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Code Issues Releases Wiki Activity

Cleanup, removing internal conversions. Testing welcome.

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git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14410 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
reimar 2005-01-06 22:56:54 +00:00
parent 6e2c3b96e5
commit 40b127ae8d
1 changed files with 15 additions and 35 deletions
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50
libao2/ao_sun.c
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@ -57,7 +57,6 @@ static int queued_bursts = 0;
static int queued_samples = 0; static int queued_samples = 0;
static int bytes_per_sample = 0; static int bytes_per_sample = 0;
static int byte_per_sec = 0; static int byte_per_sec = 0;
static int convert_u8_s8;
static int audio_fd = -1; static int audio_fd = -1;
static enum { static enum {
RTSC_UNKNOWN = 0, RTSC_UNKNOWN = 0,
@ -76,13 +75,14 @@ static int af2sunfmt(int format)
return AUDIO_ENCODING_ULAW; return AUDIO_ENCODING_ULAW;
case AF_FORMAT_A_LAW: case AF_FORMAT_A_LAW:
return AUDIO_ENCODING_ALAW; return AUDIO_ENCODING_ALAW;
case AF_FORMAT_S16_BE: case AF_FORMAT_S16_NE:
case AF_FORMAT_S16_LE:
return AUDIO_ENCODING_LINEAR; return AUDIO_ENCODING_LINEAR;
#ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1... #ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1...
case AF_FORMAT_U8: case AF_FORMAT_U8:
return AUDIO_ENCODING_LINEAR8; return AUDIO_ENCODING_LINEAR8;
#endif #endif
case AF_FORMAT_S8:
return AUDIO_ENCODING_LINEAR;
#ifdef AUDIO_ENCODING_DVI // Missing on NetBSD... #ifdef AUDIO_ENCODING_DVI // Missing on NetBSD...
case AF_FORMAT_IMA_ADPCM: case AF_FORMAT_IMA_ADPCM:
return AUDIO_ENCODING_DVI; return AUDIO_ENCODING_DVI;
@ -457,7 +457,7 @@ static int init(int rate,int channels,int format,int flags){
audio_info_t info; audio_info_t info;
int pass; int pass;
int ok; int ok;
char buf[128]; int convert_u8_s8;
setup_device_paths(); setup_device_paths();
@ -477,12 +477,15 @@ static int init(int rate,int channels,int format,int flags){
ioctl(audio_fd, AUDIO_DRAIN, 0); ioctl(audio_fd, AUDIO_DRAIN, 0);
if (af2sunfmt(format) == AUDIO_ENCODING_NONE)
format = AF_FORMAT_S16_NE;
for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */ for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */
AUDIO_INITINFO(&info); AUDIO_INITINFO(&info);
info.play.encoding = af2sunfmt(ao_data.format = format); info.play.encoding = af2sunfmt(ao_data.format = format);
info.play.precision = info.play.precision =
(format==AF_FORMAT_S16_LE || format==AF_FORMAT_S16_BE (format==AF_FORMAT_S16_NE
? AUDIO_PRECISION_16 ? AUDIO_PRECISION_16
: AUDIO_PRECISION_8); : AUDIO_PRECISION_8);
info.play.channels = ao_data.channels = channels; info.play.channels = ao_data.channels = channels;
@ -498,7 +501,9 @@ static int init(int rate,int channels,int format,int flags){
* Try S8, and if it works, use our own U8->S8 conversion before * Try S8, and if it works, use our own U8->S8 conversion before
* sending the samples to the sound driver. * sending the samples to the sound driver.
*/ */
#ifdef AUDIO_ENCODING_LINEAR8
if (info.play.encoding != AUDIO_ENCODING_LINEAR8) if (info.play.encoding != AUDIO_ENCODING_LINEAR8)
#endif
continue; continue;
info.play.encoding = AUDIO_ENCODING_LINEAR; info.play.encoding = AUDIO_ENCODING_LINEAR;
convert_u8_s8 = 1; convert_u8_s8 = 1;
@ -545,11 +550,15 @@ static int init(int rate,int channels,int format,int flags){
} }
if (!ok) { if (!ok) {
char buf[128];
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate, mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate,
channels, af_fmt2str(format, buf, 128), rate); channels, af_fmt2str(format, buf, 128), rate);
return 0; return 0;
} }
if (convert_u8_s8)
ao_data.format = AF_FORMAT_S8;
bytes_per_sample = channels * info.play.precision / 8; bytes_per_sample = channels * info.play.precision / 8;
ao_data.bps = byte_per_sec = bytes_per_sample * ao_data.samplerate; ao_data.bps = byte_per_sec = bytes_per_sample * ao_data.samplerate;
ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192; ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192;
@ -628,7 +637,7 @@ static void reset(){
AUDIO_INITINFO(&info); AUDIO_INITINFO(&info);
info.play.encoding = af2sunfmt(ao_data.format); info.play.encoding = af2sunfmt(ao_data.format);
info.play.precision = info.play.precision =
(ao_data.format==AF_FORMAT_S16_LE || ao_data.format==AF_FORMAT_S16_BE (ao_data.format==AF_FORMAT_S16_NE
? AUDIO_PRECISION_16 ? AUDIO_PRECISION_16
: AUDIO_PRECISION_8); : AUDIO_PRECISION_8);
info.play.channels = ao_data.channels; info.play.channels = ao_data.channels;
@ -696,39 +705,10 @@ static int get_space(){
// it should round it down to outburst*n // it should round it down to outburst*n
// return: number of bytes played // return: number of bytes played
static int play(void* data,int len,int flags){ static int play(void* data,int len,int flags){
#if WORDS_BIGENDIAN
int native_endian = AF_FORMAT_S16_BE;
#else
int native_endian = AF_FORMAT_S16_LE;
#endif
if (len < ao_data.outburst) return 0; if (len < ao_data.outburst) return 0;
len /= ao_data.outburst; len /= ao_data.outburst;
len *= ao_data.outburst; len *= ao_data.outburst;
/* 16-bit format using the 'wrong' byteorder? swap words */
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;
if (len > swab_len) {
if (swab_buf)
swab_buf = realloc(swab_buf, len);
else
swab_buf = malloc(len);
swab_len = len;
if (swab_buf == NULL) return 0;
}
swab(data, swab_buf, len);
data = swab_buf;
} else if (ao_data.format == AF_FORMAT_U8 && convert_u8_s8) {
int i;
unsigned char *p = data;
for (i = 0, p = data; i < len; i++, p++)
*p ^= 0x80;
}
len = write(audio_fd, data, len); len = write(audio_fd, data, len);
if(len > 0) { if(len > 0) {
queued_samples += len / bytes_per_sample; queued_samples += len / bytes_per_sample;