Hi,
below is the that I re-test with debug level=3
and the USB Audio information when I use aplay to play audio wav file
Do you have any idea?
#
#
# aplay -v -f S16_LE -c 2 -r 48000 -D default /bin/audio_src_48k_le.wav
Hardware PCM card 0 'USB AUDIO ' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 24000
period_size : 6000
period_time : 125000
tstamp_mode : NONE
period_step : 1
avail_min : 6000
period_event : 0
start_threshold : 24000
stop_threshold : 24000
silence_threshold: 0
silence_size : 0
boundary : 1572864000
appl_ptr : 0
hw_ptr : 0
#
#
# export GST_DEBUG=alsa:3,*decode*:3,wavparse:3,audioresample:3,audioconvert:3
# gst-launch-0.10 filesrc location=/bin/audio_src_48k_le.wav ! wavparse ! audioc
onvert !alsasink device="hw:0,0"
0:00:00.130000000 928 0x441000 WARN audioresample
gstaudioresample.c:1581:plugin_init: Orc d
isabled, can't benchmark int vs. float resampler
Setting pipeline to PAUSED ...
conf.c:snd_config_update_r:3661, configs=/usr/share/alsa/alsa.conf
0:00:00.900000000 931 0x4230f0 INFO wavparse
gstwavparse.c:2054:gst_wavparse_loop:<wavp
arse0> GST_WAVPARSE_START
Pipeline is PREROLLING ...
0:00:01.010000000 931 0x4230f0 INFO wavparse
gstwavparse.c:2063:gst_wavparse_loop:<wavp
arse0> GST_WAVPARSE_HEADER
0:00:01.020000000 931 0x4230f0 INFO wavparse
gstwavparse.c:1343:gst_wavparse_stream_hea
ders:<wavparse0> Got TAG: data, offset 36
0:00:01.090000000 931 0x4230f0 INFO wavparse
gstwavparse.c:1343:gst_wavparse_stream_hea
ders:<wavparse0> Got TAG: ID3x, offset 960044
0:00:01.090000000 931 0x4230f0 INFO wavparse
gstwavparse.c:1126:gst_wavparse_calculate_
duration:<wavparse0> Got datasize 960000
0:00:01.090000000 931 0x4230f0 INFO wavparse
gstwavparse.c:1130:gst_wavparse_calculate_
duration:<wavparse0> Got duration (bps) 0:00:05.000000000
0:00:01.090000000 931 0x4230f0 INFO wavparse
gstwavparse.c:2069:gst_wavparse_loop:<wavp
arse0> GST_WAVPARSE_DATA
** (gst-launch-0.10:931): WARNING **: gstwavparse.c,
gst_wavparse_stream_data,1836
** (gst-launch-0.10:931): WARNING **: offset: 44 , end: 960044 ,
dataleft: 960000
** (gst-launch-0.10:931): WARNING **: Fetching 7680 bytes of data from
the sinkpad
0:00:01.180000000 931 0x4230f0 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
0:00:01.190000000 931 0x4230f0 WARN alsa
conf.c:4692:snd_config_expand: alsalib err
or: Unknown parameters {AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
0:00:01.190000000 931 0x4230f0 WARN alsa
pcm.c:2217:snd_pcm_open_noupdate: alsalib
error: Unknown PCM iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
0:00:01.190000000 931 0x4230f0 INFO alsa
gstalsasink.c:327:gst_alsasink_getcaps:<al
sasink0> returning caps 0x49a360
0:00:01.190000000 931 0x4230f0 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-0.10:931): WARNING **: gstwavparse.c, gst_wavparse_loop,2074
0:00:01.190000000 931 0x4230f0 WARN wavparse
gstwavparse.c:2122:gst_wavparse_loop:<wavp
arse0> error: Internal data flow error.
0:00:01.190000000 931 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 ...
Freeing pipeline ...
#
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
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