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);
}
@Override
public 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_;
}
@Override
public 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);
}
@Override
public 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。
