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-amrnbenc
22  * @see_also: #GstAmrnbDec, #GstAmrnbParse
23  *
24  * AMR narrowband encoder 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.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr
31  * ]|
32  * Please note that the above stream misses the header, that is needed to play
33  * the stream.
34  * </refsect2>
35  */
36 
37 #ifdef HAVE_CONFIG_H
38 #include "config.h"
39 #endif
40 
41 #include "amrnbenc.h"
42 
43 static GType
gst_amrnbenc_bandmode_get_type(void)44 gst_amrnbenc_bandmode_get_type (void)
45 {
46   static GType gst_amrnbenc_bandmode_type = 0;
47   static const GEnumValue gst_amrnbenc_bandmode[] = {
48     {MR475, "MR475", "MR475"},
49     {MR515, "MR515", "MR515"},
50     {MR59, "MR59", "MR59"},
51     {MR67, "MR67", "MR67"},
52     {MR74, "MR74", "MR74"},
53     {MR795, "MR795", "MR795"},
54     {MR102, "MR102", "MR102"},
55     {MR122, "MR122", "MR122"},
56     {MRDTX, "MRDTX", "MRDTX"},
57     {0, NULL, NULL},
58   };
59   if (!gst_amrnbenc_bandmode_type) {
60     gst_amrnbenc_bandmode_type =
61         g_enum_register_static ("GstAmrnbEncBandMode", gst_amrnbenc_bandmode);
62   }
63   return gst_amrnbenc_bandmode_type;
64 }
65 
66 #define GST_AMRNBENC_BANDMODE_TYPE (gst_amrnbenc_bandmode_get_type())
67 
68 #define BANDMODE_DEFAULT MR122
69 enum
70 {
71   PROP_0,
72   PROP_BANDMODE
73 };
74 
75 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
76     GST_PAD_SINK,
77     GST_PAD_ALWAYS,
78     GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S16) ", "
79         "layout = (string) interleaved, "
80         "rate = (int) 8000," "channels = (int) 1")
81     );
82 
83 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
84     GST_PAD_SRC,
85     GST_PAD_ALWAYS,
86     GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1")
87     );
88 
89 GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug);
90 #define GST_CAT_DEFAULT gst_amrnbenc_debug
91 
92 static gboolean gst_amrnbenc_start (GstAudioEncoder * enc);
93 static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc);
94 static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
95     GstAudioInfo * info);
96 static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
97     GstBuffer * in_buf);
98 
99 #define gst_amrnbenc_parent_class parent_class
100 G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER);
101 
102 static void
gst_amrnbenc_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)103 gst_amrnbenc_set_property (GObject * object, guint prop_id,
104     const GValue * value, GParamSpec * pspec)
105 {
106   GstAmrnbEnc *self = GST_AMRNBENC (object);
107 
108   switch (prop_id) {
109     case PROP_BANDMODE:
110       self->bandmode = g_value_get_enum (value);
111       break;
112     default:
113       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
114       break;
115   }
116   return;
117 }
118 
119 static void
gst_amrnbenc_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)120 gst_amrnbenc_get_property (GObject * object, guint prop_id,
121     GValue * value, GParamSpec * pspec)
122 {
123   GstAmrnbEnc *self = GST_AMRNBENC (object);
124 
125   switch (prop_id) {
126     case PROP_BANDMODE:
127       g_value_set_enum (value, self->bandmode);
128       break;
129     default:
130       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
131       break;
132   }
133   return;
134 }
135 
136 static void
gst_amrnbenc_class_init(GstAmrnbEncClass * klass)137 gst_amrnbenc_class_init (GstAmrnbEncClass * klass)
138 {
139   GObjectClass *object_class = G_OBJECT_CLASS (klass);
140   GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
141   GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
142 
143   object_class->set_property = gst_amrnbenc_set_property;
144   object_class->get_property = gst_amrnbenc_get_property;
145 
146   base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbenc_start);
147   base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbenc_stop);
148   base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbenc_set_format);
149   base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbenc_handle_frame);
150 
151   g_object_class_install_property (object_class, PROP_BANDMODE,
152       g_param_spec_enum ("band-mode", "Band Mode",
153           "Encoding Band Mode (Kbps)", GST_AMRNBENC_BANDMODE_TYPE,
154           BANDMODE_DEFAULT,
155           G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
156 
157   gst_element_class_add_static_pad_template (element_class, &sink_template);
158   gst_element_class_add_static_pad_template (element_class, &src_template);
159 
160   gst_element_class_set_static_metadata (element_class, "AMR-NB audio encoder",
161       "Codec/Encoder/Audio",
162       "Adaptive Multi-Rate Narrow-Band audio encoder",
163       "Wim Taymans <wim.taymans@gmail.com>");
164 
165   GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
166       "AMR-NB audio encoder");
167 }
168 
169 static void
gst_amrnbenc_init(GstAmrnbEnc * amrnbenc)170 gst_amrnbenc_init (GstAmrnbEnc * amrnbenc)
171 {
172   GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (amrnbenc));
173 }
174 
175 static gboolean
gst_amrnbenc_start(GstAudioEncoder * enc)176 gst_amrnbenc_start (GstAudioEncoder * enc)
177 {
178   GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
179 
180   GST_DEBUG_OBJECT (amrnbenc, "start");
181 
182   if (!(amrnbenc->handle = Encoder_Interface_init (0)))
183     return FALSE;
184 
185   return TRUE;
186 }
187 
188 static gboolean
gst_amrnbenc_stop(GstAudioEncoder * enc)189 gst_amrnbenc_stop (GstAudioEncoder * enc)
190 {
191   GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
192 
193   GST_DEBUG_OBJECT (amrnbenc, "stop");
194 
195   Encoder_Interface_exit (amrnbenc->handle);
196 
197   return TRUE;
198 }
199 
200 static gboolean
gst_amrnbenc_set_format(GstAudioEncoder * enc,GstAudioInfo * info)201 gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
202 {
203   GstAmrnbEnc *amrnbenc;
204   GstCaps *copy;
205 
206   amrnbenc = GST_AMRNBENC (enc);
207 
208   /* parameters already parsed for us */
209   amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
210   amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
211 
212   /* we do not really accept other input, but anyway ... */
213   /* this is not wrong but will sound bad */
214   if (amrnbenc->channels != 1) {
215     g_warning ("amrnbdec is only optimized for mono channels");
216   }
217   if (amrnbenc->rate != 8000) {
218     g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
219   }
220 
221   /* create reverse caps */
222   copy = gst_caps_new_simple ("audio/AMR",
223       "channels", G_TYPE_INT, amrnbenc->channels,
224       "rate", G_TYPE_INT, amrnbenc->rate, NULL);
225 
226   gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
227   gst_caps_unref (copy);
228 
229   /* report needs to base class: hand one frame at a time */
230   gst_audio_encoder_set_frame_samples_min (enc, 160);
231   gst_audio_encoder_set_frame_samples_max (enc, 160);
232   gst_audio_encoder_set_frame_max (enc, 1);
233 
234   return TRUE;
235 }
236 
237 static GstFlowReturn
gst_amrnbenc_handle_frame(GstAudioEncoder * enc,GstBuffer * buffer)238 gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
239 {
240   GstAmrnbEnc *amrnbenc;
241   GstFlowReturn ret;
242   GstBuffer *out;
243   GstMapInfo in_map, out_map;
244   gsize out_size;
245 
246   amrnbenc = GST_AMRNBENC (enc);
247 
248   g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING);
249 
250   /* we don't deal with squeezing remnants, so simply discard those */
251   if (G_UNLIKELY (buffer == NULL)) {
252     GST_DEBUG_OBJECT (amrnbenc, "no data");
253     return GST_FLOW_OK;
254   }
255 
256   gst_buffer_map (buffer, &in_map, GST_MAP_READ);
257 
258   if (G_UNLIKELY (in_map.size < 320)) {
259     gst_buffer_unmap (buffer, &in_map);
260     GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT
261         " bytes", in_map.size);
262     return gst_audio_encoder_finish_frame (enc, NULL, -1);
263   }
264 
265   /* get output, max size is 32 */
266   out = gst_buffer_new_and_alloc (32);
267   /* AMR encoder actually writes into the source data buffers it gets */
268   /* should be able to handle that with what we are given */
269 
270   gst_buffer_map (out, &out_map, GST_MAP_WRITE);
271   /* encode */
272   out_size =
273       Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
274       (short *) in_map.data, out_map.data, 0);
275   gst_buffer_unmap (out, &out_map);
276   gst_buffer_resize (out, 0, out_size);
277   gst_buffer_unmap (buffer, &in_map);
278 
279   GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size);
280 
281   if (out_size) {
282     ret = gst_audio_encoder_finish_frame (enc, out, 160);
283   } else {
284     /* should not happen (without dtx or so at least) */
285     GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
286     gst_buffer_unref (out);
287     ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
288   }
289 
290   return ret;
291 }
292