Hi, Anuroop,
Do you have any idea?
the case I:
it is fail, the error is the same as previous.
====================================================
# gst-launch-0.10 -v filesrc location=/bin/audio_src_48k_le.wav ! decodebin ! al
sasink
Setting pipeline to PAUSED ...
conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3686, name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3700, lf->name=/usr/share/alsa/alsa.conf
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src:
caps = audio/x-wav
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0.GstPad:sink:
caps = audio/x-wav
Pipeline is PREROLLING ...
** (gst-launch-0.10:1083): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1083): WARNING **: offset: 44 , end: 960044 ,
dataleft: 960000
** (gst-launch-0.10:1083): WARNING **: Fetching 7680 bytes of data
from the sinkpad
conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3686, name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3700, lf->name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3686, name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3700, lf->name=/usr/share/alsa/alsa.conf
/GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:src0: caps
= audio/x-raw-int, endianness=(int)1
234, channels=(int)2, width=(int)16, depth=(int)16,
signed=(boolean)true, rate=(int)48000
/GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:src0.GstProxyPad:proxypad1:
caps = audio/x-raw-
int, endianness=(int)1234, channels=(int)2, width=(int)16,
depth=(int)16, signed=(boolean)true, rate=(int)
48000
** (gst-launch-0.10:1083): WARNING **: gstwavparse.c, gst_wavparse_loop,2074
ERROR: from element
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0:
Internal data fl
ow error.
Additional debug info:
gstwavparse.c(2122): gst_wavparse_loop ():
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavp
arse0:
streaming task paused, reason not-linked (-1)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
/GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:src0: caps = NULL
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0.GstPad:src:
caps = NULL
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0.GstPad:sink:
caps = NULL
/GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src:
caps = NULL
Freeing pipeline ...
====================================================
And
The case II:
it is workable, the log below
=========================================================
# gst-launch-0.10 -v playbin uri=file:///bin/audio_src_48k_le.wav
Setting pipeline to PAUSED ...
/GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src:
caps = audio/x-wav
/GstPlayBin:playbin0/GstDecodeBin:decodebin0/GstWavParse:wavparse0.GstPad:sink:
caps = audio/x-wav
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 44 , end: 960044 ,
dataleft: 960000
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
Pipeline is PREROLLING ...
/GstPlayBin:playbin0/GstStreamSelector:selector_audio_src0: active-pad = NULL
conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3686, name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3700, lf->name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3686, name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3700, lf->name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3686, name=/usr/share/alsa/alsa.conf
conf.c:snd_config_update_r:3700, lf->name=/usr/share/alsa/alsa.conf
** (gst-launch-0.10:1088): WARNING **: could not link ANY: -4
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPlayBin:playbin0/GstStreamSelector:selector_audio_src0.GstPlaybinSelectorPad:sink0:
caps = audio/x-raw
-int, endianness=(int)1234, channels=(int)2, width=(int)16,
depth=(int)16, signed=(boolean)true, rate=(int
)48000
/GstPlayBin:playbin0/GstDecodeBin:decodebin0.GstGhostPad:src0: caps =
audio/x-raw-int, endianness=(int)123
4, channels=(int)2, width=(int)16, depth=(int)16,
signed=(boolean)true, rate=(int)48000
/GstPlayBin:playbin0/GstDecodeBin:decodebin0.GstGhostPad:src0.GstProxyPad:proxypad1:
caps = audio/x-raw-in
t, endianness=(int)1234, channels=(int)2, width=(int)16,
depth=(int)16, signed=(boolean)true, rate=(int)48
000
/GstPlayBin:playbin0/GstStreamSelector:selector_audio_src0.GstPad:src:
caps = audio/x-raw-int, endianness=
(int)1234, channels=(int)2, width=(int)16, depth=(int)16,
signed=(boolean)true, rate=(int)48000
/GstPlayBin:playbin0/GstQueue:preroll_audio_src0.GstPad:sink: caps =
audio/x-raw-int, endianness=(int)1234
, channels=(int)2, width=(int)16, depth=(int)16, signed=(boolean)true,
rate=(int)48000
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 7724 , end: 960044 ,
dataleft: 952320
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 15404 , end: 960044 ,
dataleft: 944640
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 23084 , end: 960044 ,
dataleft: 936960
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 30764 , end: 960044 ,
dataleft: 929280
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 38444 , end: 960044 ,
dataleft: 921600
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 46124 , end: 960044 ,
dataleft: 913920
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 53804 , end: 960044 ,
dataleft: 906240
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 61484 , end: 960044 ,
dataleft: 898560
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 69164 , end: 960044 ,
dataleft: 890880
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 76844 , end: 960044 ,
dataleft: 883200
** (gst-launch-0.10:1088): WARNING **: Fetching 7680 bytes of data
from the sinkpad
** (gst-launch-0.10:1088): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:1088): WARNING **: offset: 84524 , end: 960044 ,
dataleft: 875520
=======================================================
2012/7/30 Anuroop Jesu <jesuas at gmail.com>:
> Hi Soho,
>
> I still feel it is something to do with the caps mismatch
>
> 1. Can you try
> gst-launch -v filesrc location=<file path> ! decodebin ! alsasink
> or
> gst-launch -v playbin uri=file:///path/to/somefile.wav
>
> This should work for you.
>
> With Warm Regards
> Jesu Anuroop Suresh
>
> "Any intelligent fool can make things bigger, more complex, and more
> violent. It takes a touch of genius -- and a lot of courage -- to move in
> the opposite direction."
> "Anyone who has never made a mistake has never tried anything new."
>
>
>
>
>
>
> On Mon, Jul 30, 2012 at 6:51 PM, Soho Soho123 <soho123.2012 at gmail.com>
> wrote:
>>
>> Hi Anuroop,
>>
>> It seems the issue is H/W releated.
>> When I use fakesink,
>> it is OK, like attached log.
>> How about the verification to hw alsasink?
>>
>> Thanks!
>>
>> 2012/7/30 Anuroop Jesu <jesuas at gmail.com>:
>> > Hi Soho,
>> >
>> > replace the alsasink with the fakesink which actually does not anything
>> > except accepting all sink buffers and discarding instead of actually
>> > writing
>> > to hardware.
>> >
>> > With Warm Regards
>> > Jesu Anuroop Suresh
>> >
>> > "Any intelligent fool can make things bigger, more complex, and more
>> > violent. It takes a touch of genius -- and a lot of courage -- to move
>> > in
>> > the opposite direction."
>> > "Anyone who has never made a mistake has never tried anything new."
>> >
>> >
>> >
>> >
>> >
>> >
>> > On Mon, Jul 30, 2012 at 6:36 PM, Soho Soho123 <soho123.2012 at gmail.com>
>> > wrote:
>> >>
>> >> Hi Anuroop,
>> >>
>> >> In item 2 you mentioned, I have tried. it is fail, too.
>> >> And in item 1,
>> >> Could you explain more deatil?
>> >> how to set the gst-launch command?
>> >>
>> >> Thanks!
>> >>
>> >>
>> >>
>> >>
>> >> 2012/7/30 Anuroop Jesu <jesuas at gmail.com>:
>> >> > Hi Soho,
>> >> >
>> >> > To isolate the problem further
>> >> >
>> >> > 1. We can very the pipeline to do that instead of alsasink use the
>> >> > fakesink
>> >> > if it works then pipeline is clean and we just need to check whats
>> >> > wrong
>> >> > with sink device.
>> >> >
>> >> > 2. Try the same pipeline without specifying any alsasink device let
>> >> > it
>> >> > pickup the default as in case of aplay you mentioned -D as defualt
>> >> >
>> >> > It appears to have some thing to do with the pipeline or caps
>> >> > mismatch.
>> >> >
>> >> > With Warm Regards
>> >> > Jesu Anuroop Suresh
>> >> >
>> >> > "Any intelligent fool can make things bigger, more complex, and more
>> >> > violent. It takes a touch of genius -- and a lot of courage -- to
>> >> > move
>> >> > in
>> >> > the opposite direction."
>> >> > "Anyone who has never made a mistake has never tried anything new."
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Jul 30, 2012 at 5:58 PM, Soho Soho123
>> >> > <soho123.2012 at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> Hi ,
>> >> >>
>> >> >>
>> >> >> the level 3 error log shows:
>> >> >> 0:00:11.910000000 946 0x4238f0 INFO typefindfunctions
>> >> >> gsttypefindfunctions.c:1267:mp3_type_find_
>> >> >> at_offset: audio/mpeg calculated 86 = 100 * 5 / 5 * (10000 -
>> >> >> 1676) /
>> >> >> 10000
>> >> >> 0:00:11.920000000 946 0x4238f0 INFO GST_ELEMENT_PADS
>> >> >> gstelement.c:728:gst_element_add_pad:<wavp
>> >> >> arse0> adding pad 'src'
>> >> >> 0:00:11.920000000 946 0x4238f0 INFO GST_PIPELINE
>> >> >> ./grammar.y:496:gst_parse_found_pad: tryin
>> >> >> g delayed linking wavparse0:(NULL) to audioconvert0:(NULL)
>> >> >> 0:00:11.920000000 946 0x4238f0 INFO GST_ELEMENT_PADS
>> >> >> gstutils.c:1698:gst_element_link_pads_full
>> >> >> : trying to link element wavparse0:(any) to element
>> >> >> audioconvert0:(any)
>> >> >> 0:00:11.920000000 946 0x4238f0 INFO GST_PADS
>> >> >> gstutils.c:1032:gst_pad_check_link: trying
>> >> >> to link wavparse0:src and audioconvert0:sink
>> >> >> 0:00:11.920000000 946 0x4238f0 WARN alsa
>> >> >> gstalsa.c:124:gst_alsa_detect_formats:<als
>> >> >> asink0> skipping non-int format
>> >> >> conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
>> >> >> conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
>> >> >> 0:00:11.930000000 946 0x4238f0 WARN alsa
>> >> >> conf.c:4692:snd_config_expand: alsalib err
>> >> >> or: Unknown parameters {AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
>> >> >> 0:00:11.930000000 946 0x4238f0 WARN alsa
>> >> >> pcm.c:2217:snd_pcm_open_noupdate: alsalib
>> >> >> error: Unknown PCM iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
>> >> >> 0:00:11.930000000 946 0x4238f0 INFO alsa
>> >> >> gstalsasink.c:327:gst_alsasink_getcaps:<al
>> >> >> sasink0> returning caps 0x49a960
>> >> >> 0:00:13.040000000 946 0x4238f0 INFO GST_ELEMENT_PADS
>> >> >> gstelement.c:975:gst_element_get_static_pa
>> >> >> d: found pad audioconvert0:sink
>> >> >> 0:00:13.040000000 946 0x4238f0 INFO GST_PADS
>> >> >> gstutils.c:1596:prepare_link_maybe_ghostin
>> >> >> g: wavparse0 and audioconvert0 in same bin, no need for ghost pads
>> >> >> 0:00:13.040000000 946 0x4238f0 INFO GST_PADS
>> >> >> gstpad.c:1978:gst_pad_link_prepare: trying
>> >> >> to link wavparse0:src and audioconvert0:sink
>> >> >> 0:00:13.040000000 946 0x4238f0 INFO GST_PADS
>> >> >> gstpad.c:2034:gst_pad_link_prepare: caps a
>> >> >> re incompatible
>> >> >> 0:00:13.040000000 946 0x4238f0 INFO GST_PADS
>> >> >> gstutils.c:1032:gst_pad_check_link: trying
>> >> >> to link wavparse0:src and audioconvert0:sink
>> >> >> 0:00:13.040000000 946 0x4238f0 INFO GST_ELEMENT_PADS
>> >> >> gstutils.c:1216:gst_element_get_compatible
>> >> >> _pad:<wavparse0> Could not find a compatible pad to link to
>> >> >> audioconvert0:sink
>> >> >> 0:00:13.040000000 946 0x4238f0 INFO default
>> >> >> gstutils.c:2037:gst_element_link_pads_filt
>> >> >> ered: Could not link pads: wavparse0:(null) - audioconvert0:(null)
>> >> >> 0:00:13.040000000 946 0x4238f0 INFO wavparse
>> >> >> gstwavparse.c:2039:gst_wavparse_stream_dat
>> >> >> a:<wavparse0> Error pushing on srcpad wavparse0:src, reason
>> >> >> not-linked, is linked? = 0
>> >> >>
>> >> >> ** (gst-launch-0.10:946): WARNING **: gstwavparse.c,
>> >> >> gst_wavparse_loop,2074
>> >> >> 0:00:13.050000000 946 0x4238f0 WARN wavparse
>> >> >> gstwavparse.c:2122:gst_wavparse_loop:<wavp
>> >> >> arse0> error: Internal data flow error.
>> >> >> 0:00:13.050000000 946 0x4238f0 WARN wavparse
>> >> >> gstwavparse.c:2122:gst_wavparse_loop:<wavp
>> >> >> =================================================
>> >> >>
>> >> >> What is the meaning ?
>> >> >>
>> >> >> 2012/7/30 Anuroop Jesu <jesuas at gmail.com>:
>> >> >> > Hi
>> >> >> >
>> >> >> > Use the --gst-debug-level=3 for the more detailed information for
>> >> >> > the
>> >> >> > error, in the gst-launch command
>> >> >> >
>> >> >> > Also check the your usb audio device properties with aplay using
>> >> >> > -v
>> >> >> > option.
>> >> >> >
>> >> >> > With Warm Regards
>> >> >> > Jesu Anuroop Suresh
>> >> >> >
>> >> >> > "Any intelligent fool can make things bigger, more complex, and
>> >> >> > more
>> >> >> > violent. It takes a touch of genius -- and a lot of courage -- to
>> >> >> > move
>> >> >> > in
>> >> >> > the opposite direction."
>> >> >> > "Anyone who has never made a mistake has never tried anything
>> >> >> > new."
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Mon, Jul 30, 2012 at 5:31 PM, Soho Soho123
>> >> >> > <soho123.2012 at gmail.com>
>> >> >> > wrote:
>> >> >> >>
>> >> >> >> Hi,
>> >> >> >>
>> >> >> >> after tracing the wavparse code,
>> >> >> >> the error is caused by
>> >> >> >> gst_wavparse_stream_data (GstWavParse * wav)
>> >> >> >> about line 1997, gst_pad_push get error,
>> >> >> >>
>> >> >> >> if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
>> >> >> >> goto push_error;
>> >> >> >>
>> >> >> >> Does anyone have idea about how to debug this kind of error?
>> >> >> >> Why USB Audio device cause this kind of error?
>> >> >> >> Because When I test the same audio file via I2S device, it is OK,
>> >> >> >> It is fail when I change to USB Audio device.
>> >> >> >> Anyone have idea?
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> 2012/7/30 Soho Soho123 <soho123.2012 at gmail.com>:
>> >> >> >> > Hi All,
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > Does anyone have idea about the log ?
>> >> >> >> > that gstreamer can not play wav file to usb audio alsa device.
>> >> >> >> >
>> >> >> >> > I use the command to play audio to usb alsa audio device.
>> >> >> >> > gst-launch-0.10 -v filesrc location=/bin/audio_src_48k_le.wav !
>> >> >> >> > wavparse ! audioconvert ! alsasink device="hw:0,0"
>> >> >> >> >
>> >> >> >> > It is OK by "aplay" utility, but it is fail by gstreamer launch
>> >> >> >> > ==============================================================
>> >> >> >> >
>> >> >> >> > 0:00:00.670000000 936 0x477720 DEBUG alsa
>> >> >> >> > gstalsasink.c:277:gst_alsasink_init:<GstAl
>> >> >> >> > saSink at 0x478c50> initializing alsasink
>> >> >> >> > 0:00:00.670000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x477960
>> >> >> >> > 0:00:00.670000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x477960
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c120
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c120
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c280
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c280
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c420
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c420
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c4a0
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c4a0
>> >> >> >> > 0:00:00.680000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c200
>> >> >> >> > 0:00:00.690000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c200
>> >> >> >> > 0:00:00.690000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c1a0
>> >> >> >> > 0:00:00.690000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c1a0
>> >> >> >> > 0:00:00.690000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47dfc0
>> >> >> >> > 0:00:00.690000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47dfc0
>> >> >> >> > 0:00:00.690000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c540
>> >> >> >> > 0:00:00.700000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c540
>> >> >> >> > 0:00:00.700000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47e080
>> >> >> >> > 0:00:00.700000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47e080
>> >> >> >> > 0:00:00.700000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47dd60
>> >> >> >> > 0:00:01.800000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47dd60
>> >> >> >> > 0:00:01.810000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c580
>> >> >> >> > 0:00:01.810000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c580
>> >> >> >> > 0:00:01.810000000 936 0x477720 DEBUG alsa
>> >> >> >> > gstalsasink.c:307:gst_alsasink_getcaps:<al
>> >> >> >> > sasink0> device not open, using template caps
>> >> >> >> > 0:00:01.810000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x477740
>> >> >> >> > 0:00:01.810000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x477740
>> >> >> >> > 0:00:01.810000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47ddc0
>> >> >> >> > 0:00:01.810000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47ddc0
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c1c0
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c1c0
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47dc80
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47dc80
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47dd20
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47dd20
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:611:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> step1: (2) 0x47c060
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG audioconvert
>> >> >> >> > gstaudioconvert.c:705:gst_audio_convert_tr
>> >> >> >> > ansform_caps:<audioconvert0> Caps transformed to 0x47c060
>> >> >> >> > 0:00:01.820000000 936 0x477720 DEBUG alsa
>> >> >> >> > gstalsasink.c:307:gst_alsasink_getcaps:<al
>> >> >> >> > sasink0> device not open, using template caps
>> >> >> >> > Setting pipeline to PAUSED ...
>> >> >> >> > 0:00:01.840000000 936 0x477720 LOG alsa
>> >> >> >> > gstalsasink.c:678:gst_alsasink_open:<alsas
>> >> >> >> > ink0> Opened device hw:0,0
>> >> >> >> > 0:00:01.840000000 936 0x477720 DEBUG wavparse
>> >> >> >> > gstwavparse.c:2607:gst_wavparse_sink_activ
>> >> >> >> > ate: going to pull mode
>> >> >> >> > 0:00:01.840000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:2050:gst_wavparse_loop:<wavp
>> >> >> >> > arse0> process data
>> >> >> >> > 0:00:01.840000000 940 0x4230f0 INFO wavparse
>> >> >> >> > gstwavparse.c:2054:gst_wavparse_loop:<wavp
>> >> >> >> > arse0> GST_WAVPARSE_START
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 INFO wavparse
>> >> >> >> > gstwavparse.c:2063:gst_wavparse_loop:<wavp
>> >> >> >> > arse0> GST_WAVPARSE_HEADER
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1232:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> creating the caps
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1288:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> blockalign = 4
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1289:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> width = 16
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1290:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> depth = 16
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1291:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> av_bps = 192000
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1292:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> frequency = 48000
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1293:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> channels = 2
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1294:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> bytes_per_sample = 4
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1300:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> bps = 192000
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1302:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> caps = 0x47c080
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1325:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> upstream size 982538
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 INFO wavparse
>> >> >> >> > gstwavparse.c:1343:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> Got TAG: data, offset 36
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1350:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> Got 'data' TAG, size : 960000
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1375:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> datasize = 960000
>> >> >> >> > Pipeline is PREROLLING ...
>> >> >> >> > 0:00:02.950000000 940 0x4230f0 INFO wavparse
>> >> >> >> > gstwavparse.c:1343:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> Got TAG: ID3x, offset 960044
>> >> >> >> > 0:00:02.960000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1152:gst_waveparse_ignore_ch
>> >> >> >> > unk:<wavparse0> Ignoring tag ID3x
>> >> >> >> > 0:00:02.960000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1554:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> Finished parsing headers
>> >> >> >> > 0:00:02.960000000 940 0x4230f0 INFO wavparse
>> >> >> >> > gstwavparse.c:1126:gst_wavparse_calculate_
>> >> >> >> > duration:<wavparse0> Got datasize 960000
>> >> >> >> > 0:00:02.960000000 940 0x4230f0 INFO wavparse
>> >> >> >> > gstwavparse.c:1130:gst_wavparse_calculate_
>> >> >> >> > duration:<wavparse0> Got duration (bps) 0:00:05.000000000
>> >> >> >> > 0:00:02.960000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:823:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> doing seek without event
>> >> >> >> > 0:00:02.960000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:897:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> stopped streaming at 0
>> >> >> >> > 0:00:04.060000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:916:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> cur_type =2
>> >> >> >> > 0:00:04.060000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:924:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> offset=0
>> >> >> >> > 0:00:04.060000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:926:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> offset=0
>> >> >> >> > 0:00:04.060000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:928:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> offset=44
>> >> >> >> > 0:00:04.060000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:937:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> end_offset=960000
>> >> >> >> > 0:00:04.060000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:939:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> end_offset=960000
>> >> >> >> > 0:00:04.060000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:941:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> end_offset=960044
>> >> >> >> > 0:00:04.060000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:960:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> seek: rate 1.000000, offset 44, end 960044,
>> >> >> >> > segment
>> >> >> >> > 0:00:00.000000000 -- 0:00:05.000000000
>> >> >> >> > 0:00:04.070000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:995:gst_wavparse_perform_see
>> >> >> >> > k:<wavparse0> Creating newsegment from 0 to 5000000000
>> >> >> >> > 0:00:04.070000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1600:gst_wavparse_stream_hea
>> >> >> >> > ders:<wavparse0> max buffer size 7680
>> >> >> >> > 0:00:04.070000000 940 0x4230f0 INFO wavparse
>> >> >> >> > gstwavparse.c:2069:gst_wavparse_loop:<wavp
>> >> >> >> > arse0> GST_WAVPARSE_DATA
>> >> >> >> >
>> >> >> >> > 0:00:04.070000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:1840:gst_wavparse_stream_dat
>> >> >> >> > a:<wavparse0> offset: 44 , end: 960044 , dataleft: 960000
>> >> >> >> >
>> >> >> >> > 0:00:04.070000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:1859:gst_wavparse_stream_dat
>> >> >> >> > a:<wavparse0> Fetching 7680 bytes of data from the sinkpad
>> >> >> >> >
>> >> >> >> > 0:00:04.070000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1773:gst_wavparse_add_src_pa
>> >> >> >> > d:<wavparse0> adding src pad
>> >> >> >> > 0:00:04.160000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:1783:gst_wavparse_add_src_pa
>> >> >> >> > d: typefind caps = 0x499ee0, P=86
>> >> >> >> > 0:00:04.160000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1793:gst_wavparse_add_src_pa
>> >> >> >> > d:<wavparse0> found caps 0x499ee0 for stream marked as raw PCM
>> >> >> >> > audio,
>> >> >> >> > but ignoring for now
>> >> >> >> > 0:00:04.160000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:247:gst_wavparse_create_sour
>> >> >> >> > cepad:<wavparse0> srcpad created
>> >> >> >> > 0:00:04.160000000 940 0x4230f0 WARN alsa
>> >> >> >> > gstalsa.c:124:gst_alsa_detect_formats:<als
>> >> >> >> > asink0> skipping non-int format
>> >> >> >> > 0:00:04.160000000 940 0x4230f0 LOG alsa
>> >> >> >> > gstalsa.c:30:gst_alsa_detect_rates:<alsasi
>> >> >> >> > nk0> probing sample rates ...
>> >> >> >> > 0:00:04.160000000 940 0x4230f0 DEBUG alsa
>> >> >> >> > gstalsa.c:49:gst_alsa_detect_rates:<alsasi
>> >> >> >> > nk0> Min. rate = 48000 (48000)
>> >> >> >> > 0:00:04.160000000 940 0x4230f0 DEBUG alsa
>> >> >> >> > gstalsa.c:50:gst_alsa_detect_rates:<alsasi
>> >> >> >> > nk0> Max. rate = 48000 (48000)
>> >> >> >> > 0:00:04.160000000 940 0x4230f0 LOG alsa
>> >> >> >> > gstalsa.c:265:gst_alsa_detect_channels:<al
>> >> >> >> > sasink0> probing channels ...
>> >> >> >> > 0:00:05.270000000 940 0x4230f0 DEBUG alsa
>> >> >> >> > gstalsa.c:309:gst_alsa_detect_channels:<al
>> >> >> >> > sasink0> Min. channels = 2 (2)
>> >> >> >> > 0:00:05.270000000 940 0x4230f0 DEBUG alsa
>> >> >> >> > gstalsa.c:310:gst_alsa_detect_channels:<al
>> >> >> >> > sasink0> Max. channels = 2 (2)
>> >> >> >> > 0:00:05.270000000 940 0x4230f0 DEBUG alsa
>> >> >> >> > gstalsa.c:388:gst_alsa_open_iec958_pcm:<al
>> >> >> >> > sasink0> Generated device string "iec958:{AES0 0x02 AES1 0x82
>> >> >> >> > AES2
>> >> >> >> > 0x00 AES3 0x02}"
>> >> >> >> > conf.c:snd_config_update_r:3661,
>> >> >> >> > configs=/usr/share/alsa/alsa.conf
>> >> >> >> > conf.c:snd_config_update_r:3661,
>> >> >> >> > configs=/usr/share/alsa/alsa.conf
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 WARN alsa
>> >> >> >> > conf.c:4692:snd_config_expand: alsalib err
>> >> >> >> > or: Unknown parameters {AES0 0x02 AES1 0x82 AES2 0x00 AES3
>> >> >> >> > 0x02}
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 WARN alsa
>> >> >> >> > pcm.c:2217:snd_pcm_open_noupdate: alsalib
>> >> >> >> > error: Unknown PCM iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3
>> >> >> >> > 0x02}
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 DEBUG alsa
>> >> >> >> > gstalsa.c:394:gst_alsa_open_iec958_pcm:<al
>> >> >> >> > sasink0> failed opening IEC958 device: Invalid argument
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 INFO alsa
>> >> >> >> > gstalsasink.c:327:gst_alsasink_getcaps:<al
>> >> >> >> > sasink0> returning caps 0x49a160
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 LOG alsa
>> >> >> >> > gstalsasink.c:312:gst_alsasink_getcaps:<al
>> >> >> >> > sasink0> Returning cached caps
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 LOG alsa
>> >> >> >> > gstalsasink.c:312:gst_alsasink_getcaps:<al
>> >> >> >> > sasink0> Returning cached caps
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 LOG alsa
>> >> >> >> > gstalsasink.c:312:gst_alsasink_getcaps:<al
>> >> >> >> > sasink0> Returning cached caps
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1814:gst_wavparse_add_src_pa
>> >> >> >> > d:<wavparse0> Send start segment event on newpad
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:1981:gst_wavparse_stream_dat
>> >> >> >> > a:<wavparse0> marking DISCONT
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 LOG wavparse
>> >> >> >> > gstwavparse.c:1995:gst_wavparse_stream_dat
>> >> >> >> > a:<wavparse0> Got buffer. timestamp:0:00:00.000000000 ,
>> >> >> >> > duration:0:00:00.040000000, size:7680
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 INFO wavparse
>> >> >> >> > gstwavparse.c:2039:gst_wavparse_stream_dat
>> >> >> >> > a:<wavparse0> Error pushing on srcpad wavparse0:src, reason
>> >> >> >> > not-linked, is linked? = 0
>> >> >> >> >
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 DEBUG wavparse
>> >> >> >> > gstwavparse.c:2088:gst_wavparse_loop:<wavp
>> >> >> >> > arse0> pausing task, reason not-linked
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 WARN wavparse
>> >> >> >> > gstwavparse.c:2122:gst_wavparse_loop:<wavp
>> >> >> >> > arse0> error: Internal data flow error.
>> >> >> >> > 0:00:05.280000000 940 0x4230f0 WARN wavparse
>> >> >> >> > gstwavparse.c:2122:gst_wavparse_loop:<wavp
>> >> >> >> > arse0> error: streaming task paused, reason not-linked (-1)
>> >> >> >> > ERROR: from element
>> >> >> >> > /GstPipeline:pipeline0/GstWavParse:wavparse0:
>> >> >> >> > Internal data flow error.
>> >> >> >> > Additional debug info:
>> >> >> >> > gstwavparse.c(2122): gst_wavparse_loop ():
>> >> >> >> > /GstPipeline:pipeline0/GstWavParse:wavparse0:
>> >> >> >> > streaming task paused, reason not-linked (-1)
>> >> >> >> > ERROR: pipeline doesn't want to preroll.
>> >> >> >> > Setting pipeline to NULL ...
>> >> >> >> > /GstPipeline:pipeline0/GstWavParse:wavparse0.GstPad:src: caps =
>> >> >> >> > NULL
>> >> >> >> > Freeing pipeline ...
>> >> >> >> > 0:00:06.400000000 936 0x477720 DEBUG wavparse
>> >> >> >> > gstwavparse.c:190:gst_wavparse_dispose:<wa
>> >> >> >> > vparse0> WAV: Dispose
>> >> >> >> > #
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > ============================================================================
>> >> >> >> _______________________________________________
>> >> >> >> gstreamer-devel mailing list
>> >> >> >> gstreamer-devel at lists.freedesktop.org
>> >> >> >> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > gstreamer-devel mailing list
>> >> >> > gstreamer-devel at lists.freedesktop.org
>> >> >> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >> >> >
>> >> >> _______________________________________________
>> >> >> gstreamer-devel mailing list
>> >> >> gstreamer-devel at lists.freedesktop.org
>> >> >> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >> >
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > gstreamer-devel mailing list
>> >> > gstreamer-devel at lists.freedesktop.org
>> >> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >> >
>> >> _______________________________________________
>> >> gstreamer-embedded mailing list
>> >> gstreamer-embedded at lists.freedesktop.org
>> >> http://lists.freedesktop.org/mailman/listinfo/gstreamer-embedded
>> >
>> >
>> >
>> > _______________________________________________
>> > gstreamer-devel mailing list
>> > gstreamer-devel at lists.freedesktop.org
>> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >
>>
>> _______________________________________________
>> gstreamer-embedded mailing list
>> gstreamer-embedded at lists.freedesktop.org
>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-embedded
>>
>
>
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
