[gst-devel] Problems when using osssink in ARM

Fri Sep 21 04:05:35 CEST 2007

 hi,
I tried your suggestion and got the output below, but still I can not get where the problem is, actually I don't know what kind of output is normal(some info in the end of line is not showing correct, since i just copy them from my terminal window ).
here are the info:
***************************************************************************************************************
it seems i do get some thing output from the decoder and they are put into the sink.
gst-launch-0.10 -v filesrc location="/usr/local/bin/1.mp3" ! flump3dec ! audioconvert ! audioresample ! fakesink dump=true
/pipeline0/flump3dec0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
/pipeline0/audioconvert0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
/pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
/pipeline0/audioresample0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
/pipeline0/audioresample0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
/pipeline0/fakesink0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2
/pipeline0/fakesink0: last-message = "preroll ******* "
/pipeline0/fakesink0: last-message = "event ******* E (type: 102, GstEventNewsegment, update=(boolean)false, rate=(double)1, applied_rate=(double)1, format=(GstFormat)GST_FORMAT_TIME, start=(gint64)0, stop=(gint64)-1, position=(gint64)0) 0x294a0"
/pipeline0/fakesink0: last-message = "event ******* E (type: 118, taglist, audio-codec=(string)\"MPEG\\ 1\\ Audio\\,\\ Layer\\ 3\\ \\(MP3\\)\") 0x29aa0"
/pipeline0/fakesink0: last-message = "event ******* E (type: 118, taglist, bitrate=(guint)192000) 0x29ac8"
New clock: GstSystemClock
/pipeline0/fakesink0: last-message = "chain ******* < ( 4608 bytes, timestamp: 0:00:00.000000000, duration: 0:00:00.026122448, offset: -1, offset_end: -1, flags: 0) 0x8d2d8"
00000000 (0xa6200): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000010 (0xa6210): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000020 (0xa6220): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000030 (0xa6230): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000040 (0xa6240): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000050 (0xa6250): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000060 (0xa6260): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000070 (0xa6270): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000080 (0xa6280): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
00000090 (0xa6290): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
000000a0 (0xa62a0): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
000000b0 (0xa62b0): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
000000c0 (0xa62c0): 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................
...
...
...
/pipeline0/fakesink0: last-message = "chain ******* < ( 4608 bytes, timestamp: 0:00:00.679183648, duration: 0:00:00.026122448, offset: -1, offset_end: -1, flags: 0) 0x8d4b0"
00000000 (0xa7208): 97 ff 8e ff 45 ff 97 ff f6 fe ac ff ac fe cc ff ....E...........
00000010 (0xa7218): 60 fe ed ff 0f fe 0d 00 c1 fd 31 00 7b fd 5e 00 `.........1.{.^.
00000020 (0xa7228): 3b fd 97 00 fa fc d8 00 bc fc 1a 01 85 fc 56 01 ;.............V.
00000030 (0xa7238): 55 fc 90 01 2b fc cd 01 0a fc 0b 02 f3 fb 47 02 U...+.........G.
00000040 (0xa7248): e1 fb 85 02 ce fb c4 02 c6 fb fe 02 d2 fb 31 03 ..............1.
00000050 (0xa7258): ef fb 60 03 0f fc 89 03 32 fc aa 03 60 fc c8 03 ..`.....2...`...
00000060 (0xa7268): 9b fc e6 03 dd fc 01 04 29 fd 13 04 83 fd 1e 04 ........).......
00000070 (0xa7278): e7 fd 24 04 51 fe 22 04 bf fe 14 04 32 ff fb 03 ..$.Q.".....2...
00000080 (0xa7288): aa ff db 03 27 00 b7 03 aa 00 92 03 2b 01 68 03 ....'.......+.h.
00000090 (0xa7298): a0 01 31 03 0c 02 ee 02 75 02 a9 02 de 02 65 02 ..1.....u.....e.
000000a0 (0xa72a8): 43 03 17 02 9f 03 b6 01 f1 03 4c 01 39 04 eb 00 C.........L.9...
000000b0 (0xa72b8): 78 04 96 00 ad 04 40 00 d2 04 dd ff e5 04 72 ff x..... at .......r.
000000c0 (0xa72c8): ef 04 09 ff f4 04 a4 fe f1 04 36 fe dd 04 bc fd ..........6.....
000000d0 (0xa72d8): bc 04 44 fd 8a 04 db fc 44 04 7b fc ef 03 1a fc ..D.....D.{.....
000000e0 (0xa72e8): 97 03 b9 fb 39 03 61 fb c6 02 14 fb 40 02 c8 fa ....9.a..... at ...
000000f0 (0xa72f8): ba 01 7b fa 38 01 34 fa b2 00 f9 f9 21 00 cb f9 ..{.8.4.....!...
00000100 (0xa7308): 8a ff a7 f9 ec fe 8c f9 46 fe 7f f9 a5 fd 7d f9 ........F.....}.
00000110 (0xa7318): 17 fd 83 f9 9a fc 91 f9 24 fc ab f9 b0 fb d2 f9 ........$.......
00000120 (0xa7328): 3d fb 05 fa cc fa 46 fa 64 fa 9c fa 0f fa 00 fb =.....F.d.......
00000130 (0xa7338): cd f9 67 fb 98 f9 cf fb 75 f9 40 fc 6a f9 be fc ..g.....u. at .j...
...
...
...
****************************************************************************************************************************
then i try the second command, it also seems ok, the audiosink keep writing datas, but still no sound only a flash noise.
GST_DEBUG="baseaudiosink:4, audiosink:4" gst-launch-0.10 filesrc location="/usr/local/bin/whats up.wav" ! wavparse ! audioconvert ! audioresample ! osssink
Alignment trap: gst-launch-0.10 (877) PC=0x402f7588 Instr=0xe584c008 Address=0x15a104a7 FSR 0x801
Killed
mx31# GST_DEBUG="baseaudiosink:4, audiosink:4" gst-launch-0.10 filesrc location="/usr/local/bin/whats up.wav" ! wavparse ! audioconvert ! audioresample ! okg
Feb 6 09:21:33 freescale user.warn kernel: Alignment trap: gst-launch-0.10 (877) disable codec..
PC=0x402f7588 Instr=0xe584c008 Address=0x15a104a7 FSR 0x801
Feb 6 09:21:33 freescale user.warn kernel: disable codec..
Setting pipeline to PAUSED ...
0:00:11.602708000 880 0x150c0 DEBUG audiosi
MXC Enable Codec(write)
nk gstaudiosink.c:562:gst_audio_sink_create_ringbuffer: creating ringbuffer
0:00:11.605841000 880 0x150c0 DEBUG audiosink gstaudiosink.c:564:gst_audio_sink_create_ringbuffer: created ringbuffer @0x841f8
Feb 6 09:21:45 freescale user.warn kernel:
Feb 6 09:21:45 freescale user.warn kernel: MXC Enable Codec(write)
Pipeline is PREROLLING ...
0:00:21.129583000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:407:gst_base_audio_sink_setcaps:<osssink0> release old ringbuffer
0:00:21.131907000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:412:gst_base_audio_sink_setcaps:<osssink0> parse caps
0:00:21.132613000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:423:gst_base_audio_sink_setcaps:<osssink0> acquire new ringbuffer
0:00:21.136675000 885 0xa2700 DEBUG audiosink gstaudiosink.c:210:audioringbuffer_thread_func:<osssink0> enter thread
0:00:21.137509000 885 0xa2700 DEBUG audiosink gstaudiosink.c:247:audioringbuffer_thread_func:<osssink0> signal wait
0:00:21.138044000 885 0xa2700 DEBUG audiosink gstaudiosink.c:249:audioringbuffer_thread_func:<osssink0> wait for action
0:00:21.139936000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:561:gst_base_audio_sink_event:<osssink0> new segment rate of 1.000000
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
0:00:21.144300000 880 0x150c0 DEBUG baseaudiosink gstbaseaudiosink.c:339:gst_base_audio_sink_get_time:<osssink0> processed samples: raw 0, delay 0,0
0:00:21.145791000 880 0x150c0 DEBUG baseaudiosink gstbaseaudiosink.c:1046:gst_base_audio_sink_async_play:<osssink0> ringbuffer may start now
0:00:21.148985000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:669:gst_base_audio_sink_render:<osssink0> time 0:00:00.000000000, offset 0, st0
0:00:21.153636000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:738:gst_base_audio_sink_render:<osssink0> running: start 0:00:00.000000000 - s0
0:00:21.154501000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:743:gst_base_audio_sink_render:<osssink0> base_time 0:00:00.000000000
0:00:21.155144000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:752:gst_base_audio_sink_render:<osssink0> compensating for latency 0:00:00.0000
0:00:21.155839000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:808:gst_base_audio_sink_render:<osssink0> after latency: start 0:00:00.00000000
0:00:21.156474000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:826:gst_base_audio_sink_render:<osssink0> no align possible: no previous sampln
0:00:21.157079000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:880:gst_base_audio_sink_render:<osssink0> rendering at -1 4440/4440
0:00:21.157896000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:895:gst_base_audio_sink_render:<osssink0> wrote 4440 of 4440
0:00:21.158530000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:911:gst_base_audio_sink_render:<osssink0> next sample expected at 4440
0:00:30.093297000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:669:gst_base_aumxc_audio_output_block: count = 512
dio_sink_render:<osssink0> time 0:00:00.092500000, offset 4440, start 0:00:00.000000000, samples 4458
0:00:30.096231000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:738:gst_base_audio_sink_render:<osssink0> running: start 0:00:00.092500000 - s0
0:00:30.096925000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:743:gst_base_audio_sink_render:<osssink0> base_time 0:00:00.000000000
0:00:30.097592000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:752:gst_base_audio_sink_render:<osssink0> compensating for latency 0:00:00.0000
0:00:30.098171000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:808:gst_base_audio_sink_render:<osssink0> after latency: start 0:00:00.09250000
0:00:30.098891000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:848:gst_base_audio_sink_render:<osssink0> align with prev sample, 0 < 24000
0:00:30.099434000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:880:gst_base_audio_sink_render:<osssink0> rendering at 4440 4458/4458
0:00:30.100058000 884 0x876c0 DEBUG audiosink gstaudiosink.c:441:gst_audioringbuffer_start: start, sending signal
0:00:30.100767000 885 0xa2700 DEBUG audiosink gstaudiosink.c:251:audioringbuffer_thread_func:<osssink0> got signal
0:00:30.101286000 885 0xa2700 DEBUG audiosink gstaudiosink.c:254:audioringbuffer_thread_func:<osssink0> continue running
Feb 6 09:22:04 freescale user.warn kernel: mxc_audio_output_block: count = 512
0:00:30.113969000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:895:gst_base_audio_sink_render:<osssink0> wrote 4458 of 4458
0:00:30.114612000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:911:gst_base_audio_sink_render:<osssink0> next sample expected at 8898
0:00:39.437770000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:669:gst_base_audio_sink_render:<osssink0> time 0:00:00.185375000, offset 8898,8
0:00:39.440605000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:738:gst_base_audio_sink_render:<osssink0> running: start 0:00:00.185375000 - s0
0:00:39.441296000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:743:gst_base_audio_sink_render:<osssink0> base_time 0:00:00.000000000
0:00:39.442118000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:752:gst_base_audio_sink_render:<osssink0> compensating for latency 0:00:00.0000
0:00:39.442713000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:808:gst_base_audio_sink_render:<osssink0> after latency: start 0:00:00.18537500
0:00:39.443442000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:848:gst_base_audio_sink_render:<osssink0> align with prev sample, 0 < 24000
0:00:39.444121000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:880:gst_base_audio_sink_render:<osssink0> rendering at 8898 4458/4458
0:00:39.444970000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:895:gst_base_audio_sink_render:<osssink0> wrote 4458 of 4458
0:00:39.445513000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:911:gst_base_audio_sink_render:<osssink0> next sample expected at 13356
0:00:48.765731000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:669:gst_base_audio_sink_render:<osssink0> time 0:00:00.278250000, offset 133560
0:00:48.768559000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:738:gst_base_audio_sink_render:<osssink0> running: start 0:00:00.278250000 - s6
0:00:48.769252000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:743:gst_base_audio_sink_render:<osssink0> base_time 0:00:00.000000000
0:00:48.770067000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:752:gst_base_audio_sink_render:<osssink0> compensating for latency 0:00:00.0000
0:00:48.770656000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:808:gst_base_audio_sink_render:<osssink0> after latency: start 0:00:00.27825006
0:00:48.771384000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:848:gst_base_audio_sink_render:<osssink0> align with prev sample, 0 < 24000
0:00:48.772056000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:880:gst_base_audio_sink_render:<osssink0> rendering at 13356 4460/4459
0:00:48.773422000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:895:gst_base_audio_sink_render:<osssink0> wrote 4460 of 4460
0:00:48.774038000 884 0x876c0 DEBUG baseaudiosink gstbaseaudiosink.c:911:gst_base_audio_sink_render:<osssink0> next sample expected at 17815
...
...
...   
在2007-09-21，"Ali Sabil" <ali.sabil at tandberg.com> 写道：
Hi, maybe you can first try replacing the osssink by a fakesink and setting it's property dump=true : gst-launch-0.10 -v filesrc location=~/Media/test.mp3 ! mad ! audioconvert ! audioresample ! fakesink dump=true then try the GST_DEBUG : GST_DEBUG="baseaudiosink:4, audiosink:4" gst-launch-0.10 filesrc location=~/Media/test.mp3 ! mad ! audioconvert ! audioresample ! osssink Cheers, -- Ali On Thu, 2007-09-20 at 09:50 +0800, Joyious He wrote: > Hi Jianjun, > > Here is the correct output,the previous one is copied from terminal, > and there is someything shows wrong: > > /pipeline0/flump3dec0.src: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=(int)2 > /pipeline0/audioconvert0.src: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=(int)2 > /pipeline0/audioconvert0.sink: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=(int)2 > /pipeline0/audioresample0.src: caps = audio/x-raw-int, width=(int)16, > depth=(int)16, signed=(boolean)true, endianness=(int)1234, > channels=(int)2, rate=(int)48000 > /pipeline0/audioresample0.sink: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=(int)2 > /pipeline0/osssink0.sink: caps = audio/x-raw-int, width=(int)16, > depth=(int)16, signed=(boolean)true, endianness=(int)1234, > channels=(int)2, rate=(int)48000 > New clock: GstAudioSinkClock > > > so here the audio resample src seems to be accord to my osssink, but > still the same, no sound came out.what should i do next ? > > > > > 在2007-09-19，"jianjun.yang.cn" <jianjun.yang.cn at gmail.com> 写道： > Hi Joyious, > > I think the problem lies in that audioresample fails to > convert from 44100 hz to 48000 hz. Your oss driver does not > support 44100, but the rate of 1.mp3 is 44100. So the > audioresample should resample. > But according to your output, rate of audioresample's source > pad is different with the one of osssink's sink pad. The > former is 0, while the latter is 48000. > I test the pipleline on my PC using osssink. It can work well. > > my command line: > gst-launch-0.10 -v filesrc > location= /home/jianjun/206851.mp3 ! mad ! audioconvert ! > audioresample ! osssink > > And its output: > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > /pipeline0/mad0.src: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)32, > depth=(int)32, rate=(int)44100, channels=(int)2 > /pipeline0/audioconvert0.src: caps = audio/x-raw-int, > width=(int)16, depth=(int)16, signed=(boolean)true, > endianness=(int)1234, channels=(int)2, rate=(int)44100 > /pipeline0/audioconvert0.sink: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)32, > depth=(int)32, rate=(int)44100, channels=(int)2 > /pipeline0/audioresample0.src: caps = audio/x-raw-int, > width=(int)16, depth=(int)16, signed=(boolean)true, > endianness=(int)1234, channels=(int)2, rate=(int)44100 > /pipeline0/audioresample0.sink: caps = audio/x-raw-int, > width=(int)16, depth=(int)16, signed=(boolean)true, > endianness=(int)1234, channels=(int)2, rate=(int)44100 > /pipeline0/osssink0.sink: caps = audio/x-raw-int, > width=(int)16, depth=(int)16, signed=(boolean)true, > endianness=(int)1234, channels=(int)2, rate=(int)44100 > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstAudioSinkClock > > > Regards, > Jianjun > > > 2007-09-19 > > ______________________________________________________________ > jianjun.yang.cn > > ______________________________________________________________ > 发件人： 何慧敏 > 发送时间： 2007-09-19 15:52:35 > 收件人： gstreamer-devel > 抄送： > 主题： [gst-devel] Problems when using osssink in ARM > > Hi all, > > Now I have ported the gstreamer to ARM 11, and there is a OSS > driver for this ARM board. So I am trying to make this sound > workI have installed the plugin for ossaudio.But when i trying > to play some media file, no sound came out, and the screen > just show the PLAYING message .. > > some addtional messages: > the osssink does play the audiotestsrc,and sounds a single > tone; > the audioconvert works fine when "gst-launch -v filesrc > location="/usr/local/bin/1.mp3" ! flump3dec ! audioconvert ! > wavenc ! filesink location="/usr/local/bin/1.wav" ",I can hear > the wav file on my PC,so the decodec is fine; > even no sound comes out when playing, but there are some > noises when pipeline is just prerolling just before PLAYING. > > > Here are the command and output: > *********************************************************************************************************** > mx31# gst-launch -v filesrc location="/usr/local/bin/1.mp3" ! > flump3dec ! audioconvert ! audioresample ! osssink > Setting pipeline to PAUSED ... > > MXC Enable Codec(write) > Feb 6 12:04:07 freescale user.warn kernel: > Feb 6 12:04:07 freescale user.warn kernel: MXC Enable > Codec(write) > Pipeline is PREROLLING ... > /pipeline0/flump3dec0.src: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=(int)2 > /pipeline0/audioconvert0.src: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=(i2 > /pipeline0/audioconvert0.sink: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=(2 > /pipeline0/audioresample0.src: caps = audio/x-raw-int, > width=(int)16, depth=(int)16, signed=(boolean)true, > endianness=(int)1234, channels=(int)2, rate=(int)0 > /pipeline0/audioresample0.sink: caps = audio/x-raw-int, > endianness=(int)1234, signed=(boolean)true, width=(int)16, > depth=(int)16, rate=(int)44100, channels=2 > /pipeline0/osssink0.sink: caps = audio/x-raw-int, > width=(int)16, depth=(int)16, signed=(boolean)true, > endianness=(int)1234, channels=(int)2, rate=(int)48000 > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstAudioSinkClock > mxc_audio_output_block: count = 1024 > Feb 6 12:04:37 freescale user.warn kernel: > mxc_audio_output_block: count = 1024 > > ************************************************************************************************************* > > So I'm studying the code of osssink although I am a totally > newcomer to this gstreamer and Linux things. > > Many Thanks, > > Joyious > > > > > > > > ______________________________________________________________ > 杀70万种木马病毒，瑞星2008版免费 > > > > ______________________________________________________________________ > 杀70万种木马病毒，瑞星2008版免费 > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
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