RTSP流的几个去处
写了很多关于RTSP播放和转发的blog了，今天我们做个简单的汇总，以大牛直播SDK的Android平台为例，拉取到RTSP流，除了本地播放，还有几个流向：
功能实现：
**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，同时也可以转发到轻量级RTSP服务和GB28181平台；
**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。
先说拉取数据，拉取RTSP流的时候，设置音视频数据回调。
ini
/*
* SmartRelayDemo.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;
}对应的OpenPullHandle()实现如下：
ini
private boolean OpenPullHandle()
{
//playbackUrl可自定义
//playbackUrl = "rtsp://admin:daniulive12345@192.168.0.120:554/h264/ch1/main/av_stream";
if (playbackUrl == null) {
Log.e(TAG, "playback URL is null...");
return false;
}
player_handle_ = libPlayer.SmartPlayerOpen(context_);
if (player_handle_ == 0) {
Log.e(TAG, "playerHandle is null..");
return false;
}
libPlayer.SetSmartPlayerEventCallbackV2(player_handle_,
new EventHandlePlayerV2());
libPlayer.SmartPlayerSetBuffer(player_handle_, playBuffer);
// set report download speed
libPlayer.SmartPlayerSetReportDownloadSpeed(player_handle_, 1, 3);
//设置RTSP超时时间
int rtsp_timeout = 10;
libPlayer.SmartPlayerSetRTSPTimeout(player_handle_, rtsp_timeout);
//设置RTSP TCP/UDP模式自动切换
int is_auto_switch_tcp_udp = 1;
libPlayer.SmartPlayerSetRTSPAutoSwitchTcpUdp(player_handle_, is_auto_switch_tcp_udp);
// It only used when playback RTSP stream..
//libPlayer.SmartPlayerSetRTSPTcpMode(playerHandle, 1);
libPlayer.SmartPlayerSetUrl(player_handle_, playbackUrl);
return true;
}音频处理如下：
ini
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);
}
}视频处理如下：
java
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);
}
}本地播放
ini
private boolean StartPlay()
{
if(isPlaying)
return false;
if(!isPulling)
{
if (!OpenPullHandle())
return false;
}
// 如果第二个参数设置为null，则播放纯音频
libPlayer.SmartPlayerSetSurface(player_handle_, sSurfaceView);
//libPlayer.SmartPlayerSetSurface(player_handle_, null);
libPlayer.SmartPlayerSetRenderScaleMode(player_handle_, 1);
libPlayer.SmartPlayerSetFastStartup(player_handle_, isFastStartup ? 1 : 0);
libPlayer.SmartPlayerSetAudioOutputType(player_handle_, 1);
if (isMute) {
libPlayer.SmartPlayerSetMute(player_handle_, isMute ? 1 : 0);
}
if (isHardwareDecoder)
{
int isSupportH264HwDecoder = libPlayer.SetSmartPlayerVideoHWDecoder(player_handle_, 1);
int isSupportHevcHwDecoder = libPlayer.SetSmartPlayerVideoHevcHWDecoder(player_handle_, 1);
Log.i(TAG, "isSupportH264HwDecoder: " + isSupportH264HwDecoder + ", isSupportHevcHwDecoder: " + isSupportHevcHwDecoder);
}
libPlayer.SmartPlayerSetLowLatencyMode(player_handle_, isLowLatency ? 1 : 0);
libPlayer.SmartPlayerSetRotation(player_handle_, rotate_degrees);
int iPlaybackRet = libPlayer.SmartPlayerStartPlay(player_handle_);
if (iPlaybackRet != 0 && !isPulling) {
Log.e(TAG, "StartPlay failed!");
releasePlayerHandle();
return false;
}
isPlaying = true;
return true;
}
private void StopPlay()
{
if ( !isPlaying )
return;
isPlaying = false;
if (null == libPlayer || 0 == player_handle_)
return;
libPlayer.SmartPlayerStopPlay(player_handle_);
}
转推RTMP
scss
btnRTMPPusher.setOnClickListener(new Button.OnClickListener() {
// @Override
public void onClick(View v) {
if (stream_publisher_.is_rtmp_publishing()) {
stopPush();
btnRTMPPusher.setText("推送RTMP");
return;
}
Log.i(TAG, "onClick start push rtmp..");
InitAndSetConfig();
//relayStreamUrl = "rtmp://192.168.0.108:1935/hls/stream1";
if (!stream_publisher_.SetURL(relayStreamUrl))
Log.e(TAG, "Failed to set publish stream URL..");
boolean start_ret = stream_publisher_.StartPublisher();
if (!start_ret) {
stream_publisher_.try_release();
Log.e(TAG, "Failed to start push stream..");
return;
}
btnRTMPPusher.setText("停止推送");
}
});转推轻量级RTSP服务
ini
//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/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);
}
}
//当前RTSP会话数弹出框
private void PopRtspSessionNumberDialog(int session_numbers) {
final EditText inputUrlTxt = new EditText(this);
inputUrlTxt.setFocusable(true);
inputUrlTxt.setEnabled(false);
String session_numbers_tag = "RTSP服务当前客户会话数: " + session_numbers;
inputUrlTxt.setText(session_numbers_tag);
AlertDialog.Builder builderUrl = new AlertDialog.Builder(this);
builderUrl
.setTitle("内置RTSP服务")
.setView(inputUrlTxt).setNegativeButton("确定", null);
builderUrl.show();
}
//获取RTSP会话数
class ButtonGetRtspSessionNumbersListener implements 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);
}
}
};拉取RTSP后录像
scss
btnStartStopRecorder.setOnClickListener(new Button.OnClickListener() {
// @Override
public void onClick(View v) {
if (isRecording)
{
StopRecorder();
btnStartStopRecorder.setText(" 开始录像");
} else {
Log.i(TAG, "onClick StartRecorder..");
boolean startRet = StartRecorder();
if (!startRet) {
Log.e(TAG, "Failed to call StartRecorder().");
return;
}
btnStartStopRecorder.setText("停止录像");
}
}
});转推GB28181平台
scss
class ButtonGB28181AgentListener implements OnClickListener {
public void onClick(View v) {
stopGB28181Stream();
destoryRTPSender();
if (null == gb28181_agent_ ) {
if( !initGB28181Agent() )
return;
}
if (gb28181_agent_.isRunning()) {
gb28181_agent_.terminateAllPlays(true);// 目前测试下来,发送BYE之后,有些服务器会立即发送INVITE,是否发送BYE根据实际情况看
gb28181_agent_.stop();
btnGB28181Agent.setText("启动GB28181");
}
else {
if ( gb28181_agent_.start() ) {
btnGB28181Agent.setText("停止GB28181");
}
}
}
}
//停止GB28181 媒体流
private void stopGB28181Stream() {
stream_publisher_.StopGB28181MediaStream();
stream_publisher_.try_release();
}
总结
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，如果可以跨平台，还能以SDK形式提供，会给开发者提供更大的便利！Android平台拉取RTSP流后，有了数据源，开发者可以在一个推送实例中，转推到不同的业务场景，实现高效率低延迟的数据转发。