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Android直播软件开发使用rtmp推流协议是如何实现的（二）

程序员文章站 2022-07-08 19:44:49
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MainActivity 中的代码主要是采集直播软件开发的视频、音频进行编码。然后调用jni方法进行发送。
在c层。我封装了一个类用来发送音视频数据：

class Rtmp {

private:
    int width;
    int height;
    int timeOut;
    std::string url;
    long startTime;
    RTMP *rtmp;

public:
    /**
     * 初始化
     */
    virtual int init(std::string url, int w, int h, int timeOut);

    /**
     * 发送sps、pps 帧
     */
    virtual int sendSpsAndPps(BYTE *sps, int spsLen, BYTE *pps, int ppsLen,
                              long timestamp);

    /**
     * 发送视频帧
     */
    virtual int sendVideoData(BYTE *data, int len, long timestamp);

    /**
     * 发送音频关键帧
     */
    virtual int sendAacSpec(BYTE *data, int len);

    /**
     * 发送音频数据
     */
    virtual int sendAacData(BYTE *data, int len,long timestamp);

    /**
     * 释放资源
     */
    virtual int stop() const;

    virtual ~Rtmp();
};
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实现：

//
// Created by Administrator on 1/16/2017.
//

#include "Rtmp.h"

#include "jni.h"
#include "lang.h"

#define RTMP_HEAD_SIZE (sizeof(RTMPPacket)+RTMP_MAX_HEADER_SIZE)
#define NAL_SLICE  1
#define NAL_SLICE_DPA  2
#define NAL_SLICE_DPB  3
#define NAL_SLICE_DPC  4
#define NAL_SLICE_IDR  5
#define NAL_SEI  6
#define NAL_SPS  7
#define NAL_PPS  8
#define NAL_AUD  9
#define NAL_FILLER  12

#define STREAM_CHANNEL_METADATA  0x03
#define STREAM_CHANNEL_VIDEO     0x04
#define STREAM_CHANNEL_AUDIO     0x05

int Rtmp::init(std::string url, int w, int h, int timeOut) {
    this->url = url;
    this->width = w;
    this->height = h;
    this->timeOut = timeOut;

    RTMP_LogSetLevel(RTMP_LOGDEBUG);
    rtmp = RTMP_Alloc();
    RTMP_Init(rtmp);
    rtmp->Link.timeout = timeOut;
    RTMP_SetupURL(rtmp, (char *) url.c_str());
    RTMP_EnableWrite(rtmp);

    if (RTMP_Connect(rtmp, NULL) <= 0) {
        LOGD("RTMP_Connect error");
        return -1;
    }

    if (RTMP_ConnectStream(rtmp, 0) <= 0) {
        LOGD("RTMP_ConnectStream error");
        return -1;
    }
    return 0;
}

int Rtmp::sendSpsAndPps(BYTE *sps, int spsLen, BYTE *pps, int ppsLen, long timestamp) {

    int i;
    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;//lengthSi*usOne

    /*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)) {
        RTMP_SendPacket(rtmp, packet, TRUE);
    }
    free(packet);
    return 0;
}

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;

    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;
}

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;
    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;
}

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;
}

int Rtmp::stop() const {
    RTMP_Close(rtmp);
    RTMP_Free(rtmp);
    return 0;
}

Rtmp::~Rtmp() { stop(); }
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观看

我们如果测试直播软件开发的话，我们首先需要搭建一个直播软件开发流媒体服务器。可以在本地安装一个Adobe Media Server 。然后打开它其中的一个示例。比如我安装在D盘Program File 文件夹下
那么我打开D:\Program Files\Adobe\Adobe Media Server 5\samples\videoPlayer\videoplayer.html
输入推流的地址就可以播放了。当然手机和电脑记得处于同一个局域网。
Android直播软件开发使用rtmp推流协议是如何实现的（二）

 

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相关标签： 直播APP开发 直播平台搭建 直播系统开发

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