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af_fmt2str_short

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git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14265 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
alex 2004-12-28 19:11:14 +00:00
parent e8739c6d92
commit 14a29762f2
10 changed files with 24 additions and 27 deletions
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4
libao2/ao_alsa.c
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@ -334,7 +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 (%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 @@ static int init(int rate_hz, int channels, int format, int flags)
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;

5
libao2/ao_alsa5.c
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@ -50,10 +50,9 @@ 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, af_fmt2str(format, buf, 128));
channels, af_fmt2str_short(format));
alsa_handler = NULL;
@ -112,7 +111,7 @@ 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,af_fmt2str(format,buf,128));
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str_short(format));
return(0);
default:
break;

4
libao2/ao_dsound.c
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@ -372,7 +372,7 @@ static int init(int rate, int channels, int format, int flags)
case AF_FORMAT_S8:
break;
default:
mp_msg(MSGT_AO, MSGL_V,"ao_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 @@ static int init(int rate, int channels, int format, int flags)
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

3
libao2/ao_nas.c
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@ -387,13 +387,12 @@ static int init(int rate,int channels,int format,int flags)
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;

9
libao2/ao_oss.c
View File

@ -184,8 +184,8 @@ static int init(int rate,int channels,int format,int flags){
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 @@ ac3_retry:
#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 @@ ac3_retry:
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) {

6
libao2/ao_pcm.c
View File

@ -114,9 +114,9 @@ static int init(int rate,int channels,int format,int flags){
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");

2
libao2/ao_sdl.c
View File

@ -181,7 +181,7 @@ static int init(int rate,int channels,int format,int flags){
/* 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);

3
libao2/ao_sgi.c
View File

@ -42,8 +42,7 @@ static int control(int cmd, void *arg){
// 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 */

4
libao2/ao_sun.c
View File

@ -466,8 +466,8 @@ static int init(int rate,int channels,int format,int flags){
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){

11
libao2/ao_win32.c
View File

@ -147,7 +147,6 @@ 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:
@ -156,7 +155,7 @@ 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",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 @@ static int init(int rate,int channels,int format,int flags)
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 @@ static int init(int rate,int channels,int format,int flags)
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;