[四]rtmp推流核心c代码

本文详细介绍了RTMP推流的核心过程,涉及RtmpPublisher和RtmpUitlsJava类,以及JNI调用的C++代码,包括RtmpPublisher的初始化、视频和音频数据发送等功能的实现。
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一.rtmp推流过程

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二.rtmp核心方法

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三.rtmp核心代码

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RtmpPublisher.java----->RtmpUitls.java—>publish_jni.cpp----->Ramp.cpp

1.RtmpPublisher.java

public class RtmpPublisher {

    private long cPtr;
    private long timeOffset;

    public static RtmpPublisher newInstance() {

        return new RtmpPublisher();
    }
    private RtmpPublisher(){}

    public int init(String url, int w, int h, int timeOut) {
        cPtr = RtmpUitls.init(url, w, h, timeOut);
        if (cPtr != 0) {
            return 0;
        }
        return -1;
    }

    public int sendSpsAndPps(byte[] sps, int spsLen, byte[] pps, int ppsLen, long timeOffset) {
        this.timeOffset = timeOffset;
        return RtmpUitls.sendSpsAndPps(cPtr, sps, spsLen, pps, ppsLen, 0);
    }

    public int sendVideoData(byte[] data, int len, long timestamp) {
        if(timestamp-timeOffset<=0){return -1;}
        return RtmpUitls.sendVideoData(cPtr, data, len, timestamp - timeOffset);
    }

    public int sendAacSpec(byte[] data, int len) {
        return RtmpUitls.sendAacSpec(cPtr, data, len);
    }

    public int sendAacData(byte[] data, int len, long timestamp) {
        if(timestamp-timestamp<0){return -1;}
        return RtmpUitls.sendAacData(cPtr, data, len, timestamp - timeOffset);
    }

    public int stop() {
        try {
            return RtmpUitls.stop(cPtr);
        }finally {
            cPtr=0;
        }
    }

    @Override
    protected void finalize() throws Throwable {
        super.finalize();
        if(cPtr!=0){
            stop();
        }
    }
}

2.RtmpUitls.java

public final class RtmpUitls {
    static {
        System.loadLibrary("avcore");
    }

    static native long init(String url, int w, int h, int timeOut);

    static native int sendSpsAndPps(long cptr, byte[] sps, int spsLen, byte[] pps,
                                    int ppsLen, long timestamp);

    static native int sendVideoData(long cptr, byte[] data, int len, long timestamp);

    static native int sendAacSpec(long cptr, byte[] data, int len);

    static native int sendAacData(long cptr, byte[] data, int len, long timestamp);

    static native int stop(long cptr);

}

3.publish_jni.cpp:都是jni代码

#include <jni.h>
#include <string>
#include "Rtmp.h"

#ifdef __cplusplus
extern "C" {
#endif

extern "C"
JNIEXPORT jlong JNICALL
Java_com_ivideo_avcore_rtmplive_RtmpUitls_init(JNIEnv *env, jclass type, jstring url_, jint w,
                                               jint h,
                                               jint timeOut) {
    const char *url = env->GetStringUTFChars(url_, 0);
    Rtmp *rtmp = new Rtmp();
    rtmp->init(url, w, h, timeOut);
    env->ReleaseStringUTFChars(url_, url);
    return reinterpret_cast<long> (rtmp);
}

extern "C"
JNIEXPORT jint JNICALL
Java_com_ivideo_avcore_rtmplive_RtmpUitls_sendSpsAndPps(JNIEnv *env, jclass type, jlong cptr,
                                                        jbyteArray sps_, jint spsLen,
                                                        jbyteArray pps_,
                                                        jint ppsLen, jlong timestamp) {
    jbyte *sps = env->GetByteArrayElements(sps_, NULL);
    jbyte *pps = env->GetByteArrayElements(pps_, NULL);
    Rtmp *rtmp = reinterpret_cast<Rtmp *>(cptr);
    int ret = rtmp->sendSpsAndPps((BYTE *) sps, spsLen, (BYTE *) pps, ppsLen, timestamp);

    env->ReleaseByteArrayElements(sps_, sps, 0);
    env->ReleaseByteArrayElements(pps_, pps, 0);
    return ret;
}

extern "C"
JNIEXPORT jint JNICALL
Java_com_ivideo_avcore_rtmplive_RtmpUitls_sendVideoData(JNIEnv *env, jclass type, jlong cptr,
                                                        jbyteArray data_, jint len,
                                                        jlong timestamp) {
    jbyte *data = env->GetByteArrayElements(data_, NULL);
    Rtmp *rtmp = reinterpret_cast<Rtmp *> (cptr);
    int ret = rtmp->sendVideoData((BYTE *) data, len, timestamp);

    env->ReleaseByteArrayElements(data_, data, 0);

    return ret;
}

extern "C"
JNIEXPORT jint JNICALL
Java_com_ivideo_avcore_rtmplive_RtmpUitls_sendAacSpec(JNIEnv *env, jclass type, jlong cptr,
                                                      jbyteArray data_, jint len) {
    jbyte *data = env->GetByteArrayElements(data_, NULL);

    Rtmp *rtmp = reinterpret_cast<Rtmp *> (cptr);
    int ret = rtmp->sendAacSpec((BYTE *) data, len);

    env->ReleaseByteArrayElements(data_, data, 0);
    return ret;
}

extern "C"
JNIEXPORT jint JNICALL
Java_com_ivideo_avcore_rtmplive_RtmpUitls_sendAacData(JNIEnv *env, jclass type, jlong cptr,
                                                      jbyteArray data_, jint len,
                                                      jlong timestamp) {
    jbyte *data = env->GetByteArrayElements(data_, NULL);

    Rtmp *rtmp = reinterpret_cast<Rtmp *> (cptr);
    int ret = rtmp->sendAacData((BYTE *) data, len, timestamp);

    env->ReleaseByteArrayElements(data_, data, 0);
    return ret;
}

extern "C"
JNIEXPORT jint JNICALL
Java_com_ivideo_avcore_rtmplive_RtmpUitls_stop(JNIEnv *env, jclass type, jlong cptr) {
    Rtmp *rtmp = reinterpret_cast<Rtmp *> (cptr);
    delete rtmp;
    return 0;
}

#ifdef __cplusplus
}
#endif

3.rtmp.cpp:c++代码

#include "Rtmp.h"
#include "librtmp/rtmp.h"
#include "librtmp/log.h"
/**
 * 初始化
 * @param url
 * @param w
 * @param h
 * @param timeOut
 * @return
 */
int Rtmp::init(std::string url, int w, int h, int timeOut) {
    RTMP_LogSetLevel(RTMP_LOGDEBUG);
    rtmp = RTMP_Alloc();
    RTMP_Init(rtmp);
    //设置rtmp的链接超时
    LOGI("time out = %d",timeOut);
    rtmp->Link.timeout = timeOut;
    //设置rtmp的推流网址
    RTMP_SetupURL(rtmp, (char *) url.c_str());
    //设置可写
    RTMP_EnableWrite(rtmp);
    //链接rtmp服务器
    if (!RTMP_Connect(rtmp, NULL) ) {
        LOGI("RTMP_Connect error"); //链接失败
        return -1;
    }
    LOGI("RTMP_Connect success.");//链接成功
    if (!RTMP_ConnectStream(rtmp, 0)) {//打开流
        LOGI("RTMP_ConnectStream error");
        return -1;
    }
    LOGI("RTMP_ConnectStream success.");//打开流成功

    return 0;
}
//发送视频:sps和pps
int Rtmp::sendSpsAndPps(BYTE *sps, int spsLen, BYTE *pps, int ppsLen, long timestamp) {

    int i;
    //封装RTMPPacket
    RTMPPacket *packet = (RTMPPacket *) malloc(RTMP_HEAD_SIZE + 1024);
    memset(packet, 0, RTMP_HEAD_SIZE);
    packet->m_body = (char *) packet + RTMP_HEAD_SIZE;
    BYTE *body = (BYTE *) packet->m_body;

    i = 0;
    body[i++] = 0x17; //1:keyframe 7:AVC
    body[i++] = 0x00; // AVC sequence header

    body[i++] = 0x00;
    body[i++] = 0x00;
    body[i++] = 0x00; //fill in 0

    /*AVCDecoderConfigurationRecord*/
    body[i++] = 0x01;
    body[i++] = sps[1]; //AVCProfileIndecation
    body[i++] = sps[2]; //profile_compatibilty
    body[i++] = sps[3]; //AVCLevelIndication
    body[i++] = 0xff;//lengthSizeMinusOne

    /*SPS*/
    body[i++] = 0xe1;
    body[i++] = (spsLen >> 8) & 0xff;
    body[i++] = spsLen & 0xff;
    /*sps data*/
    memcpy(&body[i], sps, spsLen);

    i += spsLen;

    /*PPS*/
    body[i++] = 0x01;
    /*sps data length*/
    body[i++] = (ppsLen >> 8) & 0xff;
    body[i++] = ppsLen & 0xff;
    memcpy(&body[i], pps, ppsLen);
    i += ppsLen;

    packet->m_packetType = RTMP_PACKET_TYPE_VIDEO;
    packet->m_nBodySize = i;
    packet->m_nChannel = 0x04;
    packet->m_nTimeStamp = 0;
    packet->m_hasAbsTimestamp = 0;
    packet->m_headerType = RTMP_PACKET_SIZE_MEDIUM;
    packet->m_nInfoField2 = rtmp->m_stream_id;

    /*发送*/
    if (RTMP_IsConnected(rtmp)) {//判断是否链接着
        RTMP_SendPacket(rtmp, packet, TRUE);//发送RTMPPacket
    }
    free(packet);
    return 0;
}
//发送视频:
int Rtmp::sendVideoData(BYTE *buf, int len, long timestamp) {
    int type;

    /*去掉帧界定符*/
    if (buf[2] == 0x00) {/*00 00 00 01*/
        buf += 4;
        len -= 4;
    } else if (buf[2] == 0x01) {
        buf += 3;
        len - 3;
    }

    type = buf[0] & 0x1f;
    //封装RTMPPacket
    RTMPPacket *packet = (RTMPPacket *) malloc(RTMP_HEAD_SIZE + len + 9);
    memset(packet, 0, RTMP_HEAD_SIZE);
    packet->m_body = (char *) packet + RTMP_HEAD_SIZE;
    packet->m_nBodySize = len + 9;


    /* send video packet*/
    BYTE *body = (BYTE *) packet->m_body;
    memset(body, 0, len + 9);

    /*key frame*/
    body[0] = 0x27;
    if (type == NAL_SLICE_IDR) {
        body[0] = 0x17; //关键帧
    }

    body[1] = 0x01;/*nal unit*/
    body[2] = 0x00;
    body[3] = 0x00;
    body[4] = 0x00;

    body[5] = (len >> 24) & 0xff;
    body[6] = (len >> 16) & 0xff;
    body[7] = (len >> 8) & 0xff;
    body[8] = (len) & 0xff;

    /*copy data*/
    memcpy(&body[9], buf, len);

    packet->m_hasAbsTimestamp = 0;
    packet->m_packetType = RTMP_PACKET_TYPE_VIDEO;
    packet->m_nInfoField2 = rtmp->m_stream_id;
    packet->m_nChannel = 0x04;
    packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
    packet->m_nTimeStamp = timestamp;

    if (RTMP_IsConnected(rtmp)) {
        RTMP_SendPacket(rtmp, packet, TRUE);
    }
    free(packet);

    return 0;
}
//发送音频
int Rtmp::sendAacSpec(BYTE *data, int spec_len) {
    RTMPPacket *packet;
    BYTE *body;
    int len = spec_len;//spec len 是2
    packet = (RTMPPacket *) malloc(RTMP_HEAD_SIZE + len + 2);
    memset(packet, 0, RTMP_HEAD_SIZE);
    packet->m_body = (char *) packet + RTMP_HEAD_SIZE;
    body = (BYTE *) packet->m_body;

    /*AF 00 +AAC RAW data*/
    body[0] = 0xAF;
    body[1] = 0x00;
    memcpy(&body[2], data, len);/*data 是AAC sequeuece header数据*/

    packet->m_packetType = RTMP_PACKET_TYPE_AUDIO;//音频
    packet->m_nBodySize = len + 2;
    packet->m_nChannel = STREAM_CHANNEL_AUDIO;
    packet->m_nTimeStamp = 0;
    packet->m_hasAbsTimestamp = 0;
    packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
    packet->m_nInfoField2 = rtmp->m_stream_id;

    if (RTMP_IsConnected(rtmp)) {
        RTMP_SendPacket(rtmp, packet, TRUE);
    }
    free(packet);

    return 0;
}
//发送音频数据
int Rtmp::sendAacData(BYTE *data, int len, long timeOffset) {
//    data += 5;
//    len += 5;
    if (len > 0) {
        RTMPPacket *packet;
        BYTE *body;
        packet = (RTMPPacket *) malloc(RTMP_HEAD_SIZE + len + 2);
        memset(packet, 0, RTMP_HEAD_SIZE);
        packet->m_body = (char *) packet + RTMP_HEAD_SIZE;
        body = (BYTE *) packet->m_body;

        /*AF 00 +AAC Raw data*/
        body[0] = 0xAF;
        body[1] = 0x01;
        memcpy(&body[2], data, len);

        packet->m_packetType = RTMP_PACKET_TYPE_AUDIO;
        packet->m_nBodySize = len + 2;
        packet->m_nChannel = STREAM_CHANNEL_AUDIO;
        packet->m_nTimeStamp = timeOffset;
        packet->m_hasAbsTimestamp = 0;
        packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
        packet->m_nInfoField2 = rtmp->m_stream_id;
        if (RTMP_IsConnected(rtmp)) {
            RTMP_SendPacket(rtmp, packet, TRUE);
        }
        LOGD("send packet body[0]=%x,body[1]=%x", body[0], body[1]);
        free(packet);

    }
    return 0;
}
//释放资源
int Rtmp::stop() const {
    RTMP_Close(rtmp);
    RTMP_Free(rtmp);
    return 0;
}

Rtmp::~Rtmp() { stop(); }
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当然,我可以为您提供一个简单的FFmpeg C++代码示例,用于将RTSP流转发到RTMP服务器。以下是一个基本的代码框架: ```cpp #include <iostream> #include <string> #include <cstdlib> #include <cstdio> extern "C" { #include <libavformat/avformat.h> #include <libavutil/opt.h> } int main() { // 初始化FFmpeg库 av_register_all(); avformat_network_init(); // 创建输入上下文 AVFormatContext* inputContext = nullptr; if (avformat_open_input(&inputContext, "rtsp://your_rtsp_url", nullptr, nullptr) != 0) { std::cerr << "无法打开RTSP输入流" << std::endl; return -1; } // 查找输入流信息 if (avformat_find_stream_info(inputContext, nullptr) < 0) { std::cerr << "无法获取输入流信息" << std::endl; return -1; } // 创建输出上下文 AVFormatContext* outputContext = nullptr; if (avformat_alloc_output_context2(&outputContext, nullptr, "flv", "rtmp://your_rtmp_url") < 0) { std::cerr << "无法创建输出上下文" << std::endl; return -1; } // 遍历输入流,复制到输出上下文 for (unsigned int i = 0; i < inputContext->nb_streams; i++) { AVStream* inStream = inputContext->streams[i]; AVStream* outStream = avformat_new_stream(outputContext, inStream->codec->codec); if (!outStream) { std::cerr << "无法创建输出流" << std::endl; return -1; } // 复制流信息 if (avcodec_copy_context(outStream->codec, inStream->codec) < 0) { std::cerr << "无法复制流信息" << std::endl; return -1; } } // 打开输出URL if (!(outputContext->oformat->flags & AVFMT_NOFILE)) { if (avio_open(&outputContext->pb, "rtmp://your_rtmp_url", AVIO_FLAG_WRITE) < 0) { std::cerr << "无法打开输出URL" << std::endl; return -1; } } // 写入输出文件头 if (avformat_write_header(outputContext, nullptr) < 0) { std::cerr << "无法写入输出文件头" << std::endl; return -1; } // 转发流数据 AVPacket packet; while (av_read_frame(inputContext, &packet) >= 0) { AVStream* inStream = inputContext->streams[packet.stream_index]; AVStream* outStream = outputContext->streams[packet.stream_index]; // 设置时间基 av_packet_rescale_ts(&packet, inStream->time_base, outStream->time_base); packet.pos = -1; // 写入输出包 if (av_interleaved_write_frame(outputContext, &packet) < 0) { std::cerr << "无法写入输出包" << std::endl; break; } av_packet_unref(&packet); } // 写入输出文件尾 av_write_trailer(outputContext); // 清理资源 avformat_close_input(&inputContext); if (outputContext && !(outputContext->oformat->flags & AVFMT_NOFILE)) avio_closep(&outputContext->pb); avformat_free_context(outputContext); return 0; } ``` 请注意,您需要根据实际情况修改以下内容: - `rtsp://your_rtsp_url`:替换为您的RTSP流URL。 - `rtmp://your_rtmp_url`:替换为您的RTMP服务器URL。 此代码使用FFmpeg库来处理RTSP流,并将其转发到RTMP服务器。它首先打开RTSP输入流,然后创建一个输出上下文,并复制输入流的信息到输出流中。然后,它打开RTMP输出URL,并开始转发流数据。最后,它写入输出文件尾,并清理资源。 希望这个代码示例对您有所帮助!如果您有任何进一步的问题,请随时提问。

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