GST alsasink problem

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GST alsasink problem

vinod james

Hi,
I have done a source installation of gstreamer and gstreamer-base plugins version 0.10.21
I am trying few things mentioned in the faq document
If I say
$aplay -v test.wav,
I am able to play the wav file through alsaplay, which shows that alsa driver is installed in my PC

If I do
$gst-inspect alsasink
shows that alsasink is installed.

And when i do
$gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! audioresample ! filesink location=test.raw
it writes the decoded raw samples into the file and I am  able to play test.raw in application like cooledit

But if I do
$gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! audioresample ! alsasink
it goes into infinite loop.
The alsasink plugin is not able to detect the alsadriver I guess.
How do I debug this problem?
Pls help me

Regards
Vinod James

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Re: GST alsasink problem

Felipe Contreras
On Fri, Dec 5, 2008 at 8:30 AM, vinod james <[hidden email]> wrote:

>
> Hi,
> I have done a source installation of gstreamer and gstreamer-base plugins
> version 0.10.21
> I am trying few things mentioned in the faq document
> If I say
> $aplay -v test.wav,
> I am able to play the wav file through alsaplay, which shows that alsa
> driver is installed in my PC
>
> If I do
> $gst-inspect alsasink
> shows that alsasink is installed.
>
> And when i do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! filesink location=test.raw
> it writes the decoded raw samples into the file and I am  able to play
> test.raw in application like cooledit
>
> But if I do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! alsasink
> it goes into infinite loop.
> The alsasink plugin is not able to detect the alsadriver I guess.
> How do I debug this problem?
> Pls help me

Try with:
gst-launch-0.10 audiotestsrc ! alsasink

And for debugging:
export GST_DEBUG=alsa:5

--
Felipe Contreras

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Re: GST alsasink problem

vinod james
Hi Felipe,
It didn't help.
No extra debug information came out
It displays and goes into infinite loop

"Setting pipeline to Paused..."
Regards
vinod

On Fri, Dec 5, 2008 at 2:17 PM, Felipe Contreras <[hidden email]> wrote:
On Fri, Dec 5, 2008 at 8:30 AM, vinod james <[hidden email]> wrote:
>
> Hi,
> I have done a source installation of gstreamer and gstreamer-base plugins
> version 0.10.21
> I am trying few things mentioned in the faq document
> If I say
> $aplay -v test.wav,
> I am able to play the wav file through alsaplay, which shows that alsa
> driver is installed in my PC
>
> If I do
> $gst-inspect alsasink
> shows that alsasink is installed.
>
> And when i do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! filesink location=test.raw
> it writes the decoded raw samples into the file and I am  able to play
> test.raw in application like cooledit
>
> But if I do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! alsasink
> it goes into infinite loop.
> The alsasink plugin is not able to detect the alsadriver I guess.
> How do I debug this problem?
> Pls help me

Try with:
gst-launch-0.10 audiotestsrc ! alsasink

And for debugging:
export GST_DEBUG=alsa:5

--
Felipe Contreras

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Vinod James

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Re: GST alsasink problem

Nie Jun
It is strange, please try
gst-launch-0.10 audiotestsrc ! alsasink --gst-debug=alsa:5,basesink:4,baseaudiosink:4,audiosink:4

It is supposed to print a lot of log messages. If there is too much log, you can change all log level to 3 or remove some log.


2008/12/5 vinod james <[hidden email]>
Hi Felipe,
It didn't help.
No extra debug information came out
It displays and goes into infinite loop

"Setting pipeline to Paused..."
Regards
vinod

On Fri, Dec 5, 2008 at 2:17 PM, Felipe Contreras <[hidden email]> wrote:
On Fri, Dec 5, 2008 at 8:30 AM, vinod james <[hidden email]> wrote:
>
> Hi,
> I have done a source installation of gstreamer and gstreamer-base plugins
> version 0.10.21
> I am trying few things mentioned in the faq document
> If I say
> $aplay -v test.wav,
> I am able to play the wav file through alsaplay, which shows that alsa
> driver is installed in my PC
>
> If I do
> $gst-inspect alsasink
> shows that alsasink is installed.
>
> And when i do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! filesink location=test.raw
> it writes the decoded raw samples into the file and I am  able to play
> test.raw in application like cooledit
>
> But if I do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! alsasink
> it goes into infinite loop.
> The alsasink plugin is not able to detect the alsadriver I guess.
> How do I debug this problem?
> Pls help me

Try with:
gst-launch-0.10 audiotestsrc ! alsasink

And for debugging:
export GST_DEBUG=alsa:5

--
Felipe Contreras

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--
Vinod James

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Re: GST alsasink problem

vinod james
Hi,
This is the log it generates( I have attached as .txt with better formatting)
0:00:00.036590000 13368  0x8578098 DEBUG                 alsa gstalsasink.c:271:gst_alsasink_init:<GstAlsaSink@0x85c2170> initializing alsasink
0:00:00.036900000 13368  0x8578098 DEBUG                 alsa gstalsasink.c:301:gst_alsasink_getcaps:<alsasink0> device not open, using template caps

Setting pipeline to PAUSED ...
0:00:00.037299000 13368  0x8578098 DEBUG            audiosink gstaudiosink.c:565:gst_audio_sink_create_ringbuffer:<alsasink0> creating
ringbuffer
0:00:00.037384000 13368  0x8578098 DEBUG            audiosink gstaudiosink.c:567:gst_audio_sink_create_ringbuffer:<alsasink0> created
ringbuffer @0x85c4908
0:00:00.051408000 13368  0x8578098 LOG                   alsa gstalsasink.c:676:gst_alsasink_open:<alsasink0> Opened device default
0:00:00.051607000 13368  0x8578098 DEBUG             basesink gstbasesink.c:3456:gst_base_sink_change_state:<alsasink0> READY to PAUSED
0:00:00.051639000 13368  0x8578098 DEBUG             basesink gstbasesink.c:3474:gst_base_sink_change_state:<alsasink0> doing async state
change
0:00:00.051683000 13368  0x8578098 DEBUG             basesink gstbasesink.c:2918:gst_base_sink_pad_activate:<alsasink0> Trying pull mode
first
0:00:00.051710000 13368  0x8578098 DEBUG             basesink gstbasesink.c:2927:gst_base_sink_pad_activate:<alsasink0> Falling back to push
mode
0:00:00.051735000 13368  0x8578098 DEBUG             basesink gstbasesink.c:2929:gst_base_sink_pad_activate:<alsasink0> Success activating
push mode
0:00:00.052103000 13368  0x8578098 WARN                  alsa gstalsa.c:124:gst_alsa_detect_formats:<alsasink0> skipping non-int format
0:00:00.052144000 13368  0x8578098 LOG                   alsa gstalsa.c:30:gst_alsa_detect_rates:<alsasink0> probing sample rates ...
0:00:00.052171000 13368  0x8578098 DEBUG                 alsa gstalsa.c:49:gst_alsa_detect_rates:<alsasink0> Min. rate = 4000 (4000)
0:00:00.052194000 13368  0x8578098 DEBUG                 alsa gstalsa.c:50:gst_alsa_detect_rates:<alsasink0> Max. rate = 2147483647 (-1)
0:00:00.052219000 13368  0x8578098 LOG                   alsa gstalsa.c:265:gst_alsa_detect_channels:<alsasink0> probing channels ...
0:00:00.052240000 13368  0x8578098 DEBUG                 alsa gstalsa.c:309:gst_alsa_detect_channels:<alsasink0> Min. channels = 1 (1)
0:00:00.052262000 13368  0x8578098 DEBUG                 alsa gstalsa.c:310:gst_alsa_detect_channels:<alsasink0> Max. channels = 8 (10000)
0:00:00.052544000 13368  0x8578098 DEBUG                 alsa gstalsa.c:388:gst_alsa_open_iec958_pcm:<alsasink0> Generated device string
"iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}"

At this point I had to press Ctrl-d as it was in infinite loop

Caught interrupt -- 0:00:05.107535000 13368  0x8578098 WARN alsa pcm_hw.c:1155:snd_pcm_hw_open: alsalib error: open /dev/snd/pcmC0D0p
failed: Interrupted system call
0:00:05.107593000 13368  0x8578098 DEBUG                 alsa gstalsa.c:394:gst_alsa_open_iec958_pcm:<alsasink0> failed opening IEC958
device: Interrupted system call
0:00:05.107623000 13368  0x8578098 INFO                  alsa gstalsasink.c:321:gst_alsasink_getcaps:<alsasink0> returning caps
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)[1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ],channels=(int)3, channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,
signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)6,
channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >;
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)[
1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ],
channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[
4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >;
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)[
1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ],
channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[
4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >;
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)[
1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ],
channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[
4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >;
audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8,rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int,
signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000,2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, signed=(boolean){ true,
false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ],channels=(int)4, channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;
audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8,rate=(int)[ 4000, 2147483647 ], channels=(int)6,
channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8,
rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >

Pipeline is PREROLLING ...
handling interrupt.
Interrupt: Stopping pipeline ...
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
0:00:05.194465000 13368  0x8578098 DEBUG             basesink
gstbasesink.c:2885:gst_base_sink_set_flushing:<alsasink0> flushing out data thread, need preroll to TRUE
0:00:05.194496000 13368  0x8578098 DEBUG             basesink
gstbasesink.c:1129:gst_base_sink_preroll_queue_flush:<alsasink0> flushing queue 0x85c2170
0:00:05.194529000 13368  0x8578098 DEBUG             basesink
gstbasesink.c:3608:gst_base_sink_change_state:<alsasink0> PAUSED to READY,
posting async-done
FREEING pipeline ...




On Fri, Dec 5, 2008 at 3:52 PM, Nie Jun <[hidden email]> wrote:
It is strange, please try
gst-launch-0.10 audiotestsrc ! alsasink --gst-debug=alsa:5,basesink:4,baseaudiosink:4,audiosink:4

It is supposed to print a lot of log messages. If there is too much log, you can change all log level to 3 or remove some log.


2008/12/5 vinod james <[hidden email]>

Hi Felipe,
It didn't help.
No extra debug information came out
It displays and goes into infinite loop

"Setting pipeline to Paused..."
Regards
vinod

On Fri, Dec 5, 2008 at 2:17 PM, Felipe Contreras <[hidden email]> wrote:
On Fri, Dec 5, 2008 at 8:30 AM, vinod james <[hidden email]> wrote:
>
> Hi,
> I have done a source installation of gstreamer and gstreamer-base plugins
> version 0.10.21
> I am trying few things mentioned in the faq document
> If I say
> $aplay -v test.wav,
> I am able to play the wav file through alsaplay, which shows that alsa
> driver is installed in my PC
>
> If I do
> $gst-inspect alsasink
> shows that alsasink is installed.
>
> And when i do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! filesink location=test.raw
> it writes the decoded raw samples into the file and I am  able to play
> test.raw in application like cooledit
>
> But if I do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! alsasink
> it goes into infinite loop.
> The alsasink plugin is not able to detect the alsadriver I guess.
> How do I debug this problem?
> Pls help me

Try with:
gst-launch-0.10 audiotestsrc ! alsasink

And for debugging:
export GST_DEBUG=alsa:5

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Re: GST alsasink problem

vinod james
Hi Felipe &  Nie,
Any ideas on how to solve this problem?
Regards
vinod

On Fri, Dec 5, 2008 at 4:49 PM, vinod james <[hidden email]> wrote:
Hi,
This is the log it generates( I have attached as .txt with better formatting)
0:00:00.036590000 13368  0x8578098 DEBUG                 alsa gstalsasink.c:271:gst_alsasink_init:<GstAlsaSink@0x85c2170> initializing alsasink
0:00:00.036900000 13368  0x8578098 DEBUG                 alsa gstalsasink.c:301:gst_alsasink_getcaps:<alsasink0> device not open, using template caps

Setting pipeline to PAUSED ...
0:00:00.037299000 13368  0x8578098 DEBUG            audiosink gstaudiosink.c:565:gst_audio_sink_create_ringbuffer:<alsasink0> creating
ringbuffer
0:00:00.037384000 13368  0x8578098 DEBUG            audiosink gstaudiosink.c:567:gst_audio_sink_create_ringbuffer:<alsasink0> created
ringbuffer @0x85c4908
0:00:00.051408000 13368  0x8578098 LOG                   alsa gstalsasink.c:676:gst_alsasink_open:<alsasink0> Opened device default
0:00:00.051607000 13368  0x8578098 DEBUG             basesink gstbasesink.c:3456:gst_base_sink_change_state:<alsasink0> READY to PAUSED
0:00:00.051639000 13368  0x8578098 DEBUG             basesink gstbasesink.c:3474:gst_base_sink_change_state:<alsasink0> doing async state
change
0:00:00.051683000 13368  0x8578098 DEBUG             basesink gstbasesink.c:2918:gst_base_sink_pad_activate:<alsasink0> Trying pull mode
first
0:00:00.051710000 13368  0x8578098 DEBUG             basesink gstbasesink.c:2927:gst_base_sink_pad_activate:<alsasink0> Falling back to push
mode
0:00:00.051735000 13368  0x8578098 DEBUG             basesink gstbasesink.c:2929:gst_base_sink_pad_activate:<alsasink0> Success activating
push mode
0:00:00.052103000 13368  0x8578098 WARN                  alsa gstalsa.c:124:gst_alsa_detect_formats:<alsasink0> skipping non-int format
0:00:00.052144000 13368  0x8578098 LOG                   alsa gstalsa.c:30:gst_alsa_detect_rates:<alsasink0> probing sample rates ...
0:00:00.052171000 13368  0x8578098 DEBUG                 alsa gstalsa.c:49:gst_alsa_detect_rates:<alsasink0> Min. rate = 4000 (4000)
0:00:00.052194000 13368  0x8578098 DEBUG                 alsa gstalsa.c:50:gst_alsa_detect_rates:<alsasink0> Max. rate = 2147483647 (-1)
0:00:00.052219000 13368  0x8578098 LOG                   alsa gstalsa.c:265:gst_alsa_detect_channels:<alsasink0> probing channels ...
0:00:00.052240000 13368  0x8578098 DEBUG                 alsa gstalsa.c:309:gst_alsa_detect_channels:<alsasink0> Min. channels = 1 (1)
0:00:00.052262000 13368  0x8578098 DEBUG                 alsa gstalsa.c:310:gst_alsa_detect_channels:<alsasink0> Max. channels = 8 (10000)
0:00:00.052544000 13368  0x8578098 DEBUG                 alsa gstalsa.c:388:gst_alsa_open_iec958_pcm:<alsasink0> Generated device string
"iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}"

At this point I had to press Ctrl-d as it was in infinite loop

Caught interrupt -- 0:00:05.107535000 13368  0x8578098 WARN alsa pcm_hw.c:1155:snd_pcm_hw_open: alsalib error: open /dev/snd/pcmC0D0p
failed: Interrupted system call
0:00:05.107593000 13368  0x8578098 DEBUG                 alsa gstalsa.c:394:gst_alsa_open_iec958_pcm:<alsasink0> failed opening IEC958
device: Interrupted system call
0:00:05.107623000 13368  0x8578098 INFO                  alsa gstalsasink.c:321:gst_alsasink_getcaps:<alsasink0> returning caps
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)[1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ],channels=(int)3, channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,
signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)6,
channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >;
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)[
1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ],
channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[
4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >;
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)[
1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ],
channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[
4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >;
audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)[
1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false}, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ],
channels=(int)3, channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234,signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[
4000, 2147483647 ], channels=(int)4,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)6,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false },
width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >;
audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8,rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int,
signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000,2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, signed=(boolean){ true,
false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ],channels=(int)4, channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >;
audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8,rate=(int)[ 4000, 2147483647 ], channels=(int)6,
channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >;audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8,
rate=(int)[ 4000, 2147483647 ], channels=(int)8,channel-positions=(GstAudioChannelPosition)<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >

Pipeline is PREROLLING ...
handling interrupt.
Interrupt: Stopping pipeline ...
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
0:00:05.194465000 13368  0x8578098 DEBUG             basesink
gstbasesink.c:2885:gst_base_sink_set_flushing:<alsasink0> flushing out data thread, need preroll to TRUE
0:00:05.194496000 13368  0x8578098 DEBUG             basesink
gstbasesink.c:1129:gst_base_sink_preroll_queue_flush:<alsasink0> flushing queue 0x85c2170
0:00:05.194529000 13368  0x8578098 DEBUG             basesink
gstbasesink.c:3608:gst_base_sink_change_state:<alsasink0> PAUSED to READY,
posting async-done
FREEING pipeline ...





On Fri, Dec 5, 2008 at 3:52 PM, Nie Jun <[hidden email]> wrote:
It is strange, please try
gst-launch-0.10 audiotestsrc ! alsasink --gst-debug=alsa:5,basesink:4,baseaudiosink:4,audiosink:4

It is supposed to print a lot of log messages. If there is too much log, you can change all log level to 3 or remove some log.


2008/12/5 vinod james <[hidden email]>

Hi Felipe,
It didn't help.
No extra debug information came out
It displays and goes into infinite loop

"Setting pipeline to Paused..."
Regards
vinod

On Fri, Dec 5, 2008 at 2:17 PM, Felipe Contreras <[hidden email]> wrote:
On Fri, Dec 5, 2008 at 8:30 AM, vinod james <[hidden email]> wrote:
>
> Hi,
> I have done a source installation of gstreamer and gstreamer-base plugins
> version 0.10.21
> I am trying few things mentioned in the faq document
> If I say
> $aplay -v test.wav,
> I am able to play the wav file through alsaplay, which shows that alsa
> driver is installed in my PC
>
> If I do
> $gst-inspect alsasink
> shows that alsasink is installed.
>
> And when i do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! filesink location=test.raw
> it writes the decoded raw samples into the file and I am  able to play
> test.raw in application like cooledit
>
> But if I do
> $gst-launch filesrc location=test.ogg ! decodebin ! audioconvert !
> audioresample ! alsasink
> it goes into infinite loop.
> The alsasink plugin is not able to detect the alsadriver I guess.
> How do I debug this problem?
> Pls help me

Try with:
gst-launch-0.10 audiotestsrc ! alsasink

And for debugging:
export GST_DEBUG=alsa:5

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Re: GST alsasink problem

Stefan Sauer
vinod james schrieb:

> Hi Felipe &  Nie,
> Any ideas on how to solve this problem?
> Regards
> vinod
>
> On Fri, Dec 5, 2008 at 4:49 PM, vinod james <[hidden email]
> <mailto:[hidden email]>> wrote:
>
>     Hi,
>     This is the log it generates( I have attached as .txt with better
>     formatting)
>     0:00:00.036590000 13368  0x8578098 DEBUG                 alsa
>     gstalsasink.c:271:gst_alsasink_init:<GstAlsaSink@0x85c2170>
>     initializing alsasink
>     0:00:00.036900000 13368  0x8578098 DEBUG                 alsa
>     gstalsasink.c:301:gst_alsasink_getcaps:<alsasink0> device not open,
>     using template caps
>
>     Setting pipeline to PAUSED ...
>
>     ...
>     At this point I had to press Ctrl-d as it was in infinite loop

better redirect to a file and sent it for analysis.
GST_DEBUG=*:2,alsa:5 GST_DEBUG_NO_COLOR=1 gst-launch-0.10 2>debug.log
audiotestsrc num-buffers=10 ! alsasink


Stefan

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Re: GST alsasink problem

Tristan Matthews-2
Just a thought, I've had issues with alsasink caps which I've resolved
by using alsa's plughw interface:

GST_DEBUG=*:2,alsa:5 GST_DEBUG_NO_COLOR=1 gst-launch-0.10 2>debug.log
audiotestsrc num-buffers=10 ! alsasink device=plughw:0


I don't know if this will resolve your issue but I'd be curious to see
the debug output.

-Tristan

Stefan Kost wrote:

> vinod james schrieb:
>  
>> Hi Felipe &  Nie,
>> Any ideas on how to solve this problem?
>> Regards
>> vinod
>>
>> On Fri, Dec 5, 2008 at 4:49 PM, vinod james <[hidden email]
>> <mailto:[hidden email]>> wrote:
>>
>>     Hi,
>>     This is the log it generates( I have attached as .txt with better
>>     formatting)
>>     0:00:00.036590000 13368  0x8578098 DEBUG                 alsa
>>     gstalsasink.c:271:gst_alsasink_init:<GstAlsaSink@0x85c2170>
>>     initializing alsasink
>>     0:00:00.036900000 13368  0x8578098 DEBUG                 alsa
>>     gstalsasink.c:301:gst_alsasink_getcaps:<alsasink0> device not open,
>>     using template caps
>>
>>     Setting pipeline to PAUSED ...
>>
>>     ...
>>     At this point I had to press Ctrl-d as it was in infinite loop
>>    
>
> better redirect to a file and sent it for analysis.
> GST_DEBUG=*:2,alsa:5 GST_DEBUG_NO_COLOR=1 gst-launch-0.10 2>debug.log
> audiotestsrc num-buffers=10 ! alsasink
>
>
> Stefan
>
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>  


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Re: GST alsasink problem

Eric Zhang-6
In reply to this post by vinod james
Hi, gstreamer-devel:

    I think you'd better add some "g_message" or "printf" codes into your gstbasesink.c and find out where the pipeline blocked, that's the key point. Blocking is a common problem when developing gstreamer applications, at least for me. And my solution is first find out where it blocks -- mostly always blocks at basesink.

    I noticed your basesink used push mode so you can first add printf codes in "gst_base_sink_chain" --> "gst_base_sink_chain_unlock" --> "gst_base_sink_queue_object_unlocked" ......

Eric Zhang

2008/12/5 vinod james <[hidden email]>

Hi,
I have done a source installation of gstreamer and gstreamer-base plugins version 0.10.21
I am trying few things mentioned in the faq document
If I say
$aplay -v test.wav,
I am able to play the wav file through alsaplay, which shows that alsa driver is installed in my PC

If I do
$gst-inspect alsasink
shows that alsasink is installed.

And when i do
$gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! audioresample ! filesink location=test.raw
it writes the decoded raw samples into the file and I am  able to play test.raw in application like cooledit

But if I do
$gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! audioresample ! alsasink
it goes into infinite loop.
The alsasink plugin is not able to detect the alsadriver I guess.
How do I debug this problem?
Pls help me

Regards
Vinod James

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Re: GST alsasink problem

Eric Zhang-6
Hi, gstreamer-devel:

    One more thing: alsasink inherits from gstbaseaudiosink which inherits from gstbasesink. GstBaseAudioSink overrides "render" function of GstBaseSink, so don't forget to take care of gstbaseaudiosink.

Eric Zhang

2008/12/9 Eric Zhang <[hidden email]>
Hi, gstreamer-devel:

    I think you'd better add some "g_message" or "printf" codes into your gstbasesink.c and find out where the pipeline blocked, that's the key point. Blocking is a common problem when developing gstreamer applications, at least for me. And my solution is first find out where it blocks -- mostly always blocks at basesink.

    I noticed your basesink used push mode so you can first add printf codes in "gst_base_sink_chain" --> "gst_base_sink_chain_unlock" --> "gst_base_sink_queue_object_unlocked" ......

Eric Zhang

2008/12/5 vinod james <[hidden email]>

Hi,
I have done a source installation of gstreamer and gstreamer-base plugins version 0.10.21
I am trying few things mentioned in the faq document
If I say
$aplay -v test.wav,
I am able to play the wav file through alsaplay, which shows that alsa driver is installed in my PC

If I do
$gst-inspect alsasink
shows that alsasink is installed.

And when i do
$gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! audioresample ! filesink location=test.raw
it writes the decoded raw samples into the file and I am  able to play test.raw in application like cooledit

But if I do
$gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! audioresample ! alsasink
it goes into infinite loop.
The alsasink plugin is not able to detect the alsadriver I guess.
How do I debug this problem?
Pls help me

Regards
Vinod James

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