynamic_pointer_cast<RtmpPlayer>(shared_from_this());
    float play_timeout_sec = (*this)[kTimeoutMS].as<int>() / 1000.0f;
    _play_timer.reset(new Timer(play_timeout_sec, [weak_self]() {
        auto strong_self = weak_self.lock();
        if (!strong_self) {
            return false;
        }
        //这里设置了回调
        strong_self->onPlayResult_l(SockException(Err_timeout, "play rtmp timeout"), false);
        return false;
    }, getPoller()));

    _metadata_got = false;
    //关键是这里，客户端连接服务器
    startConnect(host_url, iPort, play_timeout_sec);
}
//注意这里有tcpclient的回调
onConnect
onRecv

void RtmpPlayer::onConnect(const SockException &err){
    if (err.getErrCode() != Err_success) {
        onPlayResult_l(err, false);
        return;
    }
    weak_ptr<RtmpPlayer> weakSelf = dynamic_pointer_cast<RtmpPlayer>(shared_from_this());
    //在这里完成了握手协议
    startClientSession([weakSelf]() {
        auto strongSelf = weakSelf.lock();
        if (!strongSelf) {
            return;
        }
        //发送连接请求
        strongSelf->send_connect();
    });
}
void RtmpProtocol::startClientSession(const function<void()> &func) {
    //发送 C0C1
    char handshake_head = HANDSHAKE_PLAINTEXT;
    onSendRawData(obtainBuffer(&handshake_head, 1));
    RtmpHandshake c1(0);
    onSendRawData(obtainBuffer((char *) (&c1), sizeof(c1)));
    _next_step_func = [this, func](const char *data, size_t len) {
        //等待 S0+S1+S2
        return handle_S0S1S2(data, len, func);
    };
}
const char* RtmpProtocol::handle_S0S1S2(const char *data, size_t len, const function<void()> &func) {
    if (len < 1 + 2 * C1_HANDSHARK_SIZE) {
        //数据不够
        return nullptr;
    }
    if (data[0] != HANDSHAKE_PLAINTEXT) {
        throw std::runtime_error("only plaintext[0x03] handshake supported");
    }
    //发送 C2
    const char *pcC2 = data + 1;
    onSendRawData(obtainBuffer(pcC2, C1_HANDSHARK_SIZE));
    //握手结束
    _next_step_func = [this](const char *data, size_t len) {
        //握手结束并且开始进入解析命令模式
        return handle_rtmp(data, len);
    };
    //这里发送connect命令
    func();
    return data + 1 + 2 * C1_HANDSHARK_SIZE;
}

const char* RtmpProtocol::handle_rtmp(const char *data, size_t len) {
    auto ptr = data;
    while (len) {
        int offset = 0;
        uint8_t flags = ptr[0];
        size_t header_len = HEADER_LENGTH[flags >> 6];
        _now_chunk_id = flags & 0x3f;
        switch (_now_chunk_id) {
            case 0: {
                //0 值表示二字节形式，并且 ID 范围 64 - 319
                //(第二个字节 + 64)。
                if (len < 2) {
                    //need more data
                    return ptr;
                }
                _now_chunk_id = 64 + (uint8_t) (ptr[1]);
                offset = 1;
                break;
            }

            case 1: {
                //1 值表示三字节形式，并且 ID 范围为 64 - 65599
                //((第三个字节) * 256 + 第二个字节 + 64)。
                if (len < 3) {
                    //need more data
                    return ptr;
                }
                _now_chunk_id = 64 + ((uint8_t) (ptr[2]) << 8) + (uint8_t) (ptr[1]);
                offset = 2;
                break;
            }

            //带有 2 值的块流 ID 被保留，用于下层协议控制消息和命令。
            default : break;
        }

        if (len < header_len + offset) {
            //need more data
            return ptr;
        }
        RtmpHeader &header = *((RtmpHeader *) (ptr + offset));
        auto &pr = _map_chunk_data[_now_chunk_id];
        auto &now_packet = pr.first;
        auto &last_packet = pr.second;
        if (!now_packet) {
            now_packet = RtmpPacket::create();
            if (last_packet) {
                //恢复chunk上下文
                *now_packet = *last_packet;
            }
            //绝对时间戳标记复位
            now_packet->is_abs_stamp = false;
        }
        auto &chunk_data = *now_packet;
        chunk_data.chunk_id = _now_chunk_id;
        switch (header_len) {
            case 12:
                chunk_data.is_abs_stamp = true;
                chunk_data.stream_index = load_le32(header.stream_index);
            case 8:
                chunk_data.body_size = load_be24(header.body_size);
                chunk_data.type_id = header.type_id;
            case 4:
                chunk_data.ts_field = load_be24(header.time_stamp);
        }

        auto time_stamp = chunk_data.ts_field;
        if (chunk_data.ts_field == 0xFFFFFF) {
            if (len < header_len + offset + 4) {
                //need more data
                return ptr;
            }
            time_stamp = load_be32(ptr + offset + header_len);
            offset += 4;
        }

        if (chunk_data.body_size < chunk_data.buffer.size()) {
            throw std::runtime_error("非法的bodySize");
        }

        auto more = min(_chunk_size_in, (size_t) (chunk_data.body_size - chunk_data.buffer.size()));
        if (len < header_len + offset + more) {
            //need more data
            return ptr;
        }
        if (more) {
            chunk_data.buffer.append(ptr + header_len + offset, more);
        }
        ptr += header_len + offset + more;
        len -= header_len + offset + more;
        if (chunk_data.buffer.size() == chunk_data.body_size) {
            //frame is ready
            _now_stream_index = chunk_data.stream_index;
            chunk_data.time_stamp = time_stamp + (chunk_data.is_abs_stamp ? 0 : chunk_data.time_stamp);
            //保存chunk上下文
            last_packet = now_packet;
            if (chunk_data.body_size) {
                handle_chunk(std::move(now_packet));
            } else {
                now_packet = nullptr;
            }
        }
    }
    return ptr;
}

//然后发送connect
inline void RtmpPlayer::send_connect() {
    AMFValue obj(AMF_OBJECT);
    obj.set("app", _app);
    obj.set("tcUrl", _tc_url);
    //未使用代理
    obj.set("fpad", false);
    //参考librtmp,什么作用?
    obj.set("capabilities", 15);
    //SUPPORT_VID_CLIENT_SEEK 支持seek
    obj.set("videoFunction", 1);
    //只支持aac
    obj.set("audioCodecs", (double) (0x0400));
    //只支持H264
    obj.set("videoCodecs", (double) (0x0080));
    sendInvoke("connect", obj);
    addOnResultCB([this](AMFDecoder &dec) {
        //TraceL << "connect result";
        dec.load<AMFValue>();
        auto val = dec.load<AMFValue>();
        auto level = val["level"].as_string();
        auto code = val["code"].as_string();
        if (level != "status") {
            throw std::runtime_error(StrPrinter << "connect 失败:" << level << " " << code << endl);
        }
        send_createStream();
    });
}
接着创建流
inline void RtmpPlayer::send_createStream() {
    AMFValue obj(AMF_NULL);
    sendInvoke("createStream", obj);
    addOnResultCB([this](AMFDecoder &dec) {
        //TraceL << "createStream result";
        dec.load<AMFValue>();
        _stream_index = dec.load<int>();
        send_play();
    });
}
然后发送play命令

inline void RtmpPlayer::send_play() {
    AMFEncoder enc;
    enc << "play" << ++_send_req_id << nullptr << _stream_id << (double) _stream_index;
    sendRequest(MSG_CMD, enc.data());
    auto fun = [](AMFValue &val) {
        //TraceL << "play onStatus";
        auto level = val["level"].as_string();
        auto code = val["code"].as_string();
        if (level != "status") {
            throw std::runtime_error(StrPrinter << "play 失败:" << level << " " << code << endl);
        }
    };
    addOnStatusCB(fun);
    addOnStatusCB(fun);
}




//这里就是解析rtmp 包
void RtmpPlayer::onRecv(const Buffer::Ptr &buf){
    try {
        if (_benchmark_mode && !_play_timer) {
            //在性能测试模式下，如果rtmp握手完毕后，不再解析rtmp包
            _rtmp_recv_ticker.resetTime();
            return;
        }
        onParseRtmp(buf->data(), buf->size());
    } catch (exception &e) {
        SockException ex(Err_other, e.what());
        //定时器_pPlayTimer为空后表明握手结束了
        onPlayResult_l(ex, !_play_timer);
    }
}


void RtmpPlayer::onPlayResult_l(const SockException &ex, bool handshake_done) {
    if (ex.getErrCode() == Err_shutdown) {
        //主动shutdown的，不触发回调
        return;
    }

    WarnL << ex.getErrCode() << " " << ex.what();
    if (!handshake_done) {
        //开始播放阶段
        _play_timer.reset();
        //是否为性能测试模式
        _benchmark_mode = (*this)[Client::kBenchmarkMode].as<int>();
        onPlayResult(ex);
    } else if (ex) {
        //播放成功后异常断开回调
        onShutdown(ex);
    } else {
        //恢复播放
        onResume();
    }

    if (!ex) {
        //播放成功，恢复rtmp接收超时定时器
        _rtmp_recv_ticker.resetTime();
        auto timeout_ms = (*this)[kMediaTimeoutMS].as<uint64_t>();
        weak_ptr<RtmpPlayer> weakSelf = dynamic_pointer_cast<RtmpPlayer>(shared_from_this());
        auto lam = [weakSelf, timeout_ms]() {
            auto strongSelf = weakSelf.lock();
            if (!strongSelf) {
                return false;
            }
            if (strongSelf->_rtmp_recv_ticker.elapsedTime() > timeout_ms) {
                //接收rtmp媒体数据超时
                SockException ex(Err_timeout, "receive rtmp timeout");
                strongSelf->onPlayResult_l(ex, true);
                return false;
            }
            return true;
        };
        //创建rtmp数据接收超时检测定时器
        _rtmp_recv_timer = std::make_shared<Timer>(timeout_ms / 2000.0f, lam, getPoller());
    } else {
        shutdown(SockException(Err_shutdown,"teardown"));
    }
}