Patch doesn't SVN add . . .
[kugel-rb.git] / apps / codecs / aac.c
blob60460355dad0653f980516cf0e2a18c3fc6f4909
1 /***************************************************************************
2 * __________ __ ___.
3 * Open \______ \ ____ ____ | | _\_ |__ _______ ___
4 * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
5 * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
6 * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
7 * \/ \/ \/ \/ \/
8 * $Id$
10 * Copyright (C) 2005 Dave Chapman
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.
17 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
18 * KIND, either express or implied.
20 ****************************************************************************/
22 #include "codeclib.h"
23 #include "libm4a/m4a.h"
24 #include "libfaad/common.h"
25 #include "libfaad/structs.h"
26 #include "libfaad/decoder.h"
28 CODEC_HEADER
30 /* this is the codec entry point */
31 enum codec_status codec_main(void)
33 /* Note that when dealing with QuickTime/MPEG4 files, terminology is
34 * a bit confusing. Files with sound are split up in chunks, where
35 * each chunk contains one or more samples. Each sample in turn
36 * contains a number of "sound samples" (the kind you refer to with
37 * the sampling frequency).
39 size_t n;
40 static demux_res_t demux_res;
41 stream_t input_stream;
42 uint32_t sound_samples_done;
43 uint32_t elapsed_time;
44 uint32_t sample_duration;
45 uint32_t sample_byte_size;
46 int file_offset;
47 int framelength;
48 int lead_trim = 0;
49 unsigned int i;
50 unsigned char* buffer;
51 static NeAACDecFrameInfo frame_info;
52 NeAACDecHandle decoder;
53 int err;
54 uint32_t s = 0;
55 unsigned char c = 0;
57 /* Generic codec initialisation */
58 ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
59 ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
61 next_track:
62 err = CODEC_OK;
64 if (codec_init()) {
65 LOGF("FAAD: Codec init error\n");
66 err = CODEC_ERROR;
67 goto exit;
70 while (!*ci->taginfo_ready && !ci->stop_codec)
71 ci->sleep(1);
73 sound_samples_done = ci->id3->offset;
75 ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
76 codec_set_replaygain(ci->id3);
78 stream_create(&input_stream,ci);
80 /* if qtmovie_read returns successfully, the stream is up to
81 * the movie data, which can be used directly by the decoder */
82 if (!qtmovie_read(&input_stream, &demux_res)) {
83 LOGF("FAAD: File init error\n");
84 err = CODEC_ERROR;
85 goto done;
88 /* initialise the sound converter */
89 decoder = NeAACDecOpen();
91 if (!decoder) {
92 LOGF("FAAD: Decode open error\n");
93 err = CODEC_ERROR;
94 goto done;
97 NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
98 conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
99 NeAACDecSetConfiguration(decoder, conf);
101 err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
102 if (err) {
103 LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
104 err = CODEC_ERROR;
105 goto done;
108 ci->id3->frequency = s;
110 i = 0;
112 if (sound_samples_done > 0) {
113 if (alac_seek_raw(&demux_res, &input_stream, sound_samples_done,
114 &sound_samples_done, (int*) &i)) {
115 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
116 ci->set_elapsed(elapsed_time);
117 } else {
118 sound_samples_done = 0;
122 if (i == 0)
124 lead_trim = ci->id3->lead_trim;
127 /* The main decoding loop */
128 while (i < demux_res.num_sample_byte_sizes) {
129 ci->yield();
131 if (ci->stop_codec || ci->new_track) {
132 break;
135 /* Deal with any pending seek requests */
136 if (ci->seek_time) {
137 if (alac_seek(&demux_res, &input_stream,
138 ((ci->seek_time-1)/10)*(ci->id3->frequency/100),
139 &sound_samples_done, (int*) &i)) {
140 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
141 ci->set_elapsed(elapsed_time);
143 if (i == 0)
145 lead_trim = ci->id3->lead_trim;
148 ci->seek_complete();
151 /* Lookup the length (in samples and bytes) of block i */
152 if (!get_sample_info(&demux_res, i, &sample_duration,
153 &sample_byte_size)) {
154 LOGF("AAC: get_sample_info error\n");
155 err = CODEC_ERROR;
156 goto done;
159 /* There can be gaps between chunks, so skip ahead if needed. It
160 * doesn't seem to happen much, but it probably means that a
161 * "proper" file can have chunks out of order. Why one would want
162 * that an good question (but files with gaps do exist, so who
163 * knows?), so we don't support that - for now, at least.
165 file_offset = get_sample_offset(&demux_res, i);
167 if (file_offset > ci->curpos)
169 ci->advance_buffer(file_offset - ci->curpos);
171 else if (file_offset == 0)
173 LOGF("AAC: get_sample_offset error\n");
174 err = CODEC_ERROR;
175 goto done;
178 /* Request the required number of bytes from the input buffer */
179 buffer=ci->request_buffer(&n,sample_byte_size);
181 /* Decode one block - returned samples will be host-endian */
182 NeAACDecDecode(decoder, &frame_info, buffer, n);
183 /* Ignore return value, we access samples in the decoder struct
184 * directly.
186 if (frame_info.error > 0) {
187 LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
188 err = CODEC_ERROR;
189 goto done;
192 /* Advance codec buffer */
193 ci->advance_buffer(n);
195 /* Output the audio */
196 ci->yield();
198 framelength = (frame_info.samples >> 1) - lead_trim;
200 if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
202 /* Currently limited to at most one frame of tail_trim.
203 * Seems to be enough.
205 if (ci->id3->tail_trim == 0
206 && sample_duration < (frame_info.samples >> 1))
208 /* Subtract lead_trim just in case we decode a file with
209 * only one audio frame with actual data.
211 framelength = sample_duration - lead_trim;
213 else
215 framelength -= ci->id3->tail_trim;
219 if (framelength > 0)
221 ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
222 &decoder->time_out[1][lead_trim],
223 framelength);
226 if (lead_trim > 0)
228 /* frame_info.samples can be 0 for the first frame */
229 lead_trim -= (i > 0 || frame_info.samples)
230 ? (frame_info.samples >> 1) : sample_duration;
232 if (lead_trim < 0 || ci->id3->lead_trim == 0)
234 lead_trim = 0;
238 /* Update the elapsed-time indicator */
239 sound_samples_done += sample_duration;
240 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
241 ci->set_elapsed(elapsed_time);
243 /* Keep track of current position - for resuming */
244 ci->set_offset(elapsed_time);
246 i++;
249 err = CODEC_OK;
251 done:
252 LOGF("AAC: Decoded %lu samples\n", sound_samples_done);
254 if (ci->request_next_track())
255 goto next_track;
257 exit:
258 return err;