1

我在 Ubuntu 中使用以下管道流式传输 mp3，它运行良好。

发件人：

gst-launch filesrc location=/home/file.mp3 ! mad ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16, rate=44100 ! rtpL16pay  ! udpsink host=192.168.1.103 port=5000

接收者：

gst-launch udpsrc port=5000 caps="application/x-rtp, media=(string)audio, clock-rate=(int)44100, width=(int)16, height=(int)16, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-position=(int)1, payload=(int)96" ! gstrtpjitterbuffer do-lost=true ! rtpL16depay ! audioconvert ! alsasink sync=false

但是当我在 Windows 中使用它时它不起作用。我能听到它只在一开始播放，但只播放了很短的时间。目的地接收媒体，但我听不到声音。这是使用的管道，

发件人：

gst-launch -v filesrc location=C:/file.mp3 ! mad ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16, rate=44100 ! rtpL16pay  ! udpsink host=192.168.1.105 port=5000

接收者：

gst-launch -v udpsrc port=5000 caps="application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96" ! gstrtpjitterbuffer do-lost=true ! rtpL16depay ! audioconvert ! autoaudiosink sync=false

Gstreamer OSSBUILD用于 Windows。请帮我解决这个问题。

如果需要，以下是终端上的输出，

发件人：

Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstMad:mad0.GstPad:src: caps = audio/x-raw-int, endiannes
s=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100
, channels=(int)2
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-
raw-int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)1
6, rate=(int)44100, channels=(int)1
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x
-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)
32, rate=(int)44100, channels=(int)2
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x-raw-
int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)16, r
ate=(int)44100, channels=(int)1
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = audio/x-raw
-int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)44100, channels=(int)1
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:src: caps = application/x-
rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, enco
ding-params=(string)1, channels=(int)1, payload=(int)96, ssrc=(uint)1331970475,
clock-base=(uint)3177922110, seqnum-base=(uint)10029
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink: caps = audio/x-raw-i
nt, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)16, ra
te=(int)44100, channels=(int)1
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: timestamp = 3177922110
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: seqnum = 10029
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp
, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encodin
g-params=(string)1, channels=(int)1, payload=(int)96, ssrc=(uint)1331970475, clo
ck-base=(uint)3177922110, seqnum-base=(uint)10029
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Got EOS from element "pipeline0".
Execution ended after 40987345000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:src: caps = NULL
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = NULL
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = NULL
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstMad:mad0.GstPad:src: caps = NULL
Setting pipeline to NULL ...
Freeing pipeline ...

接收者：

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:src: caps = ap
plication/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(stri
ng)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96
/GstPipeline:pipeline0/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:sink: caps = a
pplication/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(str
ing)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96
/GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:src: caps = audio/x-ra
w-int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)44100, channels=(int)1, channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO >
/GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:sink: caps = applicati
on/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16,
encoding-params=(string)1, channels=(int)1, payload=(int)96
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-
raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)tru
e, channels=(int)1, rate=(int)44100, channel-positions=(GstAudioChannelPosition)
< GST_AUDIO_CHANNEL_POSITION_FRONT_MONO >
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x
-raw-int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)
16, rate=(int)44100, channels=(int)1, channel-positions=(GstAudioChannelPosition
)< GST_AUDIO_CHANNEL_POSITION_FRONT_MONO >
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstWaveFormSink:autoaudio
sink0-actual-sink-waveform.GstPad:sink: caps = audio/x-raw-int, width=(int)16, d
epth=(int)16, endianness=(int)1234, signed=(boolean)true, channels=(int)1, rate=
(int)44100, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSIT
ION_FRONT_MONO >
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink: caps =
audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boo
lean)true, channels=(int)1, rate=(int)44100, channel-positions=(GstAudioChannelP
osition)< GST_AUDIO_CHANNEL_POSITION_FRONT_MONO >
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink.GstProxy
Pad:proxypad0: caps = audio/x-raw-int, width=(int)16, depth=(int)16, endianness=
(int)1234, signed=(boolean)true, channels=(int)1, rate=(int)44100, channel-posit
ions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_MONO >
4

1 回答 1

0

为什么不转储到文件而不是自动音频同步和检查。这将告诉您管道的其余部分是否正常。我将继续从接收器管道的末端一次删除一个元素，并附加文件接收器以找出哪个元素是罪魁祸首。为什么不使用特定的音频接收器而不是自动音频接收器。这也可能是问题所在。

你有一个简单的管道：通过消除来做到这一点

  1. 确保输入确实持续不断

src -> 文件接收器

src -> depay -> 文件接收器

沿着这条路走，直到找到错误的元素。

您的文件大小应不断增长。哪个元素导致它不再工作是罪魁祸首。如果它是 audiosink，则使用一个显式的而不是 autoaudiosink。

你可以引入一个队列并检查。运行 GST_DEBUG=3 或 4 或 5 并增加调试信息以检查罪魁祸首。剩下的调试选项很多。

于 2013-05-08T09:46:47.647 回答