issue with appsrc and audioresample
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Hello,
I am trying to write a sample example where I can push and pull audio buffers via appsrc and appsink to a simple pipeline that will resample the audio. I created a simple example that sets up the pipeline and tries to push and pull a single buffer. When I run the code below, I get the following output: ./rxpipeline --gst-debug-level=2 0:00:00.148913625 22049 0x94e42c0 WARN basetransform gstbasetransform.c:1969:gst_base_transform_handle_buffer:<audiores> error: not negotiated 0:00:00.149326247 22049 0x94e42c0 WARN basetransform gstbasetransform.c:1969:gst_base_transform_handle_buffer:<audiores> error: not negotiated 0:00:00.149653838 22049 0x94e42c0 WARN basesrc gstbasesrc.c:2378:gst_base_src_loop:<appsrc> error: Internal data flow error. 0:00:00.149971963 22049 0x94e42c0 WARN basesrc gstbasesrc.c:2378:gst_base_src_loop:<appsrc> error: streaming task paused, reason not-negotiated (-4) (rxpipeline:22049): GStreamer-CRITICAL **: gst_mini_object_unref: assertion `mini_object->refcount > 0' failed I have attached the source code below. Please also note that if I remove the resample element and caps filter from the pipeline, I can successfully push and pull the buffer. Any help would be greatly appreciated. Thanks. Best regards, Nadeem --------------------------------------------- #include <gst/gst.h> #include <gst/app/gstappbuffer.h> #include <gst/app/gstappsrc.h> #include <gst/app/gstappsink.h> #define BUFFER_SIZE 160 int main (int argc, char *argv[]) { GstBuffer *app_buffer_src, *app_buffer_sink; GMainLoop *loop; GstElement *pipeline; GstElement *appsrc; GstElement *appsink; GstElement *audiores; GstElement *capsfilt_res; // initialize gst_init(&argc, &argv); // create elements loop = g_main_loop_new (NULL, FALSE); pipeline = gst_pipeline_new(NULL); appsrc = gst_element_factory_make("appsrc","appsrc"); appsink = gst_element_factory_make("appsink","appsink"); audiores = gst_element_factory_make("audioresample","audiores"); capsfilt_res = gst_element_factory_make("capsfilter","capsfilt_res"); // setup pipeline GstCaps *rescaps; rescaps = gst_caps_new_simple("audio/x-raw-int", "rate", G_TYPE_INT, (gint)48000, NULL); g_object_set(capsfilt_res,"caps",rescaps,NULL); gst_bin_add_many (GST_BIN (pipeline),appsrc,appsink,audiores,capsfilt_res,NULL); gboolean res; res = gst_element_link_many(appsrc,audiores,capsfilt_res,appsink,NULL); g_assert (res == TRUE); gst_element_set_state(pipeline,GST_STATE_PLAYING); // create buffer gpointer raw_buffer; raw_buffer = g_malloc0(BUFFER_SIZE); int ii = 0; for (ii=0; ii<BUFFER_SIZE; ii++) { ((guint8*)raw_buffer)[ii] = (guint8)ii; } // push buffer to pipeline via appsrc GstCaps *caps; caps = gst_caps_new_simple("audio/x-raw-int", "width", G_TYPE_INT, (gint)8, "depth", G_TYPE_INT, (gint)8, "channels" ,G_TYPE_INT, (gint)1, "rate",G_TYPE_INT,8000,"endianness",G_TYPE_INT,(gint)1234,NULL); app_buffer_src = gst_app_buffer_new (raw_buffer, BUFFER_SIZE, g_free, raw_buffer); GST_BUFFER_CAPS(app_buffer_src) = caps; gst_app_src_push_buffer(GST_APP_SRC(appsrc),app_buffer_src); gst_buffer_unref (app_buffer_src); // read buffers from pipeline via appsink app_buffer_sink = gst_app_sink_pull_buffer(GST_APP_SINK(appsink)); for (ii = 0; ii<GST_BUFFER_SIZE(app_buffer_sink); ii++ ) { printf("data(ii) %d\n",GST_BUFFER_DATA(app_buffer_sink)[ii]); } // cleanup gst_element_set_state(pipeline,GST_STATE_NULL); gst_object_unref(GST_OBJECT(pipeline)); return 0; } |
Re: issue with appsrc and audioresample
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On Sat, Feb 27, 2010 at 10:16 PM, na <[hidden email]> wrote:
I think your problem is here, you have to specify the full caps, the not negotiated error is right, appsrc cant tell you the kind of data you are streaming and your caps only says that it is int with a 48000 rate... but how much channels? and the depth/ width of the samples? you must give full caps or the pipeline is unable to verify what kind of data you are streaming. Only appsrc/appsink works because they don't need to know the kind of data that you are streaming, so they can negotiate with each other, but audioresample need full caps of the stream to work. Hope this helps ;) Someone correct me if i said something wrong. best regards, Katcipis g_object_set(capsfilt_res,"caps",rescaps,NULL); ------------------------------------------------------------------------------ Download Intel® Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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Hello Katcipis,
Thanks for your kind reply. I changed the caps on the appsrc buffer and caps filter to: GstCaps *rescaps; rescaps = gst_caps_new_simple("audio/x-raw-int", "rate", G_TYPE_INT, 48000, "channels" ,G_TYPE_INT, (gint)1, "endianness",G_TYPE_INT,(gint)1234, "width", G_TYPE_INT, (gint)8, "depth", G_TYPE_INT, (gint)8, "signed", G_TYPE_BOOLEAN, TRUE, NULL); g_object_set(capsfilt_res,"caps",rescaps,NULL); GstCaps *caps; caps = gst_caps_new_simple("audio/x-raw-int", "rate", G_TYPE_INT, 8000, "channels" ,G_TYPE_INT, (gint)1, "endianness",G_TYPE_INT,(gint)1234, "width", G_TYPE_INT, (gint)8, "depth", G_TYPE_INT, (gint)8, "signed", G_TYPE_BOOLEAN, TRUE, NULL); gst_buffer_set_caps(app_buffer_src,caps); However, the result is the same "not negotiated" error. I have attached the relevant debug output. If you do have any other ideas, please do let me know. Thanks Best regards, Nadeem 0:00:00.219005295 3697 0x884d008 INFO GST_EVENT gstevent.c:597:gst_event_new_new_segment_full: creating newsegment update 0, rate 1.000000, format bytes, start 0, stop -1, position 0 0:00:00.220594821 3697 0x884d008 INFO GST_STATES gstelement.c:2238:gst_element_continue_state:<appsrc> completed state change to PAUSED 0:00:00.220625877 3697 0x884d008 INFO GST_STATES gstelement.c:2251:gst_element_continue_state:<appsrc> posting state-changed READY to PAUSED 0:00:00.220644133 3697 0x884d008 INFO GST_STATES gstbin.c:2425:gst_bin_change_state_func:<pipeline0> child 'appsrc' changed state to 3(PAUSED) successfully 0:00:00.220765315 3697 0x8956458 WARN basetransform gstbasetransform.c:1969:gst_base_transform_handle_buffer:<audiores> error: not negotiated 0:00:00.220820395 3697 0x8956458 WARN basetransform gstbasetransform.c:1969:gst_base_transform_handle_buffer:<audiores> error: not negotiated 0:00:00.220877203 3697 0x8956458 INFO GST_ERROR_SYSTEM gstelement.c:1763:gst_element_message_full:<audiores> posting message: not negotiated 0:00:00.220930220 3697 0x8956458 INFO GST_ERROR_SYSTEM gstelement.c:1786:gst_element_message_full:<audiores> posted error message: not negotiated 0:00:00.220960984 3697 0x8956458 INFO basesrc gstbasesrc.c:2326:gst_base_src_loop:<appsrc> pausing after gst_pad_push() = not-negotiated 0:00:00.221001947 3697 0x8956458 WARN basesrc gstbasesrc.c:2378:gst_base_src_loop:<appsrc> error: Internal data flow error. 0:00:00.221025676 3697 0x8956458 WARN basesrc gstbasesrc.c:2378:gst_base_src_loop:<appsrc> error: streaming task paused, reason not-negotiated (-4) 0:00:00.221057825 3697 0x8956458 INFO GST_ERROR_SYSTEM gstelement.c:1763:gst_element_message_full:<appsrc> posting message: Internal data flow error. 0:00:00.221090188 3697 0x8956458 INFO GST_ERROR_SYSTEM gstelement.c:1786:gst_element_message_full:<appsrc> posted error message: Internal data flow error. 0:00:00.221205570 3697 0x8956458 INFO GST_STATES gstbin.c:2815:bin_handle_async_done:<pipeline0> committing state from READY to PAUSED, old pending PLAYING 0:00:00.221236480 3697 0x8956458 INFO GST_STATES gstbin.c:2844:bin_handle_async_done:<pipeline0> continue state change, pending PLAYING |
Re: issue with appsrc and audioresample
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Try to set the property "caps" on the appsrc instead of using a capsfilter. But setting full caps is still a need, maybe to work properly appsrc need to have the caps property defined, i never used appsrc, only appsink, and using only capsfilters usually worked when i needed then, but i remember that some problems with udpsrc where solved when i defined the "caps" property.
best regards, Katcipis On Mon, Mar 1, 2010 at 3:05 AM, na <[hidden email]> wrote:
------------------------------------------------------------------------------ Download Intel® Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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