Remove trailing whitespace from most files
[mplayer/glamo.git] / libmpcodecs / ad_faad.c
blob905decd97749db494fe8d8b5f1fe0c5860d4f288
1 /*
2 * MPlayer AAC decoder using libfaad2
4 * Copyright (C) 2002 Felix Buenemann <atmosfear at users.sourceforge.net>
6 * This file is part of MPlayer.
8 * MPlayer is free software; you can redistribute it and/or modify
9 * it under the terms of the GNU General Public License as published by
10 * the Free Software Foundation; either version 2 of the License, or
11 * (at your option) any later version.
13 * MPlayer is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16 * GNU General Public License for more details.
18 * You should have received a copy of the GNU General Public License along
19 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
23 #include <stdio.h>
24 #include <stdlib.h>
25 #include <unistd.h>
27 #include "config.h"
28 #include "ad_internal.h"
29 #include "libaf/reorder_ch.h"
31 static const ad_info_t info =
33 "AAC (MPEG2/4 Advanced Audio Coding)",
34 "faad",
35 "Felix Buenemann",
36 "faad2",
37 "uses libfaad2"
40 LIBAD_EXTERN(faad)
42 #ifndef CONFIG_FAAD_INTERNAL
43 #include <faad.h>
44 #else
45 #include "libfaad2/faad.h"
46 #endif
48 /* configure maximum supported channels, *
49 * this is theoretically max. 64 chans */
50 #define FAAD_MAX_CHANNELS 6
51 #define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS)
53 //#define AAC_DUMP_COMPRESSED
55 static faacDecHandle faac_hdec;
56 static faacDecFrameInfo faac_finfo;
58 static int preinit(sh_audio_t *sh)
60 sh->audio_out_minsize=8192*FAAD_MAX_CHANNELS;
61 sh->audio_in_minsize=FAAD_BUFFLEN;
62 return 1;
65 static int aac_probe(unsigned char *buffer, int len)
67 int i = 0, pos = 0;
68 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: %d bytes\n", len);
69 while(i <= len-4) {
70 if(
71 ((buffer[i] == 0xff) && ((buffer[i+1] & 0xf6) == 0xf0)) ||
72 (buffer[i] == 'A' && buffer[i+1] == 'D' && buffer[i+2] == 'I' && buffer[i+3] == 'F')
73 ) {
74 pos = i;
75 break;
77 mp_msg(MSGT_DECAUDIO,MSGL_V, "AUDIO PAYLOAD: %x %x %x %x\n", buffer[i], buffer[i+1], buffer[i+2], buffer[i+3]);
78 i++;
80 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: ret %d\n", pos);
81 return pos;
84 static int init(sh_audio_t *sh)
86 unsigned long faac_samplerate;
87 unsigned char faac_channels;
88 int faac_init, pos = 0;
89 faac_hdec = faacDecOpen();
91 // If we don't get the ES descriptor, try manual config
92 if(!sh->codecdata_len && sh->wf) {
93 sh->codecdata_len = sh->wf->cbSize;
94 sh->codecdata = malloc(sh->codecdata_len);
95 memcpy(sh->codecdata, sh->wf+1, sh->codecdata_len);
96 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
98 if(!sh->codecdata_len) {
99 #if 1
100 faacDecConfigurationPtr faac_conf;
101 /* Set the default object type and samplerate */
102 /* This is useful for RAW AAC files */
103 faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
104 if(sh->samplerate)
105 faac_conf->defSampleRate = sh->samplerate;
106 /* XXX: FAAD support FLOAT output, how do we handle
107 * that (FAAD_FMT_FLOAT)? ::atmos
109 if (audio_output_channels <= 2) faac_conf->downMatrix = 1;
110 switch(sh->samplesize){
111 case 1: // 8Bit
112 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
113 default:
114 sh->samplesize=2;
115 case 2: // 16Bit
116 faac_conf->outputFormat = FAAD_FMT_16BIT;
117 break;
118 case 3: // 24Bit
119 faac_conf->outputFormat = FAAD_FMT_24BIT;
120 break;
121 case 4: // 32Bit
122 faac_conf->outputFormat = FAAD_FMT_32BIT;
123 break;
125 //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
127 faacDecSetConfiguration(faac_hdec, faac_conf);
128 #endif
130 sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
131 pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
132 if(pos) {
133 sh->a_in_buffer_len -= pos;
134 memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
135 sh->a_in_buffer_len +=
136 demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]),
137 sh->a_in_buffer_size - sh->a_in_buffer_len);
138 pos = 0;
141 /* init the codec */
142 faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
143 sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
145 sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
146 // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi
148 } else { // We have ES DS in codecdata
149 faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
150 if (audio_output_channels <= 2) {
151 faac_conf->downMatrix = 1;
152 faacDecSetConfiguration(faac_hdec, faac_conf);
155 /*int i;
156 for(i = 0; i < sh_audio->codecdata_len; i++)
157 printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
159 faac_init = faacDecInit2(faac_hdec, sh->codecdata,
160 sh->codecdata_len, &faac_samplerate, &faac_channels);
162 if(faac_init < 0) {
163 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
164 faacDecClose(faac_hdec);
165 // XXX: free a_in_buffer here or in uninit?
166 return 0;
167 } else {
168 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
169 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz channels: %d\n", faac_samplerate, faac_channels);
170 sh->channels = faac_channels;
171 if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1;
172 sh->samplerate = faac_samplerate;
173 sh->samplesize=2;
174 //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
175 if(!sh->i_bps) {
176 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
177 sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
178 } else
179 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
181 return 1;
184 static void uninit(sh_audio_t *sh)
186 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n");
187 faacDecClose(faac_hdec);
190 static int aac_sync(sh_audio_t *sh)
192 int pos = 0;
193 if(!sh->codecdata_len) {
194 if(sh->a_in_buffer_len < sh->a_in_buffer_size){
195 sh->a_in_buffer_len +=
196 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
197 sh->a_in_buffer_size - sh->a_in_buffer_len);
199 pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
200 if(pos) {
201 sh->a_in_buffer_len -= pos;
202 memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
203 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos);
206 return pos;
209 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
211 switch(cmd)
213 case ADCTRL_RESYNC_STREAM:
214 aac_sync(sh);
215 return CONTROL_TRUE;
216 #if 0
217 case ADCTRL_SKIP_FRAME:
218 return CONTROL_TRUE;
219 #endif
221 return CONTROL_UNKNOWN;
224 #define MAX_FAAD_ERRORS 10
225 static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
227 int len = 0, last_dec_len = 1, errors = 0;
228 // int j = 0;
229 void *faac_sample_buffer;
231 while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) {
233 /* update buffer for raw aac streams: */
234 if(!sh->codecdata_len)
235 if(sh->a_in_buffer_len < sh->a_in_buffer_size){
236 sh->a_in_buffer_len +=
237 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
238 sh->a_in_buffer_size - sh->a_in_buffer_len);
241 #ifdef DUMP_AAC_COMPRESSED
242 {int i;
243 for (i = 0; i < 16; i++)
244 printf ("%02X ", sh->a_in_buffer[i]);
245 printf ("\n");}
246 #endif
248 if(!sh->codecdata_len){
249 // raw aac stream:
250 do {
251 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer, sh->a_in_buffer_len);
253 /* update buffer index after faacDecDecode */
254 if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) {
255 sh->a_in_buffer_len=0;
256 } else {
257 sh->a_in_buffer_len-=faac_finfo.bytesconsumed;
258 memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len);
261 if(faac_finfo.error > 0) {
262 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n",
263 faacDecGetErrorMessage(faac_finfo.error));
264 if (sh->a_in_buffer_len <= 0) {
265 errors = MAX_FAAD_ERRORS;
266 break;
268 sh->a_in_buffer_len--;
269 memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len);
270 aac_sync(sh);
271 errors++;
272 } else
273 break;
274 } while(errors < MAX_FAAD_ERRORS);
275 } else {
276 // packetized (.mp4) aac stream:
277 unsigned char* bufptr=NULL;
278 double pts;
279 int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
280 if(buflen<=0) break;
281 if (pts != MP_NOPTS_VALUE) {
282 sh->pts = pts;
283 sh->pts_bytes = 0;
285 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen);
287 //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]);
289 if(faac_finfo.error > 0) {
290 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
291 faacDecGetErrorMessage(faac_finfo.error));
292 } else if (faac_finfo.samples == 0) {
293 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
294 } else {
295 /* XXX: samples already multiplied by channels! */
296 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%ld Bytes)!\n",
297 sh->samplesize*faac_finfo.samples);
299 if (sh->channels >= 5)
300 reorder_channel_copy_nch(faac_sample_buffer,
301 AF_CHANNEL_LAYOUT_AAC_DEFAULT,
302 buf+len, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
303 sh->channels,
304 faac_finfo.samples, sh->samplesize);
305 else
306 memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
307 last_dec_len = sh->samplesize*faac_finfo.samples;
308 len += last_dec_len;
309 sh->pts_bytes += last_dec_len;
310 //printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed);
313 return len;