gstreamer/ext/gsm/gstgsmenc.c
j^ dacf8eaa18 Unify the long descriptions in the plugin details (#337263).
Original commit message from CVS:
Patch by: j^  <j at bootlab dot org>
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/arts/gst_arts.c:
* ext/artsd/gstartsdsink.c:
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/cdaudio/gstcdaudio.c:
* ext/directfb/dfbvideosink.c:
* ext/divx/gstdivxdec.c:
* ext/divx/gstdivxenc.c:
* ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
* ext/faac/gstfaac.c: (gst_faac_base_init):
* ext/faad/gstfaad.c:
* ext/gsm/gstgsmdec.c:
* ext/gsm/gstgsmenc.c:
* ext/hermes/gsthermescolorspace.c:
* ext/ivorbis/vorbisfile.c:
* ext/lcs/gstcolorspace.c:
* ext/libfame/gstlibfame.c:
* ext/libmms/gstmms.c: (gst_mms_base_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
* ext/nas/nassink.c: (gst_nassink_base_init):
* ext/neon/gstneonhttpsrc.c:
* ext/polyp/polypsink.c: (gst_polypsink_base_init):
* ext/sdl/sdlaudiosink.c:
* ext/sdl/sdlvideosink.c:
* ext/shout/gstshout.c:
* ext/snapshot/gstsnapshot.c:
* ext/sndfile/gstsf.c:
* ext/tarkin/gsttarkindec.c:
* ext/tarkin/gsttarkinenc.c:
* ext/theora/theoradec.c:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
* ext/xvid/gstxviddec.c:
* ext/xvid/gstxvidenc.c:
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init):
* gst/chart/gstchart.c:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init):
* gst/festival/gstfestival.c:
* gst/filter/gstiir.c:
* gst/filter/gstlpwsinc.c:
* gst/freeze/gstfreeze.c:
* gst/games/gstpuzzle.c: (gst_puzzle_base_init):
* gst/mixmatrix/mixmatrix.c:
* gst/mpeg1sys/gstmpeg1systemencode.c:
* gst/mpeg1videoparse/gstmp1videoparse.c:
* gst/mpeg2sub/gstmpeg2subt.c:
* gst/mpegaudioparse/gstmpegaudioparse.c:
* gst/multifilesink/gstmultifilesink.c:
* gst/overlay/gstoverlay.c:
* gst/passthrough/gstpassthrough.c:
* gst/playondemand/gstplayondemand.c:
* gst/qtdemux/qtdemux.c:
* gst/rtjpeg/gstrtjpegdec.c:
* gst/rtjpeg/gstrtjpegenc.c:
* gst/smooth/gstsmooth.c:
* gst/tta/gstttadec.c: (gst_tta_dec_base_init):
* gst/tta/gstttaparse.c: (gst_tta_parse_base_init):
* gst/videocrop/gstvideocrop.c:
* gst/videodrop/gstvideodrop.c:
* gst/virtualdub/gstxsharpen.c:
* gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init):
* gst/y4m/gsty4mencode.c:
Unify the long descriptions in the plugin details (#337263).
2006-04-06 11:35:26 +00:00

210 lines
5.5 KiB
C

/*
* Farsight
* GStreamer GSM encoder
* Copyright (C) 2005 Philippe Khalaf <burger@speedy.org>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstgsmenc.h"
GST_DEBUG_CATEGORY (gsmenc_debug);
#define GST_CAT_DEFAULT (gsmenc_debug)
/* elementfactory information */
GstElementDetails gst_gsmenc_details = GST_ELEMENT_DETAILS ("GSM audio encoder",
"Codec/Encoder/Audio",
"Encodes GSM audio",
"Philippe Khalaf <burger@speedy.org>");
/* GSMEnc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
/* FILL ME */
ARG_0
};
static void gst_gsmenc_base_init (gpointer g_class);
static void gst_gsmenc_class_init (GstGSMEnc * klass);
static void gst_gsmenc_init (GstGSMEnc * gsmenc);
static void gst_gsmenc_finalize (GObject * object);
static GstFlowReturn gst_gsmenc_chain (GstPad * pad, GstBuffer * buf);
static GstElementClass *parent_class = NULL;
GType
gst_gsmenc_get_type (void)
{
static GType gsmenc_type = 0;
if (!gsmenc_type) {
static const GTypeInfo gsmenc_info = {
sizeof (GstGSMEncClass),
gst_gsmenc_base_init,
NULL,
(GClassInitFunc) gst_gsmenc_class_init,
NULL,
NULL,
sizeof (GstGSMEnc),
0,
(GInstanceInitFunc) gst_gsmenc_init,
};
gsmenc_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstGSMEnc", &gsmenc_info, 0);
}
return gsmenc_type;
}
static GstStaticPadTemplate gsmenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
);
static GstStaticPadTemplate gsmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, "
"width = (int) 16, "
"depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
);
static void
gst_gsmenc_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 (&gsmenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gsmenc_src_template));
gst_element_class_set_details (element_class, &gst_gsmenc_details);
}
static void
gst_gsmenc_class_init (GstGSMEnc * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_gsmenc_finalize;
GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder");
}
static void
gst_gsmenc_init (GstGSMEnc * gsmenc)
{
gint use_wav49;
/* create the sink and src pads */
gsmenc->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gsmenc_sink_template), "sink");
gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->sinkpad);
gst_pad_set_chain_function (gsmenc->sinkpad, gst_gsmenc_chain);
gsmenc->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gsmenc_src_template), "src");
gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->srcpad);
gsmenc->state = gsm_create ();
/* turn on WAV49 handling */
use_wav49 = 0;
gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49);
gsmenc->adapter = gst_adapter_new ();
gsmenc->next_ts = 0;
}
static void
gst_gsmenc_finalize (GObject * object)
{
GstGSMEnc *gsmenc;
gsmenc = GST_GSMENC (object);
g_object_unref (gsmenc->adapter);
gsm_destroy (gsmenc->state);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstFlowReturn
gst_gsmenc_chain (GstPad * pad, GstBuffer * buf)
{
GstGSMEnc *gsmenc;
gsm_signal *data;
GstFlowReturn ret = GST_FLOW_OK;
gsmenc = GST_GSMENC (gst_pad_get_parent (pad));
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (gsmenc->adapter);
}
gst_adapter_push (gsmenc->adapter, buf);
while (gst_adapter_available (gsmenc->adapter) >= 320) {
GstBuffer *outbuf;
outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
gsmenc->next_ts += 20 * GST_MSECOND;
/* encode 160 16-bit samples into 33 bytes */
data = (gsm_signal *) gst_adapter_peek (gsmenc->adapter, 320);
gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
gst_adapter_flush (gsmenc->adapter, 320);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmenc->srcpad));
GST_DEBUG_OBJECT (gsmenc, "Pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_pad_push (gsmenc->srcpad, outbuf);
}
gst_object_unref (gsmenc);
return ret;
}