gstreamer/sys/osxaudio/gstosxaudiosrc.c

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/*
* GStreamer
* Copyright (C) 2005,2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* Copyright (C) 2008 Pioneers of the Inevitable <songbird@songbirdnest.com>
*
* 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-osxaudiosrc
*
* This element captures raw audio samples using the CoreAudio api.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 osxaudiosrc ! wavenc ! filesink location=audio.wav
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/gst.h>
#include <CoreAudio/CoreAudio.h>
#include <CoreAudio/AudioHardware.h>
#include "gstosxaudiosrc.h"
#include "gstosxaudioelement.h"
GST_DEBUG_CATEGORY_STATIC (osx_audiosrc_debug);
#define GST_CAT_DEFAULT osx_audiosrc_debug
/* 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) [1, MAX], " "channels = (int) [1, MAX]")
);
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 GstCaps *gst_osx_audio_src_get_caps (GstBaseSrc * src);
static GstRingBuffer *gst_osx_audio_src_create_ringbuffer (GstBaseAudioSrc *
src);
static void gst_osx_audio_src_osxelement_init (gpointer g_iface,
gpointer iface_data);
static OSStatus gst_osx_audio_src_io_proc (GstOsxRingBuffer * buf,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp, UInt32 inBusNumber,
UInt32 inNumberFrames, AudioBufferList * bufferList);
static void gst_osx_audio_src_select_device (GstOsxAudioSrc * osxsrc);
static void
gst_osx_audio_src_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_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_static_metadata (element_class, "Audio Source (OSX)",
"Source/Audio",
"Input from a sound card in OS X",
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
}
static void
gst_osx_audio_src_class_init (GstOsxAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) 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_osx_audio_src_set_property;
gobject_class->get_property = gst_osx_audio_src_get_property;
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_osx_audio_src_get_caps);
g_object_class_install_property (gobject_class, ARG_DEVICE,
g_param_spec_int ("device", "Device ID", "Device ID of input device",
0, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_osx_audio_src_create_ringbuffer);
}
static void
gst_osx_audio_src_init (GstOsxAudioSrc * src, GstOsxAudioSrcClass * gclass)
{
gst_base_src_set_live (GST_BASE_SRC (src), TRUE);
src->device_id = kAudioDeviceUnknown;
src->deviceChannels = -1;
}
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:
src->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);
switch (prop_id) {
case ARG_DEVICE:
g_value_set_int (value, src->device_id);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_osx_audio_src_get_caps (GstBaseSrc * src)
{
GstElementClass *gstelement_class;
GstOsxAudioSrc *osxsrc;
GstPadTemplate *pad_template;
GstCaps *caps;
gint min, max;
gstelement_class = GST_ELEMENT_GET_CLASS (src);
osxsrc = GST_OSX_AUDIO_SRC (src);
if (osxsrc->deviceChannels == -1) {
/* -1 means we don't know the number of channels yet. for now, return
* template caps.
*/
return NULL;
}
max = osxsrc->deviceChannels;
if (max < 1)
max = 1; /* 0 channels means 1 channel? */
min = MIN (1, max);
pad_template = gst_element_class_get_pad_template (gstelement_class, "src");
g_return_val_if_fail (pad_template != NULL, NULL);
caps = gst_caps_copy (gst_pad_template_get_caps (pad_template));
if (min == max) {
gst_caps_set_simple (caps, "channels", G_TYPE_INT, max, NULL);
} else {
gst_caps_set_simple (caps, "channels", GST_TYPE_INT_RANGE, min, max, NULL);
}
return caps;
}
static GstRingBuffer *
gst_osx_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
{
GstOsxAudioSrc *osxsrc;
GstOsxRingBuffer *ringbuffer;
osxsrc = GST_OSX_AUDIO_SRC (src);
gst_osx_audio_src_select_device (osxsrc);
GST_DEBUG ("Creating ringbuffer");
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);
ringbuffer->element = GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc);
ringbuffer->is_src = TRUE;
ringbuffer->device_id = osxsrc->device_id;
return GST_RING_BUFFER (ringbuffer);
}
static OSStatus
gst_osx_audio_src_io_proc (GstOsxRingBuffer * buf,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList)
{
OSStatus status;
guint8 *writeptr;
gint writeseg;
gint len;
gint remaining;
gint offset = 0;
status = AudioUnitRender (buf->audiounit, ioActionFlags, inTimeStamp,
inBusNumber, inNumberFrames, buf->recBufferList);
if (status) {
GST_WARNING_OBJECT (buf, "AudioUnitRender returned %d", (int) status);
return status;
}
remaining = buf->recBufferList->mBuffers[0].mDataByteSize;
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
while (remaining) {
if (!gst_ring_buffer_prepare_read (GST_RING_BUFFER (buf),
&writeseg, &writeptr, &len))
return 0;
len -= buf->segoffset;
if (len > remaining)
len = remaining;
memcpy (writeptr + buf->segoffset,
(char *) buf->recBufferList->mBuffers[0].mData + offset, len);
buf->segoffset += len;
offset += len;
remaining -= len;
if ((gint) buf->segoffset == GST_RING_BUFFER (buf)->spec.segsize) {
/* we wrote one segment */
gst_ring_buffer_advance (GST_RING_BUFFER (buf), 1);
buf->segoffset = 0;
}
}
return 0;
}
static void
gst_osx_audio_src_osxelement_init (gpointer g_iface, gpointer iface_data)
{
GstOsxAudioElementInterface *iface = (GstOsxAudioElementInterface *) g_iface;
iface->io_proc = (AURenderCallback) gst_osx_audio_src_io_proc;
}
static void
gst_osx_audio_src_select_device (GstOsxAudioSrc * osxsrc)
{
OSStatus status;
UInt32 propertySize;
if (osxsrc->device_id == kAudioDeviceUnknown) {
/* If no specific device has been selected by the user, then pick the
* default device */
GST_DEBUG_OBJECT (osxsrc, "Selecting device for OSXAudioSrc");
propertySize = sizeof (osxsrc->device_id);
status = AudioHardwareGetProperty (kAudioHardwarePropertyDefaultInputDevice,
&propertySize, &osxsrc->device_id);
if (status) {
GST_WARNING_OBJECT (osxsrc,
"AudioHardwareGetProperty returned %d", (int) status);
} else {
GST_DEBUG_OBJECT (osxsrc, "AudioHardwareGetProperty returned 0");
}
if (osxsrc->device_id == kAudioDeviceUnknown) {
GST_WARNING_OBJECT (osxsrc,
"AudioHardwareGetProperty: device_id is kAudioDeviceUnknown");
}
GST_DEBUG_OBJECT (osxsrc, "AudioHardwareGetProperty: device_id is %lu",
(long) osxsrc->device_id);
}
}