Android 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);
}
#include
#include
#include "Rtmp.h"
#ifdef __cplusplus
extern "C" {
#endif
JNIEXPORT jlong JNICALL
Java_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 JNICALL
Java_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 JNICALL
Java_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 JNICALL
Java_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 JNICALL
Java_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 JNICALL
Java_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(); }