search for: audioresampl

...i googled it but still can't find any answer: how can i set default audio track? I transcode my dvds to ogg, i use gst-launch for it: gst-launch filesrc location=stream.dump ! mpegpsdemux name=demux \ oggmux name=mux ! filesink location=out.ogg \ demux.audio_80 ! a52dec mode=2 ! audioconvert ! audioresample ! audio/x-raw-float,rate=44100,channels=2 ! audiorate ! \ taginject tags="language-code=eng" ! vorbisenc ! queue ! mux. \ demux.audio_81 ! a52dec mode=2 ! audioconvert ! audioresample ! audio/x-raw-float,rate=44100,channels=2 ! audiorate ! \ taginject tags="language-code=ger" !...

...9;t find any answer: how can i set default > audio track? > > I transcode my dvds to ogg, i use gst-launch for it: > gst-launch filesrc location=stream.dump ! mpegpsdemux name=demux \ > oggmux name=mux ! filesink location=out.ogg \ > demux.audio_80 ! a52dec mode=2 ! audioconvert ! audioresample ! > audio/x-raw-float,rate=44100,channels=2 ! audiorate ! \ > taginject tags="language-code=eng" ! vorbisenc ! queue ! mux. \ > demux.audio_81 ! a52dec mode=2 ! audioconvert ! audioresample ! > audio/x-raw-float,rate=44100,channels=2 ! audiorate ! \ > taginject tags="...

...t locally. gst-launch filesrc location=AstriDevCon_Europe_2006.mov ! qtdemux name=demux ! ffdec_h263 ! videoscale ! video/x-raw-yuv,width=352,height=288 ! ffenc_h263 rtp-payload-size=512 ! rtph263pay ! asteriskh263 ! filesink location=avout.h263 demux. ! queue ! ffdec_adpcm_ima_qt ! audioconvert ! audioresample ! audio/x-raw-int,rate=8000,channels=1 ! wavenc ! filesink location=avout.wav I was wondering if this script is expecting some specific type of file or if I am doing something else wrong. Any pointers in how to generate .h263 video files for asterisk would be helpful. ------- http://nmap.org...

...->frame) + goto start; + } /* decode next action */ action = *pc; commit 397be351d19a0f66a2b0b4836416a67b0098d233 Author: Benjamin Otte <otte at gnome.org> Date: Sun Oct 28 22:49:01 2007 +0100 rewrite to allow any output format This gets rid of long lags when audioresample takes its time diff --git a/libswfdec/swfdec_codec_gst.c b/libswfdec/swfdec_codec_gst.c index b449533..35f00e5 100644 --- a/libswfdec/swfdec_codec_gst.c +++ b/libswfdec/swfdec_codec_gst.c @@ -148,14 +148,16 @@ swfdec_gst_get_element (GstCaps *caps) static GstPad * swfdec_gst_connect_srcpad (Gst...

...e to change the Mixer settings. Also, I took off the 20 dB boost for the mic. In Sound Preferences > Devices > Audio Conferencing > Sound Capture > I have it set to OSS. Am I correct, to have that set to OSS? When I did a test, I got this error message: gconfaudiosrc ! audioconvert ! audioresample ! gconfaudiosink profile=chat: Unable to open device /dev/dsp for recording: Device or resource busy When I try to do a test call, with the Skype robot, when they would play back what I have recorded, I get 2 beeps, with only 1 or 2 seconds in between them. When I try that on MS Windows, there is...

...Otte <otte at gnome.org> Date: Fri Oct 26 21:13:20 2007 +0200 rewrite GStreamer backend to directly use the element and not create a pipeline That was Wim's suggested method, and it works fine. The only issue right now is that GStreamer also does the resampling, and a) audioresample is slow and b) the whole process takes a lot of time. This causes the app to stop for half a second when a long sample gets decoded. Not good. diff --git a/libswfdec/swfdec_codec_gst.c b/libswfdec/swfdec_codec_gst.c index 08cf154..bd8f95a 100644 --- a/libswfdec/swfdec_codec_gst.c +++ b/li...