gstreamer can not play .wav filr to usb audio device
Anuroop Jesu
jesuas at gmail.com
Mon Jul 30 05:08:00 PDT 2012
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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