gstreamer/ext/gsm/gstgsmdec.c

346 lines
9.4 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)
/* 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;
}
#define ENCODED_SAMPLES 160
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) [1, MAX], 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) [1, MAX], " "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_simple (element_class, "GSM audio decoder",
"Codec/Decoder/Audio",
"Decodes GSM encoded audio", "Philippe Khalaf <burger@speedy.org>");
}
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;
gboolean ret = FALSE;
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;
if (!gst_structure_get_int (s, "rate", &gsmdec->rate)) {
GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
goto beach;
}
/* MSGSM needs different framing */
gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
gsmdec->duration = gst_util_uint64_scale (ENCODED_SAMPLES,
GST_SECOND, gsmdec->rate);
/* Setting up src caps based on the input sample rate. */
srccaps = gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, gsmdec->rate, "channels", G_TYPE_INT, 1, NULL);
ret = gst_pad_set_caps (gsmdec->srcpad, srccaps);
gst_caps_unref (srccaps);
gst_object_unref (gsmdec);
return ret;
/* ERRORS */
wrong_caps:
GST_ERROR_OBJECT (gsmdec, "invalid caps received");
beach:
gst_object_unref (gsmdec);
return ret;
}
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 (ENCODED_SAMPLES * 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 + gsmdec->duration;
timestamp = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (outbuf) = gsmdec->duration;
GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
if (gsmdec->next_of != -1)
gsmdec->next_of += ENCODED_SAMPLES;
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;
}