10

对于我的应用程序，我必须从 Decklink 卡流式传输到 Android 应用程序（我必须是实时流，因此 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，我可以通过网络直播到另一台电脑（但它只能用RTSP工作60秒）。

这是我尝试过的：

  • 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}'

它适用于 Android VLC 0.0.11，但仅适用于 WiFi，不适用于 3G。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：

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)
  • 实时传输协议：

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

  • 尺寸：176x144

    ./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：/ ）。如果我不指定复用器，我也可以播放它 QuickTime（如果我指定复用器，ts 或 ps，我没有视频。如果我尝试另一个复用器，VLC 告诉我只允许使用 ts或 RTP 中的 ps）

如果我尝试使用 Google 视频播放器，我会在 locat 中收到这些消息：

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

我想问题出在“状态：RTSP/1.0 461 不支持的传输”上，但我看不出我可以改变什么：我已经打开了我使用的端口，并且确实在另一台计算机上接收到视频。

在安卓手机上，我可以播放我在网上找到的一些 rtsp 流，例如这个：rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov。所以应该是可以的。

如果有人可以帮助...！

4

4 回答 4

7

最后是网络问题，我通过 MacBook WiFi 共享连接我的设备，似乎它阻止了 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 不发送保持活动消息。我会尝试自己做...

于 2013-04-15T09:25:39.533 回答
0

我已经用 openRTSP 命令测试了我的 rtsp 服务器。

这是UDP端口被阻止。

如果在没有 -t 的情况下访问 rtsp：

-> $ openRTSP <rtsp_url>

我得到日志告诉我：

// 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

因此更改为 TCP：

-> $ openRTSP -t <rtsp_url>

它开始成功接收数据。

// 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。

于 2013-12-17T08:48:00.707 回答
0

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}}:gather:rtp{mp4a-latm,sdp=rtsp://0.0.0.0:5554/stream.sdp}"

和安卓代码：

@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();
        }
于 2015-01-04T04:02:26.503 回答
0

使用MediaPlayer它支持 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

于 2015-08-06T05:01:29.200 回答