diff options
-rw-r--r-- | apps/codecs/SOURCES | 1 | ||||
-rw-r--r-- | apps/codecs/codecs.make | 1 | ||||
-rw-r--r-- | apps/codecs/dnet.c | 190 | ||||
-rw-r--r-- | apps/codecs/librm/rm.c | 14 | ||||
-rw-r--r-- | apps/codecs/librm/rm.h | 7 | ||||
-rw-r--r-- | apps/metadata.c | 3 | ||||
-rw-r--r-- | apps/metadata.h | 1 | ||||
-rw-r--r-- | apps/metadata/rm.c | 22 |
8 files changed, 231 insertions, 8 deletions
diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES index dc6819d272..92e4d2d254 100644 --- a/apps/codecs/SOURCES +++ b/apps/codecs/SOURCES | |||
@@ -11,6 +11,7 @@ alac.c | |||
11 | #endif | 11 | #endif |
12 | cook.c | 12 | cook.c |
13 | raac.c | 13 | raac.c |
14 | dnet.c | ||
14 | mpc.c | 15 | mpc.c |
15 | wma.c | 16 | wma.c |
16 | sid.c | 17 | sid.c |
diff --git a/apps/codecs/codecs.make b/apps/codecs/codecs.make index b327bd7c69..41f5bdccd1 100644 --- a/apps/codecs/codecs.make +++ b/apps/codecs/codecs.make | |||
@@ -77,6 +77,7 @@ $(CODECDIR)/wavpack_enc.codec: $(CODECDIR)/libwavpack.a | |||
77 | $(CODECDIR)/asap.codec : $(CODECDIR)/libasap.a | 77 | $(CODECDIR)/asap.codec : $(CODECDIR)/libasap.a |
78 | $(CODECDIR)/cook.codec : $(CODECDIR)/libcook.a $(CODECDIR)/librm.a | 78 | $(CODECDIR)/cook.codec : $(CODECDIR)/libcook.a $(CODECDIR)/librm.a |
79 | $(CODECDIR)/raac.codec : $(CODECDIR)/libfaad.a $(CODECDIR)/librm.a | 79 | $(CODECDIR)/raac.codec : $(CODECDIR)/libfaad.a $(CODECDIR)/librm.a |
80 | $(CODECDIR)/dnet.codec : $(CODECDIR)/liba52.a $(CODECDIR)/librm.a | ||
80 | 81 | ||
81 | $(CODECS): $(CODECLIB) # this must be last in codec dependency list | 82 | $(CODECS): $(CODECLIB) # this must be last in codec dependency list |
82 | 83 | ||
diff --git a/apps/codecs/dnet.c b/apps/codecs/dnet.c new file mode 100644 index 0000000000..12352ed903 --- /dev/null +++ b/apps/codecs/dnet.c | |||
@@ -0,0 +1,190 @@ | |||
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 | unsigned long samplesdone; | ||
36 | unsigned long frequency; | ||
37 | RMContext rmctx; | ||
38 | 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 | static inline void output_audio(sample_t *samples) | ||
49 | { | ||
50 | ci->yield(); | ||
51 | ci->pcmbuf_insert(&samples[0], &samples[256], 256); | ||
52 | } | ||
53 | |||
54 | static void a52_decode_data(uint8_t *start, uint8_t *end) | ||
55 | { | ||
56 | static uint8_t *bufptr = buf; | ||
57 | static uint8_t *bufpos = buf + 7; | ||
58 | /* | ||
59 | * sample_rate and flags are static because this routine could | ||
60 | * exit between the a52_syncinfo() and the ao_setup(), and we want | ||
61 | * to have the same values when we get back ! | ||
62 | */ | ||
63 | static int sample_rate; | ||
64 | static int flags; | ||
65 | int bit_rate; | ||
66 | int len; | ||
67 | |||
68 | while (1) { | ||
69 | len = end - start; | ||
70 | if (!len) | ||
71 | break; | ||
72 | if (len > bufpos - bufptr) | ||
73 | len = bufpos - bufptr; | ||
74 | memcpy(bufptr, start, len); | ||
75 | bufptr += len; | ||
76 | start += len; | ||
77 | if (bufptr == bufpos) { | ||
78 | if (bufpos == buf + 7) { | ||
79 | int length; | ||
80 | |||
81 | length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate); | ||
82 | if (!length) { | ||
83 | //DEBUGF("skip\n"); | ||
84 | for (bufptr = buf; bufptr < buf + 6; bufptr++) | ||
85 | bufptr[0] = bufptr[1]; | ||
86 | continue; | ||
87 | } | ||
88 | bufpos = buf + length; | ||
89 | } else { | ||
90 | /* Unity gain is 1 << 26, and we want to end up on 28 bits | ||
91 | of precision instead of the default 30. | ||
92 | */ | ||
93 | level_t level = 1 << 24; | ||
94 | sample_t bias = 0; | ||
95 | int i; | ||
96 | |||
97 | /* This is the configuration for the downmixing: */ | ||
98 | flags = A52_STEREO | A52_ADJUST_LEVEL; | ||
99 | |||
100 | if (a52_frame(state, buf, &flags, &level, bias)) | ||
101 | goto error; | ||
102 | a52_dynrng(state, NULL, NULL); | ||
103 | frequency = sample_rate; | ||
104 | |||
105 | /* An A52 frame consists of 6 blocks of 256 samples | ||
106 | So we decode and output them one block at a time */ | ||
107 | for (i = 0; i < 6; i++) { | ||
108 | if (a52_block(state)) | ||
109 | goto error; | ||
110 | output_audio(a52_samples(state)); | ||
111 | samplesdone += 256; | ||
112 | } | ||
113 | ci->set_elapsed(samplesdone/(frequency/1000)); | ||
114 | bufptr = buf; | ||
115 | bufpos = buf + 7; | ||
116 | continue; | ||
117 | error: | ||
118 | //logf("Error decoding A52 stream\n"); | ||
119 | bufptr = buf; | ||
120 | bufpos = buf + 7; | ||
121 | } | ||
122 | } | ||
123 | } | ||
124 | } | ||
125 | |||
126 | |||
127 | /* this is the codec entry point */ | ||
128 | enum codec_status codec_main(void) | ||
129 | { | ||
130 | size_t n; | ||
131 | uint8_t *filebuf; | ||
132 | int retval, consumed, packet_offset; | ||
133 | |||
134 | /* Generic codec initialisation */ | ||
135 | ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED); | ||
136 | ci->configure(DSP_SET_SAMPLE_DEPTH, 28); | ||
137 | |||
138 | next_track: | ||
139 | if (codec_init()) { | ||
140 | retval = CODEC_ERROR; | ||
141 | goto exit; | ||
142 | } | ||
143 | |||
144 | while (!ci->taginfo_ready) | ||
145 | ci->yield(); | ||
146 | |||
147 | ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency); | ||
148 | codec_set_replaygain(ci->id3); | ||
149 | |||
150 | /* Intializations */ | ||
151 | state = a52_init(0); | ||
152 | ci->memset(&rmctx,0,sizeof(RMContext)); | ||
153 | ci->memset(&pkt,0,sizeof(RMPacket)); | ||
154 | init_rm(&rmctx); | ||
155 | |||
156 | /* Seek to the first packet */ | ||
157 | ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE ); | ||
158 | |||
159 | /* The main decoding loop */ | ||
160 | while(pkt.timestamp < rmctx.duration) { | ||
161 | ci->yield(); | ||
162 | if (ci->stop_codec || ci->new_track) | ||
163 | break; | ||
164 | |||
165 | if (ci->seek_time) { | ||
166 | packet_offset = ci->seek_time / (((rmctx.block_align + PACKET_HEADER_SIZE)*8*1000)/rmctx.bit_rate); | ||
167 | ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE)); | ||
168 | samplesdone = A52_SAMPLESPERFRAME * packet_offset; | ||
169 | ci->seek_complete(); | ||
170 | } | ||
171 | |||
172 | filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE); | ||
173 | consumed = rm_get_packet(&filebuf, &rmctx, &pkt); | ||
174 | if(consumed < 0) { | ||
175 | DEBUGF("rm_get_packet failed\n"); | ||
176 | return CODEC_ERROR; | ||
177 | } | ||
178 | a52_decode_data(filebuf, filebuf + rmctx.block_align); | ||
179 | ci->advance_buffer(pkt.length); | ||
180 | } | ||
181 | |||
182 | retval = CODEC_OK; | ||
183 | |||
184 | if (ci->request_next_track()) | ||
185 | goto next_track; | ||
186 | |||
187 | exit: | ||
188 | a52_free(state); | ||
189 | return retval; | ||
190 | } | ||
diff --git a/apps/codecs/librm/rm.c b/apps/codecs/librm/rm.c index c802a0c5a9..b205e7f88d 100644 --- a/apps/codecs/librm/rm.c +++ b/apps/codecs/librm/rm.c | |||
@@ -27,6 +27,8 @@ | |||
27 | #include "codeclib.h" | 27 | #include "codeclib.h" |
28 | #endif | 28 | #endif |
29 | 29 | ||
30 | #define SWAP(a, b) do{uint8_t SWAP_tmp= b; b= a; a= SWAP_tmp;}while(0) | ||
31 | |||
30 | void advance_buffer(uint8_t **buf, int val) | 32 | void advance_buffer(uint8_t **buf, int val) |
31 | { | 33 | { |
32 | *buf += val; | 34 | *buf += val; |
@@ -464,7 +466,6 @@ void rm_get_packet_fd(int fd,RMContext *rmctx, RMPacket *pkt) | |||
464 | " stream = %d\n" | 466 | " stream = %d\n" |
465 | " timestmp= %d\n",pkt->version,pkt->length,pkt->stream_number,pkt->timestamp); | 467 | " timestmp= %d\n",pkt->version,pkt->length,pkt->stream_number,pkt->timestamp); |
466 | 468 | ||
467 | //getchar(); | ||
468 | if(pkt->version == 0) | 469 | if(pkt->version == 0) |
469 | { | 470 | { |
470 | read_uint8(fd,&packet_group); | 471 | read_uint8(fd,&packet_group); |
@@ -550,7 +551,16 @@ int rm_get_packet(uint8_t **src,RMContext *rmctx, RMPacket *pkt) | |||
550 | } | 551 | } |
551 | rmctx->audio_pkt_cnt = --rmctx->sub_packet_cnt; | 552 | rmctx->audio_pkt_cnt = --rmctx->sub_packet_cnt; |
552 | } | 553 | } |
553 | } | 554 | } |
555 | |||
556 | else if (rmctx->codec_type == CODEC_AC3) { | ||
557 | /* The byte order of the data is reversed from standard AC3 */ | ||
558 | for(x = 0; x < pkt->length - PACKET_HEADER_SIZE; x+=2) { | ||
559 | SWAP((*src)[0], (*src)[1]); | ||
560 | *src += 2; | ||
561 | } | ||
562 | *src -= x; | ||
563 | } | ||
554 | rmctx->audio_pkt_cnt++; | 564 | rmctx->audio_pkt_cnt++; |
555 | }while(++(rmctx->sub_packet_cnt) < h); | 565 | }while(++(rmctx->sub_packet_cnt) < h); |
556 | 566 | ||
diff --git a/apps/codecs/librm/rm.h b/apps/codecs/librm/rm.h index 12e9b18fa3..86fe5e7f1a 100644 --- a/apps/codecs/librm/rm.h +++ b/apps/codecs/librm/rm.h | |||
@@ -28,7 +28,12 @@ | |||
28 | #define DATA_HEADER_SIZE 18 | 28 | #define DATA_HEADER_SIZE 18 |
29 | #define PACKET_HEADER_SIZE 12 | 29 | #define PACKET_HEADER_SIZE 12 |
30 | 30 | ||
31 | enum codecs{CODEC_COOK, CODEC_AAC}; | 31 | enum codecs { |
32 | CODEC_COOK, | ||
33 | CODEC_AAC, | ||
34 | CODEC_AC3 | ||
35 | }; | ||
36 | |||
32 | typedef struct rm_packet | 37 | typedef struct rm_packet |
33 | { | 38 | { |
34 | uint8_t *frames[100]; /* Pointers to ordered audio frames in buffer */ | 39 | uint8_t *frames[100]; /* Pointers to ordered audio frames in buffer */ |
diff --git a/apps/metadata.c b/apps/metadata.c index f227776c0a..63547646ca 100644 --- a/apps/metadata.c +++ b/apps/metadata.c | |||
@@ -121,6 +121,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] = | |||
121 | /* AAC in RM/RA */ | 121 | /* AAC in RM/RA */ |
122 | [AFMT_RM_AAC] = | 122 | [AFMT_RM_AAC] = |
123 | AFMT_ENTRY("RAAC", "raac", NULL, "rm\0ra\0rmvb\0" ), | 123 | AFMT_ENTRY("RAAC", "raac", NULL, "rm\0ra\0rmvb\0" ), |
124 | /* AC3 in RM/RA */ | ||
125 | [AFMT_RM_AC3] = | ||
126 | AFMT_ENTRY("AC3", "dnet", NULL, "rm\0ra\0rmvb\0" ), | ||
124 | #endif | 127 | #endif |
125 | }; | 128 | }; |
126 | 129 | ||
diff --git a/apps/metadata.h b/apps/metadata.h index cefc6c3945..c43d2c2260 100644 --- a/apps/metadata.h +++ b/apps/metadata.h | |||
@@ -63,6 +63,7 @@ enum | |||
63 | AFMT_SAP, /* Amiga 8Bit SAP Format */ | 63 | AFMT_SAP, /* Amiga 8Bit SAP Format */ |
64 | AFMT_RM_COOK, /* Cook in RM/RA */ | 64 | AFMT_RM_COOK, /* Cook in RM/RA */ |
65 | AFMT_RM_AAC, /* AAC in RM/RA */ | 65 | AFMT_RM_AAC, /* AAC in RM/RA */ |
66 | AFMT_RM_AC3, /* AC3 in RM/RA */ | ||
66 | #endif | 67 | #endif |
67 | 68 | ||
68 | /* add new formats at any index above this line to have a sensible order - | 69 | /* add new formats at any index above this line to have a sensible order - |
diff --git a/apps/metadata/rm.c b/apps/metadata/rm.c index 4be0de647f..c74acef5dd 100644 --- a/apps/metadata/rm.c +++ b/apps/metadata/rm.c | |||
@@ -160,24 +160,32 @@ static inline int real_read_audio_stream_info(int fd, RMContext *rmctx) | |||
160 | skipped += 1; | 160 | skipped += 1; |
161 | } | 161 | } |
162 | 162 | ||
163 | read_uint32be(fd, &rmctx->extradata_size); | ||
164 | skipped += 4; | ||
165 | read(fd, rmctx->codec_extradata, rmctx->extradata_size); | ||
166 | skipped += rmctx->extradata_size; | ||
167 | switch(fourcc) { | 163 | switch(fourcc) { |
168 | case FOURCC('c','o','o','k'): | 164 | case FOURCC('c','o','o','k'): |
169 | rmctx->codec_type = CODEC_COOK; | 165 | rmctx->codec_type = CODEC_COOK; |
166 | read_uint32be(fd, &rmctx->extradata_size); | ||
167 | skipped += 4; | ||
168 | read(fd, rmctx->codec_extradata, rmctx->extradata_size); | ||
169 | skipped += rmctx->extradata_size; | ||
170 | break; | 170 | break; |
171 | 171 | ||
172 | case FOURCC('r','a','a','c'): | 172 | case FOURCC('r','a','a','c'): |
173 | case FOURCC('r','a','c','p'): | 173 | case FOURCC('r','a','c','p'): |
174 | rmctx->codec_type = CODEC_AAC; | 174 | rmctx->codec_type = CODEC_AAC; |
175 | read_uint32be(fd, &rmctx->extradata_size); | ||
176 | skipped += 4; | ||
177 | read(fd, rmctx->codec_extradata, rmctx->extradata_size); | ||
178 | skipped += rmctx->extradata_size; | ||
179 | break; | ||
180 | |||
181 | case FOURCC('d','n','e','t'): | ||
182 | rmctx->codec_type = CODEC_AC3; | ||
175 | break; | 183 | break; |
176 | 184 | ||
177 | default: /* Not a supported codec */ | 185 | default: /* Not a supported codec */ |
178 | return -1; | 186 | return -1; |
179 | } | 187 | } |
180 | 188 | ||
181 | DEBUGF(" flavor = %d\n",flavor); | 189 | DEBUGF(" flavor = %d\n",flavor); |
182 | DEBUGF(" coded_frame_size = %ld\n",coded_framesize); | 190 | DEBUGF(" coded_frame_size = %ld\n",coded_framesize); |
183 | DEBUGF(" sub_packet_h = %d\n",rmctx->sub_packet_h); | 191 | DEBUGF(" sub_packet_h = %d\n",rmctx->sub_packet_h); |
@@ -407,6 +415,10 @@ bool get_rm_metadata(int fd, struct mp3entry* id3) | |||
407 | case CODEC_AAC: | 415 | case CODEC_AAC: |
408 | id3->codectype = AFMT_RM_AAC; | 416 | id3->codectype = AFMT_RM_AAC; |
409 | break; | 417 | break; |
418 | |||
419 | case CODEC_AC3: | ||
420 | id3->codectype = AFMT_RM_AC3; | ||
421 | break; | ||
410 | } | 422 | } |
411 | 423 | ||
412 | id3->bitrate = rmctx->bit_rate / 1000; | 424 | id3->bitrate = rmctx->bit_rate / 1000; |