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-rw-r--r--apps/plugins/a52towav.c217
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diff --git a/apps/plugins/a52towav.c b/apps/plugins/a52towav.c
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index f6769abd2b..0000000000
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1/***************************************************************************
2 * __________ __ ___.
3 * Open \______ \ ____ ____ | | _\_ |__ _______ ___
4 * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
5 * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
6 * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
7 * \/ \/ \/ \/ \/
8 * $Id$
9 *
10 * Copyright (C) 2005 Dave Chapman
11 *
12 * All files in this archive are subject to the GNU General Public License.
13 * See the file COPYING in the source tree root for full license agreement.
14 *
15 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
16 * KIND, either express or implied.
17 *
18 ****************************************************************************/
19
20#include "plugin.h"
21
22#if (CONFIG_HWCODEC == MASNONE)
23/* software codec platforms */
24
25#include <inttypes.h> /* Needed by a52.h */
26
27#include <codecs/liba52/config-a52.h>
28#include <codecs/liba52/a52.h>
29
30#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
31
32static struct plugin_api* rb;
33
34#ifdef WORDS_BIGENDIAN
35#define LE_S16(x) ( (uint16_t) ( ((uint16_t)(x) >> 8) | ((uint16_t)(x) << 8) ) )
36#else
37#define LE_S16(x) (x)
38#endif
39
40
41static float gain = 1;
42static a52_state_t * state;
43
44static inline int16_t convert (int32_t i)
45{
46 i >>= 15;
47 return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
48}
49
50void ao_play(file_info_struct* file_info,sample_t* samples,int flags) {
51 int i;
52 static int16_t int16_samples[256*2];
53
54 flags &= A52_CHANNEL_MASK | A52_LFE;
55
56 if (flags==A52_STEREO) {
57 for (i = 0; i < 256; i++) {
58 int16_samples[2*i] = LE_S16(convert (samples[i]));
59 int16_samples[2*i+1] = LE_S16(convert (samples[i+256]));
60 }
61 } else {
62 DEBUGF("ERROR: unsupported format: %d\n",flags);
63 }
64
65 /* FIX: Buffer the disk write to write larger amounts at one */
66 i=rb->write(file_info->outfile,int16_samples,256*2*2);
67}
68
69
70void a52_decode_data (file_info_struct* file_info, uint8_t * start, uint8_t * end)
71{
72 static uint8_t buf[3840];
73 static uint8_t * bufptr = buf;
74 static uint8_t * bufpos = buf + 7;
75
76 /*
77 * sample_rate and flags are static because this routine could
78 * exit between the a52_syncinfo() and the ao_setup(), and we want
79 * to have the same values when we get back !
80 */
81
82 static int sample_rate;
83 static int flags;
84 int bit_rate;
85 int len;
86
87 while (1) {
88 len = end - start;
89 if (!len)
90 break;
91 if (len > bufpos - bufptr)
92 len = bufpos - bufptr;
93 memcpy (bufptr, start, len);
94 bufptr += len;
95 start += len;
96 if (bufptr == bufpos) {
97 if (bufpos == buf + 7) {
98 int length;
99
100 length = a52_syncinfo (buf, &flags, &sample_rate, &bit_rate);
101 if (!length) {
102 DEBUGF("skip\n");
103 for (bufptr = buf; bufptr < buf + 6; bufptr++)
104 bufptr[0] = bufptr[1];
105 continue;
106 }
107 bufpos = buf + length;
108 } else {
109 // The following two defaults are taken from audio_out_oss.c:
110 level_t level;
111 sample_t bias;
112 int i;
113
114 /* This is the configuration for the downmixing: */
115 flags=A52_STEREO|A52_ADJUST_LEVEL|A52_LFE;
116 level=(1 << 26);
117 bias=0;
118
119 level = (level_t) (level * gain);
120
121 if (a52_frame (state, buf, &flags, &level, bias))
122 goto error;
123 file_info->frames_decoded++;
124
125 /* We assume this never changes */
126 file_info->samplerate=sample_rate;
127
128 // An A52 frame consists of 6 blocks of 256 samples
129 // So we decode and output them one block at a time
130 for (i = 0; i < 6; i++) {
131 if (a52_block (state)) {
132 goto error;
133 }
134 ao_play (file_info, a52_samples (state),flags);
135 file_info->current_sample+=256;
136 }
137 bufptr = buf;
138 bufpos = buf + 7;
139 continue;
140 error:
141 DEBUGF("error\n");
142 bufptr = buf;
143 bufpos = buf + 7;
144 }
145 }
146 }
147}
148
149
150#define BUFFER_SIZE 4096
151
152#ifdef USE_IRAM
153extern char iramcopy[];
154extern char iramstart[];
155extern char iramend[];
156#endif
157
158/* this is the plugin entry point */
159enum plugin_status plugin_start(struct plugin_api* api, void* file)
160{
161 file_info_struct file_info;
162
163 /* Generic plugin initialisation */
164
165 TEST_PLUGIN_API(api);
166 rb = api;
167
168#ifdef USE_IRAM
169 rb->memcpy(iramstart, iramcopy, iramend-iramstart);
170#endif
171
172 /* This function sets up the buffers and reads the file into RAM */
173
174 if (local_init(file,"/ac3test.wav",&file_info,api)) {
175 return PLUGIN_ERROR;
176 }
177
178 /* Intialise the A52 decoder and check for success */
179 state = a52_init (0); // Parameter is "accel"
180
181 if (state == NULL) {
182 rb->splash(HZ*2, true, "a52_init failed");
183 return PLUGIN_ERROR;
184 }
185
186 /* The main decoding loop */
187
188 file_info.start_tick=*(rb->current_tick);
189 rb->button_clear_queue();
190
191 while (file_info.curpos < file_info.filesize) {
192
193 if ((file_info.curpos+BUFFER_SIZE) < file_info.filesize) {
194 a52_decode_data(&file_info,&filebuf[file_info.curpos],&filebuf[file_info.curpos+BUFFER_SIZE]);
195 file_info.curpos+=BUFFER_SIZE;
196 } else {
197 a52_decode_data(&file_info,&filebuf[file_info.curpos],&filebuf[file_info.filesize-1]);
198 file_info.curpos=file_info.filesize;
199 }
200
201 display_status(&file_info);
202
203 if (rb->button_get(false)!=BUTTON_NONE) {
204 close_wav(&file_info);
205 return PLUGIN_OK;
206 }
207 }
208 close_wav(&file_info);
209
210 /* Cleanly close and exit */
211
212//NOT NEEDED: a52_free (state);
213
214 rb->splash(HZ*2, true, "FINISHED!");
215 return PLUGIN_OK;
216}
217#endif /* CONFIG_HWCODEC == MASNONE */
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