gstreamer/subprojects/gst-plugins-good/gst/law/mulaw-encode.c

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-mulawenc
* @title: mulawenc
*
* This element encode mulaw audio. Mulaw coding is also known as G.711.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
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#include <gst/audio/audio.h>
#include "mulaw-encode.h"
#include "mulaw-conversion.h"
extern GstStaticPadTemplate mulaw_enc_src_factory;
extern GstStaticPadTemplate mulaw_enc_sink_factory;
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define INT_FORMAT "S16LE"
#else
#define INT_FORMAT "S16BE"
#endif
GstStaticPadTemplate mulaw_enc_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " INT_FORMAT ", "
"layout = (string) interleaved, "
"rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]")
);
GstStaticPadTemplate mulaw_enc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-mulaw, "
"rate = [ 8000 , 192000 ], " "channels = [ 1 , 2 ]")
);
/* Stereo signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
static gboolean gst_mulawenc_start (GstAudioEncoder * audioenc);
static gboolean gst_mulawenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_mulawenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * buffer);
static void gst_mulawenc_set_tags (GstMuLawEnc * mulawenc);
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#define gst_mulawenc_parent_class parent_class
G_DEFINE_TYPE (GstMuLawEnc, gst_mulawenc, GST_TYPE_AUDIO_ENCODER);
GST_ELEMENT_REGISTER_DEFINE (mulawenc, "mulawenc", GST_RANK_PRIMARY,
GST_TYPE_MULAWENC);
/*static guint gst_stereo_signals[LAST_SIGNAL] = { 0 }; */
static gboolean
gst_mulawenc_start (GstAudioEncoder * audioenc)
{
GstMuLawEnc *mulawenc = GST_MULAWENC (audioenc);
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mulawenc->channels = 0;
mulawenc->rate = 0;
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return TRUE;
}
static void
gst_mulawenc_set_tags (GstMuLawEnc * mulawenc)
{
GstTagList *taglist;
guint bitrate;
/* bitrate of mulaw is 8 bits/sample * sample rate * number of channels */
bitrate = 8 * mulawenc->rate * mulawenc->channels;
taglist = gst_tag_list_new_empty ();
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_MAXIMUM_BITRATE, bitrate, NULL);
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_MINIMUM_BITRATE, bitrate, NULL);
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, bitrate, NULL);
gst_audio_encoder_merge_tags (GST_AUDIO_ENCODER (mulawenc),
taglist, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (taglist);
}
static gboolean
gst_mulawenc_set_format (GstAudioEncoder * audioenc, GstAudioInfo * info)
{
GstCaps *base_caps;
GstStructure *structure;
GstMuLawEnc *mulawenc = GST_MULAWENC (audioenc);
gboolean ret;
mulawenc->rate = info->rate;
mulawenc->channels = info->channels;
base_caps =
gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (audioenc));
g_assert (base_caps);
base_caps = gst_caps_make_writable (base_caps);
g_assert (base_caps);
structure = gst_caps_get_structure (base_caps, 0);
g_assert (structure);
gst_structure_set (structure, "rate", G_TYPE_INT, mulawenc->rate, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT, mulawenc->channels,
NULL);
gst_mulawenc_set_tags (mulawenc);
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ret = gst_audio_encoder_set_output_format (audioenc, base_caps);
gst_caps_unref (base_caps);
return ret;
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}
static GstFlowReturn
gst_mulawenc_handle_frame (GstAudioEncoder * audioenc, GstBuffer * buffer)
{
GstMuLawEnc *mulawenc;
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GstMapInfo inmap, outmap;
gint16 *linear_data;
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gsize linear_size;
guint8 *mulaw_data;
guint mulaw_size;
GstBuffer *outbuf;
GstFlowReturn ret;
if (!buffer) {
ret = GST_FLOW_OK;
goto done;
}
mulawenc = GST_MULAWENC (audioenc);
if (!mulawenc->rate || !mulawenc->channels)
goto not_negotiated;
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gst_buffer_map (buffer, &inmap, GST_MAP_READ);
linear_data = (gint16 *) inmap.data;
linear_size = inmap.size;
mulaw_size = linear_size / 2;
outbuf = gst_audio_encoder_allocate_output_buffer (audioenc, mulaw_size);
g_assert (outbuf);
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gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
mulaw_data = outmap.data;
mulaw_encode (linear_data, mulaw_data, mulaw_size);
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gst_buffer_unmap (outbuf, &outmap);
gst_buffer_unmap (buffer, &inmap);
ret = gst_audio_encoder_finish_frame (audioenc, outbuf, -1);
done:
return ret;
not_negotiated:
{
GST_DEBUG_OBJECT (mulawenc, "no format negotiated");
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
}
static void
gst_mulawenc_class_init (GstMuLawEncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *audio_encoder_class = GST_AUDIO_ENCODER_CLASS (klass);
audio_encoder_class->start = GST_DEBUG_FUNCPTR (gst_mulawenc_start);
audio_encoder_class->set_format = GST_DEBUG_FUNCPTR (gst_mulawenc_set_format);
audio_encoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_mulawenc_handle_frame);
gst_element_class_add_static_pad_template (element_class,
&mulaw_enc_src_factory);
gst_element_class_add_static_pad_template (element_class,
&mulaw_enc_sink_factory);
gst_element_class_set_static_metadata (element_class, "Mu Law audio encoder",
"Codec/Encoder/Audio",
"Convert 16bit PCM to 8bit mu law",
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
}
static void
gst_mulawenc_init (GstMuLawEnc * mulawenc)
{
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (mulawenc));
}