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rtsp to hlssink with audio

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Jerry Geis-2

rtsp to hlssink with audio

HI - all I am taking RTSP camera to hlssink and trying to get audio to work.

gst-launch-1.0 mpegtsmux name=m ! hlssink playlist-root=/ location=hlssink.camera.1636.%05d.ts target-duration=1 max-files=3 playlist-length=2 playlist-location=playlist.camera.1636.m3u8 rtspsrc latency=0 location=rtsp://192.168.2.5 name=d d. ! rtph264depay ! h264parse config-interval=-1 ! queue ! m. d. ! rtpmp4gdepay ! aacparse ! avdec_aac ! audioconvert ! audioresample ! audiorate ! avenc_aac bitrate=96000 compliance=-2 ! aacparse ! queue ! m.

This "sort" or works. I get video. Just no audio.

Below is some of the data. Not sure what the error is on libva() - that may be the whole issue.

also there is about a 3 second latency - any way to reduce that ??/

gst-inspect-1.0 | grep aac
libav: avmux_adts: libav ADTS AAC (Advanced Audio Coding) muxer (not recommended, use aacparse instead)
libav: avdec_aac_latm: libav AAC LATM (Advanced Audio Coding LATM syntax) decoder
libav: avdec_aac_fixed: libav AAC (Advanced Audio Coding) decoder
libav: avdec_aac: libav AAC (Advanced Audio Coding) decoder
libav: avenc_aac: libav AAC (Advanced Audio Coding) encoder
fluaacdec: fluaacdec: Fluendo AAC Decoder (HE enabled, stereo downmix)
voaacenc: voaacenc: AAC audio encoder
typefindfunctions: audio/aac: aac, adts, adif, loas
audioparsers: aacparse: AAC audio stream parser

jerry

libva error: va_getDriverName() failed with unknown libva error,driver_name=(null)
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://192.168.2.5
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
Progress: (request) SETUP stream 0
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager: latency = 0
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager: ntp-sync = false
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager: rfc7273-sync = false
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager: ntp-time-source = NTP time based on realtime clock
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager: drop-on-latency = false
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager: max-rtcp-rtp-time-diff = 1000
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager: buffer-mode = Slave receiver to sender clock
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc0: timeout = 5000000000
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc0: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc1: caps = application/x-rtcp
Progress: (request) SETUP stream 1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc2: timeout = 5000000000
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc2: caps = application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1, profile-level-id=(string)41, streamtype=(string)5, mode=(string)AAC-hbr, objecttype=(string)64, constantduration=(string)1024, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)12100000000000000000000000000000, ssrc=(uint)4000664930
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc3: caps = application/x-rtcp
Progress: (open) Opened Stream
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager: buffer-mode = Slave receiver to sender clock
Progress: (request) Sending PLAY request
Progress: (request) Sending PLAY request
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc0: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc2: caps = application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1, profile-level-id=(string)41, streamtype=(string)5, mode=(string)AAC-hbr, objecttype=(string)64, constantduration=(string)1024, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)12100000000000000000000000000000, ssrc=(uint)4000664930, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc0.GstPad:src: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtp_sink_0.GstProxyPad:proxypad1: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc1.GstPad:src: caps = application/x-rtcp
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstFunnel:funnel0.GstFunnelPad:funnelpad0: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtp_sink_0: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtp_sink_0: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstFunnel:funnel1.GstFunnelPad:funnelpad1: caps = application/x-rtcp
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtcp_sink_0: caps = application/x-rtcp
Progress: (request) Sent PLAY request
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc2.GstPad:src: caps = application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1, profile-level-id=(string)41, streamtype=(string)5, mode=(string)AAC-hbr, objecttype=(string)64, constantduration=(string)1024, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)12100000000000000000000000000000, ssrc=(uint)4000664930, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtp_sink_1.GstProxyPad:proxypad4: caps = application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1, profile-level-id=(string)41, streamtype=(string)5, mode=(string)AAC-hbr, objecttype=(string)64, constantduration=(string)1024, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)12100000000000000000000000000000, ssrc=(uint)4000664930, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc3.GstPad:src: caps = application/x-rtcp
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstFunnel:funnel2.GstFunnelPad:funnelpad2: caps = application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1, profile-level-id=(string)41, streamtype=(string)5, mode=(string)AAC-hbr, objecttype=(string)64, constantduration=(string)1024, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)12100000000000000000000000000000, ssrc=(uint)4000664930, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtp_sink_1: caps = application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1, profile-level-id=(string)41, streamtype=(string)5, mode=(string)AAC-hbr, objecttype=(string)64, constantduration=(string)1024, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)12100000000000000000000000000000, ssrc=(uint)4000664930, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtcp_sink_1.GstProxyPad:proxypad5: caps = application/x-rtcp
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstFunnel:funnel3.GstFunnelPad:funnelpad3: caps = application/x-rtcp
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtcp_sink_1: caps = application/x-rtcp
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc0: timeout = 0
/GstPipeline:pipeline0/GstRTSPSrc:d/GstUDPSrc:udpsrc2: timeout = 0
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstFunnel:funnel0.GstPad:src: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstRtpSession:rtpsession0.GstPad:recv_rtp_src: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstRtpSsrcDemux:rtpssrcdemux0.GstPad:sink: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstRtpSession:rtpsession0.GstPad:recv_rtp_sink: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:sink: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:sink: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager/GstRtpPtDemux:rtpptdemux0.GstPad:sink: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0.GstPad:src: caps = video/x-h264, stream-format=(string)avc, alignment=(string)au, codec_data=(buffer)0164001fffe1000f2764001fac56c05005ba6a0202020401000428ee3cb0, level=(string)3.1, profile=(string)high
/GstPipeline:pipeline0/GstH264Parse:h264parse0.GstPad:sink: caps = video/x-h264, stream-format=(string)avc, alignment=(string)au, codec_data=(buffer)0164001fffe1000f2764001fac56c05005ba6a0202020401000428ee3cb0, level=(string)3.1, profile=(string)high
/GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0.GstPad:sink: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d.GstGhostPad:recv_rtp_src_0_3065589820_96.GstProxyPad:proxypad8: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstRTSPSrc:d/GstRtpBin:manager.GstGhostPad:recv_rtp_src_0_3065589820_96.GstProxyPad:proxypad7: caps = application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)"J2QAH6xWwFAFumoCAgIE\,KO48sA\=\=", profile-level-id=(string)64001F, ssrc=(uint)3065589820, npt-start=(guint64)0, play-speed=(double)1, play-scale=(double)1
/GstPipeline:pipeline0/GstH264Parse:h264parse0.GstPad:src: caps = video/x-h264, stream-format=(string)byte-stream, alignment=(string)au, level=(string)3.1, profile=(string)high, width=(int)1280, height=(int)720, framerate=(fraction)0/1, interlace-mode=(string)progressive, parsed=(boolean)true
/GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = video/x-h264, stream-format=(string)byte-stream, alignment=(string)au, level=(string)3.1, profile=(string)high, width=(int)1280, height=(int)720, framerate=(fraction)0/1, interlace-mode=(string)progressive, parsed=(boolean)true
/GstPipeline:pipeline0/GstQueue:queue0.GstPad:src: caps = video/x-h264, stream-format=(string)byte-stream, alignment=(string)au, level=(string)3.1, profile=(string)high, width=(int)1280, height=(int)720, framerate=(fraction)0/1, interlace-mode=(string)progressive, parsed=(boolean)true

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Jerry Geis-2

Re: rtsp to hlssink with audio

If I run

gst-launch-1.0 playbin uri=rtsp://192.168.2.5

it plays with audio and video.

How do I "echo" or "export" what pipeline it being built ?

Jerry

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Michael Prendergast

Re: rtsp to hlssink with audio

Hi Jerry,

If you set the GST_DEBUG environment variable to an integer > 0 you will get more debug output printed to stderr.

There are more variables and ways to use them available here: https://gstreamer.freedesktop.org/documentation/tutorials/basic/debugging-tools.html

If you are on a *nix system I would recommend:
export GST_DEBUG=5

gst-launch-1.0 playbin uri=rtsp://192.168.2.5 2>&1 | grep 'creating element'

Regards,

Michael

On Wed, Dec 18, 2019 at 9:56 AM Jerry Geis <[hidden email]> wrote:

If I run
gst-launch-1.0 playbin uri=rtsp://192.168.2.5

it plays with audio and video.
How do I "echo" or "export" what pipeline it being built ?

Jerry

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Rand Graham-2

RE: rtsp to hlssink with audio

In reply to this post by Jerry Geis-2

Hi,

The first thing I notice is that it looks like you have an extra aacparse in your pipeline. The last aacparse before the queue and then the mux seems unneeded.

Here is an example of an hlssink pipeline that I have used with a file source.

// HLS Sink example

gst-launch-1.0 -v -e mpegtsmux name=m m. ! queue ! hlssink filesrc location=/home/rdg/data/the_canoe_h265.ts ! tsdemux name=d d. ! queue max-size-buffers=1200 max-size-buffers=0 max-size-time=0 ! h265parse ! mfxhevcdec ! videoconvert ! mfxvpp width=1280 height=720 ! mfxh264enc rate-control=cbr bitrate=1500 insert-aud=true ! "video/x-h264, stream-format=byte-stream, profile=high" ! m. d. ! queue max-size-buffers=1200 max-size-buffers=0 max-size-time=0 ! aacparse ! avdec_aac ! audioconvert ! voaacenc bitrate=128000 ! m

I use queues after the demux. (I have never used a queue prior to the mux)

Regards,

Rand

From: gstreamer-devel [mailto:[hidden email]] On Behalf Of Jerry Geis
Sent: Tuesday, December 17, 2019 2:29 PM
To: [hidden email]
Subject: rtsp to hlssink with audio

HI - all I am taking RTSP camera to hlssink and trying to get audio to work.

This "sort" or works. I get video. Just no audio.

Below is some of the data. Not sure what the error is on libva() - that may be the whole issue.

also there is about a 3 second latency - any way to reduce that ??/

jerry

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Rand Graham-2

RE: rtsp to hlssink with audio

In reply to this post by Jerry Geis-2

Hi,

Have you tried generating a pipeline graph as described here:

https://gstreamer.freedesktop.org/documentation/tutorials/basic/debugging-tools.html?gi-language=c

This may help. In my experience, the generated graph from one of the automated bins ends up being highly complex and difficult to follow.

Regards,

Rand

From: gstreamer-devel [mailto:[hidden email]] On Behalf Of Jerry Geis
Sent: Tuesday, December 17, 2019 2:56 PM
To: [hidden email]
Subject: Re: rtsp to hlssink with audio

If I run

gst-launch-1.0 playbin uri=rtsp://192.168.2.5

it plays with audio and video.

How do I "echo" or "export" what pipeline it being built ?

Jerry

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[hidden email]
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Jerry Geis-2

Re: rtsp to hlssink with audio

In reply to this post by Jerry Geis-2

So I tried a couple things.

1) removed the extra aacparse.

2) changed to use decodebin for the audio (the extra queue items are just space fillers, I presume they dont matter).

3) tried to look at playbin output | grep "creating element". It basically showed the elements I was using were correct.

gst-launch-1.0 mpegtsmux name=m ! hlssink playlist-root=/ location=hlssink.camera.1636.%05d.ts target-duration=1 max-files=3 playlist-length=2 playlist-location=playlist.camera.1636.m3u8 rtspsrc latency=0 location=rtsp://192.168.2.5 name=d d. ! rtph264depay ! h264parse config-interval=-1 ! queue ! m. d. ! decodebin ! queue ! queue ! audioconvert ! audioresample ! audiorate ! avenc_aac bitrate=96000 compliance=-2 ! queue ! m.

I am not seeing why the audio is not working here. Again playbin output to my LCD works just fine with video and audio.

Thanks for the suggestions. I could not try the other pipeline as my computer did not have those elements.

Jerry

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Rand Graham-2

RE: rtsp to hlssink with audio

I don’t know why you are not getting audio.

I did want to make a note on the queue element. My understanding of the queue element is as follows.

1) It creates a new thread.

2) Data must fill the buffer in the queue element before it can flow through the rest of the pipeline. This could impact your pipeline.

From: gstreamer-devel [mailto:[hidden email]] On Behalf Of Jerry Geis
Sent: Wednesday, December 18, 2019 10:36 AM
To: [hidden email]
Subject: Re: rtsp to hlssink with audio

So I tried a couple things.

1) removed the extra aacparse.

2) changed to use decodebin for the audio (the extra queue items are just space fillers, I presume they dont matter).

3) tried to look at playbin output | grep "creating element". It basically showed the elements I was using were correct.

gst-launch-1.0 mpegtsmux name=m ! hlssink playlist-root=/ location=hlssink.camera.1636.%05d.ts target-duration=1 max-files=3 playlist-length=2 playlist-location=playlist.camera.1636.m3u8 rtspsrc latency=0 location=rtsp://192.168.2.5 name=d d. ! rtph264depay ! h264parse config-interval=-1 ! queue ! m. d. ! decodebin ! queue ! queue ! audioconvert ! audioresample ! audiorate ! avenc_aac bitrate=96000 compliance=-2 ! queue ! m.

I am not seeing why the audio is not working here. Again playbin output to my LCD works just fine with video and audio.

Thanks for the suggestions. I could not try the other pipeline as my computer did not have those elements.

Jerry

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Jerry Geis-2

Re: rtsp to hlssink with audio

In reply to this post by Jerry Geis-2

Taking out the two extra queue did not make a difference

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Jerry Geis-2

Re: rtsp to hlssink with audio

HI All - still trying to get audio.

My camera source is an app from an Iphone called LiveReporter.

Video works fine - just cant get audio at all.

Can anyone assist in getting the audio to work with hlssink?

my command is:

Thanks,

Jerry

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David Ing

Re: rtsp to hlssink with audio

I wonder if hlssink2 would fix it... It is slightly more advanced. (When using hlssink2, mpegtsmux can be removed IIRC.)

Its just a shot in the dark... I have no idea what is wrong.

On Thu, Dec 19, 2019, 5:58 AM Jerry Geis <[hidden email]> wrote:

HI All - still trying to get audio.

My camera source is an app from an Iphone called LiveReporter.
Video works fine - just cant get audio at all.

Can anyone assist in getting the audio to work with hlssink?

my command is:
gst-launch-1.0 mpegtsmux name=m ! hlssink playlist-root=/ location=hlssink.camera.1636.%05d.ts target-duration=1 max-files=3 playlist-length=2 playlist-location=playlist.camera.1636.m3u8 rtspsrc latency=0 location=rtsp://192.168.2.5 name=d d. ! rtph264depay ! h264parse config-interval=-1 ! queue ! m. d. ! rtpmp4gdepay ! aacparse ! avdec_aac ! audioconvert ! audioresample ! audiorate ! avenc_aac bitrate=96000 compliance=-2 ! m.

Thanks,

Jerry
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Jerry Geis-2

Re: rtsp to hlssink with audio

In reply to this post by Jerry Geis-2

I am currently using 1.12.3 and hlssink2 is not present.

Thanks,

Jerry

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Jerry Geis-2

Re: rtsp to hlssink with audio

OK so If I use playbin (which I get audio from).

gst-launch-1.0 playbin uri=rtsp://192.168.2.5 --gst-debug=4 2>&1 | grep mpeg
0:00:02.814015040 53785 0x7f874003d370 INFO GST_EVENT gstevent.c:809:gst_event_new_caps: creating caps event audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, codec_data=(buffer)12100000000000000000000000000000
0:00:02.816814546 53785 0x7f874003d370 INFO GST_EVENT gstevent.c:809:gst_event_new_caps: creating caps event audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, codec_data=(buffer)12100000000000000000000000000000, framed=(boolean)true, level=(string)2, base-profile=(string)lc, profile=(string)lc, rate=(int)44100, channels=(int)2
0:00:02.839761154 53785 0x7f874003d370 INFO GST_EVENT gstevent.c:809:gst_event_new_caps: creating caps event audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, codec_data=(buffer)12100000000000000000000000000000, framed=(boolean)true, level=(string)2, base-profile=(string)lc, profile=(string)lc, rate=(int)44100, channels=(int)2

I get the above

But then if I run my hlssink command I get this for mpeg audio

0:00:00.023898648 55262 0xc798c0 INFO GST_PIPELINE gstparse.c:334:gst_parse_launch_full: parsing pipeline description 'rtspsrc name=myrtsp location=rtsp://192.168.2.5 ! queue ! rtph264depay ! h264parse config-interval=-1 ! mpegtsmux name=mymux ! hlssink playlist-root=/ location=hlssink/hlssink.camera.1636.%05d.ts target-duration=1 max-files=3 playlist-length=2 playlist-location=playlist.camera.1636.m3u8 myrtsp. ! queue ! rtpmp4gdepay ! aacparse ! avdec_aac ! audioconvert ! audioresample ! audiorate ! avenc_aac bitrate=96000 compliance=0 ! mymux. '

0:00:00.028952382 55262 0xc798c0 INFO GST_PLUGIN_LOADING gstplugin.c:843:_priv_gst_plugin_load_file_for_registry: plugin "/usr/lib64/gstreamer-1.0/libgstmpegtsmux.so" loaded
0:00:00.028958502 55262 0xc798c0 INFO GST_ELEMENT_FACTORY gstelementfactory.c:361:gst_element_factory_create: creating element "mpegtsmux"
0:00:00.035524522 55262 0xc798c0 INFO GST_TYPEFIND gsttypefind.c:72:gst_type_find_register: registering typefind function for avtype_yuv4mpegpipe
0:00:02.232670897 55262 0xf12de0 INFO GST_EVENT gstevent.c:809:gst_event_new_caps: creating caps event audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, codec_data=(buffer)12100000000000000000000000000000
0:00:02.234589109 55262 0xf12de0 INFO GST_EVENT gstevent.c:809:gst_event_new_caps: creating caps event audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, codec_data=(buffer)12100000000000000000000000000000, framed=(boolean)true, level=(string)2, base-profile=(string)lc, profile=(string)lc, rate=(int)44100, channels=(int)2
0:00:02.242827468 55262 0xf12de0 INFO libav gstavaudenc.c:271:gst_ffmpegaudenc_set_format:<avenc_aac0> Setting avcontext to bitrate 96000
0:00:02.245016298 55262 0xf12de0 INFO GST_EVENT gstevent.c:809:gst_event_new_caps: creating caps event audio/mpeg, channels=(int)1, rate=(int)44100, mpegversion=(int)4, stream-format=(string)raw, base-profile=(string)lc, level=(string)2, profile=(string)lc, codec_data=(buffer)120856e500
0:00:02.245249299 55262 0xf12de0 INFO GST_EVENT gstevent.c:809:gst_event_new_caps: creating caps event video/mpegts, systemstream=(boolean)true, packetsize=(int)188
0:00:02.245363954 55262 0xf12de0 INFO GST_EVENT gstevent.c:809:gst_event_new_caps: creating caps event video/mpegts, systemstream=(boolean)true, packetsize=(int)188, streamheader=(buffer)< 47400030a600ffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff0000b00d0001c100000001e020a2c32941, 474020308000ffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff0002b0330001c10000e042f00c050448444d5688040ffffcfc1be041f00a050848444d56ff1b443f0fe042f0060a04656e0000bd632c38 >

Does that help to tell why I am not getting audio on hlssink ?

First off - what is "parse launch full" ???

second - any clues pop up ?

Thanks,

Jerry

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