gstreamer/ext/amrnb/amrnbenc.c
Stefan Kost bebdf0837e docs/plugins/: Add new docs. Scan c++ files too.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
* docs/plugins/inspect/plugin-a52dec.xml:
* docs/plugins/inspect/plugin-amrnb.xml:
* docs/plugins/inspect/plugin-asf.xml:
* docs/plugins/inspect/plugin-dvdlpcmdec.xml:
* docs/plugins/inspect/plugin-dvdsub.xml:
* docs/plugins/inspect/plugin-iec958.xml:
* docs/plugins/inspect/plugin-lame.xml:
* docs/plugins/inspect/plugin-mad.xml:
* docs/plugins/inspect/plugin-mpeg2dec.xml:
* docs/plugins/inspect/plugin-mpegaudioparse.xml:
* docs/plugins/inspect/plugin-mpegstream.xml:
Add new docs. Scan c++ files too.
* ext/amrnb/amrnbdec.c: (gst_amrnbdec_base_init),
(gst_amrnbdec_event):
* ext/amrnb/amrnbenc.c: (gst_amrnbenc_base_init):
* ext/amrnb/amrnbparse.c: (gst_amrnbparse_base_init),
(gst_amrnbparse_loop):
Add documentation headers.
* ext/mad/gstmad.c:
* ext/mad/gstmad.h:
Refactor for docs.
2007-07-03 11:55:45 +00:00

281 lines
8.3 KiB
C

/* 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
*
* <refsect2>
* <para>
* This is an AMR narrowband encoder based on the
* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! filesink location=abc.amr
* </programlisting>
* </para>
* 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 "amrnbenc.h"
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_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@fluendo.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->finalize = gst_amrnbenc_finalize;
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, MR122, (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;
}