mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-30 13:41:48 +00:00
b8a18741d8
Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_init): * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_init): * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_init): * ext/audioresample/gstaudioresample.c: * ext/bz2/gstbz2dec.c: (gst_bz2dec_init): * ext/bz2/gstbz2enc.c: (gst_bz2enc_init): * ext/divx/gstdivxdec.c: (gst_divxdec_init): * ext/divx/gstdivxenc.c: (gst_divxenc_init): * ext/faac/gstfaac.c: (gst_faac_init): * ext/gsm/gstgsmdec.c: (gst_gsmdec_init): * ext/gsm/gstgsmenc.c: (gst_gsmenc_init): * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_init): * ext/lcs/gstcolorspace.c: (gst_colorspace_init): * ext/libfame/gstlibfame.c: (gst_fameenc_init): * ext/snapshot/gstsnapshot.c: (gst_snapshot_init): * ext/spc/gstspc.c: (gst_spc_dec_init): * ext/swfdec/gstswfdec.c: (gst_swfdec_init): * ext/xvid/gstxvidenc.c: (gst_xvidenc_init): * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init): * gst/chart/gstchart.c: (gst_chart_init): * gst/colorspace/gstcolorspace.c: (gst_colorspace_init): * gst/festival/gstfestival.c: (gst_festival_init): * gst/freeze/gstfreeze.c: (gst_freeze_init): * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_request_new_pad): * gst/mpeg1sys/gstmpeg1systemencode.c: (gst_system_encode_init): * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init): * gst/nsf/gstnsf.c: (gst_nsfdec_init): * gst/overlay/gstoverlay.c: (gst_overlay_init): * gst/passthrough/gstpassthrough.c: (passthrough_init): * gst/playondemand/gstplayondemand.c: (play_on_demand_init): * gst/smooth/gstsmooth.c: (gst_smooth_init): * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_init): * gst/speed/gstspeed.c: (speed_init): * gst/vbidec/gstvbidec.c: (gst_vbidec_init): * gst/videodrop/gstvideodrop.c: (gst_videodrop_init): * sys/dxr3/dxr3spusink.c: (dxr3spusink_init): * sys/dxr3/dxr3videosink.c: (dxr3videosink_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_init): Fix leaks.
332 lines
9 KiB
C
332 lines
9 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_STATIC (gsmdec_debug);
|
|
#define GST_CAT_DEFAULT (gsmdec_debug)
|
|
|
|
/* elementfactory information */
|
|
static const 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_setcaps (GstPad * pad, GstCaps * caps);
|
|
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; "
|
|
"audio/ms-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)
|
|
{
|
|
/* create the sink and src pads */
|
|
gsmdec->sinkpad =
|
|
gst_pad_new_from_static_template (&gsmdec_sink_template, "sink");
|
|
gst_pad_set_setcaps_function (gsmdec->sinkpad, gst_gsmdec_sink_setcaps);
|
|
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_static_template (&gsmdec_src_template, "src");
|
|
gst_element_add_pad (GST_ELEMENT (gsmdec), gsmdec->srcpad);
|
|
|
|
gsmdec->state = gsm_create ();
|
|
|
|
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_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstGSMDec *gsmdec;
|
|
GstCaps *srccaps;
|
|
GstStructure *s;
|
|
|
|
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
if (s == NULL)
|
|
goto wrong_caps;
|
|
|
|
/* figure out if we deal with plain or MSGSM */
|
|
if (gst_structure_has_name (s, "audio/x-gsm"))
|
|
gsmdec->use_wav49 = 0;
|
|
else if (gst_structure_has_name (s, "audio/ms-gsm"))
|
|
gsmdec->use_wav49 = 1;
|
|
else
|
|
goto wrong_caps;
|
|
|
|
/* MSGSM needs different framing */
|
|
gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
|
|
|
|
/* we only have one possible source caps, which is the same as our template. */
|
|
srccaps = gst_static_pad_template_get_caps (&gsmdec_src_template);
|
|
|
|
gst_pad_set_caps (gsmdec->srcpad, srccaps);
|
|
|
|
gst_object_unref (gsmdec);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
wrong_caps:
|
|
{
|
|
GST_ERROR_OBJECT (gsmdec, "invalid caps received");
|
|
gst_object_unref (gsmdec);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
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, arate;
|
|
gint64 start, stop, time;
|
|
|
|
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
|
|
&start, &stop, &time);
|
|
|
|
/* now configure the values */
|
|
gst_segment_set_newsegment_full (&gsmdec->segment, update,
|
|
rate, arate, 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;
|
|
gint needed;
|
|
|
|
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);
|
|
gsmdec->next_ts = GST_CLOCK_TIME_NONE;
|
|
/* FIXME, do some good offset */
|
|
gsmdec->next_of = 0;
|
|
}
|
|
gst_adapter_push (gsmdec->adapter, buf);
|
|
|
|
needed = 33;
|
|
/* do we have enough bytes to read a frame */
|
|
while (gst_adapter_available (gsmdec->adapter) >= needed) {
|
|
GstBuffer *outbuf;
|
|
|
|
/* always the same amount of output samples */
|
|
outbuf = gst_buffer_new_and_alloc (160 * sizeof (gsm_signal));
|
|
|
|
/* If we are not given any timestamp, interpolate from last seen
|
|
* timestamp (if any). */
|
|
if (timestamp == GST_CLOCK_TIME_NONE)
|
|
timestamp = gsmdec->next_ts;
|
|
|
|
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
|
|
|
|
/* interpolate in the next run */
|
|
if (timestamp != GST_CLOCK_TIME_NONE)
|
|
gsmdec->next_ts = timestamp + (20 * GST_MSECOND);
|
|
timestamp = GST_CLOCK_TIME_NONE;
|
|
|
|
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
|
|
GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
|
|
if (gsmdec->next_of != -1)
|
|
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, needed);
|
|
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, needed);
|
|
|
|
/* WAV49 requires alternating 33 and 32 bytes of input */
|
|
if (gsmdec->use_wav49)
|
|
needed = (needed == 33 ? 32 : 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;
|
|
}
|