技术背景
我们在做Android平台RTSP转发模块的时候，有公司提出来这样的技术需求，他们希望拉取外部RTSP摄像头的流，然后提供个轻量级RTSP服务，让内网其他终端过来拉流。实际上，这块，大牛直播SDK前几年就已经实现。
技术实现
拉流的话，很好理解，其实就是播放端，把未解码的数据，直接回调上来，如果需要预览，直接底层绘制即可。单纯的数据回调，对性能消耗不大。
回调上来的数据，可以作为轻量级RTSP服务的数据源（投递编码后数据），推送端，只要启动RTSP服务，然后发布RTSP流即可。
先说拉流，开始拉流、停止拉流实现：
/* * SmartPlayer.java * Author: daniusdk.com */private boolean StartPull(){if ( isPulling )return false;if(!isPlaying){if (!OpenPullHandle())return false;}libPlayer.SmartPlayerSetAudioDataCallback(player_handle_, new PlayerAudioDataCallback(stream_publisher_));libPlayer.SmartPlayerSetVideoDataCallback(player_handle_, new PlayerVideoDataCallback(stream_publisher_));int is_pull_trans_code = 1;libPlayer.SmartPlayerSetPullStreamAudioTranscodeAAC(player_handle_, is_pull_trans_code);int startRet = libPlayer.SmartPlayerStartPullStream(player_handle_);if (startRet != 0) {Log.e(TAG, "Failed to start pull stream!");if(!isPlaying){releasePlayerHandle();}return false;}isPulling = true;return true;}private void StopPull(){if ( !isPulling )return;isPulling = false;if (null == libPlayer || 0 == player_handle_)return;libPlayer.SmartPlayerStopPullStream(player_handle_);if ( !isPlaying){releasePlayerHandle();}}音频回调处理：
class PlayerAudioDataCallback implements NTAudioDataCallback{private WeakReference<LibPublisherWrapper> publisher_;private int audio_buffer_size = 0;private int param_info_size = 0;private ByteBuffer audio_buffer_ = null;private ByteBuffer parameter_info_ = null;public PlayerAudioDataCallback(LibPublisherWrapper publisher) {if (publisher != null)publisher_ = new WeakReference<>(publisher);}@Overridepublic ByteBuffer getAudioByteBuffer(int size){//Log.i("getAudioByteBuffer", "size: " + size);if( size < 1 ){return null;}if ( size <= audio_buffer_size && audio_buffer_ != null ){return audio_buffer_;}audio_buffer_size = size + 512;audio_buffer_size = (audio_buffer_size+0xf) & (~0xf);audio_buffer_ = ByteBuffer.allocateDirect(audio_buffer_size);// Log.i("getAudioByteBuffer", "size: " + size + " buffer_size:" + audio_buffer_size);return audio_buffer_;}@Overridepublic ByteBuffer getAudioParameterInfo(int size){//Log.i("getAudioParameterInfo", "size: " + size);if(size < 1){return null;}if ( size <= param_info_size && parameter_info_ != null ){return parameter_info_;}param_info_size = size + 32;param_info_size = (param_info_size+0xf) & (~0xf);parameter_info_ = ByteBuffer.allocateDirect(param_info_size);//Log.i("getAudioParameterInfo", "size: " + size + " buffer_size:" + param_info_size);return parameter_info_;}public void onAudioDataCallback(int ret, int audio_codec_id, int sample_size, int is_key_frame, long timestamp, int sample_rate, int channel, int parameter_info_size, long reserve){//Log.i("onAudioDataCallback", "ret: " + ret + ", audio_codec_id: " + audio_codec_id + ", sample_size: " + sample_size + ", timestamp: " + timestamp +//",sample_rate:" + sample_rate);if ( audio_buffer_ == null)return;LibPublisherWrapper publisher = publisher_.get();if (null == publisher)return;if (!publisher.is_publishing())return;audio_buffer_.rewind();publisher.PostAudioEncodedData(audio_codec_id, audio_buffer_, sample_size, is_key_frame, timestamp, parameter_info_, parameter_info_size);}}视频回调数据：
class PlayerVideoDataCallback implements NTVideoDataCallback{private WeakReference<LibPublisherWrapper> publisher_;private int video_buffer_size = 0;private ByteBuffer video_buffer_ = null;public PlayerVideoDataCallback(LibPublisherWrapper publisher) {if (publisher != null)publisher_ = new WeakReference<>(publisher);}@Overridepublic ByteBuffer getVideoByteBuffer(int size){//Log.i("getVideoByteBuffer", "size: " + size);if( size < 1 ){return null;}if ( size <= video_buffer_size && video_buffer_ != null ){return video_buffer_;}video_buffer_size = size + 1024;video_buffer_size = (video_buffer_size+0xf) & (~0xf);video_buffer_ = ByteBuffer.allocateDirect(video_buffer_size);// Log.i("getVideoByteBuffer", "size: " + size + " buffer_size:" + video_buffer_size);return video_buffer_;}public void onVideoDataCallback(int ret, int video_codec_id, int sample_size, int is_key_frame, long timestamp, int width, int height, long presentation_timestamp){//Log.i("onVideoDataCallback", "ret: " + ret + ", video_codec_id: " + video_codec_id + ", sample_size: " + sample_size + ", is_key_frame: "+ is_key_frame + ", timestamp: " + timestamp +//",presentation_timestamp:" + presentation_timestamp);if ( video_buffer_ == null)return;LibPublisherWrapper publisher = publisher_.get();if (null == publisher)return;if (!publisher.is_publishing())return;video_buffer_.rewind();publisher.PostVideoEncodedData(video_codec_id, video_buffer_, sample_size, is_key_frame, timestamp, presentation_timestamp);}}启动RTSP服务：
//启动/停止RTSP服务class ButtonRtspServiceListener implements View.OnClickListener {public void onClick(View v) {if (isRTSPServiceRunning) {stopRtspService();btnRtspService.setText("启动RTSP服务");btnRtspPublisher.setEnabled(false);isRTSPServiceRunning = false;return;}Log.i(TAG, "onClick start rtsp service..");rtsp_handle_ = libPublisher.OpenRtspServer(0);if (rtsp_handle_ == 0) {Log.e(TAG, "创建rtsp server实例失败! 请检查SDK有效性");} else {int port = 28554;if (libPublisher.SetRtspServerPort(rtsp_handle_, port) != 0) {libPublisher.CloseRtspServer(rtsp_handle_);rtsp_handle_ = 0;Log.e(TAG, "创建rtsp server端口失败! 请检查端口是否重复或者端口不在范围内!");}if (libPublisher.StartRtspServer(rtsp_handle_, 0) == 0) {Log.i(TAG, "启动rtsp server 成功!");} else {libPublisher.CloseRtspServer(rtsp_handle_);rtsp_handle_ = 0;Log.e(TAG, "启动rtsp server失败! 请检查设置的端口是否被占用!");}btnRtspService.setText("停止RTSP服务");btnRtspPublisher.setEnabled(true);isRTSPServiceRunning = true;}}}发布RTSP流：
//发布/停止RTSP流class ButtonRtspPublisherListener implements View.OnClickListener {public void onClick(View v) {if (stream_publisher_.is_rtsp_publishing()) {stopRtspPublisher();btnRtspPublisher.setText("发布RTSP流");btnGetRtspSessionNumbers.setEnabled(false);btnRtspService.setEnabled(true);return;}Log.i(TAG, "onClick start rtsp publisher..");InitAndSetConfig();String rtsp_stream_name = "stream1";stream_publisher_.SetRtspStreamName(rtsp_stream_name);stream_publisher_.ClearRtspStreamServer();stream_publisher_.AddRtspStreamServer(rtsp_handle_);if (!stream_publisher_.StartRtspStream()) {stream_publisher_.try_release();Log.e(TAG, "调用发布rtsp流接口失败!");return;}btnRtspPublisher.setText("停止RTSP流");btnGetRtspSessionNumbers.setEnabled(true);btnRtspService.setEnabled(false);}}获取RTSP Session会话数：
//获取RTSP会话数class ButtonGetRtspSessionNumbersListener implements View.OnClickListener {public void onClick(View v) {if (libPublisher != null && rtsp_handle_ != 0) {int session_numbers = libPublisher.GetRtspServerClientSessionNumbers(rtsp_handle_);Log.i(TAG, "GetRtspSessionNumbers: " + session_numbers);PopRtspSessionNumberDialog(session_numbers);}}}总结
因为RTSP外部拉流，不需要解码，配合大牛直播SDK的SmartPlayer播放器，延迟和直连的，差别不大，整体毫秒级，延迟非常低，巡检或监控类场景，都可以达到相应的技术指标。如果需要二次水印，也可以回调解码后的yuv或rgb数据，推送端添加二次文字或图片水印后，编码输出，这种在一些合成类场景，比如智慧煤矿、管廊隧道等行业，非常适用，感兴趣的开发者，可以单独跟我探讨。
