1 /* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
2  * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
3  *
4  * This library is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Library General Public
6  * License as published by the Free Software Foundation; either
7  * version 2 of the License, or (at your option) any later version.
8  *
9  * This library is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
12  * Library General Public License for more details.
13  *
14  * You should have received a copy of the GNU Library General Public
15  * License along with this library; if not, write to the
16  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17  * Boston, MA 02110-1301, USA.
18  */
19 
20 /**
21  * SECTION:element-amrwbdec
22  * @see_also: #GstAmrwbEnc
23  *
24  * AMR wideband decoder based on the
25  * <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
26  *
27  * <refsect2>
28  * <title>Example launch line</title>
29  * |[
30  * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink
31  * ]|
32  * </refsect2>
33  */
34 
35 #ifdef HAVE_CONFIG_H
36 #include "config.h"
37 #endif
38 
39 #include <gst/audio/audio.h>
40 
41 #include "amrwbdec.h"
42 
43 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
44     GST_PAD_SINK,
45     GST_PAD_ALWAYS,
46     GST_STATIC_CAPS ("audio/AMR-WB, "
47         "rate = (int) 16000, " "channels = (int) 1")
48     );
49 
50 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
51     GST_PAD_SRC,
52     GST_PAD_ALWAYS,
53     GST_STATIC_CAPS ("audio/x-raw, "
54         "format = (string) " GST_AUDIO_NE (S16) ", "
55         "layout = (string) interleaved, "
56         "rate = (int) 16000, " "channels = (int) 1")
57     );
58 
59 GST_DEBUG_CATEGORY_STATIC (gst_amrwbdec_debug);
60 #define GST_CAT_DEFAULT gst_amrwbdec_debug
61 
62 #define L_FRAME16k      320     /* Frame size at 16kHz  */
63 
64 static const unsigned char block_size[16] =
65     { 18, 24, 33, 37, 41, 47, 51, 59, 61,
66   6, 0, 0, 0, 0, 1, 1
67 };
68 
69 static gboolean gst_amrwbdec_start (GstAudioDecoder * dec);
70 static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec);
71 static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
72 static GstFlowReturn gst_amrwbdec_parse (GstAudioDecoder * dec,
73     GstAdapter * adapter, gint * offset, gint * length);
74 static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec,
75     GstBuffer * buffer);
76 
77 #define gst_amrwbdec_parent_class parent_class
78 G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_AUDIO_DECODER);
79 
80 static void
gst_amrwbdec_class_init(GstAmrwbDecClass * klass)81 gst_amrwbdec_class_init (GstAmrwbDecClass * klass)
82 {
83   GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
84   GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
85 
86   gst_element_class_add_static_pad_template (element_class, &sink_template);
87   gst_element_class_add_static_pad_template (element_class, &src_template);
88 
89   gst_element_class_set_static_metadata (element_class, "AMR-WB audio decoder",
90       "Codec/Decoder/Audio",
91       "Adaptive Multi-Rate Wideband audio decoder",
92       "Renato Araujo <renato.filho@indt.org.br>");
93 
94   base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start);
95   base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop);
96   base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format);
97   base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse);
98   base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame);
99 
100   GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0,
101       "AMR-WB audio decoder");
102 }
103 
104 static void
gst_amrwbdec_init(GstAmrwbDec * amrwbdec)105 gst_amrwbdec_init (GstAmrwbDec * amrwbdec)
106 {
107   gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (amrwbdec), TRUE);
108   gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
109       (amrwbdec), TRUE);
110   GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (amrwbdec));
111 }
112 
113 static gboolean
gst_amrwbdec_start(GstAudioDecoder * dec)114 gst_amrwbdec_start (GstAudioDecoder * dec)
115 {
116   GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
117 
118   GST_DEBUG_OBJECT (dec, "start");
119   if (!(amrwbdec->handle = D_IF_init ()))
120     return FALSE;
121 
122   amrwbdec->rate = 0;
123   amrwbdec->channels = 0;
124 
125   return TRUE;
126 }
127 
128 static gboolean
gst_amrwbdec_stop(GstAudioDecoder * dec)129 gst_amrwbdec_stop (GstAudioDecoder * dec)
130 {
131   GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
132 
133   GST_DEBUG_OBJECT (dec, "stop");
134   D_IF_exit (amrwbdec->handle);
135 
136   return TRUE;
137 }
138 
139 static gboolean
gst_amrwbdec_set_format(GstAudioDecoder * dec,GstCaps * caps)140 gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
141 {
142   GstStructure *structure;
143   GstAmrwbDec *amrwbdec;
144   GstAudioInfo info;
145 
146   amrwbdec = GST_AMRWBDEC (dec);
147 
148   structure = gst_caps_get_structure (caps, 0);
149 
150   /* get channel count */
151   gst_structure_get_int (structure, "channels", &amrwbdec->channels);
152   gst_structure_get_int (structure, "rate", &amrwbdec->rate);
153 
154   /* create reverse caps */
155   gst_audio_info_init (&info);
156   gst_audio_info_set_format (&info,
157       GST_AUDIO_FORMAT_S16, amrwbdec->rate, amrwbdec->channels, NULL);
158 
159   gst_audio_decoder_set_output_format (dec, &info);
160 
161   return TRUE;
162 }
163 
164 static GstFlowReturn
gst_amrwbdec_parse(GstAudioDecoder * dec,GstAdapter * adapter,gint * offset,gint * length)165 gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
166     gint * offset, gint * length)
167 {
168   GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec);
169   guint8 header[1];
170   guint size;
171   gboolean sync, eos;
172   gint block, mode;
173 
174   size = gst_adapter_available (adapter);
175   if (size < 1)
176     return GST_FLOW_ERROR;
177 
178   gst_audio_decoder_get_parse_state (dec, &sync, &eos);
179 
180   /* need to peek data to get the size */
181   gst_adapter_copy (adapter, header, 0, 1);
182   mode = (header[0] >> 3) & 0x0F;
183   block = block_size[mode];
184 
185   GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block);
186 
187   if (block) {
188     if (block > size)
189       return GST_FLOW_EOS;
190     *offset = 0;
191     *length = block;
192   } else {
193     /* no frame yet, skip one byte */
194     GST_LOG_OBJECT (amrwbdec, "skipping byte");
195     *offset = 1;
196     return GST_FLOW_EOS;
197   }
198 
199   return GST_FLOW_OK;
200 }
201 
202 static GstFlowReturn
gst_amrwbdec_handle_frame(GstAudioDecoder * dec,GstBuffer * buffer)203 gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
204 {
205   GstAmrwbDec *amrwbdec;
206   GstBuffer *out;
207   GstMapInfo inmap, outmap;
208 
209   amrwbdec = GST_AMRWBDEC (dec);
210 
211   /* no fancy flushing */
212   if (!buffer || !gst_buffer_get_size (buffer))
213     return GST_FLOW_OK;
214 
215   /* the library seems to write into the source data, hence the copy. */
216   /* should be no problem */
217   gst_buffer_map (buffer, &inmap, GST_MAP_READ);
218 
219   /* get output */
220   out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k);
221   gst_buffer_map (out, &outmap, GST_MAP_WRITE);
222 
223   /* decode */
224   D_IF_decode (amrwbdec->handle, (unsigned char *) inmap.data,
225       (short int *) outmap.data, _good_frame);
226 
227   gst_buffer_unmap (out, &outmap);
228   gst_buffer_unmap (buffer, &inmap);
229 
230   /* send out */
231   return gst_audio_decoder_finish_frame (dec, out, 1);
232 }
233