最近在window是平臺下，做了一功能實現通過OBS採集音視頻，並通過RTMP協議將其編碼壓縮後的數據接入到自己的程序中來，因OBS軟件自帶有很強大的遊戲錄製和桌面錄製的功能，以及輸入、輸出音頻設備數據的採集並混音的功能，目前鬥魚遊戲直播也是使用的此軟件作爲錄製工具。
OBS軟件由於使用了window sdk進行音頻的採集，所以只支持window vista版本以上平臺，故XP系統用戶是使用不了此軟件的，下載地址：https://obsproject.com/download
OBS目前只支持RTMP協議進行廣播，所以需要在自己的程序中搭建一RTMP server，讓OBS接進來，然後通過研讀OBS源碼，對接收到的RTMP音視頻數據進行反封裝，最後組裝成原始的音視頻裸碼流，如H.264、MP3、AAC。
下面詳細介紹下主要流程：
1、如何搭建rtmp server
首先用戶可以下載我上傳的librtmp庫http://down.51cto.com/data/1904438；
2、接收到視頻的處理
由於OBS在編碼後在第一幀會把pps、sps、spi等解碼關鍵信息協議開始後單獨發送一次，所以在接收端必須保存該數據內容，並加在以後的每一個關鍵幀頭部，否則不能正常的解碼。以後每接受到一個關鍵幀（通過判斷其第一個字節是否是0x17），在其頭部加上該信息，然後在所有數據幀前面加上3個字節0x00 0x00 0x01 ,大致流程如上，具體請參考代碼。
3、接收到音頻的處理
目前OBS支持MP3和AAC兩種音頻編碼格式，針對MP3編碼，接收端只需要將數據包從第二個字節開始全部存儲即可通過MP3解碼器進行正常的解碼了。針對AAC編碼，目前還沒有處理。
相關代碼如下，已經去掉了一些敏感代碼，僅供參考。
//頭文件
#include <rtmp/srs_protocol_rtmp.hpp>
#include <rtmp/srs_protocol_rtmp_stack.hpp>
#include <libs/srs_librtmp.hpp>
#include <process.h>
#define InitSockets() {\
WORD version; \
WSADATA wsaData; \
\
version = MAKEWORD(1,1); \
WSAStartup(version, &wsaData); }
#define CleanupSockets() WSACleanup()
class RtmpServer : public CThread
{
public:
RtmpServer( );
~RtmpServer();
int StartRtmpServer( );
void StopRtmpServer( );
int fmle_publish(SrsRtmpServer* rtmp, SrsRequest* req);
virtual int32_t terminate();
virtual void execute ( );
private:
RtmpCaptureDataCallback* callback_;
short m_port;
SimpleSocketStream * m_pSocket;
};
定義文件
#include "rtmpsrv.h"
#include "rtmp_input_device.h"
RtmpServer::RtmpServer( )
{
callback_ = NULL;
m_port = 1935;
m_pSocket = NULL;
}
RtmpServer::~RtmpServer()
{
return;
}
int RtmpServer::fmle_publish(SrsRtmpServer* rtmp, SrsRequest* req )
{
if (NULL == rtmp || NULL == req )
{
return -1;
}
int ret = ERROR_SUCCESS;
bool hasHead = false, bVFirst = true, bAFirst = true;
char head[128] = {0};
char commonHead[4] = {0x00, 0x00, 0x00, 0x01};
string video_data, audio_data;
int headLen = 0;
uint64_t audio_last_recv_time(0), video_last_recv_time(0);
while ( false == get_terminated() )
{
Sleep(5);
SrsMessage* msg = NULL;
if ((ret = rtmp->recv_message(&msg)) != ERROR_SUCCESS)
{
break;
}
SrsAutoFree(SrsMessage, msg);
if (msg->header.is_amf0_command() || msg->header.is_amf3_command())
{
SrsPacket* pkt = NULL;
if ((ret = rtmp->decode_message(msg, &pkt)) != ERROR_SUCCESS) {
break;
}
SrsAutoFree(SrsPacket, pkt);
if (dynamic_cast<SrsFMLEStartPacket*>(pkt)) {
SrsFMLEStartPacket* unpublish = dynamic_cast<SrsFMLEStartPacket*>(pkt);
if ((ret = rtmp->fmle_unpublish(10, unpublish->transaction_id)) != ERROR_SUCCESS) {
break;
}
break;
}
continue;
}
if (msg->header.is_audio())
{
audio_data.clear();
if (bAFirst)
{
audio_data.append((char *)msg->payload + 1, msg->size - 1);
bAFirst = false;
}else
audio_data.append((char *)msg->payload, msg->size -1);
}else if (msg->header.is_video())
{
if (!hasHead)
{
memcpy(head, commonHead, 4);
int8_t *skip = msg->payload;
int8_t *skip2 = msg->payload;
while(*(skip++) != 0x67);
while(*(skip2++) != 0x68);
int diff = skip2 - skip;
if (diff <= 0)
{
continue;
}
memcpy(head + 4, skip - 1, diff - 4); //copy sps
memcpy(head + 4 + diff - 4, commonHead, 4);
memcpy(head + 4 + diff - 4 + 4, skip2 - 1, 4); //copy pps
hasHead = true;
headLen = 4 + diff - 4 + 4 + 4;
}else
{
video_data.clear();
if (bVFirst)
{
video_last_recv_time = msg->header.timestamp;
video_data.append(head, 128);
video_data.append((const char *)msg->payload+9, msg->size - 9);
bVFirst = false;
} else
{
if (msg->header.timestamp > video_last_recv_time)
{
video_last_recv_time = msg->header.timestamp;
}
if (msg->payload[0] == 0x17) //I frame
{
video_data.append(head, headLen);
}
video_data.append(commonHead + 1, 3);
video_data.append((const char *)msg->payload+9, msg->size - 9);
}
}
}
}
return ret;
}
void RtmpServer::execute( )
{
if (NULL == m_pSocket)
{
return ;
}
m_pSocket->listen("0.0.0.0", m_port, 10);
SimpleSocketStream* server = new SimpleSocketStream;
while ( false == get_terminated() )
{
if (NULL == server)
{
SimpleSocketStream* server = new SimpleSocketStream;
}
if (NULL != m_pSocket && 0 == m_pSocket->accept(*server))
{
AUDIO_INFO("RtmpServer thread accept obs connected!");
SrsRtmpServer* rtmp = new SrsRtmpServer(server);
SrsRequest* req = new SrsRequest();
int ret = ERROR_SUCCESS;
if ((ret = rtmp->handshake()) != ERROR_SUCCESS) {
break;
}
SrsMessage* msg = NULL;
SrsConnectAppPacket* pkt = NULL;
if ((ret = srs_rtmp_expect_message<SrsConnectAppPacket>(rtmp->get_protocol(), &msg, &pkt)) != ERROR_SUCCESS) {
break;
}
if ((ret = rtmp->response_connect_app(req, 0)) != ERROR_SUCCESS) {
break;
}
while ( false == get_terminated() ) {
SrsRtmpConnType type;
if ((ret = rtmp->identify_client(10, type, req->stream, req->duration)) != ERROR_SUCCESS){
terminate();
// RtmpServer thread peer shutdown
break;
}
assert(type == SrsRtmpConnFMLEPublish);
req->strip();
rtmp->start_fmle_publish(10);
int ret = fmle_publish( rtmp, req );
if ((ret != ERROR_CONTROL_REPUBLISH) && (ret != ERROR_CONTROL_RTMP_CLOSE)) //OBS主動停止串流
{
break;
}
}
if (NULL != rtmp)
{
delete rtmp;
rtmp = NULL;
}
if(NULL != req)
{
delete req;
req = NULL;
}
if (NULL != server)
{
server->close_socket();
}
}else
{
Sleep(5);
}
}
if (NULL != server)
{
delete server;
server = NULL;
}
}
int RtmpServer::StartRtmpServer( )
{
InitSockets();
m_port = 1935;
if (NULL != m_pSocket)
{
m_pSocket->close_socket();
delete m_pSocket;
m_pSocket = NULL;
}
m_pSocket = new SimpleSocketStream;
m_pSocket->create_socket();
return start();
}
void RtmpServer::StopRtmpServer( )
{
terminate();
if (NULL != m_pSocket)
{
m_pSocket->close_socket();
delete m_pSocket;
m_pSocket = NULL;
}
CleanupSockets();
};
int32_t RtmpServer::terminate()
{
terminated_ = true;
if(thr_handle_ != NULL)
{
if(WAIT_TIMEOUT == WaitForSingleObject(thr_handle_, 100))
{
TerminateThread(thr_handle_, -1);
}
thr_handle_ = NULL;
}
return 0;
}
#endif最後在OBS廣播設定 伺服器那裏填入rtmp://ip:port