gstreamer/gst/siren/gstsirendec.c

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/*
* Siren Decoder 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-sirendec
*
* This decodes audio buffers from 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 "gstsirendec.h"
#include <string.h>
GST_DEBUG_CATEGORY (sirendec_debug);
#define GST_CAT_DEFAULT (sirendec_debug)
#define FRAME_DURATION (20 * GST_MSECOND)
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320"));
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 16, "
"depth = (int) 16, "
"endianness = (int) 1234, "
"signed = (boolean) true, "
"rate = (int) 16000, " "channels = (int) 1"));
/* signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
};
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static gboolean gst_siren_dec_start (GstAudioDecoder * dec);
static gboolean gst_siren_dec_stop (GstAudioDecoder * dec);
static gboolean gst_siren_dec_set_format (GstAudioDecoder * dec,
GstCaps * caps);
static gboolean gst_siren_dec_parse (GstAudioDecoder * dec,
GstAdapter * adapter, gint * offset, gint * length);
static GstFlowReturn gst_siren_dec_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static void
_do_init (GType type)
{
GST_DEBUG_CATEGORY_INIT (sirendec_debug, "sirendec", 0, "sirendec");
}
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GST_BOILERPLATE_FULL (GstSirenDec, gst_siren_dec, GstAudioDecoder,
GST_TYPE_AUDIO_DECODER, _do_init);
static void
gst_siren_dec_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class, &srctemplate);
gst_element_class_add_static_pad_template (element_class, &sinktemplate);
gst_element_class_set_details_simple (element_class, "Siren Decoder element",
"Codec/Decoder/Audio ",
"Decode streams encoded with the Siren7 codec into 16bit PCM",
"Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
}
static void
gst_siren_dec_class_init (GstSirenDecClass * klass)
{
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
GST_DEBUG ("Initializing Class");
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base_class->start = GST_DEBUG_FUNCPTR (gst_siren_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_dec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_dec_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (gst_siren_dec_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_dec_handle_frame);
GST_DEBUG ("Class Init done");
}
static void
gst_siren_dec_init (GstSirenDec * dec, GstSirenDecClass * klass)
{
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}
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static gboolean
gst_siren_dec_start (GstAudioDecoder * dec)
{
GstSirenDec *sdec = GST_SIREN_DEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
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sdec->decoder = Siren7_NewDecoder (16000);;
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/* no flushing please */
gst_audio_decoder_set_drainable (dec, FALSE);
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return TRUE;
}
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static gboolean
gst_siren_dec_stop (GstAudioDecoder * dec)
{
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GstSirenDec *sdec = GST_SIREN_DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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Siren7_CloseDecoder (sdec->decoder);
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return TRUE;
}
static gboolean
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gst_siren_dec_negotiate (GstSirenDec * dec)
{
gboolean res;
GstCaps *outcaps;
outcaps = gst_static_pad_template_get_caps (&srctemplate);
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res = gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), outcaps);
gst_caps_unref (outcaps);
return res;
}
static gboolean
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gst_siren_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstSirenDec *dec;
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dec = GST_SIREN_DEC (bdec);
return gst_siren_dec_negotiate (dec);
}
static GstFlowReturn
gst_siren_dec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length)
{
gint size;
GstFlowReturn ret;
size = gst_adapter_available (adapter);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
/* accept any multiple of frames */
if (size > 40) {
ret = GST_FLOW_OK;
*offset = 0;
*length = size - (size % 40);
} else {
ret = GST_FLOW_UNEXPECTED;
}
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return ret;
}
static GstFlowReturn
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gst_siren_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstSirenDec *dec;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint i, size, num_frames;
gint out_size, in_size;
gint decode_ret;
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dec = GST_SIREN_DEC (bdec);
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size = GST_BUFFER_SIZE (buf);
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GST_LOG_OBJECT (dec, "Received buffer of size %u", size);
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g_return_val_if_fail (size % 40 == 0, GST_FLOW_ERROR);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
/* process 40 input bytes into 640 output bytes */
num_frames = size / 40;
/* this is the input/output size */
in_size = num_frames * 40;
out_size = num_frames * 640;
GST_LOG_OBJECT (dec, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
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/* allow and handle un-negotiated input */
if (G_UNLIKELY (GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) == NULL)) {
gst_siren_dec_negotiate (dec);
}
/* get a buffer */
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ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), -1,
out_size, GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &out_buf);
if (ret != GST_FLOW_OK)
goto alloc_failed;
/* get the input data for all the frames */
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in_data = GST_BUFFER_DATA (buf);
out_data = GST_BUFFER_DATA (out_buf);
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (dec, "Decoding frame %u/%u", i, num_frames);
/* decode 40 input bytes to 640 output bytes */
decode_ret = Siren7_DecodeFrame (dec->decoder, in_data, out_data);
if (decode_ret != 0)
goto decode_error;
/* move to next frame */
out_data += 640;
in_data += 40;
}
GST_LOG_OBJECT (dec, "Finished decoding");
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/* might really be multiple frames,
* but was treated as one for all purposes here */
ret = gst_audio_decoder_finish_frame (bdec, out_buf, 1);
done:
return ret;
/* ERRORS */
alloc_failed:
{
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GST_DEBUG_OBJECT (dec, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
decode_error:
{
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GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("Error decoding frame: %d", decode_ret), ret);
if (ret == GST_FLOW_OK)
gst_audio_decoder_finish_frame (bdec, NULL, 1);
gst_buffer_unref (out_buf);
goto done;
}
}
gboolean
gst_siren_dec_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "sirendec",
GST_RANK_MARGINAL, GST_TYPE_SIREN_DEC);
}