对于我的应用程序，我必须从decklink卡流到Android应用程序(我必须是一个直播流，所以HLS或RTSP似乎是很好的解决方案，因为我的应用程序针对Android 3+)。我用decklink sdk重新编译了VLC，我能够通过网络直播流到另一台pc(但它只工作60秒与RTSP)。
这是我尝试的:
-
HTTP流:
./vlc -vvv decklink:// --sout '#transcode{vcodec=mp4v,acodec=mpga,vb=56,ab=24,channels=1} :standard{access=http{use-key-frames},mux=ts,dst=:3001/stream.mpeg}'
它在Android VLC 0.0.11中工作，但只能在WiFi中工作，不能在3G中工作。我无法用VideoView在应用中播放它。下面是我使用的代码和相应的错误消息:
String url = "http://134.246.63.169:5554/stream.mpeg";
VideoView videoView = (VideoView) this.findViewById(R.id.videoView);
videoView.setVideoURI(Uri.parse(url));
videoView.setMediaController(new MediaController(this));
videoView.requestFocus();
videoView.start();
错误信息:
04-08 15:26:46.272: D/MediaPlayer(16349): Couldn't open file on client side, trying server side
04-08 15:26:46.272: V/ChromiumHTTPDataSource(7680): connect on behalf of uid 1080867789
04-08 15:26:46.272: I/ChromiumHTTPDataSource(7680): connect to http://134.246.63.169:8554/ @0
04-08 15:26:46.302: I/AwesomePlayer(7680): AwesomePlayer::AwesomePlayer()in
04-08 15:26:46.302: I/AwesomePlayer(7680): AwesomePlayer::AwesomePlayer()aftermClient.connect()
04-08 15:26:46.302: I/AwesomePlayer(7680): setDataSource_l('http://134.246.63.169:5554/')
04-08 15:26:46.302: W/MediaPlayer(16349): info/warning (701, 0)
04-08 15:26:46.302: V/ChromiumHTTPDataSource(7680): connect on behalf of uid 10067
04-08 15:26:46.302: I/ChromiumHTTPDataSource(7680): connect to http://134.246.63.169:5554/ @0
04-08 15:26:46.342: I/ActivityManager(272): Displayed fr.ifremer.testrtsp/.MainActivity: +183ms
04-08 15:26:46.382: I/MediaPlayer(16349): Info (701,0)
04-08 15:27:07.592: E/MediaPlayer(16349): error (1, -2147483648)
04-08 15:27:07.592: E/MediaPlayer(16349): Error (1,-2147483648)
- RTSP:
我使用了谷歌推荐的编码选项，例如:
- 视频编解码:h264
- 音频编码:AAC
- 视频比特率:56
- 音频比特率:24
- 音频通道:1
./vlc -vvv decklink:// --sout-ffmpeg-strict=-2 --sout
'#transcode{width=176,height=144,vcodec=h264,acodec=mp4a,vb=56,ab=24,channels=1}
:rtp{dst=134.246.63.169,port-video=5554,port-audio=5556,sdp=rtsp://134.246.63.169:5554/stream.sdp}'
我能够在VLC桌面播放流，但不是在Android(甚至在Android VLC版本或默认的谷歌视频播放器:/)。如果我不指定混音器，我也可以在QuickTime播放(如果我指定混音器，ts或ps，我没有视频。如果我尝试另一个muxer, VLC告诉我，我只允许在RTP中使用ts或ps)
如果我尝试使用谷歌视频播放器，我在本地得到这些消息:
04-08 15:32:45.792: D/MediaPlayer(13688): Couldn't open file on client side, trying server side
04-08 15:32:45.802: W/MediaPlayer(13688): info/warning (701, 0)
04-08 15:32:45.812: I/MediaPlayer(13688): Info (701,0)
04-08 15:32:45.812: D/MediaPlayer(13688): getMetadata
04-08 15:32:45.812: E/MediaPlayerService(7680): getMetadata failed -38
04-08 15:32:45.852: I/MyHandler(7680): connection request completed with result 0 (Success)
04-08 15:32:45.882: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK
04-08 15:32:45.882: I/MyHandler(7680): DESCRIBE completed with result 0 (Success)
04-08 15:32:45.882: I/ASessionDescription(7680): v=0
04-08 15:32:45.882: I/ASessionDescription(7680): o=- 15352003113363922923 15352003113363922923 IN IP4 to63-169.ifremer.fr
04-08 15:32:45.882: I/ASessionDescription(7680): s=Unnamed
04-08 15:32:45.882: I/ASessionDescription(7680): i=N/A
04-08 15:32:45.882: I/ASessionDescription(7680): c=IN IP4 134.246.63.169
04-08 15:32:45.882: I/ASessionDescription(7680): t=0 0
04-08 15:32:45.882: I/ASessionDescription(7680): a=tool:vlc 2.0.5
04-08 15:32:45.882: I/ASessionDescription(7680): a=recvonly
04-08 15:32:45.882: I/ASessionDescription(7680): a=type:broadcast
04-08 15:32:45.882: I/ASessionDescription(7680): a=charset:UTF-8
04-08 15:32:45.882: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp
04-08 15:32:45.882: I/ASessionDescription(7680): m=audio 5556 RTP/AVP 96
04-08 15:32:45.882: I/ASessionDescription(7680): b=AS:24
04-08 15:32:45.882: I/ASessionDescription(7680): b=RR:0
04-08 15:32:45.882: I/ASessionDescription(7680): a=rtpmap:96 mpeg4-generic/48000
04-08 15:32:45.882: I/ASessionDescription(7680): a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr; config=118856e500; SizeLength=13; IndexLength=3; IndexDeltaLength=3; Profile=1;
04-08 15:32:45.882: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp/trackID=0
04-08 15:32:45.882: I/ASessionDescription(7680): m=video 5554 RTP/AVP 96
04-08 15:32:45.882: I/ASessionDescription(7680): b=AS:56
04-08 15:32:45.882: I/ASessionDescription(7680): b=RR:0
04-08 15:32:45.882: I/ASessionDescription(7680): a=rtpmap:96 H264/90000
04-08 15:32:45.882: I/ASessionDescription(7680): a=fmtp:96 packetization-mode=1;profile-level-id=64000b;sprop-parameter-sets=Z2QAC6zZQsTv/AC0ALBAAAADAEAAAAyjxQplgA==,aOvssiw=;
04-08 15:32:45.882: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp/trackID=1
04-08 15:32:45.982: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK
04-08 15:32:45.982: I/MyHandler(7680): SETUP(1) completed with result 0 (Success)
04-08 15:32:45.982: I/MyHandler(7680): server specified timeout of 60 secs.
04-08 15:32:45.992: W/MyHandler(7680): Missing 'source' field in Transport response. Using RTSP endpoint address.
04-08 15:32:45.992: I/APacketSource(7680): dimensions 176x144
04-08 15:32:46.012: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK
04-08 15:32:46.022: I/MyHandler(7680): SETUP(2) completed with result 0 (Success)
04-08 15:32:46.022: I/MyHandler(7680): server specified timeout of 60 secs.
04-08 15:32:46.022: W/MyHandler(7680): Missing 'source' field in Transport response. Using RTSP endpoint address.
04-08 15:32:46.022: W/MyHandler(7680): Server picked an odd RTP port, it should've picked an even one, we'll let it pass for now, but this may break in the future.
04-08 15:32:46.082: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK
04-08 15:32:46.082: D/dalvikvm(13688): GC_FOR_ALLOC freed 303K, 7% free 9289K/9927K, paused 35ms, total 36ms
04-08 15:32:46.092: I/MyHandler(7680): PLAY completed with result 0 (Success)
04-08 15:32:46.092: I/MyHandler(7680): This is a live stream
04-08 15:32:48.262: D/AudioHardware(7680): AudioHardware pcm playback is going to standby.
04-08 15:32:48.262: D/AudioHardware(7680): closePcmOut_l() mPcmOpenCnt: 1
04-08 15:32:56.092: W/MyHandler(7680): Never received any data, switching transports.
04-08 15:32:56.112: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK
04-08 15:32:56.122: I/MyHandler(7680): TEARDOWN completed with result 0 (Success)
04-08 15:32:56.122: I/MyHandler(7680): connection request completed with result 0 (Success)
04-08 15:32:56.152: I/ARTSPConnection(7680): status: RTSP/1.0 200 OK
04-08 15:32:56.152: I/MyHandler(7680): DESCRIBE completed with result 0 (Success)
04-08 15:32:56.152: I/ASessionDescription(7680): v=0
04-08 15:32:56.152: I/ASessionDescription(7680): o=- 15352003157473632156 15352003157473632156 IN IP4 to63-169.ifremer.fr
04-08 15:32:56.152: I/ASessionDescription(7680): s=Unnamed
04-08 15:32:56.152: I/ASessionDescription(7680): i=N/A
04-08 15:32:56.152: I/ASessionDescription(7680): c=IN IP4 134.246.63.169
04-08 15:32:56.152: I/ASessionDescription(7680): t=0 0
04-08 15:32:56.152: I/ASessionDescription(7680): a=tool:vlc 2.0.5
04-08 15:32:56.152: I/ASessionDescription(7680): a=recvonly
04-08 15:32:56.152: I/ASessionDescription(7680): a=type:broadcast
04-08 15:32:56.152: I/ASessionDescription(7680): a=charset:UTF-8
04-08 15:32:56.152: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp
04-08 15:32:56.152: I/ASessionDescription(7680): m=audio 5556 RTP/AVP 96
04-08 15:32:56.152: I/ASessionDescription(7680): b=AS:24
04-08 15:32:56.152: I/ASessionDescription(7680): b=RR:0
04-08 15:32:56.152: I/ASessionDescription(7680): a=rtpmap:96 mpeg4-generic/48000
04-08 15:32:56.152: I/ASessionDescription(7680): a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr; config=118856e500; SizeLength=13; IndexLength=3; IndexDeltaLength=3; Profile=1;
04-08 15:32:56.152: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp/trackID=0
04-08 15:32:56.152: I/ASessionDescription(7680): m=video 5554 RTP/AVP 96
04-08 15:32:56.152: I/ASessionDescription(7680): b=AS:56
04-08 15:32:56.152: I/ASessionDescription(7680): b=RR:0
04-08 15:32:56.152: I/ASessionDescription(7680): a=rtpmap:96 H264/90000
04-08 15:32:56.152: I/ASessionDescription(7680): a=fmtp:96 packetization-mode=1;profile-level-id=64000b;sprop-parameter-sets=Z2QAC6zZQsTv/AC0ALBAAAADAEAAAAyjxQplgA==,aOvssiw=;
04-08 15:32:56.152: I/ASessionDescription(7680): a=control:rtsp://134.246.63.169:5554/stream.sdp/trackID=1
04-08 15:32:56.222: I/ARTSPConnection(7680): status: RTSP/1.0 461 Unsupported transport
04-08 15:32:56.222: I/MyHandler(7680): SETUP(1) completed with result 0 (Success)
04-08 15:32:56.222: I/APacketSource(7680): dimensions 176x144
04-08 15:32:56.242: I/ARTSPConnection(7680): status: RTSP/1.0 461 Unsupported transport
04-08 15:32:56.252: I/MyHandler(7680): SETUP(2) completed with result 0 (Success)
04-08 15:32:56.272: E/MediaPlayer(13688): error (1, -2147483648)
04-08 15:32:56.272: E/MediaPlayer(13688): Error (1,-2147483648)
04-08 15:32:56.272: D/VideoView(13688): Error: 1,-2147483648
我猜问题是指向"状态:RTSP/1.0 461不支持的传输"，但我看不出我能改变什么:我已经打开了我使用的端口，并且我确实在另一台计算机上接收视频。
在Android手机上，我可以播放一些我在网上找到的rtsp流，例如:rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov。所以应该是可能的。
如果有人能帮忙…div !
最后是网络问题，我通过MacBook WiFi共享连接我的设备，它似乎阻止了RTSP流。现在我正在使用路由器，它在RTSP中工作(我仍然无法在Android VideoView中接收HTTP流)。尽管如此，我仍然有一个超时问题:RTSP流在60秒后停止，因为VideoView不发送keep alive消息。
我已经用openRTSP命令测试了我的rtsp服务器。
UDP端口被阻塞。
如果访问rtsp不带-t:
-> $ openRTSP <rtsp_url>
我得到的日志告诉我:
// omit lots of lines..
Created receiver for "video/H264" subsession (client ports 63346-63347)
Sending request: SETUP rtsp://61.218.52.250:554/live/ch00_0/trackID=0 RTSP/1.0
CSeq: 4
User-Agent: openRTSP (LIVE555 Streaming Media v2013.12.16)
Transport: RTP/AVP;unicast;client_port=63346-63347
Received 47 new bytes of response data.
Received a complete SETUP response:
RTSP/1.0 461 Unsupported Transport
CSeq: 4
Failed to setup "video/H264" subsession: 461 Unsupported Transport
所以改成TCP:
-> $ openRTSP -t <rtsp_url>
it start receive data successfully.
// omit lots of lines..
Opened URL "rtsp://61.218.52.250:554/live/ch00_0", returning a SDP description:
v=0
o=- 1 1 IN IP4 127.0.0.1
s=Ubiquiti Live
i=UBNT Streaming Media
c=IN IP4 0.0.0.0
t=0 0
m=video 0 RTP/AVP 99
b=AS:50000
a=framerate:25
a=x-dimensions:1280,720
a=x-vendor-id:ubnt,a521
a=x-rtp-ts:4617405454576779984
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42A01E;packetization-mode=1;sprop-parameter-sets=Z0IAKOkAoAt1xIAG3dAAzf5gDYgQlA==,aM4xUg==
a=control:trackID=0
Sending request: SETUP rtsp://61.218.52.250:554/live/ch00_0/trackID=0 RTSP/1.0
CSeq: 4
User-Agent: openRTSP (LIVE555 Streaming Media v2013.12.16)
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Received 107 new bytes of response data.
Received a complete SETUP response:
RTSP/1.0 200 OK
CSeq: 4
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Session: E090B5503236A1BFB7CE
Setup "video/H264" subsession (client ports 54884-54885)
Sending request: PLAY rtsp://61.218.52.250:554/live/ch00_0/ RTSP/1.0
CSeq: 5
User-Agent: openRTSP (LIVE555 Streaming Media v2013.12.16)
Session: E090B5503236A1BFB7CE
Range: npt=0.000-
Received 159 new bytes of response data.
Received a complete PLAY response:
RTSP/1.0 200 OK
CSeq: 5
Session: E090B5503236A1BFB7CE
Range: npt=now-
RTP-Info: url=rtsp://61.218.52.250:554/live/ch00_0//trackID=0;seq=41402;rtptime=0
Started playing session
Data is being streamed (signal with "kill -HUP 96432" or "kill -USR1 96432" to terminate)...
Received 47 new bytes of response data.
Received 1424 new bytes of response data.
Received 1424 new bytes of response data.
Received 1424 new bytes of response data.
Received 1424 new bytes of response data.
Received 1448 new bytes of response data.
Received 1448 new bytes of response data.
参考openRTSP基础。
现在我要弄清楚如何在Android中自动切换到TCP
请尝试VLC:
vlc some_file.mp4 -I http——sout "#transcode{overlay,ab=128,samplerate=44100,channels=2,acodec=mp4a,vcodec=h264,width=480,height=270,vfilter="canvas{width=480,height=270,aspect=16:9}"，fps=25,vb=800,venc=x264{level=12,no-cabac,subme=20,threads=4,bframes=0,min-keyint=1,keyint=50}}:gather:rtp{mp4a-latm,sdp=rtsp://0.0.0.0:5554/stream.sdp}"
和android代码:
@Override
protected void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_main);
final VideoView vidView = (VideoView)findViewById(R.id.myVideo);
MediaController vidControl = new MediaController(this);
vidControl.setAnchorView(vidView);
vidView.setMediaController(vidControl);
vidView.setVideoPath("rtsp://137.110.92.231:5554/stream.sdp");
vidView.start();
}
使用MediaPlayer它支持HTTP和RTSP网络协议。http://developer.android.com/guide/topics/media/mediaplayer.html#mediaplayerhttp://developer.android.com/guide/appendix/media-formats.html建议