使用 Gstreamer 接收音频流导致 reason not negotiated 错误
Recieving audio stream with Gstreamer results in reason not negotiated error
我想使用 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
...
交易结果如下
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 管道有问题，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，然后 rx 播放了麦克风音频！这意味着接收设备无法播放 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
您需要在接收中添加上限，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
我想使用 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 ...
交易结果如下
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 管道有问题，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，然后 rx 播放了麦克风音频！这意味着接收设备无法播放 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
您需要在接收中添加上限，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