gstreamer/sys/osxaudio/gstosxaudiosrc.c

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/*
* GStreamer
* Copyright 2005,2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
*
* 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-plugin
*
* <refsect2>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch -v -m osxaudiosrc ! fakesink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/gst.h>
#include <CoreAudio/CoreAudio.h>
#include "gstosxaudiosrc.h"
#include "gstosxaudioelement.h"
GST_DEBUG_CATEGORY_STATIC (osx_audiosrc_debug);
#define GST_CAT_DEFAULT osx_audiosrc_debug
static GstElementDetails gst_osx_audio_src_details =
GST_ELEMENT_DETAILS ("Audio Source (OSX)",
"Source/Audio",
"Input from a sound card in OS X",
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_DEVICE
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, "
"signed = (boolean) { TRUE }, "
"width = (int) 32, "
"depth = (int) 32, " "rate = (int) 44100, " "channels = (int) 2")
);
static void gst_osx_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_osx_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstRingBuffer *gst_osx_audio_src_create_ringbuffer (GstBaseAudioSrc *
src);
/*static GstCaps* gst_osx_audio_sink_getcaps (GstBaseSink * bsrc);*/
static void gst_osx_audio_src_osxelement_init (gpointer g_iface,
gpointer iface_data);
OSStatus gst_osx_audio_src_io_proc (AudioDeviceID inDevice,
const AudioTimeStamp * inNow, const AudioBufferList * inInputData,
const AudioTimeStamp * inInputTime, AudioBufferList * outOutputData,
const AudioTimeStamp * inOutputTime, void *inClientData);
static void
gst_osx_audio_src_osxelement_do_init (GType type)
{
static const GInterfaceInfo osxelement_info = {
gst_osx_audio_src_osxelement_init,
NULL,
NULL
};
GST_DEBUG_CATEGORY_INIT (osx_audiosrc_debug, "osxaudiosrc", 0,
"OSX Audio Src");
GST_DEBUG ("Adding static interface");
g_type_add_interface_static (type, GST_OSX_AUDIO_ELEMENT_TYPE,
&osxelement_info);
}
GST_BOILERPLATE_FULL (GstOsxAudioSrc, gst_osx_audio_src, GstBaseAudioSrc,
GST_TYPE_BASE_AUDIO_SRC, gst_osx_audio_src_osxelement_do_init);
static void
gst_osx_audio_src_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details (element_class, &gst_osx_audio_src_details);
}
/* initialize the plugin's class */
static void
gst_osx_audio_src_class_init (GstOsxAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_class_init): * ext/esd/esdsink.c: (gst_esdsink_class_init): * ext/flac/gstflactag.c: (gst_flac_tag_class_init): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init): * ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init): * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init): * ext/libmng/gstmngdec.c: (gst_mngdec_class_init): * ext/libmng/gstmngenc.c: (gst_mngenc_class_init): * ext/libpng/gstpngdec.c: (gst_pngdec_class_init): * ext/libpng/gstpngenc.c: (gst_pngenc_class_init): * ext/mikmod/gstmikmod.c: (gst_mikmod_class_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init): * ext/speex/gstspeexenc.c: (gst_speexenc_class_init): * gst/alpha/gstalpha.c: (gst_alpha_class_init): * gst/avi/gstavimux.c: (gst_avimux_class_init): * gst/debug/efence.c: (gst_efence_class_init): * gst/debug/negotiation.c: (gst_negotiation_class_init): * gst/flx/gstflxdec.c: (gst_flxdec_class_init): * gst/goom/gstgoom.c: (gst_goom_class_init): * gst/id3demux/gstid3demux.c: (gst_id3demux_class_init): * gst/interleave/deinterleave.c: (deinterleave_class_init): * gst/interleave/interleave.c: (interleave_class_init): * gst/law/alaw-decode.c: (gst_alawdec_class_init): * gst/law/alaw-encode.c: (gst_alawenc_class_init): * gst/law/mulaw-encode.c: (gst_mulawenc_class_init): * gst/median/gstmedian.c: (gst_median_class_init): * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init): * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init): * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init): * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init): * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init): * gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init): * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init): * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init): * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init): * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init): * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init): * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init): * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init): * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init): * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init): * gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init): * gst/smpte/gstsmpte.c: (gst_smpte_class_init): * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init): * gst/udp/gstudpsink.c: (gst_udpsink_class_init): * gst/videomixer/videomixer.c: (gst_videomixer_class_init): * gst/wavenc/gstwavenc.c: (gst_wavenc_class_init): * sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init): * sys/oss/gstosssink.c: (gst_oss_sink_class_init): * sys/osxaudio/gstosxaudioelement.c: (gst_osxaudioelement_class_init): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init): * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:21:45 +00:00
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_osx_audio_src_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_osx_audio_src_get_property);
g_object_class_install_property (gobject_class, ARG_DEVICE,
g_param_spec_int ("device", "Device ID", "Device ID of output device",
0, G_MAXINT, 0, G_PARAM_READWRITE));
gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_osx_audio_src_create_ringbuffer);
}
/* initialize the new element
* instantiate pads and add them to element
* set functions
* initialize structure
*/
static void
gst_osx_audio_src_init (GstOsxAudioSrc * src, GstOsxAudioSrcClass * gclass)
{
/* GstElementClass *klass = GST_ELEMENT_GET_CLASS (sink); */
gst_base_src_set_live (GST_BASE_SRC (src), TRUE);
src->ringbuffer = NULL;
GST_DEBUG ("Initialising object");
gst_osx_audio_src_create_ringbuffer (GST_BASE_AUDIO_SRC (src));
}
static void
gst_osx_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (object);
switch (prop_id) {
case ARG_DEVICE:
if (src->ringbuffer)
src->ringbuffer->device_id = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_osx_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (object);
int val = 0;
switch (prop_id) {
case ARG_DEVICE:
if (src->ringbuffer)
val = src->ringbuffer->device_id;
g_value_set_int (value, val);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstElement vmethod implementations */
/* GstBaseAudioSink vmethod implementations */
static GstRingBuffer *
gst_osx_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
{
GstOsxAudioSrc *osxsrc;
osxsrc = GST_OSX_AUDIO_SRC (src);
if (!osxsrc->ringbuffer) {
GST_DEBUG ("Creating ringbuffer");
osxsrc->ringbuffer = g_object_new (GST_TYPE_OSX_RING_BUFFER, NULL);
GST_DEBUG ("osx src 0x%p element 0x%p ioproc 0x%p", osxsrc,
GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc),
(void *) gst_osx_audio_src_io_proc);
osxsrc->ringbuffer->element = GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc);
}
return GST_RING_BUFFER (osxsrc->ringbuffer);
}
OSStatus
gst_osx_audio_src_io_proc (AudioDeviceID inDevice, const AudioTimeStamp * inNow,
const AudioBufferList * inInputData, const AudioTimeStamp * inInputTime,
AudioBufferList * outOutputData, const AudioTimeStamp * inOutputTime,
void *inClientData)
{
GstOsxRingBuffer *buf = GST_OSX_RING_BUFFER (inClientData);
guint8 *writeptr;
gint writeseg;
gint len;
close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
2006-03-13 15:49:08 +00:00
if (gst_ring_buffer_prepare_read (GST_RING_BUFFER (buf), &writeseg, &writeptr,
&len)) {
memcpy (writeptr, (char *) inInputData->mBuffers[0].mData, len);
/* clear written samples */
/*gst_ring_buffer_clear (GST_RING_BUFFER(buf), writeseg); */
/* we wrote one segment */
gst_ring_buffer_advance (GST_RING_BUFFER (buf), 1);
}
return 0;
}
static void
gst_osx_audio_src_osxelement_init (gpointer g_iface, gpointer iface_data)
{
GstOsxAudioElementInterface *iface = (GstOsxAudioElementInterface *) g_iface;
iface->io_proc = gst_osx_audio_src_io_proc;
}