Android(安卓)使用librtmp推流【音视频传输】

1.通过jni调用librtmp

下面是jni中提供给java调用的本地方法：

public final class PublishJni {    static {        System.loadLibrary("publish");    }    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);}

publish.so包中导出的调用接口

#include #include #include "Rtmp.h"#ifdef __cplusplusextern "C" {#endifJNIEXPORT jlong JNICALLJava_com_blueberry_media_PublishJni_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 (rtmp);}JNIEXPORT jint JNICALLJava_com_blueberry_media_PublishJni_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(cptr);    int ret = rtmp->sendSpsAndPps((BYTE *) sps, spsLen, (BYTE *) pps, ppsLen, timestamp);    env->ReleaseByteArrayElements(sps_, sps, 0);    env->ReleaseByteArrayElements(pps_, pps, 0);    return ret;}JNIEXPORT jint JNICALLJava_com_blueberry_media_PublishJni_sendVideoData(JNIEnv *env, jclass type, jlong cptr,                                                  jbyteArray data_, jint len, jlong timestamp) {    jbyte *data = env->GetByteArrayElements(data_, NULL);    Rtmp *rtmp = reinterpret_cast (cptr);    int ret = rtmp->sendVideoData((BYTE *) data, len, timestamp);    env->ReleaseByteArrayElements(data_, data, 0);    return ret;}JNIEXPORT jint JNICALLJava_com_blueberry_media_PublishJni_sendAacSpec(JNIEnv *env, jclass type, jlong cptr,                                                jbyteArray data_, jint len) {    jbyte *data = env->GetByteArrayElements(data_, NULL);    Rtmp *rtmp = reinterpret_cast (cptr);    int ret = rtmp->sendAacSpec((BYTE *) data, len);    env->ReleaseByteArrayElements(data_, data, 0);    return ret;}JNIEXPORT jint JNICALLJava_com_blueberry_media_PublishJni_sendAacData(JNIEnv *env, jclass type, jlong cptr,                                                jbyteArray data_, jint len, jlong timestamp) {    jbyte *data = env->GetByteArrayElements(data_, NULL);    Rtmp *rtmp = reinterpret_cast (cptr);    int ret = rtmp->sendAacData((BYTE *) data, len, timestamp);    env->ReleaseByteArrayElements(data_, data, 0);    return ret;}JNIEXPORT jint JNICALLJava_com_blueberry_media_PublishJni_stop(JNIEnv *env, jclass type, jlong cptr) {    Rtmp *rtmp = reinterpret_cast (cptr);    delete rtmp;    return 0;}#ifdef __cplusplus}#endif

Rtmp具体接口实现：

#include "Rtmp.h"int Rtmp::init(std::string url, int w, int h, int timeOut) {    RTMP_LogSetLevel(RTMP_LOGDEBUG);    rtmp = RTMP_Alloc();    RTMP_Init(rtmp);    LOGI("time out = %d",timeOut);    rtmp->Link.timeout = timeOut;    RTMP_SetupURL(rtmp, (char *) url.c_str());    RTMP_EnableWrite(rtmp);    if (!RTMP_Connect(rtmp, NULL) ) {        LOGI("RTMP_Connect error");        return -1;    }    LOGI("RTMP_Connect success.");    if (!RTMP_ConnectStream(rtmp, 0)) {        LOGI("RTMP_ConnectStream error");        return -1;    }    LOGI("RTMP_ConnectStream success.");    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;    //VideoTagHeader[TagType==9]    //Frame Type 4bits | CodecID 4bits | AVCPacketType 8bit | CompositionTime 24bits|    //A)Frame Type, Type of video frame. The following values are defined:    //1 = key frame (for AVC, a seekable frame)    //2 = inter frame (for AVC, a non-seekable frame)    //3 = disposable inter frame (H.263 only)    //4 = generated key frame (reserved for server use only)    //5 = video info/command frame    //B)CodecID, Codec Identifier. The following values are defined:    //2 = Sorenson H.263    //3 = Screen video    //4 = On2 VP6    //5 = On2 VP6 with alpha channel    //6 = Screen video version 2    //7 = AVC    //C)AVCPacketType, IF CodecID == 7:    //0 = AVC sequence header    //1 = AVC NALU    //2 = AVC end of sequence (lower level NALU sequence ender is not required or supported)    //D)CompositionTime, IF CodecID == 7:    //IF AVCPacketType == 1    //Composition time offset    //ELSE    //0    //See ISO 14496-12, 8.15.3 for an explanation of composition    //times. The offset in an FLV file is always in milliseconds.    i = 0;    body[i++] = 0x17; //1:keyframe(I帧) 7:AVC    body[i++] = 0x00; //AVC packet type：AVC sequence header     body[i++] = 0x00; //composition time：3字节，AVC时无意义，全为0     body[i++] = 0x00;    body[i++] = 0x00; //fill in 0    /*AVCDecoderConfigurationRecord*/    body[i++] = 0x01;   //configurationVersion  1(byte)    版本    body[i++] = sps[1]; //AVCProfileIndecation  1(byte)    sps[1]    body[i++] = sps[2]; //profile_compatibilty  1(byte)    sps[2]    body[i++] = sps[3]; //AVCLevelIndication    1(byte)    sps[3]    body[i++] = 0xff;   //lengthSizeMinusOne    1(byte)    FLV中NALU包长数据所使用的字节数，包长= （lengthSizeMinusOne & 3） + 1    /*SPS*/    body[i++] = 0xe1;                  //numOfSequenceParameterSets    1(byte)    SPS个数，通常为0xe1 个数= numOfSequenceParameterSets & 01F    body[i++] = (spsLen >> 8) & 0xff;  //sequenceParameterSetLength    2(byte)    SPS长度    body[i++] = spsLen & 0xff;             /*sps data*/    memcpy(&body[i], sps, spsLen);     //sequenceParameterSetNALUnits  x(byte)    SPS内容    i += spsLen;    /*PPS*/    body[i++] = 0x01;                   //numOfPictureParameterSets  1(byte)         PPS个数，通常为0x01    /*sps data length*/    body[i++] = (ppsLen >> 8) & 0xff;   //pictureParameterSetLength  2(byte)         PPS长度    body[i++] = ppsLen & 0xff;    memcpy(&body[i], pps, ppsLen);      //sequenceParameterSetNALUnits ppslen(byte)  PPS内容    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;    //sps 与 pps 的帧界定符都是 00 00 00 01，而普通帧可能是 00 00 00 01 也有可能 00 00 01    /*去掉帧界定符*/    if (buf[2] == 0x00) {//startcode:/*00 00 00 01*/         buf += 4;        len -= 4;    } else if (buf[2] == 0x01) {//startcode:/*00 00 01*/         buf += 3;        len - 3;    }    type = buf[0] & 0x1f; //1bit:forbidden,2bit:priority,5bit:type    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;                 //2:innerframe(P帧) 7:AVC     if (type == NAL_SLICE_IDR) {        body[0] = 0x17;             //1:keyframe(I帧) 7:AVC    }    body[1] = 0x01;    //AVC packet type：AVC NALU  /*nal unit*/    body[2] = 0x00;    //composition time：3字节，AVC时无意义，全为0     body[3] = 0x00;    body[4] = 0x00;    body[5] = (len >> 24) & 0xff;    //NALU length：(lengthSizeMinusOne & 3) + 1字节 NALU长度     body[6] = (len >> 16) & 0xff;    body[7] = (len >> 8) & 0xff;    body[8] = (len) & 0xff;    /*copy data*/    memcpy(&body[9], buf, len);      //NALU Data    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, AAC sequence header存放的是AudioSpecificConfig结构    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;    //AudioTagHeader[TagType==8]    //SoundFormat 4bits | SoundRate 2bits | SoundSize 1bit | SoundType 1bits| AACPacketType 8bits|    //A)SoundFormat, 音频格式. The following values are defined:    //0 = Linear PCM, platform endian    //1 = ADPCM    //2 = MP3    //3 = Linear PCM, little endian    //4 = Nellymoser 16 kHz mono    //5 = Nellymoser 8 kHz mono    //6 = Nellymoser    //7 = G.711 A-law logarithmic PCM    //8 = G.711 mu-law logarithmic PCM    //9 = reserved    //10 = AAC    //11 = Speex    //14 = MP3 8 kHz    //15 = Device-specific sound    //Formats 7, 8, 14, and 15 are reserved.    //AAC is supported in Flash Player 9,0,115,0 and higher.    //Speex is supported in Flash Player 10 and higher.    //B)SoundRate, 采样率. The following values are defined:    //0 = 5.5 kHz    //1 = 11 kHz    //2 = 22 kHz    //3 = 44 kHz    //C)SoundSize, 采样精度. Size of each audio sample. This parameter only pertains to uncompressed formats. Compressed formats always decode to 16 bits internally.    //0 = 8-bit samples    //1 = 16-bit samples    //D)SoundType, 声道数. Mono or stereo sound    //0 = Mono sound    //1 = Stereo sound    //E)AACPacketType, AACAUDIODATA的类型. IF SoundFormat == 10    //只有音频格式为AAC（0x0A），AudioTagHeader中才会多出1个字节的数据AACPacketType    //0 = AAC sequence header    //1 = AAC raw    //1)第一个字节af，a就是10代表的意思是AAC.    //后四位f代表如下:    //前2个bit的含义 抽样频率，二进制11，代表44kHZ.    //第3个bit，采样精度, 1代表音频用16位的.    //第4个bit代表声道 1代表Stereo.    //2)第2个字节00, 代表AACPacketType：    //0 = AAC sequence header，1 = AAC raw。    //第一个音频包用0，后面的都用1    /*AF 00 +AAC sequence header*/    body[0] = 0xAF;        body[1] = 0x00;    memcpy(&body[2], data, len);/*data 是AAC sequeuece header数据*/    //AudioSpecificConfig(2byte)结构定义：    //audioObjectType:5bit，表示编码结构类型，AAC main编码为1，LOW低复杂度编码为2，SSR为3。    //samplingFrequencyIndex:4bit，表示采样率。    //0x00   96000    //0x01   88200    //0x02   64000    //0x03   48000    //0x04   44100    //0x05   32000    //0x06   24000    //0x07   22050    //0x08   16000    //0x09   12000    //0x0A   11025    //0x0B     8000    //0x0C   reserved    //0x0D   reserved    //0x0E   reserved    //0x0F   escape value    // 按理说，应该是：0 ~ 96000， 1~88200， 2~64000， 3~48000， 4~44100， 5~32000， 6~24000， 7~ 22050， 8~16000...)，    //通常aac固定选中44100，即应该对应为4，但是试验结果表明，当音频采样率小于等于44100时，应该选择3，而当音频采样率为48000时，应该选择2；    //channelConfiguration:4bit，表示声道数。    //frameLengthFlag:1bit    //dependsOnCoreCoder:1bit    //extensionFlag:1bit    //最后3bit，固定为0    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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