gstreamer/ext/amrnb/amrnbenc.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-amrnbenc
* @see_also: #GstAmrnbDec, #GstAmrnbParse
*
* AMR narrowband 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 ! amrnbenc ! filesink location=abc.amr
* ]|
* Please note that the above stream misses the header, that is needed to play
* the stream.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "amrnbenc.h"
static GType
gst_amrnbenc_bandmode_get_type ()
{
static GType gst_amrnbenc_bandmode_type = 0;
static const GEnumValue gst_amrnbenc_bandmode[] = {
{MR475, "MR475", "MR475"},
{MR515, "MR515", "MR515"},
{MR59, "MR59", "MR59"},
{MR67, "MR67", "MR67"},
{MR74, "MR74", "MR74"},
{MR795, "MR795", "MR795"},
{MR102, "MR102", "MR102"},
{MR122, "MR122", "MR122"},
{MRDTX, "MRDTX", "MRDTX"},
{0, NULL, NULL},
};
if (!gst_amrnbenc_bandmode_type) {
gst_amrnbenc_bandmode_type =
g_enum_register_static ("GstAmrnbEncBandMode", gst_amrnbenc_bandmode);
}
return gst_amrnbenc_bandmode_type;
}
#define GST_AMRNBENC_BANDMODE_TYPE (gst_amrnbenc_bandmode_get_type())
#define BANDMODE_DEFAULT MR122
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) 8000," "channels = (int) 1")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1")
);
GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug);
#define GST_CAT_DEFAULT gst_amrnbenc_debug
static void gst_amrnbenc_finalize (GObject * object);
static GstFlowReturn gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps);
static GstStateChangeReturn gst_amrnbenc_state_change (GstElement * element,
GstStateChange transition);
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0, "AMR-NB audio encoder");
GST_BOILERPLATE_FULL (GstAmrnbEnc, gst_amrnbenc, GstElement, GST_TYPE_ELEMENT,
_do_init);
static void
gst_amrnbenc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAmrnbEnc *self = GST_AMRNBENC (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_amrnbenc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAmrnbEnc *self = GST_AMRNBENC (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_amrnbenc_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstElementDetails details = GST_ELEMENT_DETAILS ("AMR-NB audio encoder",
"Codec/Encoder/Audio",
"Adaptive Multi-Rate Narrow-Band audio encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>, "
"Wim Taymans <wim.taymans@gmail.com>");
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 (element_class, &details);
}
static void
gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
object_class->set_property = gst_amrnbenc_set_property;
object_class->get_property = gst_amrnbenc_get_property;
object_class->finalize = gst_amrnbenc_finalize;
g_object_class_install_property (object_class, PROP_BANDMODE,
g_param_spec_enum ("band-mode", "Band Mode",
"Encoding Band Mode (Kbps)", GST_AMRNBENC_BANDMODE_TYPE,
BANDMODE_DEFAULT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrnbenc_state_change);
}
static void
gst_amrnbenc_init (GstAmrnbEnc * amrnbenc, GstAmrnbEncClass * klass)
{
/* create the sink pad */
amrnbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_setcaps_function (amrnbenc->sinkpad, gst_amrnbenc_setcaps);
gst_pad_set_chain_function (amrnbenc->sinkpad, gst_amrnbenc_chain);
gst_element_add_pad (GST_ELEMENT (amrnbenc), amrnbenc->sinkpad);
/* create the src pad */
amrnbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (amrnbenc->srcpad);
gst_element_add_pad (GST_ELEMENT (amrnbenc), amrnbenc->srcpad);
amrnbenc->adapter = gst_adapter_new ();
/* init rest */
amrnbenc->handle = NULL;
}
static void
gst_amrnbenc_finalize (GObject * object)
{
GstAmrnbEnc *amrnbenc;
amrnbenc = GST_AMRNBENC (object);
g_object_unref (G_OBJECT (amrnbenc->adapter));
amrnbenc->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps)
{
GstStructure *structure;
GstAmrnbEnc *amrnbenc;
GstCaps *copy;
amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
/* get channel count */
gst_structure_get_int (structure, "channels", &amrnbenc->channels);
gst_structure_get_int (structure, "rate", &amrnbenc->rate);
/* this is not wrong but will sound bad */
if (amrnbenc->channels != 1) {
g_warning ("amrnbdec is only optimized for mono channels");
}
if (amrnbenc->rate != 8000) {
g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
}
/* create reverse caps */
copy = gst_caps_new_simple ("audio/AMR",
"channels", G_TYPE_INT, amrnbenc->channels,
"rate", G_TYPE_INT, amrnbenc->rate, NULL);
/* precalc duration as it's constant now */
amrnbenc->duration = gst_util_uint64_scale_int (160, GST_SECOND,
amrnbenc->rate * amrnbenc->channels);
gst_pad_set_caps (amrnbenc->srcpad, copy);
gst_caps_unref (copy);
return TRUE;
}
static GstFlowReturn
gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer)
{
GstAmrnbEnc *amrnbenc;
GstFlowReturn ret;
amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad));
g_return_val_if_fail (amrnbenc->handle, GST_FLOW_WRONG_STATE);
if (amrnbenc->rate == 0 || amrnbenc->channels == 0)
goto not_negotiated;
/* 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 (amrnbenc->adapter);
amrnbenc->ts = 0;
}
/* take latest timestamp, FIXME timestamp is the one of the
* first buffer in the adapter. */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
amrnbenc->ts = GST_BUFFER_TIMESTAMP (buffer);
ret = GST_FLOW_OK;
gst_adapter_push (amrnbenc->adapter, buffer);
/* Collect samples until we have enough for an output frame */
while (gst_adapter_available (amrnbenc->adapter) >= 320) {
GstBuffer *out;
guint8 *data;
gint outsize;
/* get output, max size is 32 */
out = gst_buffer_new_and_alloc (32);
GST_BUFFER_DURATION (out) = amrnbenc->duration;
GST_BUFFER_TIMESTAMP (out) = amrnbenc->ts;
if (amrnbenc->ts != -1)
amrnbenc->ts += amrnbenc->duration;
gst_buffer_set_caps (out, GST_PAD_CAPS (amrnbenc->srcpad));
/* The AMR encoder actually writes into the source data buffers it gets */
data = gst_adapter_take (amrnbenc->adapter, 320);
/* encode */
outsize =
Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
(short *) data, (guint8 *) GST_BUFFER_DATA (out), 0);
g_free (data);
GST_BUFFER_SIZE (out) = outsize;
/* play */
if ((ret = gst_pad_push (amrnbenc->srcpad, out)) != GST_FLOW_OK)
break;
}
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (amrnbenc, STREAM, TYPE_NOT_FOUND,
(NULL), ("unknown type"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
static GstStateChangeReturn
gst_amrnbenc_state_change (GstElement * element, GstStateChange transition)
{
GstAmrnbEnc *amrnbenc;
GstStateChangeReturn ret;
amrnbenc = GST_AMRNBENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!(amrnbenc->handle = Encoder_Interface_init (0)))
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
amrnbenc->rate = 0;
amrnbenc->channels = 0;
amrnbenc->ts = 0;
gst_adapter_clear (amrnbenc->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
Encoder_Interface_exit (amrnbenc->handle);
break;
default:
break;
}
return ret;
}