gstreamer/ext/gsm/gstgsmdec.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

288 lines
7.8 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 "gstgsmdec.h"
GST_DEBUG_CATEGORY (gsmdec_debug);
#define GST_CAT_DEFAULT (gsmdec_debug)
/* elementfactory information */
GstElementDetails gst_gsmdec_details = GST_ELEMENT_DETAILS ("GSM audio decoder",
"Codec/Decoder/Audio",
"Decodes GSM encoded audio",
"Philippe Khalaf <burger@speedy.org>");
/* GSMDec signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
/* FILL ME */
ARG_0
};
static void gst_gsmdec_base_init (gpointer g_class);
static void gst_gsmdec_class_init (GstGSMDec * klass);
static void gst_gsmdec_init (GstGSMDec * gsmdec);
static void gst_gsmdec_finalize (GObject * object);
static gboolean gst_gsmdec_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_gsmdec_chain (GstPad * pad, GstBuffer * buf);
static GstElementClass *parent_class = NULL;
/*static guint gst_gsmdec_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_gsmdec_get_type (void)
{
static GType gsmdec_type = 0;
if (!gsmdec_type) {
static const GTypeInfo gsmdec_info = {
sizeof (GstGSMDecClass),
gst_gsmdec_base_init,
NULL,
(GClassInitFunc) gst_gsmdec_class_init,
NULL,
NULL,
sizeof (GstGSMDec),
0,
(GInstanceInitFunc) gst_gsmdec_init,
};
gsmdec_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstGSMDec", &gsmdec_info, 0);
}
return gsmdec_type;
}
static GstStaticPadTemplate gsmdec_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
);
static GstStaticPadTemplate gsmdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
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_gsmdec_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 (&gsmdec_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gsmdec_src_template));
gst_element_class_set_details (element_class, &gst_gsmdec_details);
}
static void
gst_gsmdec_class_init (GstGSMDec * 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_gsmdec_finalize;
GST_DEBUG_CATEGORY_INIT (gsmdec_debug, "gsmdec", 0, "GSM Decoder");
}
static void
gst_gsmdec_init (GstGSMDec * gsmdec)
{
gint use_wav49;
/* create the sink and src pads */
gsmdec->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gsmdec_sink_template), "sink");
gst_pad_set_event_function (gsmdec->sinkpad, gst_gsmdec_sink_event);
gst_pad_set_chain_function (gsmdec->sinkpad, gst_gsmdec_chain);
gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->sinkpad);
gsmdec->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gsmdec_src_template), "src");
gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->srcpad);
gsmdec->state = gsm_create ();
/* turn on WAV49 handling */
use_wav49 = 0;
gsm_option (gsmdec->state, GSM_OPT_WAV49, &use_wav49);
gsmdec->adapter = gst_adapter_new ();
gsmdec->next_of = 0;
gsmdec->next_ts = 0;
}
static void
gst_gsmdec_finalize (GObject * object)
{
GstGSMDec *gsmdec;
gsmdec = GST_GSMDEC (object);
g_object_unref (gsmdec->adapter);
gsm_destroy (gsmdec->state);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_gsmdec_sink_event (GstPad * pad, GstEvent * event)
{
gboolean res;
GstGSMDec *gsmdec;
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
res = gst_pad_push_event (gsmdec->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&gsmdec->segment, GST_FORMAT_UNDEFINED);
res = gst_pad_push_event (gsmdec->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
GstFormat format;
gdouble rate;
gint64 start, stop, time;
gst_event_parse_new_segment (event, &update, &rate, &format, &start,
&stop, &time);
/* now configure the values */
gst_segment_set_newsegment (&gsmdec->segment, update,
rate, format, start, stop, time);
/* and forward */
res = gst_pad_push_event (gsmdec->srcpad, event);
break;
}
case GST_EVENT_EOS:
default:
res = gst_pad_push_event (gsmdec->srcpad, event);
break;
}
gst_object_unref (gsmdec);
return res;
}
static GstFlowReturn
gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
{
GstGSMDec *gsmdec;
gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
timestamp = GST_BUFFER_TIMESTAMP (buf);
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (gsmdec->adapter);
}
gst_adapter_push (gsmdec->adapter, buf);
/* do we have enough bytes to read a header */
while (gst_adapter_available (gsmdec->adapter) >= 33) {
GstBuffer *outbuf;
outbuf = gst_buffer_new_and_alloc (160 * sizeof (gsm_signal));
/* TODO take new segment in consideration, if not given restart
* timestamps at 0 */
if (timestamp == GST_CLOCK_TIME_NONE) {
/* If we are not given any timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = gsmdec->next_ts;
if (gsmdec->next_ts != GST_CLOCK_TIME_NONE)
gsmdec->next_ts += 20 * GST_MSECOND;
}
else {
/* upstream gave a timestamp, use it. */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
gsmdec->next_ts = timestamp + 20 * GST_MSECOND;
/* and make sure we interpollate in the next run */
timestamp = GST_CLOCK_TIME_NONE;
}
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
gsmdec->next_of += 160;
GST_BUFFER_OFFSET_END (outbuf) = gsmdec->next_of;
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmdec->srcpad));
/* now encode frame into the output buffer */
data = (gsm_byte *) gst_adapter_peek (gsmdec->adapter, 33);
if (gsm_decode (gsmdec->state, data,
(gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
/* invalid frame */
GST_WARNING_OBJECT (gsmdec, "tried to decode an invalid frame, skipping");
}
gst_adapter_flush (gsmdec->adapter, 33);
GST_DEBUG_OBJECT (gsmdec, "Pushing buffer of size %d ts %" GST_TIME_FORMAT,
GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
/* push */
ret = gst_pad_push (gsmdec->srcpad, outbuf);
}
gst_object_unref (gsmdec);
return ret;
}