[MPlayer-cvslog] CVS: main/libmpcodecs ad_liba52.c,1.14,1.15
Zoltan Hidvegi CVS
syncmail at mplayerhq.hu
Fri Feb 25 12:07:23 CET 2005
CVS change done by Zoltan Hidvegi CVS
Update of /cvsroot/mplayer/main/libmpcodecs
In directory mail:/var2/tmp/cvs-serv17355/libmpcodecs
Modified Files:
ad_liba52.c
Log Message:
Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
supported output formats. Add ADCTRL_SET_VOLUME (not yet used).
Index: ad_liba52.c
===================================================================
RCS file: /cvsroot/mplayer/main/libmpcodecs/ad_liba52.c,v
retrieving revision 1.14
retrieving revision 1.15
diff -u -r1.14 -r1.15
--- ad_liba52.c 21 Sep 2004 20:34:46 -0000 1.14
+++ ad_liba52.c 25 Feb 2005 11:07:21 -0000 1.15
@@ -2,6 +2,7 @@
#include <stdlib.h>
#include <unistd.h>
#include <math.h>
+#include <assert.h>
#include "config.h"
#ifdef USE_LIBA52
@@ -13,17 +14,28 @@
#include "cpudetect.h"
+#include "../libaf/af_format.h"
+
#include "../liba52/a52.h"
#include "../liba52/mm_accel.h"
static sample_t * a52_samples;
static a52_state_t a52_state;
static uint32_t a52_flags=0;
+/** Used by a52_resample_float, it defines the mapping between liba52
+ * channels and output channels. The ith nibble from the right in the
+ * hex representation of channel_map is the index of the source
+ * channel corresponding to the ith output channel. Source channels are
+ * indexed 1-6. Silent output channels are marked by 0xf. */
+static uint32_t channel_map;
#define DRC_NO_ACTION 0
#define DRC_NO_COMPRESSION 1
#define DRC_CALLBACK 2
+/** The output is multiplied by this var. Used for volume control */
+static sample_t a52_level = 1;
+/** The value of the -a52drc switch. */
float a52_drc_level = 1.0;
static int a52_drc_action = DRC_NO_ACTION;
@@ -67,7 +79,7 @@
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate);
sh_audio->samplerate=sample_rate;
sh_audio->i_bps=bit_rate/8;
- sh_audio->samplesize=2;
+ sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2;
demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8);
if(sh_audio->format!=0x2000)
swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
@@ -114,15 +126,40 @@
{
/* Dolby AC3 audio: */
/* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
- sh->audio_out_minsize=audio_output_channels*2*256*6;
+ sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6;
sh->audio_in_minsize=3840;
+ a52_level = 1.0;
return 1;
}
+/**
+ * \brief Function to convert the "planar" float format used by liba52
+ * into the interleaved float format used by libaf/libao2.
+ * \param in the input buffer containing the planar samples.
+ * \param out the output buffer where the interleaved result is stored.
+ */
+static int a52_resample_float(float *in, int16_t *out)
+{
+ unsigned long i;
+ float *p = (float*) out;
+ for (i = 0; i != 256; i++) {
+ unsigned long map = channel_map;
+ do {
+ unsigned long ch = map & 15;
+ if (ch == 15)
+ *p = 0;
+ else
+ *p = in[i + ((ch-1)<<8)];
+ p++;
+ } while ((map >>= 4));
+ }
+ return (int16_t*) p - out;
+}
+
static int init(sh_audio_t *sh_audio)
{
uint32_t a52_accel=0;
- sample_t level=1, bias=384;
+ sample_t level=a52_level, bias=384;
int flags=0;
/* Dolby AC3 audio:*/
if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
@@ -178,6 +215,36 @@
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
/* frame decoded, let's init resampler:*/
+ channel_map = 0;
+ if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
+ if (!(flags & A52_LFE)) {
+ switch ((flags<<3) | sh_audio->channels) {
+ case (A52_MONO << 3) | 1: channel_map = 0x1; break;
+ case (A52_CHANNEL << 3) | 2:
+ case (A52_STEREO << 3) | 2:
+ case (A52_DOLBY << 3) | 2: channel_map = 0x21; break;
+ case (A52_2F1R << 3) | 3: channel_map = 0x321; break;
+ case (A52_2F2R << 3) | 4: channel_map = 0x4321; break;
+ case (A52_3F << 3) | 5: channel_map = 0x2ff31; break;
+ case (A52_3F2R << 3) | 5: channel_map = 0x25431; break;
+ }
+ } else if (sh_audio->channels == 6) {
+ switch (flags & ~A52_LFE) {
+ case A52_MONO : channel_map = 0x12ffff; break;
+ case A52_CHANNEL:
+ case A52_STEREO :
+ case A52_DOLBY : channel_map = 0x1fff32; break;
+ case A52_3F : channel_map = 0x13ff42; break;
+ case A52_2F1R : channel_map = 0x1f4432; break;
+ case A52_2F2R : channel_map = 0x1f5432; break;
+ case A52_3F2R : channel_map = 0x136542; break;
+ }
+ }
+ if (channel_map) {
+ a52_resample = a52_resample_float;
+ break;
+ }
+ } else
if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
--sh_audio->channels; /* try to decrease no. of channels*/
}
@@ -199,15 +266,28 @@
case ADCTRL_SKIP_FRAME:
a52_fillbuff(sh); break; // skip AC3 frame
return CONTROL_TRUE;
+ case ADCTRL_SET_VOLUME: {
+ float vol = *(float*)arg;
+ if (vol > 60.0) vol = 60.0;
+ a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0);
+ return CONTROL_TRUE;
+ }
+ case ADCTRL_QUERY_FORMAT:
+ if (*(int*)arg == AF_FORMAT_S16_NE ||
+ *(int*)arg == AF_FORMAT_FLOAT_NE)
+ return CONTROL_TRUE;
+ return CONTROL_FALSE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
- sample_t level=1, bias=384;
+ sample_t level=a52_level, bias=384;
int flags=a52_flags|A52_ADJUST_LEVEL;
int i,len=-1;
+ if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
+ bias = 0;
if(!sh_audio->a_in_buffer_len)
if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
sh_audio->a_in_buffer_len=0;
@@ -232,6 +312,7 @@
}
len+=2*a52_resample(a52_samples,(int16_t *)&buf[len]);
}
+ assert(len <= maxlen);
return len;
}
#endif
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