mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 20:21:24 +00:00
c49cf83ee3
Original commit message from CVS: * docs/plugins/gst-plugins-ugly-plugins-docs.sgml: * docs/plugins/gst-plugins-ugly-plugins-sections.txt: * ext/a52dec/gsta52dec.c: * ext/amrnb/amrnbdec.c: * ext/amrnb/amrnbenc.c: * ext/amrnb/amrnbparse.c: * ext/lame/gstlame.c: * ext/mad/gstmad.c: * ext/sidplay/gstsiddec.cc: * gst/asfdemux/gstrtspwms.c: * gst/mpegaudioparse/gstxingmux.c: * gst/realmedia/rademux.c: * gst/realmedia/rdtmanager.c: * gst/realmedia/rtspreal.c: * gst/synaesthesia/gstsynaesthesia.c: Add missing elements to docs. Restore alphabetical order in section file. Document mad (it was included in docs already). Fix doc-markup: use convinience syntax for examples (produces valid docbook), add several refsec2 when we have several titles. Fix some types.
351 lines
10 KiB
C
351 lines
10 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
|
|
*
|
|
* 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;
|
|
}
|