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Fedora ships these libraries as part of the main distribution now, and they are decades old anyway so don't implement any of the newer features. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4512>
234 lines
6.7 KiB
C
234 lines
6.7 KiB
C
/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
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* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-amrwbdec
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* @title: amrwbdec
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* @see_also: #GstAmrwbEnc
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*
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* AMR wideband decoder based on the
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* [opencore codec implementation](http://sourceforge.net/projects/opencore-amr).
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink
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* ]|
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/audio/audio.h>
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#include "amrwbdec.h"
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/AMR-WB, "
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"rate = (int) 16000, " "channels = (int) 1")
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);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) 16000, " "channels = (int) 1")
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);
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GST_DEBUG_CATEGORY_STATIC (gst_amrwbdec_debug);
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#define GST_CAT_DEFAULT gst_amrwbdec_debug
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#define L_FRAME16k 320 /* Frame size at 16kHz */
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static const unsigned char block_size[16] =
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{ 18, 24, 33, 37, 41, 47, 51, 59, 61,
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6, 0, 0, 0, 0, 1, 1
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};
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static gboolean gst_amrwbdec_start (GstAudioDecoder * dec);
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static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec);
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static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
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static GstFlowReturn gst_amrwbdec_parse (GstAudioDecoder * dec,
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GstAdapter * adapter, gint * offset, gint * length);
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static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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#define gst_amrwbdec_parent_class parent_class
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G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_AUDIO_DECODER);
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GST_ELEMENT_REGISTER_DEFINE (amrwbdec, "amrwbdec",
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GST_RANK_PRIMARY, GST_TYPE_AMRWBDEC);
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static void
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gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
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{
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_set_static_metadata (element_class, "AMR-WB audio decoder",
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"Codec/Decoder/Audio",
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"Adaptive Multi-Rate Wideband audio decoder",
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"Renato Araujo <renato.filho@indt.org.br>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame);
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GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0,
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"AMR-WB audio decoder");
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}
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static void
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gst_amrwbdec_init (GstAmrwbDec * amrwbdec)
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{
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (amrwbdec), TRUE);
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(amrwbdec), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (amrwbdec));
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}
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static gboolean
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gst_amrwbdec_start (GstAudioDecoder * dec)
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{
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GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
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if (!(amrwbdec->handle = D_IF_init ()))
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return FALSE;
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amrwbdec->rate = 0;
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amrwbdec->channels = 0;
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return TRUE;
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}
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static gboolean
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gst_amrwbdec_stop (GstAudioDecoder * dec)
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{
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GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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D_IF_exit (amrwbdec->handle);
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return TRUE;
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}
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static gboolean
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gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstStructure *structure;
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GstAmrwbDec *amrwbdec;
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GstAudioInfo info;
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amrwbdec = GST_AMRWBDEC (dec);
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structure = gst_caps_get_structure (caps, 0);
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/* get channel count */
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gst_structure_get_int (structure, "channels", &amrwbdec->channels);
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gst_structure_get_int (structure, "rate", &amrwbdec->rate);
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/* create reverse caps */
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info,
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GST_AUDIO_FORMAT_S16, amrwbdec->rate, amrwbdec->channels, NULL);
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gst_audio_decoder_set_output_format (dec, &info);
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return TRUE;
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}
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static GstFlowReturn
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gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
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guint8 header[1];
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guint size;
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gboolean sync, eos;
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gint block, mode;
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size = gst_adapter_available (adapter);
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if (size < 1)
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return GST_FLOW_ERROR;
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gst_audio_decoder_get_parse_state (dec, &sync, &eos);
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/* need to peek data to get the size */
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gst_adapter_copy (adapter, header, 0, 1);
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mode = (header[0] >> 3) & 0x0F;
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block = block_size[mode];
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GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block);
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if (block) {
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if (block > size)
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return GST_FLOW_EOS;
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*offset = 0;
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*length = block;
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} else {
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/* no frame yet, skip one byte */
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GST_LOG_OBJECT (amrwbdec, "skipping byte");
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*offset = 1;
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return GST_FLOW_EOS;
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}
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
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{
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GstAmrwbDec *amrwbdec;
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GstBuffer *out;
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GstMapInfo inmap, outmap;
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amrwbdec = GST_AMRWBDEC (dec);
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/* no fancy flushing */
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if (!buffer || !gst_buffer_get_size (buffer))
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return GST_FLOW_OK;
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/* the library seems to write into the source data, hence the copy. */
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/* should be no problem */
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gst_buffer_map (buffer, &inmap, GST_MAP_READ);
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/* get output */
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out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k);
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gst_buffer_map (out, &outmap, GST_MAP_WRITE);
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/* decode */
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D_IF_decode (amrwbdec->handle, (unsigned char *) inmap.data,
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(short int *) outmap.data, _good_frame);
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gst_buffer_unmap (out, &outmap);
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gst_buffer_unmap (buffer, &inmap);
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/* send out */
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return gst_audio_decoder_finish_frame (dec, out, 1);
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}
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