Hi,
the audiosink/audiosrc is now working,
but there is an unexpected bug while stopping the pipeline.
** (test_audio_plugin:11726): [1;35mCRITICAL [0m **: [34m20:48:28.984 [0m:
Resources for ring buffer 0x555555a40040 still acquired
Thread #1 [test_audio_plug] 11726 [core: 1] (Suspended : Signal :
SIGTRAP:Trace/breakpoint trap)
0x7ffff7582c41
g_logv() at 0x7ffff7583f2b
g_log() at 0x7ffff758407f
gst_audio_ring_buffer_close_device() at 0x7ffff4adcc9b
0x7ffff4affc8e
Gst::Element_Class::change_state_vfunc_callback() at 0x7ffff7258281
gst_element_change_state() at gstelement.c:2,952 0x7ffff7af8bce
gst_element_set_state_func() at gstelement.c:2,906 0x7ffff7af9309
Gst::Element_Class::set_state_vfunc_callback() at 0x7ffff72581e1
gst_bin_element_set_state() at gstbin.c:2,602 0x7ffff7ad6942
gst_bin_change_state_func() at gstbin.c:2,944 0x7ffff7ad6942
gst_element_change_state() at gstelement.c:2,952 0x7ffff7af8bce
gst_element_set_state_func() at gstelement.c:2,906 0x7ffff7af9309
gst_bin_element_set_state() at gstbin.c:2,602 0x7ffff7ad6942
gst_bin_change_state_func() at gstbin.c:2,944 0x7ffff7ad6942
gst_pipeline_change_state() at gstpipeline.c:508 0x7ffff7b1eac2
gst_element_change_state() at gstelement.c:2,952 0x7ffff7af8bce
gst_element_continue_state() at gstelement.c:2,660 0x7ffff7af95ef
gst_element_change_state() at gstelement.c:2,991 0x7ffff7af8eb5
gst_element_continue_state() at gstelement.c:2,660 0x7ffff7af95ef
gst_element_change_state() at gstelement.c:2,998 0x7ffff7af8c26
gst_element_set_state_func() at gstelement.c:2,906 0x7ffff7af9309
main() at test_audio_plugin.cpp:51 0x55555555d1e2
i had to override create_ring_buffer_vfunc
Glib::RefPtr<Gst::AudioRingBuffer> create_ring_buffer_vfunc() override
{
const auto base =
static_cast<BaseClassType*>(g_type_class_peek_parent(
G_OBJECT_GET_CLASS(gobject_)) // Get the parent class of
the object class (The original underlying C class).
);
const auto newbase = (_GstAudioBaseSrcClass *) base;
if (newbase && newbase->create_ringbuffer) {
auto retval = Glib::wrap(
(*newbase->create_ringbuffer)(
(_GstAudioBaseSrc *) gobj()),true);
return retval;
}
using RType = Glib::RefPtr<Gst::AudioRingBuffer>;
return RType();
}
the default behavior was causing (test_audio_plugin:11956):
GStreamer-CRITICAL **: 20:55:06.510: gst_object_set_parent: assertion
'GST_IS_OBJECT (object)' failed
Thread #1 [test_audio_plug] 11987 [core: 4] (Suspended : Signal :
SIGTRAP:Trace/breakpoint trap)
0x7ffff7582c41
g_logv() at 0x7ffff7583f2b
g_log() at 0x7ffff758407f
gst_object_set_parent() at gstobject.c:670 0x7ffff7acbeaa
gst_audio_base_src_create_ringbuffer() at 0x7ffff4aff82d
0x7ffff4aff94c
Gst::Element_Class::change_state_vfunc_callback() at 0x7ffff7258281
gst_element_change_state() at gstelement.c:2,952 0x7ffff7af8bce
gst_element_set_state_func() at gstelement.c:2,906 0x7ffff7af9309
Gst::Element_Class::set_state_vfunc_callback() at 0x7ffff72581e1
gst_bin_element_set_state() at gstbin.c:2,602 0x7ffff7ad6942
gst_bin_change_state_func() at gstbin.c:2,944 0x7ffff7ad6942
gst_element_change_state() at gstelement.c:2,952 0x7ffff7af8bce
gst_element_set_state_func() at gstelement.c:2,906 0x7ffff7af9309
gst_bin_element_set_state() at gstbin.c:2,602 0x7ffff7ad6942
gst_bin_change_state_func() at gstbin.c:2,944 0x7ffff7ad6942
gst_element_change_state() at gstelement.c:2,952 0x7ffff7af8bce
gst_element_set_state_func() at gstelement.c:2,906 0x7ffff7af9309
main() at test_audio_plugin.cpp:49 0x55555555cf7a
On Mon, Jun 18, 2018 at 7:49 PM, Ankur Deep Jaiswal <[email protected]>
wrote:
> Hi,
>
> got it,
> we also have to override prepare_audiosink_vfunc
> example .
>
> bool prepare_audiosink_vfunc(Gst::AudioRingBufferSpec& spec) override
> {
> auto srccaps = Gst::Caps::create_from_string("audio/x-raw,
> format=(string)S16LE, rate=(int)16000,channels=(int)
> 1,channel-mask=(bitmask)0x1,layout=(string)interleaved");
> spec.set_caps(srccaps);
> return true;
> }
>
>
>
> On Mon, Jun 18, 2018 at 7:03 PM, Ankur Deep Jaiswal <[email protected]
> > wrote:
>
>> Hi,
>>
>> I'm trying to create a custom audio sink plugin for gstreamer using the
>> Gst::AudioSink as a base class.,
>> i have gone through a previous mail on april and created a sample plugin.
>>
>> but plugin fails by audiobasesink gstaudiobasesink.c:1193:gst_au
>> dio_base_sink_preroll:<customaudiosink0> [00m error: sink not negotiated.
>>
>> i have looked into gstaudiobasesink.c
>> it fails at !gst_audio_ring_buffer_is_acquired.
>>
>> am i missing something here.
>>
>> i am including the sample here
>>
>> --------- .h
>>
>> #include <gstreamermm.h>
>> #include <gstreamermm/audiosink.h>
>> #include <gstreamermm/private/audiosink_p.h>
>> #include <iostream>
>>
>> class CustomAudioSink: public Gst::AudioSink {
>> public:
>> explicit CustomAudioSink(GstAudioSink * gobj);
>> virtual ~CustomAudioSink();
>>
>> static void class_init(Gst::ElementClass<CustomAudioSink> *klass);
>> static bool register_element(Glib::RefPtr<Gst::Plugin> plugin);
>>
>> bool open_vfunc() override;
>> bool close_vfunc() override;
>> int write_vfunc(gpointer data, guint length) override;
>> guint get_delay_vfunc() const override;
>>
>> Glib::RefPtr<Gst::AudioRingBuffer> create_ring_buffer_vfunc()
>> override
>> {
>> const auto base = static_cast<BaseClassType*>(g_
>> type_class_peek_parent(
>> G_OBJECT_GET_CLASS(gobject_)) // Get the parent class of
>> the object class (The original underlying C class).
>> );
>>
>> const auto newbase = (_GstAudioBaseSinkClass *) base;
>>
>> if (newbase && newbase->create_ringbuffer) {
>> Glib::RefPtr<Gst::AudioRingBuffer> retval(
>> Glib::wrap(
>> (*newbase->create_ringbuffer)(
>> (_GstAudioBaseSink *) gobj()), true));
>> return retval;
>> }
>>
>> using RType = Glib::RefPtr<Gst::AudioRingBuffer>;
>> return RType();
>> }
>> };
>>
>> ---------- .cpp
>>
>> #include "CustomAudioSink.h"
>> #include <iostream>
>>
>> CustomAudioSink::CustomAudioSink(GstAudioSink * gobj) :
>> Glib::ObjectBase(typeid(CustomAudioSink)), Gst::AudioSink(gobj) {
>>
>> }
>>
>> CustomAudioSink::~CustomAudioSink() {
>> // TODO Auto-generated destructor stub
>> }
>>
>> void CustomAudioSink::class_init(Gst::ElementClass<CustomAudioSink>
>> *klass) {
>> klass->set_metadata("foo_longname", "foo_classification",
>> "foo_detail_description", "foo_detail_author");
>> //channel-mask=(bitmask)0x1,layout=(string)interleaved
>> klass->add_pad_template(
>> Gst::PadTemplate::create("sink", Gst::PadDirection::PAD_SINK,
>> Gst::PAD_ALWAYS, Gst::Caps::create_any()));
>> }
>>
>> bool CustomAudioSink::register_element(Glib::RefPtr<Gst::Plugin> plugin)
>> {
>> Gst::ElementFactory::register_element(plugin, "customaudiosink", 0,
>> Gst::register_mm_type<CustomAudioSink>("customaudiosink"));
>> return true;
>> }
>>
>> bool CustomAudioSink::open_vfunc() {
>> std::cout << "*** CustomAudioSink: " << __FUNCTION__ << " Called ***"
>> << std::endl;
>> return true;
>> }
>> bool CustomAudioSink::close_vfunc() {
>> std::cout << "*** CustomAudioSink: " << __FUNCTION__ << " Called ***"
>> << std::endl;
>> return true;
>> }
>> int CustomAudioSink::write_vfunc(gpointer data, guint length) {
>> std::cout << "Writing Audio : " << length << std::endl;
>> return length;
>> }
>> guint CustomAudioSink::get_delay_vfunc() const {
>> return 0;
>> }
>>
>> ------ main.cpp
>>
>> #include <gstreamermm.h>
>> #include <giomm.h>
>> #include <iostream>
>> #include "CustomAudioSink.h"
>>
>> using namespace std;
>>
>> bool on_bus_message(const Glib::RefPtr<Gst::Bus> & bus, const
>> Glib::RefPtr<Gst::Message> & message)
>> {
>> std::cout << "Bus Message : " << message->get_message_type() << ":" <<
>> GST_MESSAGE_TYPE_NAME(message.get())
>> << std::endl;
>> return true;
>> }
>>
>> int main() {
>> cout << "!!!Hello World!!!" << endl; // prints !!!Hello World!!!
>> Gst::init();
>> Gio::init();
>> auto mainloop = Glib::MainLoop::create();
>>
>> Gst::Plugin::register_static(GST_VERSION_MAJOR, GST_VERSION_MINOR,
>> "customaudiosink", "customaudiosink",
>> sigc::ptr_fun(&CustomAudioSink::register_element), "0.123",
>> "LGPL", "source?", "package?", "abcd.com");
>>
>> std::string description =
>> "audiotestsrc blocksize=16000 num-buffers=100 !
>> audio/x-raw,format=(string)S16LE,rate=(int)16000,channels=(int)1 !
>> customaudiosink";
>> auto bin = Gst::Parse::create_bin(description, true);
>> auto pipeline = Gst::Pipeline::create();
>> pipeline->get_bus()->add_watch(sigc::ptr_fun(&on_bus_message));
>> pipeline->add(bin);
>>
>> pipeline->set_state(Gst::State::STATE_PLAYING);
>> mainloop->run();
>> pipeline->set_state(Gst::State::STATE_NULL);
>>
>> return 0;
>> }
>>
>> --
>> Regards
>> Ankur Deep Jaiswal
>>
>>
>
>
> --
> Regards
>
> Ankur Deep Jaiswal
> Software Architect
> Techgentsia Software Technologies Private Limited
> Ernakulam
>
--
Regards
Ankur Deep Jaiswal
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