应用错误收集

我想通过Gstreamer从MIC流式传输音频数据。但是我无法用rx播放MIC音频。如何播放MIC输入的音频流？

tx：gst-launch-1.0 -v alsasrc device =＆＃34; hw：0＆＃34; ！ decodebin！ audioconvert   ！ rtpL16pay！排队！ udpsink host = 239.0.0.1 auto-multicast = true   端口= 5004

rx：gst-launch-1.0 udpsrc multicast-group = 239.0.0.1 port = 5004   帽=＆＃34;应用程序/ x-RTP＆＃34; ！ rtpL16depay！ alsasink

rx结果：将管道设置为PAUSED ...管道是实时的   不需要PREROLL ...将管道设置为PLAYING ...新时钟：   GstSystemClock错误：来自元素   / GstPipeline：pipeline0 / GstUDPSrc：udpsrc0：内部数据流错误。   其他调试信息：   ../../../../gstreamer-1.8.1/libs/gst/base/gstbasesrc.c(2948）：   gst_base_src_loop（）：/ GstPipeline：pipeline0 / GstUDPSrc：udpsrc0：   流媒体任务暂停，原因未协商（-4）执行结束   在0：00：00.009364000之后将管道设置为PAUSED ...设置   管道到READY ...将管道设置为NULL ...释放管道   ...

结果如下。

将管道设置为PAUSED ...管道是实时的，不需要 PREROLL ...将管道设置为PLAYING ...新时钟： GstAudioSrcClock / GstPipeline：pipeline0 / GstAlsaSrc：alsasrc0：实际缓冲时间= 200000 / GstPipeline：pipeline0 / GstAlsaSrc：alsasrc0：actual-latency-time = 10000 /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src：caps = ＆＃34; audio / x-raw \，\ format \ =（string）S16LE \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ channels \ =（int）2 \，\ channel-mask \ =（bitmask）0x0000000000000003＆＃34; /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:sink.GstProxyPad:proxypad0： caps =＆＃34; audio / x-raw \，\ format \ =（string）S16LE \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ channels \ =（int）2 \，\ channel-mask \ =（bitmask）0x0000000000000003＆＃34; /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src： caps =＆＃34; audio / x-raw \，\ format \ =（string）S16LE \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ channels \ =（int）2 \，\ channel-mask \ =（bitmask）0x0000000000000003＆＃34; 重新分配延迟...... /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src:cap =＆＃34; audio / x-raw \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ format \ =（string）S16BE \，\ channels \ =（int）2 \，\ 通道掩模\ =（位掩码）0x0000000000000003＆＃34; /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:src:caps = ＆＃34; 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＆＃34; /GstPipeline:pipeline0/GstQueue:queue0.GstPad:src：caps = ＆＃34; 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＆＃34; /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink：caps = ＆＃34; 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＆＃34; /GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink：caps = ＆＃34; 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＆＃34; /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink：caps = ＆＃34; audio / x-raw \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ format \ =（string）S16BE \，\ channels \ =（int）2 \，\ 通道掩模\ =（位掩码）0x0000000000000003＆＃34; /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink:cap =＆＃34; audio / x-raw \，\ format \ =（string）S16LE \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ channels \ =（int）2 \，\ channel-mask \ =（bitmask）0x0000000000000003＆＃34; /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstDecodePad:src_0.GstProxyPad:proxypad1： caps =＆＃34; audio / x-raw \，\ format \ =（string）S16LE \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ channels \ =（int）2 \，\ channel-mask \ =（bitmask）0x0000000000000003＆＃34; /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:sink： caps =＆＃34; audio / x-raw \，\ format \ =（string）S16LE \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ channels \ =（int）2 \，\ channel-mask \ =（bitmask）0x0000000000000003＆＃34; /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:sink:cap =＆＃34; audio / x-raw \，\ format \ =（string）S16LE \，\ layout \ =（string）interleaved \，\ rate \ =（int）44100 \，\ channels \ =（int）2 \，\ channel-mask \ =（bitmask）0x0000000000000003＆＃34; / GstPipeline：pipeline0 / GstRtpL16Pay：rtpl16pay0：timestamp = 725507323 / GstPipeline：pipeline0 / GstRtpL16Pay：rtpl16pay0：seqnum = 20783

我认为rx管道是错误的，但我找不到解决方案。请告诉我如何制作管道。

PS：我尝试了以下命令，并且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 =＆＃34; hw：0＆＃34; ！ decodebin！ audioconvert   ！听觉样本！ alawenc！ rtppcmapay！排队！ 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 =＆＃34; application / x-rtp，media =（string）audio，clock-rate =（int）8000，   encoding-name =（string）PCMA，encoding-params =（string）2，   channels =（int）1，payload =（int）8＆＃34; ！ rtppcmadepay！ alawdec！ alsasink

使用Gstreamer接收音频流导致原因没有协商错误

1 个答案: