[MPlayer-cvslog] CVS: main/libao2 ao_sun.c,1.30,1.31
Alex Beregszaszi
syncmail at mplayerhq.hu
Mon Dec 27 18:49:55 CET 2004
CVS change done by Alex Beregszaszi
Update of /cvsroot/mplayer/main/libao2
In directory mail:/var2/tmp/cvs-serv3589
Modified Files:
ao_sun.c
Log Message:
10l
Index: ao_sun.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_sun.c,v
retrieving revision 1.30
retrieving revision 1.31
diff -u -r1.30 -r1.31
--- ao_sun.c 7 Dec 2004 02:24:15 -0000 1.30
+++ ao_sun.c 27 Dec 2004 17:49:52 -0000 1.31
@@ -25,7 +25,7 @@
#include "audio_out.h"
#include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
#include "mp_msg.h"
#include "help_mp.h"
@@ -69,22 +69,22 @@
// convert an OSS audio format specification into a sun audio encoding
-static int oss2sunfmt(int oss_format)
+static int af2sunfmt(int format)
{
- switch (oss_format){
- case AFMT_MU_LAW:
+ switch (format){
+ case AF_FORMAT_MU_LAW:
return AUDIO_ENCODING_ULAW;
- case AFMT_A_LAW:
+ case AF_FORMAT_A_LAW:
return AUDIO_ENCODING_ALAW;
- case AFMT_S16_BE:
- case AFMT_S16_LE:
+ case AF_FORMAT_S16_BE:
+ case AF_FORMAT_S16_LE:
return AUDIO_ENCODING_LINEAR;
#ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1...
- case AFMT_U8:
+ case AF_FORMAT_U8:
return AUDIO_ENCODING_LINEAR8;
#endif
#ifdef AUDIO_ENCODING_DVI // Missing on NetBSD...
- case AFMT_IMA_ADPCM:
+ case AF_FORMAT_IMA_ADPCM:
return AUDIO_ENCODING_DVI;
#endif
default:
@@ -465,20 +465,6 @@
enable_sample_timing = realtime_samplecounter_available(audio_dev);
}
-#define AF_FILTER_TEST 0
-#if AF_FILTER_TEST
- /* test code to force use of the audio filter modules */
- {
- char *s;
- if (s = getenv("AF_RATE"))
- rate = atoi(s);
- if (s = getenv("AF_CHANNELS"))
- channels = atoi(s);
- if (s = getenv("AF_BITS"))
- format = atoi(s) == 16 ? AFMT_S16_NE : AFMT_U8;
- }
-#endif
-
// printf("ao2: %d Hz %d chans %s [0x%X]\n",
// rate,channels,audio_out_format_name(format),format);
@@ -495,7 +481,7 @@
AUDIO_INITINFO(&info);
info.play.encoding = oss2sunfmt(ao_data.format = format);
info.play.precision =
- (format==AFMT_S16_LE || format==AFMT_S16_BE
+ (format==AF_FORMAT_S16_LE || format==AF_FORMAT_S16_BE
? AUDIO_PRECISION_16
: AUDIO_PRECISION_8);
info.play.channels = ao_data.channels = channels;
@@ -578,7 +564,7 @@
ao_data.buffersize=0;
#ifdef HAVE_AUDIO_SELECT
data = malloc(ao_data.outburst);
- memset(data, format==AFMT_U8 ? 0x80 : 0, ao_data.outburst);
+ memset(data, format==AF_FORMAT_U8 ? 0x80 : 0, ao_data.outburst);
while(ao_data.buffersize<0x40000){
fd_set rfds;
struct timeval tv;
@@ -641,7 +627,7 @@
AUDIO_INITINFO(&info);
info.play.encoding = oss2sunfmt(ao_data.format);
info.play.precision =
- (ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_S16_BE
+ (ao_data.format==AF_FORMAT_S16_LE || ao_data.format==AF_FORMAT_S16_BE
? AUDIO_PRECISION_16
: AUDIO_PRECISION_8);
info.play.channels = ao_data.channels;
@@ -710,9 +696,9 @@
// return: number of bytes played
static int play(void* data,int len,int flags){
#if WORDS_BIGENDIAN
- int native_endian = AFMT_S16_BE;
+ int native_endian = AF_FORMAT_S16_BE;
#else
- int native_endian = AFMT_S16_LE;
+ int native_endian = AF_FORMAT_S16_LE;
#endif
if (len < ao_data.outburst) return 0;
@@ -720,7 +706,7 @@
len *= ao_data.outburst;
/* 16-bit format using the 'wrong' byteorder? swap words */
- if ((ao_data.format == AFMT_S16_LE || ao_data.format == AFMT_S16_BE)
+ if ((ao_data.format == AF_FORMAT_S16_LE || ao_data.format == AF_FORMAT_S16_BE)
&& ao_data.format != native_endian) {
static void *swab_buf;
static int swab_len;
@@ -734,7 +720,7 @@
}
swab(data, swab_buf, len);
data = swab_buf;
- } else if (ao_data.format == AFMT_U8 && convert_u8_s8) {
+ } else if (ao_data.format == AF_FORMAT_U8 && convert_u8_s8) {
int i;
unsigned char *p = data;
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