如何使用 gstreamer 通过 RTSP 流式传输 MIC 输入
How to stream MIC input by RTSP with gstreamer
我想从 RTSP 服务器流式传输麦克风输入。
我使用 gstreamer1.0 和 gstreamer1.0-rtsp-server (v1.12.3)。
我尝试了以下命令，但管道创建的 RTSP 服务器没有响应。如何流式传输？
Tx: GST_DEBUG=3 test-launch "( alsasrc device="hw:0" ! audioconvert ! rtpL16pay )"
Rx: gst-launch-1.0 -q rtspsrc location=rtsp://10.5.41.11:8554/test ! decodebin ! audioconvert ! alsasink
结果（tx）是
stream ready at rtsp://127.0.0.1:8554/test
0:00:08.267067376 800 0x7601dd80 FIXME default gstutils.c:3902:gst_pad_create_stream_id_internal:<alsasrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:00:08.274445501 800 0x7601dd80 WARN alsa pcm_hw.c:1250:snd_pcm_hw_get_chmap: alsalib error: Cannot read Channel Map ctl
: No such file or directory
0:00:08.299559918 800 0x7601dd80 WARN basesrc gstbasesrc.c:2939:gst_base_src_loop:<alsasrc0> error: Internal data stream error.
0:00:08.299722543 800 0x7601dd80 WARN basesrc gstbasesrc.c:2939:gst_base_src_loop:<alsasrc0> error: streaming stopped, reason not-linked (-1)
0:00:08.300421793 800 0x7601d920 WARN rtspmedia rtsp-media.c:2433:default_handle_message: 0x7600e7d0: got error Internal data stream error. (../../../../gstreamer-1.12.3/libs/gst/base/gstbasesrc.c(2939): gst_base_src_loop (): /GstPipeline:media-pipeline/GstBin:bin0/GstAlsaSrc:alsasrc0:
streaming stopped, reason not-linked (-1))
0:00:08.300855501 800 0xdef890 WARN rtspmedia rtsp-media.c:2728:wait_preroll: failed to preroll pipeline
0:00:08.301006459 800 0xdef890 WARN rtspmedia rtsp-media.c:3032:gst_rtsp_media_prepare: failed to preroll pipeline
0:00:08.314116334 800 0xdef890 ERROR rtspclient rtsp-client.c:977:find_media: client 0xdcfb40: can't prepare media
0:00:08.316345126 800 0xdef890 ERROR rtspclient rtsp-client.c:2696:handle_describe_request: client 0xdcfb40: no media
我也尝试了下面的命令。
Tx: GST_DEBUG=3 test-launch "( alsasrc device="hw:0" ! audio decodebin ! audioconvert ! rtpL16pay )"
Rx: gst-launch-1.0 -q rtspsrc location=rtsp://10.5.41.11:8554/test ! decodebin ! audioconvert ! alsasink
结果(tx):
stream ready at rtsp://127.0.0.1:8554/test
0:00:03.216280583 805 0x75f105b0 FIXME default gstutils.c:3902:gst_pad_create_stream_id_internal:<alsasrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:00:03.222776083 805 0x75f105b0 WARN alsa pcm_hw.c:1250:snd_pcm_hw_get_chmap: alsalib error: Cannot read Channel Map ctl
: No such file or directory
0:00:03.238019083 805 0x1b82890 FIXME rtspmedia rtsp-media.c:3581:gst_rtsp_media_suspend: suspend for dynamic pipelines needs fixing
0:00:03.256477500 805 0x75f105b0 WARN basesrc gstbasesrc.c:2939:gst_base_src_loop:<alsasrc0> error: Internal data stream error.
0:00:03.256641250 805 0x75f105b0 WARN basesrc gstbasesrc.c:2939:gst_base_src_loop:<alsasrc0> error: streaming stopped, reason not-linked (-1)
0:00:03.257567125 805 0x75f10150 WARN rtspmedia rtsp-media.c:2433:default_handle_message: 0x75f0ea70: got error Internal data stream error. (../../../../gstreamer-1.12.3/libs/gst/base/gstbasesrc.c(2939): gst_base_src_loop (): /GstPipeline:media-pipeline/GstBin:bin0/GstAlsaSrc:alsasrc0:
streaming stopped, reason not-linked (-1))
PS:
第一个tx管道不包含decodebin，rx错误是503服务不可用。所以我认为 tx 管道是错误的。
第二个结果如下
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not get/set settings from/on resource.
Additional debug info:
../../../gst-plugins-good-1.8.3/gst/rtsp/gstrtspsrc.c(6361): gst_rtspsrc_setup_streams (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
SDP contains no streams
ERROR: pipeline doesn't want to preroll.
我认为这个错误表明我应该将 SDP 添加到 rx 或 tx 管道。
如何将 SDP 添加到管道？
为了回答评论，我在这里描述命令。
Tx: GST_DEBUG=3 test-launch "( alsasrc device="hw:0" ! avenc_g722 ! rtpg722pay name=pay0 )"
Rx: gst-launch-1.0 rtspsrc location=rtsp://10.5.41.11:8554/test rtpjitterbuffer latency=0 ! fakesink
结果（发送）：
0:00:30.110267253 1979 0xe62830 WARN rtspmedia rtsp-media.c:243 3:default_handle_message: 0xe90d20: got error Internal data stream error. (../.. /../../gstreamer-1.12.3/libs/gst/base/gstbasesrc.c(2939): gst_base_src_loop (): /GstPipeline:media-pipeline/GstBin:bin2/GstAlsaSrc:alsasrc2:
streaming stopped, reason not-negotiated (-4))
0:00:30.118076170 1979 0x75f3b4f0 ERROR rtspclient rtsp-client.c:97 7:find_media: client 0xe42c60: can't prepare media
0:00:30.119548961 1979 0x75f3b4f0 ERROR rtspclient rtsp-client.c:26 96:handle_describe_request: client 0xe42c60: no media
结果（Rx）：
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://10.5.41.11:8554/test
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not read from resource.
Additional debug info:
../../../gst-plugins-good-1.8.3/gst/rtsp/gstrtspsrc.c(5670): gst_rtspsrc_send (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Got error response: 503 (Service Unavailable).
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
您需要为 rtpL16pay 设置名称，请为 TX 尝试以下管道：
对于测试，最初从 audiotestsrc 开始：
test-launch "(audiotestsrc ! audioconvert ! rtpL16pay name=pay0 )"
从你的麦克风（如果它连接在 0 使用以下管道）：
test-launch "(alsasrc device="hw:0" ! audioparse ! decodebin ! audioconvert ! audioresample ! avenc_g722 ! rtpg722pay name=pay0 )"
对于 RX，请尝试以下管道：
gst-launch-1.0 -v -e rtspsrc location=rtsp://127.0.0.1:8554/test !
rtpjitterbuffer latency=100 ! rtpL16depay ! audioconvert ! alsasink
我想从 RTSP 服务器流式传输麦克风输入。 我使用 gstreamer1.0 和 gstreamer1.0-rtsp-server (v1.12.3)。
我尝试了以下命令，但管道创建的 RTSP 服务器没有响应。如何流式传输？
Tx: GST_DEBUG=3 test-launch "( alsasrc device="hw:0" ! audioconvert ! rtpL16pay )"
Rx: gst-launch-1.0 -q rtspsrc location=rtsp://10.5.41.11:8554/test ! decodebin ! audioconvert ! alsasink
结果（tx）是
stream ready at rtsp://127.0.0.1:8554/test
0:00:08.267067376 800 0x7601dd80 FIXME default gstutils.c:3902:gst_pad_create_stream_id_internal:<alsasrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:00:08.274445501 800 0x7601dd80 WARN alsa pcm_hw.c:1250:snd_pcm_hw_get_chmap: alsalib error: Cannot read Channel Map ctl
: No such file or directory
0:00:08.299559918 800 0x7601dd80 WARN basesrc gstbasesrc.c:2939:gst_base_src_loop:<alsasrc0> error: Internal data stream error.
0:00:08.299722543 800 0x7601dd80 WARN basesrc gstbasesrc.c:2939:gst_base_src_loop:<alsasrc0> error: streaming stopped, reason not-linked (-1)
0:00:08.300421793 800 0x7601d920 WARN rtspmedia rtsp-media.c:2433:default_handle_message: 0x7600e7d0: got error Internal data stream error. (../../../../gstreamer-1.12.3/libs/gst/base/gstbasesrc.c(2939): gst_base_src_loop (): /GstPipeline:media-pipeline/GstBin:bin0/GstAlsaSrc:alsasrc0:
streaming stopped, reason not-linked (-1))
0:00:08.300855501 800 0xdef890 WARN rtspmedia rtsp-media.c:2728:wait_preroll: failed to preroll pipeline
0:00:08.301006459 800 0xdef890 WARN rtspmedia rtsp-media.c:3032:gst_rtsp_media_prepare: failed to preroll pipeline
0:00:08.314116334 800 0xdef890 ERROR rtspclient rtsp-client.c:977:find_media: client 0xdcfb40: can't prepare media
0:00:08.316345126 800 0xdef890 ERROR rtspclient rtsp-client.c:2696:handle_describe_request: client 0xdcfb40: no media
我也尝试了下面的命令。
Tx: GST_DEBUG=3 test-launch "( alsasrc device="hw:0" ! audio decodebin ! audioconvert ! rtpL16pay )"
Rx: gst-launch-1.0 -q rtspsrc location=rtsp://10.5.41.11:8554/test ! decodebin ! audioconvert ! alsasink
结果(tx):
stream ready at rtsp://127.0.0.1:8554/test
0:00:03.216280583 805 0x75f105b0 FIXME default gstutils.c:3902:gst_pad_create_stream_id_internal:<alsasrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id
0:00:03.222776083 805 0x75f105b0 WARN alsa pcm_hw.c:1250:snd_pcm_hw_get_chmap: alsalib error: Cannot read Channel Map ctl
: No such file or directory
0:00:03.238019083 805 0x1b82890 FIXME rtspmedia rtsp-media.c:3581:gst_rtsp_media_suspend: suspend for dynamic pipelines needs fixing
0:00:03.256477500 805 0x75f105b0 WARN basesrc gstbasesrc.c:2939:gst_base_src_loop:<alsasrc0> error: Internal data stream error.
0:00:03.256641250 805 0x75f105b0 WARN basesrc gstbasesrc.c:2939:gst_base_src_loop:<alsasrc0> error: streaming stopped, reason not-linked (-1)
0:00:03.257567125 805 0x75f10150 WARN rtspmedia rtsp-media.c:2433:default_handle_message: 0x75f0ea70: got error Internal data stream error. (../../../../gstreamer-1.12.3/libs/gst/base/gstbasesrc.c(2939): gst_base_src_loop (): /GstPipeline:media-pipeline/GstBin:bin0/GstAlsaSrc:alsasrc0:
streaming stopped, reason not-linked (-1))
PS:
第一个tx管道不包含decodebin，rx错误是503服务不可用。所以我认为 tx 管道是错误的。
第二个结果如下
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not get/set settings from/on resource.
Additional debug info:
../../../gst-plugins-good-1.8.3/gst/rtsp/gstrtspsrc.c(6361): gst_rtspsrc_setup_streams (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
SDP contains no streams
ERROR: pipeline doesn't want to preroll.
我认为这个错误表明我应该将 SDP 添加到 rx 或 tx 管道。 如何将 SDP 添加到管道？
为了回答评论，我在这里描述命令。
Tx: GST_DEBUG=3 test-launch "( alsasrc device="hw:0" ! avenc_g722 ! rtpg722pay name=pay0 )"
Rx: gst-launch-1.0 rtspsrc location=rtsp://10.5.41.11:8554/test rtpjitterbuffer latency=0 ! fakesink
结果（发送）：
0:00:30.110267253 1979 0xe62830 WARN rtspmedia rtsp-media.c:243 3:default_handle_message: 0xe90d20: got error Internal data stream error. (../.. /../../gstreamer-1.12.3/libs/gst/base/gstbasesrc.c(2939): gst_base_src_loop (): /GstPipeline:media-pipeline/GstBin:bin2/GstAlsaSrc:alsasrc2:
streaming stopped, reason not-negotiated (-4))
0:00:30.118076170 1979 0x75f3b4f0 ERROR rtspclient rtsp-client.c:97 7:find_media: client 0xe42c60: can't prepare media
0:00:30.119548961 1979 0x75f3b4f0 ERROR rtspclient rtsp-client.c:26 96:handle_describe_request: client 0xe42c60: no media
结果（Rx）：
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://10.5.41.11:8554/test
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not read from resource.
Additional debug info:
../../../gst-plugins-good-1.8.3/gst/rtsp/gstrtspsrc.c(5670): gst_rtspsrc_send (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Got error response: 503 (Service Unavailable).
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
您需要为 rtpL16pay 设置名称，请为 TX 尝试以下管道：
对于测试，最初从 audiotestsrc 开始：
test-launch "(audiotestsrc ! audioconvert ! rtpL16pay name=pay0 )"
从你的麦克风（如果它连接在 0 使用以下管道）：
test-launch "(alsasrc device="hw:0" ! audioparse ! decodebin ! audioconvert ! audioresample ! avenc_g722 ! rtpg722pay name=pay0 )"
对于 RX，请尝试以下管道：
gst-launch-1.0 -v -e rtspsrc location=rtsp://127.0.0.1:8554/test ! rtpjitterbuffer latency=100 ! rtpL16depay ! audioconvert ! alsasink