gstreamer/ext/amrwbenc/gstamrwbenc.c

387 lines
11 KiB
C

/* GStreamer Adaptive Multi-Rate Wide-Band (AMR-WB) plugin
* Copyright (C) 2006 Edgard Lima <edgard.lima@indt.org.br>
*
* 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-amrwbenc
* @see_also: #GstAmrwbDec, #GstAmrwbParse
*
* AMR wideband encoder based on the
* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrwbenc ! filesink location=abc.amr
* ]|
* Please not that the above stream misses the header, that is needed to play
* the stream.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstamrwbenc.h"
/* these defines are not in all .h files */
#ifndef MR660
#define MR660 0
#define MR885 1
#define MR1265 2
#define MR1425 2
#define MR1585 3
#define MR1825 4
#define MR1985 5
#define MR2305 6
#define MR2385 7
#define MRDTX 8
#endif
static GType
gst_amrwbenc_bandmode_get_type ()
{
static GType gst_amrwbenc_bandmode_type = 0;
static GEnumValue gst_amrwbenc_bandmode[] = {
{MR660, "MR660", "MR660"},
{MR885, "MR885", "MR885"},
{MR1265, "MR1265", "MR1265"},
{MR1425, "MR1425", "MR1425"},
{MR1585, "MR1585", "MR1585"},
{MR1825, "MR1825", "MR1825"},
{MR1985, "MR1985", "MR1985"},
{MR2305, "MR2305", "MR2305"},
{MR2385, "MR2385", "MR2385"},
{MRDTX, "MRDTX", "MRDTX"},
{0, NULL, NULL},
};
if (!gst_amrwbenc_bandmode_type) {
gst_amrwbenc_bandmode_type =
g_enum_register_static ("GstAmrWbEncBandMode", gst_amrwbenc_bandmode);
}
return gst_amrwbenc_bandmode_type;
}
#define GST_AMRWBENC_BANDMODE_TYPE (gst_amrwbenc_bandmode_get_type())
#define BANDMODE_DEFAULT MR660
enum
{
PROP_0,
PROP_BANDMODE
};
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
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")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/AMR-WB, "
"rate = (int) 16000, " "channels = (int) 1")
);
GST_DEBUG_CATEGORY_STATIC (gst_amrwbenc_debug);
#define GST_CAT_DEFAULT gst_amrwbenc_debug
static void gst_amrwbenc_finalize (GObject * object);
static GstFlowReturn gst_amrwbenc_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_amrwbenc_setcaps (GstPad * pad, GstCaps * caps);
static GstStateChangeReturn gst_amrwbenc_state_change (GstElement * element,
GstStateChange transition);
static void
_do_init (GType object_type)
{
const GInterfaceInfo preset_interface_info = {
NULL, /* interface init */
NULL, /* interface finalize */
NULL /* interface_data */
};
g_type_add_interface_static (object_type, GST_TYPE_PRESET,
&preset_interface_info);
GST_DEBUG_CATEGORY_INIT (gst_amrwbenc_debug, "amrwbenc", 0,
"AMR-WB audio encoder");
}
#define GstAmrWbEnc GstAmrwbEnc
#define GstAmrWbEncClass GstAmrwbEncClass
GST_BOILERPLATE_FULL (GstAmrWbEnc, gst_amrwbenc, GstElement, GST_TYPE_ELEMENT,
_do_init);
static void
gst_amrwbenc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAmrwbEnc *self = GST_AMRWBENC (object);
switch (prop_id) {
case PROP_BANDMODE:
self->bandmode = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static void
gst_amrwbenc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAmrwbEnc *self = GST_AMRWBENC (object);
switch (prop_id) {
case PROP_BANDMODE:
g_value_set_enum (value, self->bandmode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static void
gst_amrwbenc_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details_simple (element_class, "AMR-WB audio encoder",
"Codec/Encoder/Audio",
"Adaptive Multi-Rate Wideband audio encoder",
"Renato Araujo <renato.filho@indt.org.br>");
}
static void
gst_amrwbenc_class_init (GstAmrwbEncClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
object_class->finalize = gst_amrwbenc_finalize;
object_class->set_property = gst_amrwbenc_set_property;
object_class->get_property = gst_amrwbenc_get_property;
g_object_class_install_property (object_class, PROP_BANDMODE,
g_param_spec_enum ("band-mode", "Band Mode",
"Encoding Band Mode (Kbps)", GST_AMRWBENC_BANDMODE_TYPE,
BANDMODE_DEFAULT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrwbenc_state_change);
}
static void
gst_amrwbenc_init (GstAmrwbEnc * amrwbenc, GstAmrwbEncClass * klass)
{
/* create the sink pad */
amrwbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_setcaps_function (amrwbenc->sinkpad, gst_amrwbenc_setcaps);
gst_pad_set_chain_function (amrwbenc->sinkpad, gst_amrwbenc_chain);
gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->sinkpad);
/* create the src pad */
amrwbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (amrwbenc->srcpad);
gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->srcpad);
amrwbenc->adapter = gst_adapter_new ();
/* init rest */
amrwbenc->handle = NULL;
amrwbenc->channels = 0;
amrwbenc->rate = 0;
amrwbenc->ts = 0;
}
static void
gst_amrwbenc_finalize (GObject * object)
{
GstAmrwbEnc *amrwbenc;
amrwbenc = GST_AMRWBENC (object);
g_object_unref (G_OBJECT (amrwbenc->adapter));
amrwbenc->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_amrwbenc_setcaps (GstPad * pad, GstCaps * caps)
{
GstStructure *structure;
GstAmrwbEnc *amrwbenc;
GstCaps *copy;
amrwbenc = GST_AMRWBENC (GST_PAD_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
/* get channel count */
gst_structure_get_int (structure, "channels", &amrwbenc->channels);
gst_structure_get_int (structure, "rate", &amrwbenc->rate);
/* this is not wrong but will sound bad */
if (amrwbenc->channels != 1) {
GST_WARNING ("amrwbdec is only optimized for mono channels");
}
if (amrwbenc->rate != 16000) {
GST_WARNING ("amrwbdec is only optimized for 16000 Hz samplerate");
}
/* create reverse caps */
copy = gst_caps_new_simple ("audio/AMR-WB",
"channels", G_TYPE_INT, amrwbenc->channels,
"rate", G_TYPE_INT, amrwbenc->rate, NULL);
gst_pad_set_caps (amrwbenc->srcpad, copy);
gst_caps_unref (copy);
return TRUE;
}
static GstFlowReturn
gst_amrwbenc_chain (GstPad * pad, GstBuffer * buffer)
{
GstAmrwbEnc *amrwbenc;
GstFlowReturn ret = GST_FLOW_OK;
const int buffer_size = sizeof (Word16) * L_FRAME16k;
amrwbenc = GST_AMRWBENC (gst_pad_get_parent (pad));
g_return_val_if_fail (amrwbenc->handle, GST_FLOW_WRONG_STATE);
if (amrwbenc->rate == 0 || amrwbenc->channels == 0) {
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
/* discontinuity clears adapter, FIXME, maybe we can set some
* encoder flag to mask the discont. */
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (amrwbenc->adapter);
amrwbenc->ts = 0;
amrwbenc->discont = TRUE;
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
amrwbenc->ts = GST_BUFFER_TIMESTAMP (buffer);
ret = GST_FLOW_OK;
gst_adapter_push (amrwbenc->adapter, buffer);
/* Collect samples until we have enough for an output frame */
while (gst_adapter_available (amrwbenc->adapter) >= buffer_size) {
GstBuffer *out;
guint8 *data;
gint outsize;
out = gst_buffer_new_and_alloc (buffer_size);
GST_BUFFER_DURATION (out) = GST_SECOND * L_FRAME16k /
(amrwbenc->rate * amrwbenc->channels);
GST_BUFFER_TIMESTAMP (out) = amrwbenc->ts;
if (amrwbenc->ts != -1) {
amrwbenc->ts += GST_BUFFER_DURATION (out);
}
if (amrwbenc->discont) {
GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
amrwbenc->discont = FALSE;
}
gst_buffer_set_caps (out, gst_pad_get_caps (amrwbenc->srcpad));
data = (guint8 *) gst_adapter_peek (amrwbenc->adapter, buffer_size);
/* encode */
outsize =
E_IF_encode (amrwbenc->handle, amrwbenc->bandmode, (Word16 *) data,
(UWord8 *) GST_BUFFER_DATA (out), 0);
gst_adapter_flush (amrwbenc->adapter, buffer_size);
GST_BUFFER_SIZE (out) = outsize;
/* play */
if ((ret = gst_pad_push (amrwbenc->srcpad, out)) != GST_FLOW_OK)
break;
}
done:
gst_object_unref (amrwbenc);
return ret;
}
static GstStateChangeReturn
gst_amrwbenc_state_change (GstElement * element, GstStateChange transition)
{
GstAmrwbEnc *amrwbenc;
GstStateChangeReturn ret;
amrwbenc = GST_AMRWBENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!(amrwbenc->handle = E_IF_init ()))
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
amrwbenc->rate = 0;
amrwbenc->channels = 0;
amrwbenc->ts = 0;
amrwbenc->discont = FALSE;
gst_adapter_clear (amrwbenc->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
E_IF_exit (amrwbenc->handle);
break;
default:
break;
}
return ret;
}