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/*
* Siren Encoder Gst Element
*
* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
*
* 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.
*
*/
/**
* SECTION:element-sirenenc
*
* This encodes audio buffers into the Siren 16 codec (a 16khz extension of
* G.722.1) that is meant to be compatible with the Microsoft Windows Live
* Messenger(tm) implementation.
*
* Ref: http://www.polycom.com/company/about_us/technology/siren_g7221/index.html
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstsirenenc.h"
#include <string.h>
GST_DEBUG_CATEGORY (sirenenc_debug);
#define GST_CAT_DEFAULT (sirenenc_debug)
#define FRAME_DURATION (20 * GST_MSECOND)
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320"));
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, format = (string) \"S16LE\", "
"rate = (int) 16000, " "channels = (int) 1"));
static gboolean gst_siren_enc_start (GstAudioEncoder * enc);
static gboolean gst_siren_enc_stop (GstAudioEncoder * enc);
static gboolean gst_siren_enc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_siren_enc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
G_DEFINE_TYPE (GstSirenEnc, gst_siren_enc, GST_TYPE_AUDIO_ENCODER);
static void
gst_siren_enc_class_init (GstSirenEncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
GST_DEBUG ("Initializing Class");
GST_DEBUG_CATEGORY_INIT (sirenenc_debug, "sirenenc", 0, "sirenenc");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&srctemplate));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sinktemplate));
gst_element_class_set_details_simple (element_class, "Siren Encoder element",
"Codec/Encoder/Audio ",
"Encode 16bit PCM streams into the Siren7 codec",
"Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
base_class->start = GST_DEBUG_FUNCPTR (gst_siren_enc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_enc_handle_frame);
GST_DEBUG ("Class Init done");
}
static void
gst_siren_enc_init (GstSirenEnc * enc)
{
}
static gboolean
gst_siren_enc_start (GstAudioEncoder * enc)
{
GstSirenEnc *senc = GST_SIREN_ENC (enc);
GST_DEBUG_OBJECT (enc, "start");
senc->encoder = Siren7_NewEncoder (16000);
return TRUE;
}
static gboolean
gst_siren_enc_stop (GstAudioEncoder * enc)
{
GstSirenEnc *senc = GST_SIREN_ENC (enc);
GST_DEBUG_OBJECT (senc, "stop");
Siren7_CloseEncoder (senc->encoder);
return TRUE;
}
static gboolean
gst_siren_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
gboolean res;
GstCaps *outcaps;
outcaps = gst_static_pad_template_get_caps (&srctemplate);
res = gst_audio_encoder_set_output_format (benc, outcaps);
gst_caps_unref (outcaps);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, 320);
gst_audio_encoder_set_frame_samples_max (benc, 320);
/* no remainder or flushing please */
gst_audio_encoder_set_hard_min (benc, TRUE);
gst_audio_encoder_set_drainable (benc, FALSE);
return res;
}
static GstFlowReturn
gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSirenEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint i, size, num_frames;
gint out_size, in_size;
gint encode_ret;
GstMapInfo inmap, outmap;
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
enc = GST_SIREN_ENC (benc);
size = gst_buffer_get_size (buf);
GST_LOG_OBJECT (enc, "Received buffer of size %d", size);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
/* we need to process 640 input bytes to produce 40 output bytes */
/* calculate the amount of frames we will handle */
num_frames = size / 640;
/* this is the input/output size */
in_size = num_frames * 640;
out_size = num_frames * 40;
GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
/* get a buffer */
out_buf = gst_audio_encoder_allocate_output_buffer (benc, out_size);
if (out_buf == NULL)
goto alloc_failed;
/* get the input data for all the frames */
gst_buffer_map (buf, &inmap, GST_MAP_READ);
gst_buffer_map (out_buf, &outmap, GST_MAP_READ);
in_data = inmap.data;
out_data = outmap.data;
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
/* encode 640 input bytes to 40 output bytes */
encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
if (encode_ret != 0)
goto encode_error;
/* move to next frame */
out_data += 40;
in_data += 640;
}
gst_buffer_unmap (buf, &inmap);
gst_buffer_unmap (out_buf, &outmap);
GST_LOG_OBJECT (enc, "Finished encoding");
/* we encode all we get, pass it along */
ret = gst_audio_encoder_finish_frame (benc, out_buf, -1);
done:
return ret;
/* ERRORS */
alloc_failed:
{
GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
encode_error:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("Error encoding frame: %d", encode_ret));
ret = GST_FLOW_ERROR;
gst_buffer_unref (out_buf);
goto done;
}
}
gboolean
gst_siren_enc_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "sirenenc",
GST_RANK_MARGINAL, GST_TYPE_SIREN_ENC);
}
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