Re: I tried to replace audio sink at runtime, but failed
Locked
Re: I tried to replace audio sink at runtime, but failed
 
|
Re: I tried to replace audio sink at runtime, but failed
|
|
After I added block for the src pad of previous element (mp3 decoder), I still got some warning of the new audio sink.
" <pulse_play_audio> warning: Internal data flow problem." "<pulse_play_audio> warning: Received buffer without a new-segment.Assuming timestamps start from 0." >-----Original Message----- >From: Antonio Marqués [mailto:[hidden email]] >Sent: 2009年3月2日 18:43 >To: Discussion of the development of GStreamer >Subject: Re: [gst-devel] I tried to replace audio sink at runtime, but failed > >I think you should first block the src pad of the previous element, and >then unlink the element. >It's explained in: >http://www.sfr-fresh.com/unix/privat/gstreamer-0.10.22.tar.gz:a/gstream >er-0.10.22/docs/design/part-block.txt >Regards >On Mon, 2009-03-02 at 10:13 +0100, Zhao, Halley wrote: >> Hi all: >> >> I try to replace audio sink as following, but failed, could you give >> me some advice >> >> >> >> /* pause */ >> >> gst_element_set_state (pipeline, GST_STATE_PAUSED); >> >> /* unlink and remove former alsa sink */ >> >> gst_element_unlink(decoder, alsaaudiosink); >> >> gst_bin_remove (GST_BIN (pipeline), alsaaudiosink); >> >> /* link to pulse audio sink */ >> >> pulseaudiosink = gst_element_factory_make ("pulsesink", >> "pulse_play_audio"); >> >> gst_bin_add (GST_BIN (pipeline), pulseaudiosink); >> >> gst_element_link(decoder, pulseaudiosink); >> >> /* start playing */ >> >> gst_element_set_state (pipeline, GST_STATE_PLAYING); >> >> >> >> >> >> >> >> ====complete source code==== >> >> /* example-begin helloworld.c */ >> >> #include <gst/gst.h> >> >> >> >> int >> >> main (int argc, char *argv[]) >> >> { >> >> GstElement *pipeline, *filesrc, *decoder, *alsaaudiosink = NULL, >> *pulseaudiosink = NULL; >> >> >> >> gst_init(&argc, &argv); >> >> >> >> if (argc != 2) { >> >> g_print ("usage: %s <mp3 filename>\n", argv[0]); >> >> exit (-1); >> >> } >> >> >> >> /* create a new pipeline to hold the elements */ >> >> pipeline = gst_pipeline_new ("pipeline"); >> >> >> >> /* create a disk reader */ >> >> filesrc = gst_element_factory_make ("filesrc", "disk_source"); >> >> g_object_set (G_OBJECT (filesrc), "location", argv[1], NULL); >> >> >> >> /* now it's time to get the decoder */ >> >> decoder = gst_element_factory_make ("mad", "decoder"); >> >> >> >> >> >> /* and an audio sink */ >> >> alsaaudiosink = gst_element_factory_make ("alsasink", >> "alsa_play_audio"); >> >> >> >> /* add objects to the main pipeline */ >> >> gst_bin_add_many (GST_BIN (pipeline), filesrc, decoder, >> alsaaudiosink, NULL); >> >> >> >> /* link src to sink */ >> >> gst_element_link(filesrc, decoder); >> >> gst_element_link(decoder,alsaaudiosink); >> >> >> >> /* start playing */ >> >> gst_element_set_state (pipeline, GST_STATE_PLAYING); >> >> >> >> static int is_playing = 1; >> >> static int is_quiting = 0; >> >> static int is_alsasink = 1; >> >> >> >> while (1) { >> >> if(!(gst_bin_iterate_elements (GST_BIN (pipeline)))) break; >> >> >> >> printf(" q:Quit, p:Pause/Play, t:Test: "); >> >> char ch =0 ; >> >> ch=getchar(); >> >> >> >> switch (ch) { >> >> case 'p': >> >> if(is_playing) { >> >> /* pause */ >> >> gst_element_set_state (pipeline, GST_STATE_PAUSED); >> >> >> >> } >> >> else { >> >> /* start playing */ >> >> gst_element_set_state (pipeline, GST_STATE_PLAYING); >> >> >> >> } >> >> is_playing = !is_playing; >> >> break; >> >> case 't': >> >> >> >printf("=============================================================== >=\n"); >> >> if(is_alsasink) { >> >> /* pause */ >> >> gst_element_set_state (pipeline, GST_STATE_PAUSED); >> >> // gst_element_set_state (pipeline, GST_STATE_NULL); >> >> sleep(1); >> >> >> >> gst_element_unlink(decoder, alsaaudiosink); >> >> gst_bin_remove (GST_BIN (pipeline), alsaaudiosink); >> >> gst_element_set_state (alsaaudiosink, GST_STATE_NULL); >> >> gst_object_unref (GST_OBJECT (alsaaudiosink)); >> >> >> >> pulseaudiosink = gst_element_factory_make ("pulsesink", >> "pulse_play_audio"); >> >> gst_bin_add (GST_BIN (pipeline), pulseaudiosink); >> >> gst_element_link(decoder, pulseaudiosink); >> >> sleep(1); >> >> >> >> /* start playing */ >> >> printf("pulse sink prepare to play:\n"); >> >> gst_element_set_state (pipeline, GST_STATE_PLAYING); >> >> } >> >> else { >> >> /* pause */ >> >> gst_element_set_state (pipeline, GST_STATE_PAUSED); >> >> // gst_element_set_state (pipeline, GST_STATE_NULL); >> >> sleep(1); >> >> >> >> gst_element_unlink(decoder, pulseaudiosink); >> >> gst_bin_remove (GST_BIN (pipeline), pulseaudiosink); >> >> gst_element_set_state (pulseaudiosink, GST_STATE_NULL); >> >> gst_object_unref (GST_OBJECT (pulseaudiosink)); >> >> >> >> alsaaudiosink = gst_element_factory_make ("alsasink", >> "alsa_play_audio"); >> >> gst_bin_add (GST_BIN (pipeline), alsaaudiosink); >> >> gst_element_link(decoder, alsaaudiosink); >> >> sleep(1); >> >> >> >> /* start playing */ >> >> printf("alsa sink prepare to play:\n"); >> >> gst_element_set_state (pipeline, GST_STATE_PLAYING); >> >> } >> >> >> >> is_alsasink = !is_alsasink; >> >> >> >> break; >> >> case 'q': >> >> is_quiting = 1; >> >> break; >> >> default: >> >> break; >> >> } >> >> >> >> if(is_quiting) break; >> >> >> >> >> >> GstFormat fmt = GST_FORMAT_TIME; >> >> gint64 pos, len; >> >> >> >> if (gst_element_query_position (pipeline, &fmt, &pos) >> >> && gst_element_query_duration (pipeline, &fmt, &len)) { >> >> g_print ("Time: %" GST_TIME_FORMAT " / %" GST_TIME_FORMAT >> "\n", >> >> GST_TIME_ARGS (pos), GST_TIME_ARGS (len)); >> >> } >> >> >> >> sleep(2); >> >> } >> >> >> >> >> >> /* stop the pipeline */ >> >> gst_element_set_state (pipeline, GST_STATE_NULL); >> >> >> >> /* we don't need a reference to these objects anymore */ >> >> gst_object_unref (GST_OBJECT (pipeline)); >> >> /* unreffing the pipeline unrefs the contained elements as well */ >> >> >> >> exit (0); >> >> } >> >> /* example-end helloworld.c */ >> >> ZHAO, Halley (Aihua) >> >> Email: [hidden email] >> >> Tel: +86(21)61166476 >> >> iNet: 8821-6476 >> >> SSG/OTC/UMD 3W033 >> >> >> >> >> >-- >Toni Marqués Marqués >Telefónica I+D >División de Tecnologías de Video >[hidden email] >933653188 > > > > > >----------------------------------------------------------------------- >------- >Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, >CA >-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise >-Strategies to boost innovation and cut costs with open source participation >-Receive a $600 discount off the registration fee with the source code: SFAD >http://p.sf.net/sfu/XcvMzF8H >_______________________________________________ >gstreamer-devel mailing list >[hidden email] >https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
mp3.c (7K)
Re: I tried to replace audio sink at runtime, but failed
|
|
Re: I tried to replace audio sink at runtime, but failed
|
|
| Free forum by Nabble | Edit this page |