gstreamer/subprojects/gst-plugins-bad/ext/gsm/gstgsmenc.c

190 lines
5.2 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstgsmenc.h"
GST_DEBUG_CATEGORY_STATIC (gsmenc_debug);
#define GST_CAT_DEFAULT (gsmenc_debug)
/* GSMEnc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
/* FILL ME */
ARG_0
};
static gboolean gst_gsmenc_start (GstAudioEncoder * enc);
static gboolean gst_gsmenc_stop (GstAudioEncoder * enc);
static gboolean gst_gsmenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_gsmenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
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, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) 8000, channels = (int) 1")
);
G_DEFINE_TYPE (GstGSMEnc, gst_gsmenc, GST_TYPE_AUDIO_ENCODER);
GST_ELEMENT_REGISTER_DEFINE (gsmenc, "gsmenc", GST_RANK_PRIMARY,
GST_TYPE_GSMENC);
static void
gst_gsmenc_class_init (GstGSMEncClass * klass)
{
GstElementClass *element_class;
GstAudioEncoderClass *base_class;
element_class = (GstElementClass *) klass;
base_class = (GstAudioEncoderClass *) klass;
gst_element_class_add_static_pad_template (element_class,
&gsmenc_sink_template);
gst_element_class_add_static_pad_template (element_class,
&gsmenc_src_template);
gst_element_class_set_static_metadata (element_class, "GSM audio encoder",
"Codec/Encoder/Audio", "Encodes GSM audio",
"Philippe Khalaf <burger@speedy.org>");
base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmenc_handle_frame);
GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder");
}
static void
gst_gsmenc_init (GstGSMEnc * gsmenc)
{
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (gsmenc));
}
static gboolean
gst_gsmenc_start (GstAudioEncoder * enc)
{
GstGSMEnc *gsmenc = GST_GSMENC (enc);
gint use_wav49;
GST_DEBUG_OBJECT (enc, "start");
gsmenc->state = gsm_create ();
/* turn off WAV49 handling */
use_wav49 = 0;
gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49);
return TRUE;
}
static gboolean
gst_gsmenc_stop (GstAudioEncoder * enc)
{
GstGSMEnc *gsmenc = GST_GSMENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
gsm_destroy (gsmenc->state);
return TRUE;
}
static gboolean
gst_gsmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstCaps *srccaps;
srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (benc), srccaps);
gst_caps_unref (srccaps);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, 160);
gst_audio_encoder_set_frame_samples_max (benc, 160);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE;
}
static GstFlowReturn
gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
{
GstGSMEnc *gsmenc;
gsm_signal *data;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
GstMapInfo map, omap;
gsmenc = GST_GSMENC (benc);
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (gsmenc, "no data");
goto done;
}
gst_buffer_map (buffer, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < 320)) {
GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d", (gint) map.size);
gst_buffer_unmap (buffer, &map);
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto done;
}
outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
/* encode 160 16-bit samples into 33 bytes */
data = (gsm_signal *) map.data;
gsm_encode (gsmenc->state, data, (gsm_byte *) omap.data);
GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", (gint) omap.size);
gst_buffer_unmap (buffer, &map);
gst_buffer_unmap (outbuf, &omap);
ret = gst_audio_encoder_finish_frame (benc, outbuf, 160);
done:
return ret;
}