gstreamer/ext/amrwbdec/amrwbdec.c

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/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.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-amrwbdec
* @see_also: #GstAmrwbEnc
*
* AMR wideband decoder based on the
* <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! audioresample ! audioconvert ! alsasink
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "amrwbdec.h"
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/AMR-WB, "
"rate = (int) 16000, " "channels = (int) 1")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 16, "
"depth = (int) 16, "
"signed = (boolean) TRUE, "
"endianness = (int) BYTE_ORDER, "
"rate = (int) 16000, " "channels = (int) 1")
);
GST_DEBUG_CATEGORY_STATIC (gst_amrwbdec_debug);
#define GST_CAT_DEFAULT gst_amrwbdec_debug
#define L_FRAME16k 320 /* Frame size at 16kHz */
static const unsigned char block_size[16] =
{ 18, 24, 33, 37, 41, 47, 51, 59, 61,
6, 0, 0, 0, 0, 1, 1
};
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static gboolean gst_amrwbdec_start (GstAudioDecoder * dec);
static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec);
static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
static gboolean gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length);
static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0, "AMR-WB audio decoder");
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GST_BOILERPLATE_FULL (GstAmrwbDec, gst_amrwbdec, GstAudioDecoder,
GST_TYPE_AUDIO_DECODER, _do_init);
static void
gst_amrwbdec_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class,
&sink_template);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_details_simple (element_class, "AMR-WB audio decoder",
"Codec/Decoder/Audio",
"Adaptive Multi-Rate Wideband audio decoder",
"Renato Araujo <renato.filho@indt.org.br>");
}
static void
gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
{
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame);
}
static void
gst_amrwbdec_init (GstAmrwbDec * amrwbdec, GstAmrwbDecClass * klass)
{
}
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static gboolean
gst_amrwbdec_start (GstAudioDecoder * dec)
{
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GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
if (!(amrwbdec->handle = D_IF_init ()))
return FALSE;
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amrwbdec->rate = 0;
amrwbdec->channels = 0;
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return TRUE;
}
static gboolean
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gst_amrwbdec_stop (GstAudioDecoder * dec)
{
GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
GST_DEBUG_OBJECT (dec, "stop");
D_IF_exit (amrwbdec->handle);
return TRUE;
}
static gboolean
gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
GstStructure *structure;
GstAmrwbDec *amrwbdec;
GstCaps *copy;
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amrwbdec = GST_AMRWBDEC (dec);
structure = gst_caps_get_structure (caps, 0);
/* get channel count */
gst_structure_get_int (structure, "channels", &amrwbdec->channels);
gst_structure_get_int (structure, "rate", &amrwbdec->rate);
/* create reverse caps */
copy = gst_caps_new_simple ("audio/x-raw-int",
"channels", G_TYPE_INT, amrwbdec->channels,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", G_TYPE_INT, amrwbdec->rate, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (amrwbdec), copy);
gst_caps_unref (copy);
return TRUE;
}
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static GstFlowReturn
gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length)
{
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GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
const guint8 *data;
guint size;
gboolean sync, eos;
gint block, mode;
size = gst_adapter_available (adapter);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
gst_audio_decoder_get_parse_state (dec, &sync, &eos);
/* need to peek data to get the size */
if (gst_adapter_available (adapter) < 1)
return GST_FLOW_ERROR;
data = gst_adapter_peek (adapter, 1);
mode = (data[0] >> 3) & 0x0F;
block = block_size[mode];
GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block);
if (block) {
*offset = 0;
*length = block;
} else {
/* no frame yet, skip one byte */
GST_LOG_OBJECT (amrwbdec, "skipping byte");
*offset = 1;
return GST_FLOW_UNEXPECTED;
}
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return GST_FLOW_OK;
}
static GstFlowReturn
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gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstAmrwbDec *amrwbdec;
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GstBuffer *out;
const guint8 *data;
amrwbdec = GST_AMRWBDEC (dec);
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/* no fancy flushing */
if (!buffer || !GST_BUFFER_SIZE (buffer))
return GST_FLOW_OK;
if (amrwbdec->rate == 0 || amrwbdec->channels == 0)
goto not_negotiated;
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/* the library seems to write into the source data, hence the copy. */
/* should be no problem */
data = GST_BUFFER_DATA (buffer);
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/* get output */
out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k);
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/* decode */
D_IF_decode (amrwbdec->handle, (unsigned char *) data,
(Word16 *) GST_BUFFER_DATA (out), _good_frame);
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/* send out */
return gst_audio_decoder_finish_frame (dec, out, 1);
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (amrwbdec, STREAM, TYPE_NOT_FOUND, (NULL),
("Decoder is not initialized"));
return GST_FLOW_NOT_NEGOTIATED;
}
}