diff -bur g2/audio/afmt.c g2.my/audio/afmt.c --- g2/audio/afmt.c Sun Nov 17 18:35:31 2002 +++ g2.my/audio/afmt.c Mon May 12 18:37:41 2003 @@ -4,7 +4,7 @@ #include "../config.h" #include "afmt.h" -char *audio_out_format_name(int format) +char *get_afmt_name(int format) { switch (format) { @@ -49,7 +49,7 @@ } // return number of bits for 1 sample in one channel, or 8 bits for compressed -int audio_out_format_bits(int format){ +int get_afmt_bits(int format){ switch (format) { case AFMT_S16_LE: diff -bur g2/audio/afmt.h g2.my/audio/afmt.h --- g2/audio/afmt.h Wed Apr 16 22:38:44 2003 +++ g2.my/audio/afmt.h Mon May 12 18:45:02 2003 @@ -1,3 +1,5 @@ +#ifndef __AFMT_H +#define __AFMT_H /* Defines that AFMT_ stuff */ @@ -52,4 +54,9 @@ #ifndef AFMT_FLOAT # define AFMT_FLOAT 0x00004000 +#endif + +char *get_afmt_name(int format); +int get_afmt_bits(int format); + #endif diff -bur g2/libao2/ao_alsa5.c g2.my/libao2/ao_alsa5.c --- g2/libao2/ao_alsa5.c Sun Mar 23 20:13:41 2003 +++ g2.my/libao2/ao_alsa5.c Mon May 12 18:40:13 2003 @@ -51,7 +51,7 @@ snd_pcm_channel_info_t chninfo; mp_msg(MSGT_AO, MSGL_INFO, "alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz, - channels, audio_out_format_name(format)); + channels, get_afmt_name(format)); alsa_handler = NULL; @@ -111,7 +111,7 @@ break; case -1: mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: invalid format (%s) requested - output disabled\n", - audio_out_format_name(format)); + get_afmt_name(format)); return(0); default: break; diff -bur g2/libao2/ao_alsa9.c g2.my/libao2/ao_alsa9.c --- g2/libao2/ao_alsa9.c Sun Mar 23 01:00:26 2003 +++ g2.my/libao2/ao_alsa9.c Mon May 12 18:40:22 2003 @@ -210,7 +210,7 @@ printf("alsa-init: testing and bugreports are welcome.\n"); printf("alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz, - channels, audio_out_format_name(format)); + channels, get_afmt_name(format)); alsa_handler = NULL; @@ -280,7 +280,7 @@ break; case -1: printf("alsa-init: invalid format (%s) requested - output disabled\n", - audio_out_format_name(format)); + get_afmt_name(format)); return(0); default: break; diff -bur g2/libao2/ao_nas.c g2.my/libao2/ao_nas.c --- g2/libao2/ao_nas.c Sun Mar 23 01:00:26 2003 +++ g2.my/libao2/ao_nas.c Mon May 12 18:41:50 2003 @@ -402,7 +402,7 @@ memset(nas_data, 0, sizeof(struct ao_nas_data)); mp_msg(MSGT_AO, MSGL_V, "ao2: %d Hz %d chans %s\n",rate,channels, - audio_out_format_name(format)); + get_afmt_name(format)); ao_data.format = format; ao_data.samplerate = rate; diff -bur g2/libao2/ao_oss.c g2.my/libao2/ao_oss.c --- g2/libao2/ao_oss.c Thu Apr 17 10:36:07 2003 +++ g2.my/libao2/ao_oss.c Mon May 12 18:42:13 2003 @@ -96,7 +96,7 @@ static int init(int rate,int channels,int format,int flags){ mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels, - audio_out_format_name(format)); + get_afmt_name(format)); if (ao_subdevice) dsp = ao_subdevice; @@ -146,10 +146,10 @@ 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)); + get_afmt_name(ao_data.format), get_afmt_name(format)); #if 0 if(ao_data.format!=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)); + 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",get_afmt_name(format)); #endif ao_data.channels = channels; diff -bur g2/libao2/ao_pcm.c g2.my/libao2/ao_pcm.c --- g2/libao2/ao_pcm.c Thu Apr 17 10:47:16 2003 +++ g2.my/libao2/ao_pcm.c Mon May 12 18:42:21 2003 @@ -114,7 +114,7 @@ printf("PCM: File: %s (%s)\n" "PCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename, (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, - (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format)); + (channels > 1) ? "Stereo" : "Mono", get_afmt_name(format)); printf("PCM: Info: fastest dumping is achieved with -vc null -vo null\n" "PCM: Info: to write WAVE files use -waveheader (default); " "for RAW PCM -nowaveheader.\n"); diff -bur g2/libao2/ao_sdl.c g2.my/libao2/ao_sdl.c --- g2/libao2/ao_sdl.c Thu Apr 17 10:35:14 2003 +++ g2.my/libao2/ao_sdl.c Mon May 12 18:42:32 2003 @@ -153,7 +153,7 @@ /* Allocate ring-buffer memory */ for(i=0;i 1) ? "Stereo" : "Mono", audio_out_format_name(format)); + mp_msg(MSGT_AO,MSGL_INFO,"SDL: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", get_afmt_name(format)); if(ao_subdevice) { setenv("SDL_AUDIODRIVER", ao_subdevice, 1); diff -bur g2/libao2/ao_sgi.c g2.my/libao2/ao_sgi.c --- g2/libao2/ao_sgi.c Sun Mar 23 01:00:26 2003 +++ g2.my/libao2/ao_sgi.c Mon May 12 18:42:41 2003 @@ -38,7 +38,7 @@ // return: 1=success 0=fail static int init(int rate, int channels, int format, int flags) { - printf("ao_sgi, init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format)); + printf("ao_sgi, init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", get_afmt_name(format)); { /* from /usr/share/src/dmedia/audio/setrate.c */ diff -bur g2/libao2/ao_sun.c g2.my/libao2/ao_sun.c --- g2/libao2/ao_sun.c Sun Mar 23 01:00:26 2003 +++ g2.my/libao2/ao_sun.c Mon May 12 18:42:53 2003 @@ -478,7 +478,7 @@ #endif // printf("ao2: %d Hz %d chans %s [0x%X]\n", -// rate,channels,audio_out_format_name(format),format); +// rate,channels,get_afmt_name(format),format); audio_fd=open(audio_dev, O_WRONLY); if(audio_fd<0){ @@ -557,7 +557,7 @@ if (!ok) { printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n", - channels, audio_out_format_name(format), rate); + channels, get_afmt_name(format), rate); return 0; } diff -bur g2/libao2/ao_win32.c g2.my/libao2/ao_win32.c --- g2/libao2/ao_win32.c Sun Mar 23 01:00:26 2003 +++ g2.my/libao2/ao_win32.c Mon May 12 18:43:05 2003 @@ -108,11 +108,11 @@ ao_data.bps*=2; if(ao_data.buffersize==-1) { - ao_data.buffersize=audio_out_format_bits(format)/8; + ao_data.buffersize=get_afmt_bits(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, get_afmt_name(format)); mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize); //fill waveformatex @@ -121,7 +121,7 @@ wformat.wFormatTag = WAVE_FORMAT_PCM; wformat.nChannels = channels; wformat.nSamplesPerSec = rate; - wformat.wBitsPerSample = audio_out_format_bits(format); + wformat.wBitsPerSample = get_afmt_bits(format); wformat.nBlockAlign = wformat.nChannels * (wformat.wBitsPerSample >> 3); wformat.nAvgBytesPerSec = wformat.nSamplesPerSec * wformat.nBlockAlign; diff -bur g2/test-codecs.c g2.my/test-codecs.c --- g2/test-codecs.c Sat May 10 21:17:31 2003 +++ g2.my/test-codecs.c Mon May 12 18:39:46 2003 @@ -168,12 +168,12 @@ sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize, // output: &ao_data.samplerate, &ao_data.channels, &ao_data.format, - audio_out_format_bits(ao_data.format)/8)){ + get_afmt_bits(ao_data.format)/8)){ mp_msg(MSGT_CPLAYER,MSGL_ERR,"Error at audio filter chain pre-init!\n"); } else { mp_msg(MSGT_CPLAYER,MSGL_INFO,"AF_pre: %dHz %dch %s\n", ao_data.samplerate, ao_data.channels, - audio_out_format_name(ao_data.format)); + get_afmt_name(ao_data.format)); } #endif @@ -192,8 +192,8 @@ mp_msg(MSGT_CPLAYER,MSGL_INFO,"AO: [%s] %dHz %dch %s (%d bps)\n", audio_out->info->short_name, ao_data.samplerate, ao_data.channels, - audio_out_format_name(ao_data.format), - audio_out_format_bits(ao_data.format)/8 ); + get_afmt_name(ao_data.format), + get_afmt_bits(ao_data.format)/8 ); mp_msg(MSGT_CPLAYER,MSGL_V,MSGTR_AODescription_AOAuthor, audio_out->info->name, audio_out->info->author); if(strlen(audio_out->info->comment) > 0) @@ -205,7 +205,7 @@ (int)(sh_audio->samplerate*playback_speed), sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize, ao_data.samplerate, ao_data.channels, ao_data.format, - audio_out_format_bits(ao_data.format)/8, /* ao_data.bps, */ + get_afmt_bits(ao_data.format)/8, /* ao_data.bps, */ ao_data.outburst*4, ao_data.buffersize)){ mp_msg(MSGT_CPLAYER,MSGL_ERR,"Couldn't find matching filter / ao format!\n"); } diff -bur g2/test-play.c g2.my/test-play.c --- g2/test-play.c Sun May 11 14:12:06 2003 +++ g2.my/test-play.c Mon May 12 18:39:21 2003 @@ -146,12 +146,12 @@ sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize, // output: &ao_data.samplerate, &ao_data.channels, &ao_data.format, - audio_out_format_bits(ao_data.format)/8)){ + get_afmt_bits(ao_data.format)/8)){ mp_msg(MSGT_CPLAYER,MSGL_ERR,"Error at audio filter chain pre-init!\n"); } else { mp_msg(MSGT_CPLAYER,MSGL_INFO,"AF_pre: %dHz %dch %s\n", ao_data.samplerate, ao_data.channels, - audio_out_format_name(ao_data.format)); + get_afmt_name(ao_data.format)); } // Find capable audio-out driver: if(!(audio_out=init_best_audio_out(audio_driver_list, @@ -168,7 +168,7 @@ (int)(sh_audio->samplerate*playback_speed), sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize, ao_data.samplerate, ao_data.channels, ao_data.format, - audio_out_format_bits(ao_data.format)/8, /* ao_data.bps, */ + get_afmt_bits(ao_data.format)/8, /* ao_data.bps, */ ao_data.outburst*4, ao_data.buffersize)){ mp_msg(MSGT_CPLAYER,MSGL_ERR,"Couldn't find matching filter / ao format!\n"); }