如何实现多路海康大华等RTSP数据转RTMP推送

一个好的转发模块，首先要低延迟！其次足够稳定、灵活、有状态反馈机制、资源占用低，跨平台，最好以接口形式提供，便于第三方系统集成。

以Windows平台为例，我们的考虑的点如下

1. 拉流：通过RTSP直播播放SDK的数据回调接口，拿到音视频数据；

2. 转推：通过RTMP直播推送SDK的编码后数据输入接口，把回调上来的数据，传给RTMP直播推送模块，实现RTSP数据流到RTMP服务器的转发；

3. 录像：如果需要录像，借助RTSP直播播放SDK，拉到音视频数据后，直接存储MP4文件即可；

4. 快照：如果需要实时快照，拉流后，解码调用播放端快照接口，生成快照，因为快照涉及到video数据解码，如无必要，可不必开启，不然会额外消耗性能。

5. 拉流预览：如需预览拉流数据，只要调用播放端的播放接口，即可实现拉流数据预览；

6. 数据转AAC后转发：考虑到好多监控设备出来的音频可能是PCMA/PCMU的，如需要更通用的音频格式，可以转AAC后，在通过RTMP推送；

7. 转推RTMP实时静音：只需要在传audio数据的地方，加个判断即可；

8. 拉流速度反馈：通过RTSP播放端的实时码率反馈event，拿到实时带宽占用即可；

9. 整体网络状态反馈：考虑到有些摄像头可能会临时或异常关闭，RTMP服务器亦是，可以通过推拉流的event回调状态，查看那整体网络情况，如此界定：是拉不到流，还是推不到RTMP服务器。

系统设计架构图

Windows转发demo分析

大牛直播SDK的转发demo，Windows平台，对应C++ demo工程：WIN-RelaySDK-CPP-Demo，如需下载demo源码，参看 Github

1. 拉流：拉流和播放有些类似，但不需要播放（也就是说不要解码，资源消耗非常低），在做过基础的参数配置之后（对应demo里面OpenPullHandle()），设置音视频数据回调，然后调用StartPullStream()即可：

1.1 基础参数设置：

bool nt_stream_relay_wrapper::OpenPullHandle(const std::string& url, bool is_rtsp_tcp_mode, bool is_mute)
{if ( pull_handle_ != NULL )return true;if ( url.empty() )return false;duration_ = 0;NT_HANDLE pull_handle = NULL;ASSERT( pull_api_ != NULL );if (NT_ERC_OK != pull_api_->Open(&pull_handle, render_wnd_, 0, NULL)){return false;}ASSERT(pull_handle != NULL);pull_api_->SetEventCallBack(pull_handle, this, &NT_Pull_SDKEventHandle);pull_api_->SetBuffer(pull_handle, 0);pull_api_->SetFastStartup(pull_handle, 1);pull_api_->SetRTSPTcpMode(pull_handle, is_rtsp_tcp_mode ? 1 : 0);pull_api_->SetMute(pull_handle, is_mute ? 1 : 0);if ( NT_ERC_OK != pull_api_->SetURL(pull_handle, url.c_str()) ){pull_api_->Close(pull_handle);pull_handle = NULL;return false;}if ( setting_pos_ >= 0ll ){pull_api_->SetPos(pull_handle, setting_pos_);}pull_handle_ = pull_handle;return true;
}

1.2 设置音视频数据回调：

 pull_api_->SetPullStreamVideoDataCallBack(pull_handle_, this, &SP_SDKPullStreamVideoDataHandle);pull_api_->SetPullStreamAudioDataCallBack(pull_handle_, this, &SP_SDKPullStreamAudioDataHandle);

1.3 开始拉流：

 auto ret = pull_api_->StartPullStream(pull_handle_);if ( NT_ERC_OK != ret ){if ( !is_playing_ ){pull_api_->Close(pull_handle_);pull_handle_ = NULL;}return false;}

拉流整体代码如下：

bool nt_stream_relay_wrapper::StartPull(const std::string& url, bool is_rtsp_tcp_mode, bool is_transcode_aac)
{if ( is_pulling_ )return false;if ( !OpenPullHandle(url, is_rtsp_tcp_mode) )return false;pull_api_->SetPullStreamVideoDataCallBack(pull_handle_, this, &SP_SDKPullStreamVideoDataHandle);pull_api_->SetPullStreamAudioDataCallBack(pull_handle_, this, &SP_SDKPullStreamAudioDataHandle);pull_api_->SetPullStreamAudioTranscodeAAC(pull_handle_, is_transcode_aac? 1: 0);auto ret = pull_api_->StartPullStream(pull_handle_);if ( NT_ERC_OK != ret ){if ( !is_playing_ ){pull_api_->Close(pull_handle_);pull_handle_ = NULL;}return false;}is_pulling_ = true;return true;
}

2. 停止拉流：

停止拉流流程比较简单，先判断是否在拉流状态，如果拉流，调用StopPullStream() 即可，如没有预览画面，调用Close()接口关闭拉流实例。

void nt_stream_relay_wrapper::StopPull()
{if ( !is_pulling_ )return;pull_api_->StopPullStream(pull_handle_);if ( !is_playing_ ){pull_api_->Close(pull_handle_);pull_handle_ = NULL;}is_pulling_ = false;
}

3. 拉流端预览：

拉流端预览，说白了就是播放拉流数据，流程比较简单，demo调用如下，如不需要播放声音，调用SetMute()，实时打开/关闭即可：

bool nt_stream_relay_wrapper::StartPlay(const std::string& url, bool is_rtsp_tcp_mode, bool is_mute)
{if ( is_playing_ )return false;if ( !OpenPullHandle(url, is_rtsp_tcp_mode, is_mute) )return false;pull_api_->SetMute(pull_handle_, is_mute ? 1 : 0);auto ret = pull_api_->StartPlay(pull_handle_);if ( NT_ERC_OK != ret ){if ( !is_pulling_ ){pull_api_->Close(pull_handle_);pull_handle_ = NULL;}return false;}is_playing_ = true;return true;
}

4. 拉流端关闭预览：

void nt_stream_relay_wrapper::StopPlay()
{if ( !is_playing_ )return;pull_api_->StopPlay(pull_handle_);if ( !is_pulling_ ){pull_api_->Close(pull_handle_);pull_handle_ = NULL;}is_playing_ = false;
}

5. 开始推流到RTMP服务器：

推流的流程，如之前所述，调用RTMP推送模块，然后数据源传编码后的音视频数据即可，下图的demo源码，同时展示了，RTSP流获取到后，转推RTMP的时候，数据解密的处理：

bool nt_stream_relay_wrapper::StartPush(const std::string& url)
{if ( is_pushing_ )return false;if ( url.empty() )return false;if ( !OpenPushHandle() )return false;auto push_handle = GetPushHandle();ASSERT(push_handle != nullptr);ASSERT(push_api_ != NULL);if ( NT_ERC_OK != push_api_->SetURL(push_handle, url.c_str(), NULL) ){if ( !is_started_rtsp_stream_ ){push_api_->Close(push_handle);SetPushHandle(nullptr);}return false;}// 加密测试 +++// push_api_->SetRtmpEncryptionOption(push_handle, url.c_str(), 1, 1);// NT_BYTE test_key[16] = {'1', '2', '3'};// push_api_->SetRtmpEncryptionKey(push_handle, url.c_str(), test_key, 16);// 加密测试 --if ( NT_ERC_OK != push_api_->StartPublisher(push_handle, NULL) ){if ( !is_started_rtsp_stream_ ){push_api_->Close(push_handle);SetPushHandle(nullptr);}return false;}// // test push rtsp ++// push_api_->SetPushRtspTransportProtocol(push_handle, 1);// // push_api_->SetPushRtspTransportProtocol(push_handle, 2);// push_api_->SetPushRtspURL(push_handle, "rtsp://player.daniulive.com:554/liverelay111.sdp");// push_api_->StartPushRtsp(push_handle, 0);// // test push rtsp--is_pushing_ = true;return true;
}

6. 传递转推RTMP数据：

void nt_stream_relay_wrapper::OnVideoDataHandle(NT_HANDLE handle, NT_UINT32 video_codec_id, NT_BYTE* data, NT_UINT32 size, NT_SP_PullStreamVideoDataInfo* info)
{if (!is_pushing_ && !is_started_rtsp_stream_)return;if ( pull_handle_ != handle )return;if (data == NULL)return;if (size < 1)return;if (info == NULL)return;std::unique_lock<std::recursive_mutex> lock(push_handle_mutex_);if (!is_pushing_ && !is_started_rtsp_stream_)return;if (push_handle_ == NULL)return;push_api_->PostVideoEncodedDataV2(push_handle_, video_codec_id,data, size, info->is_key_frame_, info->timestamp_, info->presentation_timestamp_);
}void nt_stream_relay_wrapper::OnAudioDataHandle(NT_HANDLE handle, NT_UINT32 auido_codec_id,NT_BYTE* data, NT_UINT32 size, NT_SP_PullStreamAuidoDataInfo* info)
{if (!is_pushing_ && !is_started_rtsp_stream_)return;if (pull_handle_ != handle)return;if (data == NULL)return;if (size < 1)return;if (info == NULL)return;std::unique_lock<std::recursive_mutex> lock(push_handle_mutex_);if (!is_pushing_ && !is_started_rtsp_stream_)return;if (push_handle_ == NULL)return;push_api_->PostAudioEncodedData(push_handle_, auido_codec_id, data, size,info->is_key_frame_, info->timestamp_, info->parameter_info_, info->parameter_info_size_);
}

7. 关闭实时RTMP转推

void nt_stream_relay_wrapper::StopPush()
{if ( !is_pushing_ )return;is_pushing_ = false;std::unique_lock<std::recursive_mutex> lock(push_handle_mutex_);if ( nullptr == push_handle_ )return;push_api_->StopPublisher(push_handle_);// // test push rtsp ++// push_api_->StopPushRtsp(push_handle_);// // test push rtsp--if ( !is_started_rtsp_stream_ ){push_api_->Close(push_handle_);push_handle_ = nullptr;}
}

以上就是RTSP或RTMP流转RTMP推送的流程，感兴趣的开发者，可做设计参考。