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author | Sean Bartell <wingedtachikoma@gmail.com> | 2011-06-25 21:32:25 -0400 |
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committer | Nils Wallménius <nils@rockbox.org> | 2012-04-25 22:13:20 +0200 |
commit | f40bfc9267b13b54e6379dfe7539447662879d24 (patch) | |
tree | 9b20069d5e62809ff434061ad730096836f916f2 /lib/rbcodec/codecs/a52_rm.c | |
parent | a0009907de7a0107d49040d8a180f140e2eff299 (diff) | |
download | rockbox-f40bfc9267b13b54e6379dfe7539447662879d24.tar.gz rockbox-f40bfc9267b13b54e6379dfe7539447662879d24.zip |
Add codecs to librbcodec.
Change-Id: Id7f4717d51ed02d67cb9f9cb3c0ada4a81843f97
Reviewed-on: http://gerrit.rockbox.org/137
Reviewed-by: Nils Wallménius <nils@rockbox.org>
Tested-by: Nils Wallménius <nils@rockbox.org>
Diffstat (limited to 'lib/rbcodec/codecs/a52_rm.c')
-rw-r--r-- | lib/rbcodec/codecs/a52_rm.c | 227 |
1 files changed, 227 insertions, 0 deletions
diff --git a/lib/rbcodec/codecs/a52_rm.c b/lib/rbcodec/codecs/a52_rm.c new file mode 100644 index 0000000000..2db126f5b1 --- /dev/null +++ b/lib/rbcodec/codecs/a52_rm.c | |||
@@ -0,0 +1,227 @@ | |||
1 | /*************************************************************************** | ||
2 | * __________ __ ___. | ||
3 | * Open \______ \ ____ ____ | | _\_ |__ _______ ___ | ||
4 | * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / | ||
5 | * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < | ||
6 | * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ | ||
7 | * \/ \/ \/ \/ \/ | ||
8 | * $Id$ | ||
9 | * | ||
10 | * Copyright (C) 2009 Mohamed Tarek | ||
11 | * | ||
12 | * This program is free software; you can redistribute it and/or | ||
13 | * modify it under the terms of the GNU General Public License | ||
14 | * as published by the Free Software Foundation; either version 2 | ||
15 | * of the License, or (at your option) any later version. | ||
16 | * | ||
17 | * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY | ||
18 | * KIND, either express or implied. | ||
19 | * | ||
20 | ****************************************************************************/ | ||
21 | |||
22 | #include "codeclib.h" | ||
23 | #include <codecs/librm/rm.h> | ||
24 | #include <inttypes.h> /* Needed by a52.h */ | ||
25 | #include <codecs/liba52/config-a52.h> | ||
26 | #include <codecs/liba52/a52.h> | ||
27 | |||
28 | CODEC_HEADER | ||
29 | |||
30 | #define BUFFER_SIZE 4096 | ||
31 | |||
32 | #define A52_SAMPLESPERFRAME (6*256) | ||
33 | |||
34 | static a52_state_t *state; | ||
35 | static unsigned long samplesdone; | ||
36 | static unsigned long frequency; | ||
37 | static RMContext rmctx; | ||
38 | static RMPacket pkt; | ||
39 | |||
40 | static void init_rm(RMContext *rmctx) | ||
41 | { | ||
42 | memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext)); | ||
43 | } | ||
44 | |||
45 | /* used outside liba52 */ | ||
46 | static uint8_t buf[3840] IBSS_ATTR; | ||
47 | |||
48 | /* The following two functions, a52_decode_data and output_audio are taken from a52.c */ | ||
49 | static inline void output_audio(sample_t *samples) | ||
50 | { | ||
51 | ci->yield(); | ||
52 | ci->pcmbuf_insert(&samples[0], &samples[256], 256); | ||
53 | } | ||
54 | |||
55 | static void a52_decode_data(uint8_t *start, uint8_t *end) | ||
56 | { | ||
57 | static uint8_t *bufptr = buf; | ||
58 | static uint8_t *bufpos = buf + 7; | ||
59 | /* | ||
60 | * sample_rate and flags are static because this routine could | ||
61 | * exit between the a52_syncinfo() and the ao_setup(), and we want | ||
62 | * to have the same values when we get back ! | ||
63 | */ | ||
64 | static int sample_rate; | ||
65 | static int flags; | ||
66 | int bit_rate; | ||
67 | int len; | ||
68 | |||
69 | while (1) { | ||
70 | len = end - start; | ||
71 | if (!len) | ||
72 | break; | ||
73 | if (len > bufpos - bufptr) | ||
74 | len = bufpos - bufptr; | ||
75 | memcpy(bufptr, start, len); | ||
76 | bufptr += len; | ||
77 | start += len; | ||
78 | if (bufptr == bufpos) { | ||
79 | if (bufpos == buf + 7) { | ||
80 | int length; | ||
81 | |||
82 | length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate); | ||
83 | if (!length) { | ||
84 | //DEBUGF("skip\n"); | ||
85 | for (bufptr = buf; bufptr < buf + 6; bufptr++) | ||
86 | bufptr[0] = bufptr[1]; | ||
87 | continue; | ||
88 | } | ||
89 | bufpos = buf + length; | ||
90 | } else { | ||
91 | /* Unity gain is 1 << 26, and we want to end up on 28 bits | ||
92 | of precision instead of the default 30. | ||
93 | */ | ||
94 | level_t level = 1 << 24; | ||
95 | sample_t bias = 0; | ||
96 | int i; | ||
97 | |||
98 | /* This is the configuration for the downmixing: */ | ||
99 | flags = A52_STEREO | A52_ADJUST_LEVEL; | ||
100 | |||
101 | if (a52_frame(state, buf, &flags, &level, bias)) | ||
102 | goto error; | ||
103 | a52_dynrng(state, NULL, NULL); | ||
104 | frequency = sample_rate; | ||
105 | |||
106 | /* An A52 frame consists of 6 blocks of 256 samples | ||
107 | So we decode and output them one block at a time */ | ||
108 | for (i = 0; i < 6; i++) { | ||
109 | if (a52_block(state)) | ||
110 | goto error; | ||
111 | output_audio(a52_samples(state)); | ||
112 | samplesdone += 256; | ||
113 | } | ||
114 | ci->set_elapsed(samplesdone/(frequency/1000)); | ||
115 | bufptr = buf; | ||
116 | bufpos = buf + 7; | ||
117 | continue; | ||
118 | error: | ||
119 | //logf("Error decoding A52 stream\n"); | ||
120 | bufptr = buf; | ||
121 | bufpos = buf + 7; | ||
122 | } | ||
123 | } | ||
124 | } | ||
125 | } | ||
126 | |||
127 | /* this is the codec entry point */ | ||
128 | enum codec_status codec_main(enum codec_entry_call_reason reason) | ||
129 | { | ||
130 | if (reason == CODEC_LOAD) { | ||
131 | /* Generic codec initialisation */ | ||
132 | ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED); | ||
133 | ci->configure(DSP_SET_SAMPLE_DEPTH, 28); | ||
134 | } | ||
135 | else if (reason == CODEC_UNLOAD) { | ||
136 | if (state) | ||
137 | a52_free(state); | ||
138 | } | ||
139 | |||
140 | return CODEC_OK; | ||
141 | } | ||
142 | |||
143 | /* this is called for each file to process */ | ||
144 | enum codec_status codec_run(void) | ||
145 | { | ||
146 | size_t n; | ||
147 | uint8_t *filebuf; | ||
148 | int consumed, packet_offset; | ||
149 | int playback_on = -1; | ||
150 | size_t resume_offset; | ||
151 | intptr_t param; | ||
152 | enum codec_command_action action = CODEC_ACTION_NULL; | ||
153 | |||
154 | if (codec_init()) { | ||
155 | return CODEC_ERROR; | ||
156 | } | ||
157 | |||
158 | resume_offset = ci->id3->offset; | ||
159 | |||
160 | ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency); | ||
161 | codec_set_replaygain(ci->id3); | ||
162 | |||
163 | ci->seek_buffer(ci->id3->first_frame_offset); | ||
164 | |||
165 | /* Intializations */ | ||
166 | state = a52_init(0); | ||
167 | ci->memset(&rmctx,0,sizeof(RMContext)); | ||
168 | ci->memset(&pkt,0,sizeof(RMPacket)); | ||
169 | init_rm(&rmctx); | ||
170 | |||
171 | samplesdone = 0; | ||
172 | |||
173 | /* check for a mid-track resume and force a seek time accordingly */ | ||
174 | if(resume_offset > rmctx.data_offset + DATA_HEADER_SIZE) { | ||
175 | resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE; | ||
176 | /* put number of subpackets to skip in resume_offset */ | ||
177 | resume_offset /= (rmctx.block_align + PACKET_HEADER_SIZE); | ||
178 | param = (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate); | ||
179 | action = CODEC_ACTION_SEEK_TIME; | ||
180 | } | ||
181 | else { | ||
182 | /* Seek to the first packet */ | ||
183 | ci->set_elapsed(0); | ||
184 | ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE ); | ||
185 | } | ||
186 | |||
187 | /* The main decoding loop */ | ||
188 | while((unsigned)rmctx.audio_pkt_cnt < rmctx.nb_packets) { | ||
189 | if (action == CODEC_ACTION_NULL) | ||
190 | action = ci->get_command(¶m); | ||
191 | |||
192 | if (action == CODEC_ACTION_HALT) | ||
193 | break; | ||
194 | |||
195 | if (action == CODEC_ACTION_SEEK_TIME) { | ||
196 | packet_offset = param / ((rmctx.block_align*8*1000)/rmctx.bit_rate); | ||
197 | ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + | ||
198 | packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE)); | ||
199 | rmctx.audio_pkt_cnt = packet_offset; | ||
200 | samplesdone = (rmctx.sample_rate/1000 * param); | ||
201 | ci->set_elapsed(samplesdone/(frequency/1000)); | ||
202 | ci->seek_complete(); | ||
203 | } | ||
204 | |||
205 | action = CODEC_ACTION_NULL; | ||
206 | |||
207 | filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE); | ||
208 | consumed = rm_get_packet(&filebuf, &rmctx, &pkt); | ||
209 | |||
210 | if(consumed < 0 && playback_on != 0) { | ||
211 | if(playback_on == -1) { | ||
212 | /* Error only if packet-parsing failed and playback hadn't started */ | ||
213 | DEBUGF("rm_get_packet failed\n"); | ||
214 | return CODEC_ERROR; | ||
215 | } | ||
216 | else { | ||
217 | break; | ||
218 | } | ||
219 | } | ||
220 | |||
221 | playback_on = 1; | ||
222 | a52_decode_data(filebuf, filebuf + rmctx.block_align); | ||
223 | ci->advance_buffer(pkt.length); | ||
224 | } | ||
225 | |||
226 | return CODEC_OK; | ||
227 | } | ||