<div dir="ltr">Hi Mike, Here is the complete code of the plugin. Sorry there are lot of commented part in it. I have revisited it several times but not able to find what could be wrong. Please help me on this. /* * GStreamer * Copyright (C) 2005 Thomas Vander Stichele <<a href="mailto:thomas@apestaart.org">thomas@apestaart.org</a>> * Copyright (C) 2005 Ronald S. Bultje <<a href="mailto:rbultje@ronald.bitfreak.net">rbultje@ronald.bitfreak.net</a>> * Copyright (C) 2010 durgesh <<<a href="mailto:user@hostname.org">user@hostname.org</a>>> * * Permission is hereby granted, free of charge, to any person obtaining a * copy of this software and associated documentation files (the "Software"), * to deal in the Software without restriction, including without limitation * the rights to use, copy, modify, merge, publish, distribute, sublicense, * and/or sell copies of the Software, and to permit persons to whom the * Software is furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER * DEALINGS IN THE SOFTWARE. * * Alternatively, the contents of this file may be used under the * GNU Lesser General Public License Version 2.1 (the "LGPL"), in * which case the following provisions apply instead of the ones * mentioned above: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:element-droidaudiosrc * * FIXME:Describe droidaudiosrc here. * * <refsect2> * <title>Example launch line</title> * |[ * gst-launch -v -m fakesrc ! droidaudiosrc ! fakesink silent=TRUE * ]| * </refsect2> */ #ifdef HAVE_CONFIG_H # include <config.h> #endif #include <gst/gst.h> #include <gst/gstutils.h> #include "gstdroidaudiosrc.h" GST_DEBUG_CATEGORY_STATIC (gst_droidaudiosrc_debug); #define GST_CAT_DEFAULT gst_droidaudiosrc_debug #define DEFAULT_BUFFERTIME (500*GST_MSECOND) / (GST_USECOND) #define DEFAULT_LATENCYTIME (50*GST_MSECOND) / (GST_USECOND) /* Filter signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { PROP_0, PROP_SILENT }; /* the capabilities of the inputs and outputs. * * describe the real formats here. */ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("ANY") ); /* static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("ANY") );*/ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, //GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "signed = (boolean) { TRUE, FALSE }, " "width = (int) 16, " "depth = (int) 16, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ];") ); //static GstElementClass *parent_class = NULL; GST_BOILERPLATE (Gstdroidaudiosrc, gst_droidaudiosrc, GstElement, GST_TYPE_ELEMENT); static void gst_droidaudiosrc_reset (Gstdroidaudiosrc * asrc); static void gst_droidaudiosrc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_droidaudiosrc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_droidaudiosrc_set_caps (GstPad * pad, GstCaps * caps); static GstFlowReturn gst_droidaudiosrc_chain (GstPad * pad, GstBuffer * buf); static GstCaps *gst_droidsrc_getcaps (GstBaseSrc * bsrc); static gboolean gst_droidsrc_open (GstAudioSrc * droidsrc); static gboolean gst_droidsrc_prepare (GstAudioSrc * droidsrc, GstRingBufferSpec * spec); static gboolean gst_droidsrc_unprepare (GstAudioSrc * droidsrc); static gboolean gst_droidsrc_close (GstAudioSrc * droidsrc); static guint gst_droidsrc_read (GstAudioSrc * droidsrc, gpointer data, guint length); static guint gst_droidsrc_delay (GstAudioSrc * droidsrc); static void gst_droidsrc_reset (GstAudioSrc * droidsrc); static void gst_droidaudiosrc_finalise(GObject* object); static void gst_droidaudiosrc_dispose (GObject * object); /* GObject vmethod implementations */ #define PDB static void gst_droidaudiosrc_base_init (gpointer gclass) { #ifdef PDB printf("base_init started"); #endif GstElementClass *element_class = GST_ELEMENT_CLASS (gclass); gst_element_class_set_details_simple(element_class, "droidaudiosrc", "Src/Audio", "Input from android audio record", "durgesh o mishra <a href="mailto:durgesh.mishra@gmail.com">durgesh.mishra@gmail.com</a>"); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_factory)); #ifdef PDB printf("base_init done"); #endif /* gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_factory));*/ } /* initialize the droidaudiosrc's class */ static void gst_droidaudiosrc_class_init (GstdroidaudiosrcClass * klass) { #ifdef PDB printf("class_init started"); #endif GObjectClass *gobject_class; GstElementClass *gstelement_class; //Durgesh GstBaseSrcClass *gstbasesrc_class; GstBaseAudioSrcClass *gstbaseaudiosrc_class; GstAudioSrcClass *gstaudiosrc_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesrc_class = (GstBaseSrcClass *) klass; gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; gstaudiosrc_class = (GstAudioSrcClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_droidaudiosrc_dispose); gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_droidaudiosrc_finalise); gobject_class->set_property = gst_droidaudiosrc_set_property; gobject_class->get_property = gst_droidaudiosrc_get_property; gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_droidsrc_getcaps); gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_droidaudiosrc_set_caps); // gstaudiosrc_class-> gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_droidsrc_open); gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_droidsrc_prepare); gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_droidsrc_unprepare); gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_droidsrc_close); gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_droidsrc_read); gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_droidsrc_delay); gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_droidsrc_reset); #ifdef PDB printf("class_init done"); #endif /* g_object_class_install_property (gobject_class, PROP_SILENT, g_param_spec_boolean ("silent", "Silent", "Produce verbose output ?", FALSE, G_PARAM_READWRITE));*/ } static void gst_droidaudiosrc_dispose (GObject * object) { #ifdef PDB printf("droidaudiosrc_dispose started"); #endif Gstdroidaudiosrc *asrc = GST_DROIDAUDIOSRC (object); if (asrc->probed_caps) { gst_caps_unref (asrc->probed_caps); asrc->probed_caps = NULL; } G_OBJECT_CLASS (parent_class)->dispose (object); #ifdef PDB printf("droidaudiosrc_dispose done"); #endif } static guint gst_droidsrc_delay (GstAudioSrc * droidsrc) { //Do nothing #ifdef PDB printf("delay done"); #endif return 0; } static void gst_droidaudiosrc_finalise (GObject * object) { #ifdef PDB printf("finalize started"); #endif Gstdroidaudiosrc *asrc = GST_DROIDAUDIOSRC (object); gst_droidaudiosrc_reset (asrc); G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object)); #ifdef PDB printf("finalise done"); #endif } /* initialize the new element * instantiate pads and add them to element * set pad calback functions * initialize instance structure */ static void gst_droidaudiosrc_reset (Gstdroidaudiosrc * asrc) { #ifdef PDB printf("droidaudiosrc_reset started"); #endif GST_DEBUG_OBJECT (asrc, "enter"); if (asrc->audiorecord_device != NULL) { audiorecord_device_release (asrc->audiorecord_device); asrc->audiorecord_device = NULL; } asrc->audiorecord_device = NULL; asrc->m_init = FALSE; asrc->m_audiosrc = NULL; #ifdef PDB printf("reset done"); #endif } static void gst_droidsrc_reset (GstAudioSrc * droidsrc) { #ifdef PDB printf("droidsrc_reset started"); #endif Gstdroidaudiosrc *asrc = GST_DROIDAUDIOSRC (droidsrc); GST_DEBUG_OBJECT(asrc,"enter"); gst_droidaudiosrc_reset(asrc); #ifdef PDB printf("droidsrc reset done"); #endif } static gboolean gst_droidsrc_close (GstAudioSrc * droidsrc) { #ifdef PDB printf("droidsrc_close started"); #endif Gstdroidaudiosrc *asrc = GST_DROIDAUDIOSRC (droidsrc); GST_DEBUG_OBJECT (asrc, "enter"); if (asrc->audiorecord_device != NULL) { GST_DEBUG_OBJECT (asrc, "release audio record device"); audiorecord_device_stop (asrc->audiorecord_device); audiorecord_device_release (asrc->audiorecord_device); asrc->audiorecord_device = NULL; } #ifdef PDB printf("droidsrc close done"); #endif return TRUE; } static void gst_droidaudiosrc_init (Gstdroidaudiosrc * asrc, GstdroidaudiosrcClass * gclass) { #ifdef PDB printf("droidaudiosrc_init started"); #endif GST_DEBUG_OBJECT (asrc, "initializing android audio source"); //gst_droidaudiosrc_reset(asrc->audiorecord_device); gst_droidaudiosrc_reset(asrc); //This code may require to be deleted. GstPad* src_pad = gst_pad_new_from_static_template (&src_factory, "src"); gst_pad_set_getcaps_function (src_pad, GST_DEBUG_FUNCPTR(gst_pad_proxy_getcaps)); gst_element_add_pad (GST_ELEMENT (asrc), src_pad); #ifdef PDB printf("droid src init done"); #endif /* alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE); alsasrc->cached_caps = NULL; */ // alsasrc->alsa_lock = g_mutex_new (); /* filter->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); gst_pad_set_setcaps_function (filter->sinkpad, GST_DEBUG_FUNCPTR(gst_droidaudiosrc_set_caps)); gst_pad_set_getcaps_function (filter->sinkpad, GST_DEBUG_FUNCPTR(gst_pad_proxy_getcaps)); gst_pad_set_chain_function (filter->sinkpad, GST_DEBUG_FUNCPTR(gst_droidaudiosrc_chain)); filter->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); gst_pad_set_getcaps_function (filter->srcpad, GST_DEBUG_FUNCPTR(gst_pad_proxy_getcaps)); gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad); gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad); filter->silent = FALSE;*/ } static void gst_droidaudiosrc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { /*Gstdroidaudiosrc *filter = GST_DROIDAUDIOSRC (object); switch (prop_id) { case PROP_SILENT: filter->silent = g_value_get_boolean (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; }*/ #ifdef PDB printf("set property done"); #endif // Do Nothing } static void gst_droidaudiosrc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { /*Gstdroidaudiosrc *filter = GST_DROIDAUDIOSRC (object); switch (prop_id) { case PROP_SILENT: g_value_set_boolean (value, filter->silent); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; }*/ //Do Nothing #ifdef PDB printf("get property done"); #endif } /* GstElement vmethod implementations */ /* this function handles the link with other elements */ static gboolean gst_droidaudiosrc_set_caps (GstPad * pad, GstCaps * caps) { /* Gstdroidaudiosrc *filter; GstPad *otherpad; filter = GST_DROIDAUDIOSRC (gst_pad_get_parent (pad)); otherpad = (pad == filter->srcpad) ? filter->sinkpad : filter->srcpad; gst_object_unref (filter); return gst_pad_set_caps (otherpad, caps);*/ //do nothing #ifdef PDB printf("set caps done"); #endif return FALSE; } static GstCaps * gst_droidsrc_getcaps (GstBaseSrc * bsrc) { GstElementClass *element_class; GstPadTemplate *pad_template; Gstdroidaudiosrc *asrc; GstCaps *caps; #ifdef PDB printf("droidaudiosrc_getcaps started"); #endif asrc = GST_DROIDAUDIOSRC (bsrc); GST_DEBUG_OBJECT (asrc, "enter,%p", asrc->audiorecord_device); /*if(asrc->audiorecord_device == NULL || asrc->m_init == FALSE) { caps = gst_caps_copy(gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD(bsrc))); } else*/ if (asrc->probed_caps) caps = gst_caps_copy(asrc->probed_caps); else { caps = gst_caps_new_simple ("audio/x-raw-int", "signed",G_TYPE_BOOLEAN, FALSE, "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "rate", G_TYPE_INT,8000 , "channels", G_TYPE_INT, 2, NULL); if (caps && !gst_caps_is_empty(caps)) { asrc->probed_caps = gst_caps_copy(caps); } } #ifdef PDB printf("get caps done"); #endif return caps; } /* chain function * this function does the actual processing */ /* static GstFlowReturn gst_droidaudiosrc_chain (GstPad * pad, GstBuffer * buf) { Gstdroidaudiosrc *filter; filter = GST_DROIDAUDIOSRC (GST_OBJECT_PARENT (pad)); if (filter->silent == FALSE) g_print ("I'm plugged, therefore I'm in.\n"); return gst_pad_push (filter->srcpad, buf); }*/ /* entry point to initialize the plug-in * initialize the plug-in itself * register the element factories and other features */ static gboolean droidaudiosrc_init (GstPlugin * droidaudiosrc) { #ifdef PDB printf("droidaudiosrc_init started"); #endif GST_DEBUG_CATEGORY_INIT (gst_droidaudiosrc_debug, "droidaudiosrc", 0, "Template droidaudiosrc"); #ifdef PDB printf("audiosrc init done"); #endif return gst_element_register (droidaudiosrc, "droidaudiosrc", GST_RANK_NONE, GST_TYPE_DROIDAUDIOSRC); } static gboolean gst_droidsrc_open (GstAudioSrc * droidsrc) { Gstdroidaudiosrc *asrc = GST_DROIDAUDIOSRC (droidsrc); GstBaseAudioSrc *baseaudiosrc = (GstBaseAudioSrc *) asrc; #ifdef PDB printf("droidsrc_open started"); #endif GST_DEBUG_OBJECT (asrc, "enter"); g_return_val_if_fail (asrc != NULL, FALSE); baseaudiosrc->buffer_time = DEFAULT_BUFFERTIME; baseaudiosrc->latency_time = DEFAULT_LATENCYTIME; if (asrc->audiorecord_device == NULL) { if (asrc->m_audiosrc) { if (!(asrc->audiorecord_device = audiorecord_device_open(asrc->m_audiosrc))) goto failed_creation; GST_DEBUG_OBJECT (asrc, "open an existed audio record, %p", asrc->audiorecord_device); } else { if (!(asrc->audiorecord_device = audiorecord_device_create ())) goto failed_creation; GST_DEBUG_OBJECT (asrc, "create a new audio record, %p", asrc->audiorecord_device); } } return TRUE; /* ERRORS */ failed_creation: { GST_ELEMENT_ERROR (asrc, RESOURCE, SETTINGS, (NULL), ("Failed to create AudioRecord")); return FALSE; } } static gboolean gst_droidsrc_prepare (GstAudioSrc * droidsrc, GstRingBufferSpec * spec) { Gstdroidaudiosrc *asrc; asrc = GST_DROIDAUDIOSRC (droidsrc); GST_DEBUG_OBJECT (asrc, "enter"); if( audiorecord_device_set (asrc->audiorecord_device, 3, spec->channels, spec->rate, spec->segsize) == -1) goto failed_creation; asrc->m_init = TRUE; //audioflinger->m_init = TRUE; // gst_audioflinger_sink_set_volume (audioflinger, audioflinger->m_volume); // gst_audioflinger_sink_set_mute (audioflinger, audioflinger->m_mute); spec->bytes_per_sample = (spec->width / 8) * spec->channels; asrc->bytes_per_sample = spec->bytes_per_sample; GST_DEBUG_OBJECT (asrc, "channels: %d, rate: %d, width: %d, got segsize: %d, segtotal: %d, " "frame count: %d, frame size: %d", spec->channels, spec->rate, spec->width, spec->segsize, spec->segtotal, audiorecord_device_frameCount (asrc->audiorecord_device), audiorecord_device_frameSize (asrc->audiorecord_device) ); /* GST_DEBUG_OBJECT (asrc, "pause device"); audiorecord_device_pause (asrc->audiorecord_device); */ return TRUE; /* ERRORS */ failed_creation: { GST_ELEMENT_ERROR (asrc, RESOURCE, SETTINGS, (NULL), ("Failed to create Audiorecord for format %d", spec->format)); return FALSE; } dodgy_width: { GST_ELEMENT_ERROR (asrc, RESOURCE, SETTINGS, (NULL), ("Unhandled width %d", spec->width)); return FALSE; } } static gboolean gst_droidsrc_unprepare (GstAudioSrc * droidsrc) { Gstdroidaudiosrc *asrc; asrc = GST_DROIDAUDIOSRC (droidsrc); GST_DEBUG_OBJECT (asrc, "enter"); //audioflinger = GST_AUDIOFLINGERSINK (asink); if (asrc->audiorecord_device != NULL) { GST_DEBUG_OBJECT (asrc, "release flinger device"); audiorecord_device_stop(asrc->audiorecord_device); asrc->m_init = FALSE; } return TRUE; } static guint gst_droidsrc_read (GstAudioSrc * droidsrc, gpointer data, guint length) { Gstdroidaudiosrc *asrc; asrc = GST_DROIDAUDIOSRC (droidsrc); //GstAudioFlingerSink *audioflinger; guint ret = 0; // audioflinger = GST_AUDIOFLINGERSINK (asink); GST_INFO_OBJECT (asrc, "read length=%d", length); if (asrc->audiorecord_device == NULL || asrc->m_init == FALSE) { GST_WARNING_OBJECT (asrc, "Cannot write when device isn't initialized. Consume data"); return length; } /* FIXME : Instead of using write, we should be using obtainBuffer/releaseBuffer * which gives us a more granular feedback. */ ret = audiorecord_device_read (asrc->audiorecord_device, data, length); if (ret == 0) { GST_WARNING_OBJECT (asrc, "read failure"); ret = length; } GST_INFO_OBJECT (asrc, "read=%u", ret); return ret; } /*static guint gst_droidsrc_delay (GstAudioSrc * asrc); static void gst_droidsrc_reset (GstAudioSrc * asrc);*/ /* PACKAGE: this is usually set by autotools depending on some _INIT macro * in <a href="http://configure.ac">configure.ac</a> and then written into and defined in config.h, but we can * just set it ourselves here in case someone doesn't use autotools to * compile this code. GST_PLUGIN_DEFINE needs PACKAGE to be defined. */ #ifndef PACKAGE #define PACKAGE "myfirstdroidaudiosrc" #endif GST_PLUGIN_DEFINE ( GST_VERSION_MAJOR, GST_VERSION_MINOR, "droidaudiosrc", "Template droidaudiosrc", droidaudiosrc_init, VERSION, "LGPL", "GStreamer", "<a href="http://gstreamer.net/">http://gstreamer.net/</a>" ) Kind Regards -Durgesh O Mishra <div class="gmail_quote">On 18 February 2010 23:41, Michael Smith <<a href="mailto:msmith@xiph.org">msmith@xiph.org</a>> wrote: <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">On Thu, Feb 18, 2010 at 9:55 AM, Durgesh Mishra <div class="im"><<a href="mailto:durgesh.mishra@gmail.com">durgesh.mishra@gmail.com</a>> wrote: > Hi Mike, > </div><div class="im">> The problem is resolved now. Here is something that I did. > > I took reference of AlsaSrc plugin for writing this DroidAudioSrc plugin. > > I did write these statements in droidaudiosrc_init() function. > > GstPad* src_pad = gst_pad_new_from_static_template (&src_factory, "src"); > gst_pad_set_getcaps_function (src_pad, > > GST_DEBUG_FUNCPTR(gst_pad_proxy_getcaps)); > gst_element_add_pad (GST_ELEMENT (asrc), src_pad); </div>You should not be adding the src pad yourself - if you're needing this, then that means there's something else wrong in your plugin - the base source should be adding the pad for you. If it's not doing that, then something isn't initialising properly - and you'll probably have many more problems. So you should revert that change, and figure out the real cause of your bug. Mike ------------------------------------------------------------------------------ Download Intel&#174; 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. <a href="http://p.sf.net/sfu/intel-sw-dev" target="_blank">http://p.sf.net/sfu/intel-sw-dev</a> <div><div></div><div class="h5">_______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> </div></div></blockquote></div> -- "If you only have a hammer, you tend to see every problem as a nail." </div>