我想使用Gstreamer流式传输来自MIC的音频数据。
但是我无法用rx播放MIC音频。
如何播放来自MIC输入的音频流？
tx: gst-launch-1.0 -v alsasrc device="hw:0" ! decodebin ! audioconvert ! rtpL16pay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004
rx: gst-launch-1.0 udpsrc multicast-group=239.0.0.1 port=5004 caps="application/x-rtp" ! rtpL16depay ! alsasink
rx result: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error. Additional debug info: ../../../../gstreamer-1.8.1/libs/gst/base/gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: streaming task paused, reason not-negotiated (-4) Execution ended after 0:00:00.009364000 Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ...
tx结果如下。
Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000 /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000 /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:sink.GstProxyPad:proxypad0: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" Redistribute latency... /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = "audio/x-raw\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ format\=(string)S16BE\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:src: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)3961155089\,\ timestamp-offset\=(uint)725507323\,\ seqnum-offset\=(uint)20783" /GstPipeline:pipeline0/GstQueue:queue0.GstPad:src: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)3961155089\,\ timestamp-offset\=(uint)725507323\,\ seqnum-offset\=(uint)20783" /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)3961155089\,\ timestamp-offset\=(uint)725507323\,\ seqnum-offset\=(uint)20783" /GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)3961155089\,\ timestamp-offset\=(uint)725507323\,\ seqnum-offset\=(uint)20783" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink: caps = "audio/x-raw\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ format\=(string)S16BE\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstDecodePad:src_0.GstProxyPad:proxypad1: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:sink: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:sink: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: timestamp = 725507323 /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: seqnum = 20783
我认为rx管道是错误的，但是我找不到解决方案。
请告诉我如何制作管道。
PS:
我尝试执行以下命令，然后RX播放麦克风音频!这意味着接收器设备无法播放L16音频？
tx: gst-launch-1.0 -v alsasrc device="hw:0" ! decodebin ! audioconvert ! audioresample ! alawenc ! rtppcmapay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004
rx: gst-launch-1.0 udpsrc multicast-group=239.0.0.1 port=5004 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, encoding-params=(string)2, channels=(int)1, payload=(int)8" ! rtppcmadepay ! alawdec ! alsasink
最佳答案
您需要在接收中添加上限，请尝试以下管道:
gst-launch-1.0 udpsrc multicast-group=239.0.0.1 port=5004 caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)2, channels=(int)2, payload=(int)96' ! rtpL16depay ! audioconvert ! alsasink
关于audio - 使用Gstreamer接收音频流导致原因未协商错误，我们在Stack Overflow上找到一个类似的问题： https://stackoverflow.com/questions/48221472/