gstreamer can not play .wav filr to usb audio device
Anuroop Jesu
jesuas at gmail.com
Mon Jul 30 05:54:04 PDT 2012
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
> >> > #
> >> >
> >> >
> ============================================================================
> >> _______________________________________________
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> >> 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
> >
> _______________________________________________
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> gstreamer-devel at lists.freedesktop.org
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
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