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/*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, 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 gboolean gst_gsmdec_start (GstAudioDecoder * dec);
static gboolean gst_gsmdec_stop (GstAudioDecoder * dec);
static gboolean gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
static GstFlowReturn gst_gsmdec_parse (GstAudioDecoder * dec,
GstAdapter * adapter, gint * offset, gint * length);
static GstFlowReturn gst_gsmdec_handle_frame (GstAudioDecoder * dec,
GstBuffer * in_buf);
/*static guint gst_gsmdec_signals[LAST_SIGNAL] = { 0 }; */
#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, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) [1, MAX], channels = (int) 1")
);
G_DEFINE_TYPE (GstGSMDec, gst_gsmdec, GST_TYPE_AUDIO_DECODER);
static void
gst_gsmdec_class_init (GstGSMDecClass * klass)
{
GstElementClass *element_class;
GstAudioDecoderClass *base_class;
element_class = (GstElementClass *) klass;
base_class = (GstAudioDecoderClass *) klass;
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_static_metadata (element_class, "GSM audio decoder",
"Codec/Decoder/Audio",
"Decodes GSM encoded audio", "Philippe Khalaf <burger@speedy.org>");
base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmdec_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (gst_gsmdec_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmdec_handle_frame);
GST_DEBUG_CATEGORY_INIT (gsmdec_debug, "gsmdec", 0, "GSM Decoder");
}
static void
gst_gsmdec_init (GstGSMDec * gsmdec)
{
}
static gboolean
gst_gsmdec_start (GstAudioDecoder * dec)
{
GstGSMDec *gsmdec = GST_GSMDEC (dec);
GST_DEBUG_OBJECT (dec, "start");
gsmdec->state = gsm_create ();
return TRUE;
}
static gboolean
gst_gsmdec_stop (GstAudioDecoder * dec)
{
GstGSMDec *gsmdec = GST_GSMDEC (dec);
GST_DEBUG_OBJECT (dec, "stop");
gsm_destroy (gsmdec->state);
return TRUE;
}
static gboolean
gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
GstGSMDec *gsmdec;
GstStructure *s;
gboolean ret = FALSE;
gint rate;
GstAudioInfo info;
gsmdec = GST_GSMDEC (dec);
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;
gsmdec->needed = 33;
if (!gst_structure_get_int (s, "rate", &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);
/* Setting up src caps based on the input sample rate. */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, rate, 1, NULL);
ret = gst_audio_decoder_set_output_format (dec, &info);
return ret;
/* ERRORS */
wrong_caps:
GST_ERROR_OBJECT (gsmdec, "invalid caps received");
beach:
return ret;
}
static GstFlowReturn
gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length)
{
GstGSMDec *gsmdec = GST_GSMDEC (dec);
guint size;
size = gst_adapter_available (adapter);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
if (size < gsmdec->needed)
return GST_FLOW_EOS;
*offset = 0;
*length = gsmdec->needed;
/* WAV49 requires alternating 33 and 32 bytes of input */
if (gsmdec->use_wav49) {
gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstGSMDec *gsmdec;
gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
GstMapInfo map, omap;
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
gsmdec = GST_GSMDEC (dec);
/* always the same amount of output samples */
outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
/* now encode frame into the output buffer */
gst_buffer_map (buffer, &map, GST_MAP_READ);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
data = (gsm_byte *) map.data;
if (gsm_decode (gsmdec->state, data, (gsm_signal *) omap.data) < 0) {
/* invalid frame */
GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
("tried to decode an invalid frame"), ret);
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
outbuf = NULL;
} else {
gst_buffer_unmap (outbuf, &omap);
}
gst_buffer_unmap (buffer, &map);
gst_audio_decoder_finish_frame (dec, outbuf, 1);
return ret;
}
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