13 #include "ad_internal.h"
15 #include "cpudetect.h"
17 #include "libaf/af_format.h"
19 #include "liba52/a52.h"
20 #include "liba52/mm_accel.h"
22 static sample_t
* a52_samples
;
23 static a52_state_t a52_state
;
24 static uint32_t a52_flags
=0;
25 /** Used by a52_resample_float, it defines the mapping between liba52
26 * channels and output channels. The ith nibble from the right in the
27 * hex representation of channel_map is the index of the source
28 * channel corresponding to the ith output channel. Source channels are
29 * indexed 1-6. Silent output channels are marked by 0xf. */
30 static uint32_t channel_map
;
32 #define DRC_NO_ACTION 0
33 #define DRC_NO_COMPRESSION 1
34 #define DRC_CALLBACK 2
36 /** The output is multiplied by this var. Used for volume control */
37 static sample_t a52_level
= 1;
38 /** The value of the -a52drc switch. */
39 float a52_drc_level
= 1.0;
40 static int a52_drc_action
= DRC_NO_ACTION
;
44 static ad_info_t info
=
46 "AC3 decoding with liba52",
55 extern int audio_output_channels
;
57 int a52_fillbuff(sh_audio_t
*sh_audio
){
63 sh_audio
->a_in_buffer_len
=0;
66 while(sh_audio
->a_in_buffer_len
<8){
67 int c
=demux_getc(sh_audio
->ds
);
68 if(c
<0) return -1; /* EOF*/
69 sh_audio
->a_in_buffer
[sh_audio
->a_in_buffer_len
++]=c
;
71 if(sh_audio
->format
!=0x2000) swab(sh_audio
->a_in_buffer
,sh_audio
->a_in_buffer
,8);
72 length
= a52_syncinfo (sh_audio
->a_in_buffer
, &flags
, &sample_rate
, &bit_rate
);
73 if(length
>=7 && length
<=3840) break; /* we're done.*/
74 /* bad file => resync*/
75 if(sh_audio
->format
!=0x2000) swab(sh_audio
->a_in_buffer
,sh_audio
->a_in_buffer
,8);
76 memmove(sh_audio
->a_in_buffer
,sh_audio
->a_in_buffer
+1,7);
77 --sh_audio
->a_in_buffer_len
;
79 mp_msg(MSGT_DECAUDIO
,MSGL_DBG2
,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length
,flags
,sample_rate
,bit_rate
);
80 sh_audio
->samplerate
=sample_rate
;
81 sh_audio
->i_bps
=bit_rate
/8;
82 sh_audio
->samplesize
=sh_audio
->sample_format
==AF_FORMAT_FLOAT_NE
? 4 : 2;
83 demux_read_data(sh_audio
->ds
,sh_audio
->a_in_buffer
+8,length
-8);
84 if(sh_audio
->format
!=0x2000)
85 swab(sh_audio
->a_in_buffer
+8,sh_audio
->a_in_buffer
+8,length
-8);
87 if(crc16_block(sh_audio
->a_in_buffer
+2,length
-2)!=0)
88 mp_msg(MSGT_DECAUDIO
,MSGL_STATUS
,"a52: CRC check failed! \n");
93 /* returns: number of available channels*/
94 static int a52_printinfo(sh_audio_t
*sh_audio
){
95 int flags
, sample_rate
, bit_rate
;
98 a52_syncinfo (sh_audio
->a_in_buffer
, &flags
, &sample_rate
, &bit_rate
);
99 switch(flags
&A52_CHANNEL_MASK
){
100 case A52_CHANNEL
: mode
="channel"; channels
=2; break;
101 case A52_MONO
: mode
="mono"; channels
=1; break;
102 case A52_STEREO
: mode
="stereo"; channels
=2; break;
103 case A52_3F
: mode
="3f";channels
=3;break;
104 case A52_2F1R
: mode
="2f+1r";channels
=3;break;
105 case A52_3F1R
: mode
="3f+1r";channels
=4;break;
106 case A52_2F2R
: mode
="2f+2r";channels
=4;break;
107 case A52_3F2R
: mode
="3f+2r";channels
=5;break;
108 case A52_CHANNEL1
: mode
="channel1"; channels
=2; break;
109 case A52_CHANNEL2
: mode
="channel2"; channels
=2; break;
110 case A52_DOLBY
: mode
="dolby"; channels
=2; break;
112 mp_msg(MSGT_DECAUDIO
,MSGL_INFO
,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n",
113 channels
, (flags
&A52_LFE
)?1:0,
114 mode
, (flags
&A52_LFE
)?"+lfe":"",
115 sample_rate
, bit_rate
*0.001f
);
116 return (flags
&A52_LFE
) ? (channels
+1) : channels
;
119 sample_t
dynrng_call (sample_t c
, void *data
) {
120 // fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, (double)a52_drc_level, (double)pow((double)c, a52_drc_level));
121 return pow((double)c
, a52_drc_level
);
125 static int preinit(sh_audio_t
*sh
)
127 /* Dolby AC3 audio: */
128 /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
129 if (sh
->samplesize
< 2) sh
->samplesize
= 2;
130 sh
->audio_out_minsize
=audio_output_channels
*sh
->samplesize
*256*6;
131 sh
->audio_in_minsize
=3840;
137 * \brief Function to convert the "planar" float format used by liba52
138 * into the interleaved float format used by libaf/libao2.
139 * \param in the input buffer containing the planar samples.
140 * \param out the output buffer where the interleaved result is stored.
142 static int a52_resample_float(float *in
, int16_t *out
)
145 float *p
= (float*) out
;
146 for (i
= 0; i
!= 256; i
++) {
147 unsigned long map
= channel_map
;
149 unsigned long ch
= map
& 15;
153 *p
= in
[i
+ ((ch
-1)<<8)];
155 } while ((map
>>= 4));
157 return (int16_t*) p
- out
;
160 static int init(sh_audio_t
*sh_audio
)
162 uint32_t a52_accel
=0;
163 sample_t level
=a52_level
, bias
=384;
165 /* Dolby AC3 audio:*/
166 if(gCpuCaps
.hasSSE
) a52_accel
|=MM_ACCEL_X86_SSE
;
167 if(gCpuCaps
.hasMMX
) a52_accel
|=MM_ACCEL_X86_MMX
;
168 if(gCpuCaps
.hasMMX2
) a52_accel
|=MM_ACCEL_X86_MMXEXT
;
169 if(gCpuCaps
.has3DNow
) a52_accel
|=MM_ACCEL_X86_3DNOW
;
170 if(gCpuCaps
.has3DNowExt
) a52_accel
|=MM_ACCEL_X86_3DNOWEXT
;
171 if(gCpuCaps
.hasAltiVec
) a52_accel
|=MM_ACCEL_PPC_ALTIVEC
;
172 a52_samples
=a52_init (a52_accel
);
173 if (a52_samples
== NULL
) {
174 mp_msg(MSGT_DECAUDIO
,MSGL_ERR
,"A52 init failed\n");
177 if(a52_fillbuff(sh_audio
)<0){
178 mp_msg(MSGT_DECAUDIO
,MSGL_ERR
,"A52 sync failed\n");
183 /* Init a52 dynrng */
184 if (a52_drc_level
< 0.001) {
185 /* level == 0 --> no compression, init library without callback */
186 a52_drc_action
= DRC_NO_COMPRESSION
;
187 } else if (a52_drc_level
> 0.999) {
188 /* level == 1 --> full compression, do nothing at all (library default = full compression) */
189 a52_drc_action
= DRC_NO_ACTION
;
191 a52_drc_action
= DRC_CALLBACK
;
193 /* Library init for dynrng has to be done for each frame, see decode_audio() */
196 /* 'a52 cannot upmix' hotfix:*/
197 a52_printinfo(sh_audio
);
198 sh_audio
->channels
=audio_output_channels
;
199 while(sh_audio
->channels
>0){
200 switch(sh_audio
->channels
){
201 case 1: a52_flags
=A52_MONO
; break;
202 /* case 2: a52_flags=A52_STEREO; break;*/
203 case 2: a52_flags
=A52_DOLBY
; break;
204 /* case 3: a52_flags=A52_3F; break;*/
205 case 3: a52_flags
=A52_2F1R
; break;
206 case 4: a52_flags
=A52_2F2R
; break; /* 2+2*/
207 case 5: a52_flags
=A52_3F2R
; break;
208 case 6: a52_flags
=A52_3F2R
|A52_LFE
; break; /* 5.1*/
211 flags
=a52_flags
|A52_ADJUST_LEVEL
;
212 mp_msg(MSGT_DECAUDIO
,MSGL_V
,"A52 flags before a52_frame: 0x%X\n",flags
);
213 if (a52_frame (&a52_state
, sh_audio
->a_in_buffer
, &flags
, &level
, bias
)){
214 mp_msg(MSGT_DECAUDIO
,MSGL_ERR
,"a52: error decoding frame -> nosound\n");
217 mp_msg(MSGT_DECAUDIO
,MSGL_V
,"A52 flags after a52_frame: 0x%X\n",flags
);
218 /* frame decoded, let's init resampler:*/
220 if (sh_audio
->sample_format
== AF_FORMAT_FLOAT_NE
) {
221 if (!(flags
& A52_LFE
)) {
222 switch ((flags
<<3) | sh_audio
->channels
) {
223 case (A52_MONO
<< 3) | 1: channel_map
= 0x1; break;
224 case (A52_CHANNEL
<< 3) | 2:
225 case (A52_STEREO
<< 3) | 2:
226 case (A52_DOLBY
<< 3) | 2: channel_map
= 0x21; break;
227 case (A52_2F1R
<< 3) | 3: channel_map
= 0x321; break;
228 case (A52_2F2R
<< 3) | 4: channel_map
= 0x4321; break;
229 case (A52_3F
<< 3) | 5: channel_map
= 0x2ff31; break;
230 case (A52_3F2R
<< 3) | 5: channel_map
= 0x25431; break;
232 } else if (sh_audio
->channels
== 6) {
233 switch (flags
& ~A52_LFE
) {
234 case A52_MONO
: channel_map
= 0x12ffff; break;
237 case A52_DOLBY
: channel_map
= 0x1fff32; break;
238 case A52_3F
: channel_map
= 0x13ff42; break;
239 case A52_2F1R
: channel_map
= 0x1f4432; break;
240 case A52_2F2R
: channel_map
= 0x1f5432; break;
241 case A52_3F2R
: channel_map
= 0x136542; break;
245 a52_resample
= a52_resample_float
;
249 if(a52_resample_init(a52_accel
,flags
,sh_audio
->channels
)) break;
250 --sh_audio
->channels
; /* try to decrease no. of channels*/
252 if(sh_audio
->channels
<=0){
253 mp_msg(MSGT_DECAUDIO
,MSGL_ERR
,"a52: no resampler. try different channel setup!\n");
259 static void uninit(sh_audio_t
*sh
)
263 static int control(sh_audio_t
*sh
,int cmd
,void* arg
, ...)
267 case ADCTRL_RESYNC_STREAM
:
268 case ADCTRL_SKIP_FRAME
:
271 case ADCTRL_SET_VOLUME
: {
272 float vol
= *(float*)arg
;
273 if (vol
> 60.0) vol
= 60.0;
274 a52_level
= vol
<= -200.0 ? 0 : pow(10.0,vol
/20.0);
277 case ADCTRL_QUERY_FORMAT
:
278 if (*(int*)arg
== AF_FORMAT_S16_NE
||
279 *(int*)arg
== AF_FORMAT_FLOAT_NE
)
281 return CONTROL_FALSE
;
283 return CONTROL_UNKNOWN
;
286 static int decode_audio(sh_audio_t
*sh_audio
,unsigned char *buf
,int minlen
,int maxlen
)
288 sample_t level
=a52_level
, bias
=384;
289 int flags
=a52_flags
|A52_ADJUST_LEVEL
;
291 if (maxlen
/ sh_audio
->samplesize
/ 256 / sh_audio
->channels
< 6) {
292 mp_msg(MSGT_DECAUDIO
, MSGL_V
, "maxlen too small in decode_audio\n");
295 if (sh_audio
->sample_format
== AF_FORMAT_FLOAT_NE
)
297 if(!sh_audio
->a_in_buffer_len
)
298 if(a52_fillbuff(sh_audio
)<0) return len
; /* EOF */
299 sh_audio
->a_in_buffer_len
=0;
300 if (a52_frame (&a52_state
, sh_audio
->a_in_buffer
, &flags
, &level
, bias
)){
301 mp_msg(MSGT_DECAUDIO
,MSGL_WARN
,"a52: error decoding frame\n");
306 if (a52_drc_action
!= DRC_NO_ACTION
) {
307 if (a52_drc_action
== DRC_NO_COMPRESSION
)
308 a52_dynrng(&a52_state
, NULL
, NULL
);
310 a52_dynrng(&a52_state
, dynrng_call
, NULL
);
314 for (i
= 0; i
< 6; i
++) {
315 if (a52_block (&a52_state
, a52_samples
)){
316 mp_msg(MSGT_DECAUDIO
,MSGL_WARN
,"a52: error at resampling\n");
319 len
+=2*a52_resample(a52_samples
,(int16_t *)&buf
[len
]);
321 assert(len
<= maxlen
);