long run gstreamer stops working without message
Thomas DEBESSE
thomas.debesse at rcf.fr
Wed Dec 5 02:53:29 PST 2012
Hello, I always encounters strange behavior when I use gstreamer for
long durations.
For example, if I use this pipeline :
gst-launch-0.10 jackaudiosrc ! "audio/x-raw-float" ! audioconvert !
audioresample ! "audio/x-raw-int, rate=44100, width=16" ! alsasink
device="hw:1" sync=0
On this example, the alsasink module uses the soundcard directly,
without alsa any module (no plug, no dmix).
After about 14 hours, the pipeline stop to play sound. And logs say nothing !
I have no other solutions : I must restart the pipeline regularly (set
gst.STATE_NULL then set gst.STATE_PLAYING).
I encounter similar problems with various pipelines, as I wrote a few
months ago. The common point is that it breaks after several hours of
usage, and I use the modules audioresample, audioconvert, and
alsasink.
I use audioresample and audioconvert in other pipelines without
alsasink, and I have no problem when I use some others modules that
are not alsasink (I don't have problems with jackaudiosink and
udpsink).
Does someone have an idea ?
2012/7/6 Thomas DEBESSE <thomas.debesse at rcf.fr>:
> Hello, I have a strange problem : I use gstreamer to stream sound
> between two computer, it's work nicely but after a few hours, I get
> silence without information in the logs. I use also gstreamer to
> connect two audio interfaces (one play the sound recorded by the
> other). It's work nicely but after a few hours, I get noise (the sound
> is completely distorted).
>
> ## First case
>
> On a computer (sender)
>
> export GST_DEBUG="3"
>
> gst-launch-0.10 -v alsasrc device=intel_in \
> ! audioconvert \
> ! audioresample \
> ! queue max-size-buffers=100 leaky=downstream \
> ! "audio/x-raw-int,rate=44100,width=16,channels=2" \
> ! celtenc cbr=true bitrate=64 framesize=128 \
> ! rtpceltpay \
> ! udpsink host=prjevalski port=4951
>
> I have also tested with "queue max-size-buffers=100 leaky=downstream"
> after "audioresample" element.
>
> on another computer (receiver)
>
> export GST_DEBUG="3"
>
> caps="application/x-rtp, media=(string)audio, clock-rate=(int)44100,
> encoding-name=(string)CELT, encoding-params=(string)2,
> frame-size=(string)480, payload=(int)96"
>
> gst-launch-0.10 -v udpsrc uri="udp://prjevalski:4951" caps="$caps" \
> ! gstrtpjitterbuffer latency=60 drop-on-latency=true \
> ! rtpceltdepay \
> ! celtdec \
> ! audioconvert \
> ! audioresample \
> ! "audio/x-raw-int,rate=44100,width=16,channels=2" \
> ! alsasink device=nvidia_front sync=false
>
> I've also tested with "queue max-size-buffers=100 leaky=downstream"
> before "audioconvert" element.
>
> The sound is transmitted for several hours before I get silence.
>
> For example, at 5 PM yesterday I launched these commands, the last
> gstreamer's output on sender and receiver are form 5 am, but this
> morning but no sound came out.
> In tcpdump, I still see a rtp stream
>
> I built the same pipeline using python.
> When silence occur, a simple (stopping and replaying the receiver
> gstreamer pipeline)
>
> receiver.set_state(gst.STATE_NULL)
> receiver.set_state(gst.STATE_PLAYING)
>
> gives me the sound.
>
> Both computers are on a local network, (I first had this problem using
> internet but using a local network the bug also occurs)
> I configure my alsa interface in this way (using both plug and
> dmix/dsnoop module) :
>
> pcm.intel_front {
> type plug
> slave.pcm {
> type dmix
> ipc_key 1024
> slave {
> pcm "front:0,0"
> }
> }
> }
>
> pcm.intel_in {
> type plug
> slave.pcm {
> type dsnoop
> ipc_key 1024
> slave {
> pcm "hw:0,0"
> }
> }
> }
>
> I tested these alsa interfaces :
>
> lspci :
> 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM
> (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01)
> 05:04.0 Multimedia audio controller: Creative Labs CA0106 Soundblaster
> # Audigy SE
> 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2)
> lsusb :
> Bus 003 Device 003: ID 06f8:c001 Guillemot Corp. # Muse Pocket LT3
>
> I get the bug with all of them.
>
> ## Second case
>
> Also, I use another pipeline :
> export GST_DEBUG="3"
>
> gst-launch-0.10 -v alsasrc device=intel_in \
> ! audioconvert \
> ! audioresample \
> ! "audio/x-raw-int,rate=44100,width=16,channels=2" \
> ! alsasink device=audigy_front sync=false \
>
> I have also tested with a "queue" after "audioresample" element.
>
> The purpose of this pipeline is simply to play on a alsa interface
> which is recorded from another alsa interface.
> The sound is okay for several hours until it becomes suddenly a very
> ugly mixture.
> I've launched this pipeline yesterday at about 5 PM, the sound was
> ugly this morning, and in a recording the sound is good and then the
> sound deteriorates suddenly.
>
> The logs report nothing since 5 PM :
>
> 0:00:00.446042222 27052 0x98e6640 INFO GST_STATES
> gstbin.c:2443:gst_bin_change_state_func:<pipeline0> child 'alsasink0'
> changed state to 4(PLAYING) successfully
> 0:00:00.446067645 27052 0x98e6640 INFO GST_STATES
> gstelement.c:2236:gst_element_continue_state:<capsfilter0> completed
> state change to PLAYING
> 0:00:00.446083010 27052 0x98e6640 INFO GST_STATES
> gstbin.c:2443:gst_bin_change_state_func:<pipeline0> child
> 'capsfilter0' changed state to 4(PLAYING) successfully
> 0:00:00.446100890 27052 0x98e6640 INFO GST_STATES
> gstelement.c:2236:gst_element_continue_state:<queue0> completed state
> change to PLAYING
> 0:00:00.446114859 27052 0x98e6640 INFO GST_STATES
> gstbin.c:2443:gst_bin_change_state_func:<pipeline0> child 'queue0'
> changed state to 4(PLAYING) successfully
> 0:00:00.446132459 27052 0x98e6640 INFO GST_STATES
> gstelement.c:2236:gst_element_continue_state:<audioresample0>
> completed state change to PLAYING
> 0:00:00.446146428 27052 0x98e6640 INFO GST_STATES
> gstbin.c:2443:gst_bin_change_state_func:<pipeline0> child
> 'audioresample0' changed state to 4(PLAYING) successfully
> 0:00:00.446163749 27052 0x98e6640 INFO GST_STATES
> gstelement.c:2236:gst_element_continue_state:<audioconvert0> completed
> state change to PLAYING
> 0:00:00.446177159 27052 0x98e6640 INFO GST_STATES
> gstbin.c:2443:gst_bin_change_state_func:<pipeline0> child
> 'audioconvert0' changed state to 4(PLAYING) successfully
> 0:00:00.446193642 27052 0x98e6640 INFO GST_STATES
> gstelement.c:2236:gst_element_continue_state:<alsasrc0> completed
> state change to PLAYING
> 0:00:00.446207331 27052 0x98e6640 INFO GST_STATES
> gstbin.c:2443:gst_bin_change_state_func:<pipeline0> child 'alsasrc0'
> changed state to 4(PLAYING) successfully
> 0:00:00.446226608 27052 0x98e6640 INFO GST_STATES
> gstelement.c:2236:gst_element_continue_state:<pipeline0> completed
> state change to PLAYING
> 0:00:00.446240297 27052 0x98e6640 INFO GST_STATES
> gstelement.c:2249:gst_element_continue_state:<pipeline0> posting
> state-changed PAUSED to PLAYING
> [then nothing]
>
> ## Some system information
>
> I'm using Debian Squeeze with Gstreamer 0.10
>
> some "apt-cache show" output :
>
> libgstreamer0.10-0
> Version: 0.10.30-1
> libcelt0-0
> Version: 0.7.1-1
> libasound2
> Version: 1.0.23-2.1
>
> some "gst-inspect-0.10" output :
>
> alsasrc
> Long name: Audio source (ALSA)
> Filename: /usr/lib/gstreamer-0.10/libgstalsa.so
> Version: 0.10.30
> Source module: gst-plugins-base
> alsasink
> Long name: Audio sink (ALSA)
> Filename: /usr/lib/gstreamer-0.10/libgstalsa.so
> Version: 0.10.30
> Source module: gst-plugins-base
> audioconvert
> Long name: Audio converter
> Filename: /usr/lib/gstreamer-0.10/libgstaudioconvert.so
> Version: 0.10.30
> Source module: gst-plugins-base
> audioresample
> Long name: Audio resampler
> Filename: /usr/lib/gstreamer-0.10/libgstaudioresample.so
> Version: 0.10.30
> Source module: gst-plugins-base
> queue
> Long name: Queue
> Filename: /usr/lib/gstreamer-0.10/libgstcoreelements.so
> Version: 0.10.30
> Source module: gstreamer
> celtenc
> Long name: Celt audio encoder
> Filename: /usr/lib/gstreamer-0.10/libgstcelt.so
> Version: 0.10.19
> Source module: gst-plugins-bad
> celtdec
> Long name: Celt audio decoder
> Filename: /usr/lib/gstreamer-0.10/libgstcelt.so
> Version: 0.10.19
> Source module: gst-plugins-bad
> rtpceltpay
> Long name: RTP CELT payloader
> Filename: /usr/lib/gstreamer-0.10/libgstrtp.so
> Version: 0.10.24
> Source module: gst-plugins-good
> rtpceltdepay
> Long name: RTP CELT depayloader
> Filename: /usr/lib/gstreamer-0.10/libgstrtp.so
> Version: 0.10.24
> Source module: gst-plugins-good
> gstrtpjitterbuffer
> Long name: RTP packet jitter-buffer
> Filename: /usr/lib/gstreamer-0.10/libgstrtpmanager.so
> Version: 0.10.24
> Source module: gst-plugins-good
> udpsrc
> Long name: UDP packet receiver
> Filename: /usr/lib/gstreamer-0.10/libgstudp.so
> Version: 0.10.24
> Source module: gst-plugins-good
> udpsink
> Long name: UDP packet sender
> Filename: /usr/lib/gstreamer-0.10/libgstudp.so
> Version: 0.10.24
> Source module: gst-plugins-good
>
> Do you think I forgot a few items or settings, or have I found a bug?
>
> Thanks!
>
> --
> Thomas DEBESSE
> RCF Méditerrannée
--
Thomas DEBESSE
RCF Méditerrannée
More information about the gstreamer-devel
mailing list