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-rw-r--r--libao2/ao_alsa.c10
-rw-r--r--libao2/ao_alsa5.c6
-rw-r--r--libao2/ao_dsound.c10
-rw-r--r--libao2/ao_sgi.c5
-rw-r--r--libao2/ao_sun.c7
-rw-r--r--libao2/ao_win32.c3
6 files changed, 21 insertions, 20 deletions
diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
index 539fd0fa7c..db23f1b532 100644
--- a/libao2/ao_alsa.c
+++ b/libao2/ao_alsa.c
@@ -256,8 +256,8 @@ static int init(int rate_hz, int channels, int format, int flags)
// make sure alsa_device is null-terminated even when using strncpy etc.
memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
- mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz,
- channels, audio_out_format_name(format));
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
+ channels, format);
alsa_handler = NULL;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
@@ -334,8 +334,7 @@ static int init(int rate_hz, int channels, int format, int flags)
ao_data.bps *= 4;
break;
case -1:
- mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%s) requested - output disabled\n",
- audio_out_format_name(format));
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%x) requested - output disabled\n",format);
return(0);
break;
default:
@@ -587,8 +586,7 @@ static int init(int rate_hz, int channels, int format, int flags)
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_INFO,
- "alsa-init: format %s are not supported by hardware, trying default\n",
- audio_out_format_name(format));
+ "alsa-init: format %x are not supported by hardware, trying default\n", format);
alsa_format = SND_PCM_FORMAT_S16_LE;
ao_data.format = AF_FORMAT_S16_LE;
ao_data.bps = channels * rate_hz * 2;
diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c
index ed073f55ae..687b48ac8c 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -50,9 +50,10 @@ static int init(int rate_hz, int channels, int format, int flags)
snd_pcm_channel_setup_t setup;
snd_pcm_info_t info;
snd_pcm_channel_info_t chninfo;
+ char buf[128];
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_InitInfo, rate_hz,
- channels, audio_out_format_name(format));
+ channels, af_fmt2str(format));
alsa_handler = NULL;
@@ -111,8 +112,7 @@ static int init(int rate_hz, int channels, int format, int flags)
ao_data.bps *= 2;
break;
case -1:
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,
- audio_out_format_name(format));
+ mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str(format,&buf,128));
return(0);
default:
break;
diff --git a/libao2/ao_dsound.c b/libao2/ao_dsound.c
index 39ab901df8..bcdce0b1a0 100644
--- a/libao2/ao_dsound.c
+++ b/libao2/ao_dsound.c
@@ -274,7 +274,7 @@ static int write_buffer(unsigned char *data, int len)
int i, j;
int numsamp,sampsize;
- sampsize = audio_out_format_bits(ao_data.format)>>3; // bytes per sample
+ sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample
numsamp = dwBytes1 / (ao_data.channels * sampsize); // number of samples for each channel in this buffer
for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
@@ -372,16 +372,16 @@ static int init(int rate, int channels, int format, int flags)
case AF_FORMAT_S8:
break;
default:
- mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",audio_out_format_name(format));
+ mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %x not supported defaulting to Signed 16-bit Little-Endian\n",format);
format=AF_FORMAT_S16_LE;
}
//fill global ao_data
ao_data.channels = channels;
ao_data.samplerate = rate;
ao_data.format = format;
- ao_data.bps = channels * rate * (audio_out_format_bits(format)>>3);
+ ao_data.bps = channels * rate * (af_fmt2bits(format)>>3);
if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec
- mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, audio_out_format_name(format));
+ mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%x\n", rate, channels, format);
mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
//fill waveformatex
@@ -395,7 +395,7 @@ static int init(int rate, int channels, int format, int flags)
wformat.Format.nBlockAlign = 4;
} else {
wformat.Format.wFormatTag = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
- wformat.Format.wBitsPerSample = audio_out_format_bits(format);
+ wformat.Format.wBitsPerSample = af_fmt2bits(format);
wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
}
diff --git a/libao2/ao_sgi.c b/libao2/ao_sgi.c
index d1c6ef2d1d..796691ab4e 100644
--- a/libao2/ao_sgi.c
+++ b/libao2/ao_sgi.c
@@ -41,8 +41,9 @@ static int control(int cmd, void *arg){
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {
-
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
+
+ char buf[128];
+ mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str(format, &buf, 128));
{ /* from /usr/share/src/dmedia/audio/setrate.c */
diff --git a/libao2/ao_sun.c b/libao2/ao_sun.c
index b9b6a3a9f4..2e102ccfa1 100644
--- a/libao2/ao_sun.c
+++ b/libao2/ao_sun.c
@@ -457,6 +457,7 @@ static int init(int rate,int channels,int format,int flags){
audio_info_t info;
int pass;
int ok;
+ char buf[128];
setup_device_paths();
@@ -479,7 +480,7 @@ static int init(int rate,int channels,int format,int flags){
for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */
AUDIO_INITINFO(&info);
- info.play.encoding = oss2sunfmt(ao_data.format = format);
+ info.play.encoding = af2sunfmt(ao_data.format = format);
info.play.precision =
(format==AF_FORMAT_S16_LE || format==AF_FORMAT_S16_BE
? AUDIO_PRECISION_16
@@ -545,7 +546,7 @@ static int init(int rate,int channels,int format,int flags){
if (!ok) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate,
- channels, audio_out_format_name(format), rate);
+ channels, af_fmt2str(format, &buf, 128), rate);
return 0;
}
@@ -625,7 +626,7 @@ static void reset(){
ioctl(audio_fd, AUDIO_DRAIN, 0);
AUDIO_INITINFO(&info);
- info.play.encoding = oss2sunfmt(ao_data.format);
+ info.play.encoding = af2sunfmt(ao_data.format);
info.play.precision =
(ao_data.format==AF_FORMAT_S16_LE || ao_data.format==AF_FORMAT_S16_BE
? AUDIO_PRECISION_16
diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c
index e589ced8c1..ff07a9e73c 100644
--- a/libao2/ao_win32.c
+++ b/libao2/ao_win32.c
@@ -147,6 +147,7 @@ static int init(int rate,int channels,int format,int flags)
MMRESULT result;
unsigned char* buffer;
int i;
+ char buf[128];
switch(format){
case AF_FORMAT_AC3:
@@ -155,7 +156,7 @@ static int init(int rate,int channels,int format,int flags)
case AF_FORMAT_S8:
break;
default:
- mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",audio_out_format_name(format));
+ mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str(format, &buf, 128));
format=AF_FORMAT_S16_LE;
}
//fill global ao_data
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-12 17:20:49 +0000