mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-23 16:50:47 +00:00
234 lines
6.7 KiB
C
234 lines
6.7 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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
/**
|
|
* SECTION:element-amrwbdec
|
|
* @title: amrwbdec
|
|
* @see_also: #GstAmrwbEnc
|
|
*
|
|
* AMR wideband decoder based on the
|
|
* [opencore codec implementation](http://sourceforge.net/projects/opencore-amr).
|
|
*
|
|
* ## Example launch line
|
|
* |[
|
|
* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink
|
|
* ]|
|
|
*
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <gst/audio/audio.h>
|
|
|
|
#include "amrwbdec.h"
|
|
|
|
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/AMR-WB, "
|
|
"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/x-raw, "
|
|
"format = (string) " GST_AUDIO_NE (S16) ", "
|
|
"layout = (string) interleaved, "
|
|
"rate = (int) 16000, " "channels = (int) 1")
|
|
);
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (gst_amrwbdec_debug);
|
|
#define GST_CAT_DEFAULT gst_amrwbdec_debug
|
|
|
|
#define L_FRAME16k 320 /* Frame size at 16kHz */
|
|
|
|
static const unsigned char block_size[16] =
|
|
{ 18, 24, 33, 37, 41, 47, 51, 59, 61,
|
|
6, 0, 0, 0, 0, 1, 1
|
|
};
|
|
|
|
static gboolean gst_amrwbdec_start (GstAudioDecoder * dec);
|
|
static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec);
|
|
static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
|
|
static GstFlowReturn gst_amrwbdec_parse (GstAudioDecoder * dec,
|
|
GstAdapter * adapter, gint * offset, gint * length);
|
|
static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec,
|
|
GstBuffer * buffer);
|
|
|
|
#define gst_amrwbdec_parent_class parent_class
|
|
G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_AUDIO_DECODER);
|
|
GST_ELEMENT_REGISTER_DEFINE (amrwbdec, "amrwbdec",
|
|
GST_RANK_PRIMARY, GST_TYPE_AMRWBDEC);
|
|
|
|
static void
|
|
gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
|
|
{
|
|
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
|
|
|
|
gst_element_class_add_static_pad_template (element_class, &sink_template);
|
|
gst_element_class_add_static_pad_template (element_class, &src_template);
|
|
|
|
gst_element_class_set_static_metadata (element_class, "AMR-WB audio decoder",
|
|
"Codec/Decoder/Audio",
|
|
"Adaptive Multi-Rate Wideband audio decoder",
|
|
"Renato Araujo <renato.filho@indt.org.br>");
|
|
|
|
base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start);
|
|
base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop);
|
|
base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format);
|
|
base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse);
|
|
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0,
|
|
"AMR-WB audio decoder");
|
|
}
|
|
|
|
static void
|
|
gst_amrwbdec_init (GstAmrwbDec * amrwbdec)
|
|
{
|
|
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (amrwbdec), TRUE);
|
|
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
|
|
(amrwbdec), TRUE);
|
|
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (amrwbdec));
|
|
}
|
|
|
|
static gboolean
|
|
gst_amrwbdec_start (GstAudioDecoder * dec)
|
|
{
|
|
GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
|
|
|
|
GST_DEBUG_OBJECT (dec, "start");
|
|
if (!(amrwbdec->handle = D_IF_init ()))
|
|
return FALSE;
|
|
|
|
amrwbdec->rate = 0;
|
|
amrwbdec->channels = 0;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_amrwbdec_stop (GstAudioDecoder * dec)
|
|
{
|
|
GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
|
|
|
|
GST_DEBUG_OBJECT (dec, "stop");
|
|
D_IF_exit (amrwbdec->handle);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
|
|
{
|
|
GstStructure *structure;
|
|
GstAmrwbDec *amrwbdec;
|
|
GstAudioInfo info;
|
|
|
|
amrwbdec = GST_AMRWBDEC (dec);
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
/* get channel count */
|
|
gst_structure_get_int (structure, "channels", &amrwbdec->channels);
|
|
gst_structure_get_int (structure, "rate", &amrwbdec->rate);
|
|
|
|
/* create reverse caps */
|
|
gst_audio_info_init (&info);
|
|
gst_audio_info_set_format (&info,
|
|
GST_AUDIO_FORMAT_S16, amrwbdec->rate, amrwbdec->channels, NULL);
|
|
|
|
gst_audio_decoder_set_output_format (dec, &info);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
|
|
gint * offset, gint * length)
|
|
{
|
|
GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
|
|
guint8 header[1];
|
|
guint size;
|
|
gboolean sync, eos;
|
|
gint block, mode;
|
|
|
|
size = gst_adapter_available (adapter);
|
|
if (size < 1)
|
|
return GST_FLOW_ERROR;
|
|
|
|
gst_audio_decoder_get_parse_state (dec, &sync, &eos);
|
|
|
|
/* need to peek data to get the size */
|
|
gst_adapter_copy (adapter, header, 0, 1);
|
|
mode = (header[0] >> 3) & 0x0F;
|
|
block = block_size[mode];
|
|
|
|
GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block);
|
|
|
|
if (block) {
|
|
if (block > size)
|
|
return GST_FLOW_EOS;
|
|
*offset = 0;
|
|
*length = block;
|
|
} else {
|
|
/* no frame yet, skip one byte */
|
|
GST_LOG_OBJECT (amrwbdec, "skipping byte");
|
|
*offset = 1;
|
|
return GST_FLOW_EOS;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
|
|
{
|
|
GstAmrwbDec *amrwbdec;
|
|
GstBuffer *out;
|
|
GstMapInfo inmap, outmap;
|
|
|
|
amrwbdec = GST_AMRWBDEC (dec);
|
|
|
|
/* no fancy flushing */
|
|
if (!buffer || !gst_buffer_get_size (buffer))
|
|
return GST_FLOW_OK;
|
|
|
|
/* the library seems to write into the source data, hence the copy. */
|
|
/* should be no problem */
|
|
gst_buffer_map (buffer, &inmap, GST_MAP_READ);
|
|
|
|
/* get output */
|
|
out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k);
|
|
gst_buffer_map (out, &outmap, GST_MAP_WRITE);
|
|
|
|
/* decode */
|
|
D_IF_decode (amrwbdec->handle, (unsigned char *) inmap.data,
|
|
(short int *) outmap.data, _good_frame);
|
|
|
|
gst_buffer_unmap (out, &outmap);
|
|
gst_buffer_unmap (buffer, &inmap);
|
|
|
|
/* send out */
|
|
return gst_audio_decoder_finish_frame (dec, out, 1);
|
|
}
|