技術背景
我們在做Android平臺RTSP轉發模組的時候，有公司提出來這樣的技術需求，他們希望拉取外部RTSP攝像頭的流，然後提供個輕量級RTSP服務，讓內網其他終端過來拉流。實際上，這塊，大牛直播SDK前幾年就已經實現。
技術實現
拉流的話，很好理解，其實就是播放端，把未解碼的資料，直接回調上來，如果需要預覽，直接底層繪製即可。單純的資料回撥，對效能消耗不大。
回撥上來的資料，可以作為輕量級RTSP服務的資料來源（投遞編碼後資料），推送端，只要啟動RTSP服務，然後釋出RTSP流即可。
先說拉流，開始拉流、停止拉流實現：
/*
* 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();
}
}
音訊回撥處理：
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);
}
}
影片回撥資料：
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);
}
}
啟動RTSP服務：
//啟動/停止RTSP服務
class ButtonRtspServiceListener implements View.OnClickListener {
public void onClick(View v) {
if (isRTSPServiceRunning) {
stopRtspService();
btnRtspService.setText("啟動RTSP服務");
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, "建立rtsp server例項失敗! 請檢查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, "建立rtsp server埠失敗! 請檢查埠是否重複或者埠不在範圍內!");
}
if (libPublisher.StartRtspServer(rtsp_handle_, 0) == 0) {
Log.i(TAG, "啟動rtsp server 成功!");
} else {
libPublisher.CloseRtspServer(rtsp_handle_);
rtsp_handle_ = 0;
Log.e(TAG, "啟動rtsp server失敗! 請檢查設定的埠是否被佔用!");
}
btnRtspService.setText("停止RTSP服務");
btnRtspPublisher.setEnabled(true);
isRTSPServiceRunning = true;
}
}
}
釋出RTSP流：
//釋出/停止RTSP流
class ButtonRtspPublisherListener implements View.OnClickListener {
public void onClick(View v) {
if (stream_publisher_.is_rtsp_publishing()) {
stopRtspPublisher();
btnRtspPublisher.setText("釋出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, "呼叫釋出rtsp流介面失敗!");
return;
}
btnRtspPublisher.setText("停止RTSP流");
btnGetRtspSessionNumbers.setEnabled(true);
btnRtspService.setEnabled(false);
}
}
獲取RTSP Session會話數：
//獲取RTSP會話數
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);
}
}
}
總結
因為RTSP外部拉流，不需要解碼，配合大牛直播SDK的SmartPlayer播放器，延遲和直連的，差別不大，整體毫秒級，延遲非常低，巡檢或監控類場景，都可以達到相應的技術指標。如果需要二次水印，也可以回撥解碼後的yuv或rgb資料，推送端新增二次文字或圖片水印後，編碼輸出，這種在一些合成類場景，比如智慧煤礦、管廊隧道等行業，非常適用，感興趣的開發者，可以單獨跟我探討。
