From: Shenhong W. <qc...@ho...> - 2008-06-18 06:20:52
|
Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline |
From: Zhao Liang-E. <E3...@mo...> - 2008-06-18 06:29:26
|
Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> |
From: Shenhong W. <qc...@ho...> - 2008-06-18 08:43:33
|
Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 14:29:27 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best RegardsZhao Liang From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline |
From: Zhao Liang-E. <E3...@mo...> - 2008-06-18 08:49:08
|
Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best Regards Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PM To: Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 14:29:27 +0800 From: E3...@mo... To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> |
From: Shenhong W. <qc...@ho...> - 2008-06-18 08:54:39
|
Zhao Liang:Thanks! Now we use a queue before the aac decoder &alsasink. How to check the queue is empty and pause/restart pipeline? hehe...thanks! Best Regards! Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:49:08 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best RegardsZhao Liang From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PMTo: Zhao Liang-E3423C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang:Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards!Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 14:29:27 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best RegardsZhao Liang From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx |
From: Zhao Bin-E. <bi...@mo...> - 2008-06-18 08:55:39
|
yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423C Sent: Wednesday, June 18, 2008 4:49 PM To: Shenhong Wang; gst...@li... Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best Regards Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PM To: Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 14:29:27 +0800 From: E3...@mo... To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> |
From: Shenhong W. <qc...@ho...> - 2008-06-18 09:04:45
|
Thanks! Brad. However I use two queues for audio and video separately but one pipeline. So it would be impossible for me to pause the pipeline? because the application can play video very well even the audio is blocked. Why the alsasink will drop all packets(frames) after a break or so? thanks again Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:55:38 +0800From: bi...@mo...To: E3...@mo...; qc...@ho...; gst...@li... yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423CSent: Wednesday, June 18, 2008 4:49 PMTo: Shenhong Wang; gst...@li...Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best RegardsZhao Liang From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PMTo: Zhao Liang-E3423C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang:Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards!Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 14:29:27 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best RegardsZhao Liang From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline |
From: Stefan K. <en...@ho...> - 2008-06-22 05:56:39
|
hi, you can use multiqueue, which is severl synchronized queues. But I belive you problem needs to be fixed elsewhere. Can you tell us how the whole pipeline looks like? You said your aac-decoder does not set timestamps? Is it putting GST_CLOCK_TIME_NONE there (don't use 0)? You should implement qos and timestamps. The sink will tell you whats the next expected timestamp and your aac decoder could skip packets that come late already. Stefan Shenhong Wang schrieb: > Thanks! Brad. > However I use two queues for audio and video separately but one > pipeline. So it would be impossible for me to pause the pipeline? > because the application can play video very well even the audio is blocked. > Why the alsasink will drop all packets(frames) after a break or so? > thanks again > > Shenhong > > > > > ------------------------------------------------------------------------ > Subject: RE: [gst-embedded] Question on gst_plugin alsasink > Date: Wed, 18 Jun 2008 16:55:38 +0800 > From: bi...@mo... > To: E3...@mo...; qc...@ho...; > gst...@li... > > > > yes, you can refernce how to use queue. you can set water mark in > queue.And then post message to bus if lower than mater mark. in your > main app you can recieve the message to pause the pipeline. > > if higher water mark, you can use the same mechanism. > > > > > ------------------------------------------------------------------------ > *From:* gst...@li... > [mailto:gst...@li...] *On Behalf > Of *Zhao Liang-E3423C > *Sent:* Wednesday, June 18, 2008 4:49 PM > *To:* Shenhong Wang; gst...@li... > *Subject:* Re: [gst-embedded] Question on gst_plugin alsasink > > Hi shenhong, > > A simply solution you can try. > > Put a queue before alsasink, when queue is dry, pause pipeline, and > restart pipeline when queue bufferred enough data. > > > > *Best Regards > Zhao Liang * > > ------------------------------------------------------------------------ > *From:* Shenhong Wang [mailto:qc...@ho...] > *Sent:* Wednesday, June 18, 2008 4:44 PM > *To:* Zhao Liang-E3423C; gst...@li... > *Subject:* RE: [gst-embedded] Question on gst_plugin alsasink > > Hi, Zhao Liang: > Generally, the aacdec &alsasink will not play out any audio > frames(packets) after its source element has a break to send audio > frames (packets) to them. It looks the alsasink drops all > frames(packets) from the break. The break is needed because we have > more video frames and sometime the wireless signal is not good. > It looks the aacdec is slower than the expectation from alsasink.If > so, how to fix the issue? thanks! > > best Regards! > Shenhong > > > > > > > > > ------------------------------------------------------------------------ > Subject: RE: [gst-embedded] Question on gst_plugin alsasink > Date: Wed, 18 Jun 2008 14:29:27 +0800 > From: E3...@mo... > To: qc...@ho...; gst...@li... > > Hi Shenhong, > > Your issue is very similar with the issue I even met. I think it > is due to gstbaseaudiosink/gstaudiosink, it will drop the > packets by gstringbuffer when read rate is bigger than write > rate in ringbuffer, please see gstringbuffer.c > gst_ring_buffer_commit_full (). > > For the rootcause, I think maybe the alsasink audiodevice buffer > is too big or your aac decoder is too slow. > > > *Best Regards > Zhao Liang* > > ------------------------------------------------------------------------ > *From:* gst...@li... > [mailto:gst...@li...] *On > Behalf Of *Shenhong Wang > *Sent:* Wednesday, June 18, 2008 2:21 PM > *To:* gst...@li... > *Subject:* [gst-embedded] Question on gst_plugin alsasink > > > Dear all, > Now we are using alsasink to play audio on Marvell PXA310 board. > The audio is aac format. The audio frames(packets) > are frequently sent to the aac decoder & alsasink to play out. > Unfortunately only the begining frames can be played out and > then nothing is played out. > If we save those audio frames into a file, the aac > decoder&alsasink can be successfully played out. It means the > audio frames are ok. > Could anyone tell me what's the difference for alsasink to > process audio packets and files? How to fix the above issue? > thank you very much! > > Best Regards! > Shenhong WANG > > ------------------------------------------------------------------------ > Connect to the next generation of MSN Messenger Get it now! > <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline> > > > ------------------------------------------------------------------------ > Connect to the next generation of MSN Messenger Get it now! > <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline> > > > ------------------------------------------------------------------------ > Connect to the next generation of MSN Messenger Get it now! > <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline> > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > > > ------------------------------------------------------------------------ > > _______________________________________________ > Gstreamer-embedded mailing list > Gst...@li... > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded |
From: Zhao Bin-E. <bi...@mo...> - 2008-06-18 09:08:13
|
I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). Please check code in gstbaseaudiosink.c and gstaudiosink.c i remember the sig_write is lower than sig_done,sink will drop the buffer. ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 5:05 PM To: Zhao Bin-E6223C; Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Thanks! Brad. However I use two queues for audio and video separately but one pipeline. So it would be impossible for me to pause the pipeline? because the application can play video very well even the audio is blocked. Why the alsasink will drop all packets(frames) after a break or so? thanks again Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 16:55:38 +0800 From: bi...@mo... To: E3...@mo...; qc...@ho...; gst...@li... yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423C Sent: Wednesday, June 18, 2008 4:49 PM To: Shenhong Wang; gst...@li... Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best Regards Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PM To: Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 14:29:27 +0800 From: E3...@mo... To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> |
From: Shenhong W. <qc...@ho...> - 2008-06-19 01:46:41
|
Hi, Brad or Zhao Liang: Is it possible for you to publish an example - how to post a message to bus and pause/play pipeline? thanks a lot! Best Regards! Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 17:08:09 +0800From: bi...@mo...To: qc...@ho...; E3...@mo...; gst...@li... I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). Please check code in gstbaseaudiosink.c and gstaudiosink.c i remember the sig_write is lower than sig_done,sink will drop the buffer. From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 5:05 PMTo: Zhao Bin-E6223C; Zhao Liang-E3423C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Thanks! Brad.However I use two queues for audio and video separately but one pipeline. So it would be impossible for me to pause the pipeline? because the application can play video very well even the audio is blocked. Why the alsasink will drop all packets(frames) after a break or so? thanks again Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:55:38 +0800From: bi...@mo...To: E3...@mo...; qc...@ho...; gst...@li... yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423CSent: Wednesday, June 18, 2008 4:49 PMTo: Shenhong Wang; gst...@li...Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best RegardsZhao Liang From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PMTo: Zhao Liang-E3423C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang:Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards!Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 14:29:27 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best RegardsZhao Liang From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! Connect to the next generation of MSN Messenger Get it now! Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx |
From: Zhao Liang-E. <E3...@mo...> - 2008-06-19 05:08:21
|
please check queue signals "underrun" "overrun" .... Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Thursday, June 19, 2008 9:47 AM To: Zhao Bin-E6223C; Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Brad or Zhao Liang: Is it possible for you to publish an example - how to post a message to bus and pause/play pipeline? thanks a lot! Best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 17:08:09 +0800 From: bi...@mo... To: qc...@ho...; E3...@mo...; gst...@li... I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). Please check code in gstbaseaudiosink.c and gstaudiosink.c i remember the sig_write is lower than sig_done,sink will drop the buffer. ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 5:05 PM To: Zhao Bin-E6223C; Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Thanks! Brad. However I use two queues for audio and video separately but one pipeline. So it would be impossible for me to pause the pipeline? because the application can play video very well even the audio is blocked. Why the alsasink will drop all packets(frames) after a break or so? thanks again Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 16:55:38 +0800 From: bi...@mo... To: E3...@mo...; qc...@ho...; gst...@li... yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423C Sent: Wednesday, June 18, 2008 4:49 PM To: Shenhong Wang; gst...@li... Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best Regards Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PM To: Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 14:29:27 +0800 From: E3...@mo... To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Get news, entertainment and everything you care about at Live.com. Check it out! <http://www.live.com/getstarted.aspx> |
From: Zhao Bin-E. <bi...@mo...> - 2008-06-18 07:01:08
|
Hi, your issue seems that audio frame is delayed when arrivering alsasink. basesink will drop the delayed buffer and you could'nt hear any sound. Please check your timestamp of buffer. Brad ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> |
From: Shenhong W. <qc...@ho...> - 2008-06-18 08:38:55
|
Brad, thanks! At the moment I didn't put any timestamp on the frames(buffer) yet. Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drop all frames(packets) from the break. Why? How to fix it? Thanks! Best Regards! Shenhong WANG Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 15:01:06 +0800From: bi...@mo...To: qc...@ho...; gst...@li... Hi, your issue seems that audio frame is delayed when arrivering alsasink. basesink will drop the delayed buffer and you could'nt hear any sound. Please check your timestamp of buffer. Brad From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE |
From: Zhao Bin-E. <bi...@mo...> - 2008-06-18 08:42:35
|
Why your source element has a break? do you use live source? I suggest you pause the pipleline at the moment. Brad ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:39 PM To: Zhao Bin-E6223C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Brad, thanks! At the moment I didn't put any timestamp on the frames(buffer) yet. Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drop all frames(packets) from the break. Why? How to fix it? Thanks! Best Regards! Shenhong WANG ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 15:01:06 +0800 From: bi...@mo... To: qc...@ho...; gst...@li... Hi, your issue seems that audio frame is delayed when arrivering alsasink. basesink will drop the delayed buffer and you could'nt hear any sound. Please check your timestamp of buffer. Brad ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Explore the seven wonders of the world Learn more! <http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBR E> |
From: Shenhong W. <qc...@ho...> - 2008-06-18 08:45:08
|
Brad, Yes, now I am using a live source via wireless signal. Questions: a) I don't know when the break will happen b) I don't know how to pause the pipeline How to move it forward? thanks! Best Regards! Shenhong WANG Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:42:35 +0800From: bi...@mo...To: qc...@ho...; gst...@li... Why your source element has a break? do you use live source? I suggest you pause the pipleline at the moment. Brad From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:39 PMTo: Zhao Bin-E6223C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Brad,thanks! At the moment I didn't put any timestamp on the frames(buffer) yet.Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drop all frames(packets) from the break. Why? How to fix it? Thanks! Best Regards!Shenhong WANG Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 15:01:06 +0800From: bi...@mo...To: qc...@ho...; gst...@li... Hi, your issue seems that audio frame is delayed when arrivering alsasink. basesink will drop the delayed buffer and you could'nt hear any sound. Please check your timestamp of buffer. Brad From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! Explore the seven wonders of the world Learn more! _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE |