VLC RTSP实时流到Android


VLC RTSP live stream to Android

对于我的应用程序，我必须从卡链接卡流式传输到Android应用程序(我必须是实时流式传输，因此HLS或RTSP似乎是不错的解决方案，因为我的应用程序针对Android 3+。)我使用了decklink sdk重新编译了VLC，并且可以通过网络将视频流实时传输到另一台PC(但是在RTSP下它只能工作60秒)。

这是我尝试的：

  • HTTP流：

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    ./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}'

它可以在Android VLC 0.0.11中使用，但只能在WiFi中使用，而不能在3G中使用。而且我无法使用VideoView在我的应用程序中播放它。这是我使用的代码和相应的错误消息：

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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();

错误讯息：

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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：

我在此页面上使用了Google推荐的编码选项，例如：

  • 视频编解码器：h264
  • 音频编解码器：AAC
  • 视频比特率：56
  • 音频比特率：24
  • 音频通道：1个

  • 尺寸：176x144

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    ./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}'

我可以在VLC桌面上播放流，但不能在Android中播放(即使在Android VLC版本或默认的Google视频播放器：/中)。
如果我未指定复用器，也可以在QuickTime中播放它(如果我指定ts或ps的复用器，则没有视频。如果尝试其他复用器，VLC告诉我只能使用ts或RTP中的ps)

如果我尝试使用Google视频播放器，则会在locat中收到以下消息：

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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

我猜问题出在"状态：RTSP / 1.0 461不支持的传输"，但是我看不到我可以更改什么：我已经打开了我使用的端口，并且在另一台计算机上收到了视频。

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。
因此应该有可能。

如果有人可以帮助...！

相关讨论

最终这是一个网络问题，我通过MacBook WiFi共享连接设备，似乎阻止了RTSP流。现在，我正在使用路由器，它可以在RTSP中使用(我仍然无法在Android VideoView中接收HTTP流)。
不过，我仍然存在超时问题：RTSP流在60秒后停止，因为VideoView不会发送保持活动的消息。我会尝试自己做的...

相关讨论

使用MediaPlayer，它同时支持HTTP和RTSP网络协议。
http://developer.android.com/guide/topics/media/mediaplayer.html#mediaplayer
http://developer.android.com/guide/appendix/media-formats.html#recommendations

请尝试VLC：

vlc some_file.mp4 -I http --sout" #transcode {soverlay，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}}：聚集：rtp {mp4a-latm，sdp = rtsp：//0.0.0.0：5554 / stream.sdp}"

和android代码：

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@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();
        }

我已经使用openRTSP命令测试了rtsp服务器。

UDP端口被阻止。

如果不带-t访问rtsp：

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-> $ openRTSP <rtsp_url>

我收到日志告诉我：

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// 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

因此，通过以下方式更改为TCP：

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-> $ openRTSP -t <rtsp_url>

它开始成功接收数据。

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// 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基础知识。

现在，我必须弄清楚如何在Android中自动切换到TCP。

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