Hi,
I am developing a live audio streaming application using the gstreamer API’s. I have tried with the following API’s of gstreamer,[gst_element_factory_make()] which accepts file as the argument. But require gstreamer API which accepts byte* buffer as an argument as we are managing the connections in a different layer and my streaming-application will receive the live audio streaming buffer periodically from other components. I could have missed something as I am new to gstreamer and exploring this great library. Your help in this regard is highly appreciated, waiting for your response. Thanks & Regards, Gowri _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Hi,
I am developing a live audio streaming application using the gstreamer API’s. I have tried with the following API’s of gstreamer,[gst_element_factory_make()] which accepts file as the argument. But require gstreamer API which accepts byte* buffer as an argument as we are managing the connections in a different layer and my streaming-application will receive the live audio streaming buffer periodically from other components. I could have missed something as I am new to gstreamer and exploring this great library. Your help in this regard is highly appreciated, waiting for your response. Thanks & Regards, Gowri _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On Fri, 2011-11-25 at 10:14 +0530, Sree Gowri wrote:
> > I am developing a live audio streaming application using the gstreamer > API’s. I have tried with the following API’s of > gstreamer,[gst_element_factory_make()] which accepts file as the > argument. But require gstreamer API which accepts byte* buffer as an > argument as we are managing the connections in a different layer and > my streaming-application will receive the live audio streaming buffer > periodically from other components. Maybe you're looking for the appsrc element? (from gst-plugins-base) Cheers -Tim _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On Fri, Nov 25, 2011 at 4:55 PM, Tim-Philipp Müller <[hidden email]> wrote:
Hi, Thanks tim for your reply. I used appsrc to pass the buffer to the pipeline. From the file, buffer is read of size READBUF and then passed to gst_app_buffer_new. I have defined READBUF as 188*1000. What is the ideal way of the buffer size to be? Is there any standard size of the buffer to be passed to the gst_app_buffer_new. code snippet as follows " const int READBUF = 188*1000; bytes_read = fread(data, 1, READBUF, fp); buf =( GstBuffer *) gst_app_buffer_new (data, READBUF, g_free, data); " and the pipeline is "pipeline= gst_parse_launch("appsrc name=mysource !audio/mpeg ! mad ! alsasink", NULL);" However I can listen only to some portion of audio from the file and then gets segmentation fault. Not sure why the entire buffer is not playing. I am investigating on this. Kindly let me know if I am missing anything. Soon will post the entire code. Currently buffer contains only mp3 data. In case if buffer contains mp3 data+metadata, does gstreamer skips the metadata and plays only audio data?? or do we need to remove the metadata and then give only audio data to gstreamer. Kindly let me know on this. Thanks and Regards, Gowri
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