[MPlayer-cvslog] CVS: main/libao2 ao_alsa.c, 1.13, 1.14 ao_alsa5.c, 1.23, 1.24 ao_dsound.c, 1.7, 1.8 ao_nas.c, 1.19, 1.20 ao_oss.c, 1.47, 1.48 ao_pcm.c, 1.23, 1.24 ao_sdl.c, 1.39, 1.40 ao_sgi.c, 1.10, 1.11 ao_sun.c, 1.33, 1.34 ao_win32.c, 1.20, 1.21
Alex Beregszaszi
syncmail at mplayerhq.hu
Tue Dec 28 20:11:17 CET 2004
CVS change done by Alex Beregszaszi
Update of /cvsroot/mplayer/main/libao2
In directory mail:/var2/tmp/cvs-serv21245
Modified Files:
ao_alsa.c ao_alsa5.c ao_dsound.c ao_nas.c ao_oss.c ao_pcm.c
ao_sdl.c ao_sgi.c ao_sun.c ao_win32.c
Log Message:
af_fmt2str_short
Index: ao_alsa.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_alsa.c,v
retrieving revision 1.13
retrieving revision 1.14
diff -u -r1.13 -r1.14
--- ao_alsa.c 27 Dec 2004 18:14:03 -0000 1.13
+++ ao_alsa.c 28 Dec 2004 19:11:14 -0000 1.14
@@ -334,7 +334,7 @@
ao_data.bps *= 4;
break;
case -1:
- mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%x) requested - output disabled\n",format);
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%s) requested - output disabled\n",af_fmt2str_short(format));
return(0);
break;
default:
@@ -586,7 +586,7 @@
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_INFO,
- "alsa-init: format %x are not supported by hardware, trying default\n", format);
+ "alsa-init: format %s are not supported by hardware, trying default\n", af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
ao_data.format = AF_FORMAT_S16_LE;
ao_data.bps = channels * rate_hz * 2;
Index: ao_alsa5.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_alsa5.c,v
retrieving revision 1.23
retrieving revision 1.24
diff -u -r1.23 -r1.24
--- ao_alsa5.c 27 Dec 2004 19:43:13 -0000 1.23
+++ ao_alsa5.c 28 Dec 2004 19:11:14 -0000 1.24
@@ -50,10 +50,9 @@
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, af_fmt2str(format, buf, 128));
+ channels, af_fmt2str_short(format));
alsa_handler = NULL;
@@ -112,7 +111,7 @@
ao_data.bps *= 2;
break;
case -1:
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str(format,buf,128));
+ mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str_short(format));
return(0);
default:
break;
Index: ao_dsound.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_dsound.c,v
retrieving revision 1.7
retrieving revision 1.8
diff -u -r1.7 -r1.8
--- ao_dsound.c 27 Dec 2004 18:10:30 -0000 1.7
+++ ao_dsound.c 28 Dec 2004 19:11:14 -0000 1.8
@@ -372,7 +372,7 @@
case AF_FORMAT_S8:
break;
default:
- mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %x not supported defaulting to Signed 16-bit Little-Endian\n",format);
+ mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
format=AF_FORMAT_S16_LE;
}
//fill global ao_data
@@ -381,7 +381,7 @@
ao_data.format = format;
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:%x\n", rate, channels, format);
+ mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, af_fmt2str_short(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
Index: ao_nas.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_nas.c,v
retrieving revision 1.19
retrieving revision 1.20
diff -u -r1.19 -r1.20
--- ao_nas.c 27 Dec 2004 19:43:13 -0000 1.19
+++ ao_nas.c 28 Dec 2004 19:11:14 -0000 1.20
@@ -387,13 +387,12 @@
int bytes_per_sample = channels * AuSizeofFormat(auformat);
int buffer_size;
char *server;
- char buf[128];
nas_data=malloc(sizeof(struct ao_nas_data));
memset(nas_data, 0, sizeof(struct ao_nas_data));
mp_msg(MSGT_AO, MSGL_V, "ao2: %d Hz %d chans %s\n",rate,channels,
- af_fmt2str(format,buf,128));
+ af_fmt2str_short(format));
ao_data.format = format;
ao_data.samplerate = rate;
Index: ao_oss.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_oss.c,v
retrieving revision 1.47
retrieving revision 1.48
diff -u -r1.47 -r1.48
--- ao_oss.c 28 Dec 2004 01:59:12 -0000 1.47
+++ ao_oss.c 28 Dec 2004 19:11:14 -0000 1.48
@@ -184,8 +184,8 @@
char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
int oss_format;
-// mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels,
-// audio_out_format_name(format));
+ mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels,
+ af_fmt2str_short(format));
if (ao_subdevice)
dsp = ao_subdevice;
@@ -275,8 +275,6 @@
#endif
goto ac3_retry;
}
-// mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
-// audio_out_format_name(ao_data.format), audio_out_format_name(format));
#if 0
if(oss_format!=format2oss(format))
mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %s sample format! Broken audio or bad playback speed are possible! Try with '-aop list=format'\n",audio_out_format_name(format));
@@ -284,6 +282,9 @@
ao_data.format = oss2format(oss_format);
if (ao_data.format == -1) return 0;
+
+ mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
+ af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
ao_data.channels = channels;
if(format != AF_FORMAT_AC3) {
Index: ao_pcm.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_pcm.c,v
retrieving revision 1.23
retrieving revision 1.24
diff -u -r1.23 -r1.24
--- ao_pcm.c 27 Dec 2004 17:30:14 -0000 1.23
+++ ao_pcm.c 28 Dec 2004 19:11:14 -0000 1.24
@@ -114,9 +114,9 @@
wavhdr.data_length=le2me_32(0x7ffff000);
wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
-// mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
-// (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
-// (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
+ mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
+ (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
+ (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
fp = fopen(ao_outputfilename, "wb");
Index: ao_sdl.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_sdl.c,v
retrieving revision 1.39
retrieving revision 1.40
diff -u -r1.39 -r1.40
--- ao_sdl.c 27 Dec 2004 17:30:14 -0000 1.39
+++ ao_sdl.c 28 Dec 2004 19:11:14 -0000 1.40
@@ -181,7 +181,7 @@
/* Allocate ring-buffer memory */
buffer = (unsigned char *) malloc(BUFFSIZE);
-// mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
+ mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
if(ao_subdevice) {
setenv("SDL_AUDIODRIVER", ao_subdevice, 1);
Index: ao_sgi.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_sgi.c,v
retrieving revision 1.10
retrieving revision 1.11
diff -u -r1.10 -r1.11
--- ao_sgi.c 27 Dec 2004 19:43:13 -0000 1.10
+++ ao_sgi.c 28 Dec 2004 19:11:14 -0000 1.11
@@ -42,8 +42,7 @@
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {
- char buf[128];
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str(format, buf, 128));
+ mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
{ /* from /usr/share/src/dmedia/audio/setrate.c */
Index: ao_sun.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_sun.c,v
retrieving revision 1.33
retrieving revision 1.34
diff -u -r1.33 -r1.34
--- ao_sun.c 27 Dec 2004 19:43:13 -0000 1.33
+++ ao_sun.c 28 Dec 2004 19:11:14 -0000 1.34
@@ -466,8 +466,8 @@
enable_sample_timing = realtime_samplecounter_available(audio_dev);
}
-// printf("ao2: %d Hz %d chans %s [0x%X]\n",
-// rate,channels,audio_out_format_name(format),format);
+ printf("ao2: %d Hz %d chans %s [0x%X]\n",
+ rate,channels,af_fmt2str_short(format),format);
audio_fd=open(audio_dev, O_WRONLY);
if(audio_fd<0){
Index: ao_win32.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_win32.c,v
retrieving revision 1.20
retrieving revision 1.21
diff -u -r1.20 -r1.21
--- ao_win32.c 27 Dec 2004 18:10:30 -0000 1.20
+++ ao_win32.c 28 Dec 2004 19:11:14 -0000 1.21
@@ -147,7 +147,6 @@
MMRESULT result;
unsigned char* buffer;
int i;
- char buf[128];
switch(format){
case AF_FORMAT_AC3:
@@ -156,7 +155,7 @@
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",af_fmt2str(format, &buf, 128));
+ mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
format=AF_FORMAT_S16_LE;
}
//fill global ao_data
@@ -168,11 +167,11 @@
ao_data.bps*=2;
if(ao_data.buffersize==-1)
{
- ao_data.buffersize=audio_out_format_bits(format)/8;
+ ao_data.buffersize=af_fmt2bits(format)/8;
ao_data.buffersize*= channels;
ao_data.buffersize*= SAMPLESIZE;
}
- mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, audio_out_format_name(format));
+ mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format));
mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
//fill waveformatex
@@ -189,14 +188,14 @@
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);
}
if(channels>2)
{
wformat.dwChannelMask = channel_mask[channels-3];
wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
- wformat.Samples.wValidBitsPerSample=audio_out_format_bits(format);
+ wformat.Samples.wValidBitsPerSample=af_fmt2bits(format);
}
wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
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