Mercurial > mplayer.hg
changeset 14245:815f03b7cee5
removing AFMT_ dependancy
line wrap: on
line diff
--- a/Makefile Mon Dec 27 11:09:49 2004 +0000 +++ b/Makefile Mon Dec 27 17:30:15 2004 +0000 @@ -20,7 +20,7 @@ endif SRCS_COMMON = cpudetect.c codec-cfg.c spudec.c playtree.c playtreeparser.c asxparser.c vobsub.c subreader.c sub_cc.c find_sub.c m_config.c m_option.c parser-cfg.c m_struct.c edl.c -SRCS_MENCODER = mencoder.c mp_msg-mencoder.c $(SRCS_COMMON) libao2/afmt.c divx4_vbr.c libvo/aclib.c libvo/osd.c libvo/sub.c libvo/font_load.c libvo/font_load_ft.c xvid_vbr.c parser-mecmd.c +SRCS_MENCODER = mencoder.c mp_msg-mencoder.c $(SRCS_COMMON) divx4_vbr.c libvo/aclib.c libvo/osd.c libvo/sub.c libvo/font_load.c libvo/font_load_ft.c xvid_vbr.c parser-mecmd.c SRCS_MPLAYER = mplayer.c mp_msg.c $(SRCS_COMMON) mixer.c parser-mpcmd.c ifeq ($(UNRARLIB),yes)
--- a/libaf/Makefile Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/Makefile Mon Dec 27 17:30:15 2004 +0000 @@ -2,7 +2,7 @@ LIBNAME = libaf.a -SRCS=af.c af_mp.c af_dummy.c af_delay.c af_channels.c af_format.c af_resample.c \ +SRCS=af.c af_dummy.c af_delay.c af_channels.c af_format.c af_resample.c \ window.c filter.c af_volume.c af_equalizer.c af_tools.c af_comp.c af_gate.c \ af_pan.c af_surround.c af_sub.c af_export.c af_volnorm.c af_extrastereo.c \ af_lavcresample.c af_sweep.c af_hrtf.c $(OPTIONAL_SRCS)
--- a/libaf/af_comp.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_comp.c Mon Dec 27 17:30:15 2004 +0000 @@ -44,7 +44,7 @@ af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; - af->data->format = AF_FORMAT_F | AF_FORMAT_NE; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; // Time constant set to 0.1s
--- a/libaf/af_equalizer.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_equalizer.c Mon Dec 27 17:30:15 2004 +0000 @@ -87,7 +87,7 @@ af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; - af->data->format = AF_FORMAT_NE | AF_FORMAT_F; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; // Calculate number of active filters @@ -173,7 +173,7 @@ float* end = in + c->len/4; // Block loop end while(in < end){ - register uint32_t k = 0; // Frequency band index + register int k = 0; // Frequency band index register float yt = *in; // Current input sample in+=nch;
--- a/libaf/af_export.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_export.c Mon Dec 27 17:30:15 2004 +0000 @@ -72,7 +72,7 @@ // Accept only int16_t as input format (which sucks) af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; - af->data->format = AF_FORMAT_SI | AF_FORMAT_NE; + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; // If buffer length isn't set, set it to the default value @@ -163,7 +163,6 @@ */ static void uninit( struct af_instance_s* af ) { - int i; if (af->data){ free(af->data); af->data = NULL;
--- a/libaf/af_extrastereo.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_extrastereo.c Mon Dec 27 17:30:15 2004 +0000 @@ -37,7 +37,7 @@ af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = 2; - af->data->format = AF_FORMAT_SI | AF_FORMAT_NE; + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; return af_test_output(af,(af_data_t*)arg);
--- a/libaf/af_format.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_format.c Mon Dec 27 17:30:15 2004 +0000 @@ -43,7 +43,7 @@ static void int2float(void* in, void* out, int len, int bps); // Convert from string to format -static int str2fmt(char* str) +int af_str2fmt(char* str) { int format=0; // Scan for endianess @@ -87,16 +87,34 @@ return format; } +inline int af_fmt2bits(int format) +{ + return (format & AF_FORMAT_BITS_MASK)+8; +// return (((format & AF_FORMAT_BITS_MASK)>>3)+1) * 8; +#if 0 + switch(format & AF_FORMAT_BITS_MASK) + { + case AF_FORMAT_8BIT: return 8; + case AF_FORMAT_16BIT: return 16; + case AF_FORMAT_24BIT: return 24; + case AF_FORMAT_32BIT: return 32; + case AF_FORMAT_48BIT: return 48; + } +#endif + return -1; +} + /* Convert format to str input str is a buffer for the converted string, size is the size of the buffer */ -char* fmt2str(int format, char* str, size_t size) +char* af_fmt2str(int format, char* str, int size) { int i=0; - // Print endinaness + + // Endianess if(AF_FORMAT_LE == (format & AF_FORMAT_END_MASK)) - i+=snprintf(str,size,"little endian "); + i+=snprintf(str,size-i,"little endian "); else - i+=snprintf(str,size,"big endian "); + i+=snprintf(str,size-i,"big endian "); if(format & AF_FORMAT_SPECIAL_MASK){ switch(format & AF_FORMAT_SPECIAL_MASK){ @@ -108,12 +126,17 @@ i+=snprintf(&str[i],size-i,"MPEG 2 "); break; case(AF_FORMAT_AC3): i+=snprintf(&str[i],size-i,"AC3 "); break; + default: + printf("Unknown special\n"); } } else{ + // Bits + i+=snprintf(&str[i],size-i,"%d-bit ", af_fmt2bits(format)); + // Point if(AF_FORMAT_F == (format & AF_FORMAT_POINT_MASK)) - i+=snprintf(&str[i],size,"float "); + i+=snprintf(&str[i],size-i,"float "); else{ // Sign if(AF_FORMAT_US == (format & AF_FORMAT_SIGN_MASK)) @@ -121,7 +144,7 @@ else i+=snprintf(&str[i],size-i,"signed "); - i+=snprintf(&str[i],size,"int "); + i+=snprintf(&str[i],size-i,"int "); } } return str; @@ -148,7 +171,7 @@ case(AF_FORMAT_MPEG2): case(AF_FORMAT_AC3): af_msg(AF_MSG_ERROR,"[format] Sample format %s not yet supported \n", - fmt2str(format,buf,255)); + af_fmt2str(format,buf,255)); return AF_ERROR; } return AF_OK; @@ -173,9 +196,9 @@ (AF_OK != check_format(af->data->format))) return AF_ERROR; - af_msg(AF_MSG_VERBOSE,"[format] Changing sample format from %ibit %sto %ibit %s \n", - ((af_data_t*)arg)->bps*8,fmt2str(((af_data_t*)arg)->format,buf1,255), - af->data->bps*8,fmt2str(af->data->format,buf2,255)); + af_msg(AF_MSG_VERBOSE,"[format] Changing sample format from %sto %s \n", + af_fmt2str(((af_data_t*)arg)->format,buf1,255), + af_fmt2str(af->data->format,buf2,255)); af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; @@ -190,7 +213,7 @@ str[0] = '\0'; sscanf((char*)arg,"%i:%s",&bps,str); // Convert string to format - format = str2fmt(str); + format = af_str2fmt(str); // Automatic correction of errors switch(format & AF_FORMAT_SPECIAL_MASK){
--- a/libaf/af_format.h Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_format.h Mon Dec 27 17:30:15 2004 +0000 @@ -1,6 +1,8 @@ /* The sample format system used lin libaf is based on bitmasks. The format definition only refers to the storage format not the resolution. */ +#ifndef __af_format_h__ +#define __af_format_h__ // Endianess #define AF_FORMAT_BE (0<<0) // Big Endian @@ -14,8 +16,8 @@ #endif // Signed/unsigned -#define AF_FORMAT_SI (0<<1) // SIgned -#define AF_FORMAT_US (1<<1) // Un Signed +#define AF_FORMAT_SI (0<<1) // Signed +#define AF_FORMAT_US (1<<1) // Unsigned #define AF_FORMAT_SIGN_MASK (1<<1) // Fixed or floating point @@ -23,13 +25,63 @@ #define AF_FORMAT_F (1<<2) // Foating point #define AF_FORMAT_POINT_MASK (1<<2) +// Bits used +#define AF_FORMAT_8BIT (0<<3) +#define AF_FORMAT_16BIT (1<<3) +#define AF_FORMAT_24BIT (2<<3) +#define AF_FORMAT_32BIT (3<<3) +#define AF_FORMAT_40BIT (4<<3) +#define AF_FORMAT_48BIT (5<<3) +#define AF_FORMAT_BITS_MASK (7<<3) + // Special flags refering to non pcm data -#define AF_FORMAT_MU_LAW (1<<3) // -#define AF_FORMAT_A_LAW (2<<3) // -#define AF_FORMAT_MPEG2 (3<<3) // MPEG(2) audio -#define AF_FORMAT_AC3 (4<<3) // Dolby Digital AC3 -#define AF_FORMAT_IMA_ADPCM AF_FORMAT_LE|AF_FORMAT_SI // Same as 16 bit signed int -#define AF_FORMAT_SPECIAL_MASK (7<<3) +#define AF_FORMAT_MU_LAW (1<<6) +#define AF_FORMAT_A_LAW (2<<6) +#define AF_FORMAT_MPEG2 (3<<6) // MPEG(2) audio +#define AF_FORMAT_AC3 (4<<6) // Dolby Digital AC3 +#define AF_FORMAT_IMA_ADPCM (5<<6) +#define AF_FORMAT_SPECIAL_MASK (7<<6) + +// PREDEFINED formats + +#define AF_FORMAT_U8 (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_8BIT|AF_FORMAT_NE) +#define AF_FORMAT_S8 (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_8BIT|AF_FORMAT_NE) +#define AF_FORMAT_U16_LE (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_16BIT|AF_FORMAT_LE) +#define AF_FORMAT_U16_BE (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_16BIT|AF_FORMAT_BE) +#define AF_FORMAT_S16_LE (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_16BIT|AF_FORMAT_LE) +#define AF_FORMAT_S16_BE (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_16BIT|AF_FORMAT_BE) +#define AF_FORMAT_U24_LE (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_24BIT|AF_FORMAT_LE) +#define AF_FORMAT_U24_BE (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_24BIT|AF_FORMAT_BE) +#define AF_FORMAT_S24_LE (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_24BIT|AF_FORMAT_LE) +#define AF_FORMAT_S24_BE (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_24BIT|AF_FORMAT_BE) +#define AF_FORMAT_U32_LE (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_32BIT|AF_FORMAT_LE) +#define AF_FORMAT_U32_BE (AF_FORMAT_I|AF_FORMAT_US|AF_FORMAT_32BIT|AF_FORMAT_BE) +#define AF_FORMAT_S32_LE (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_32BIT|AF_FORMAT_LE) +#define AF_FORMAT_S32_BE (AF_FORMAT_I|AF_FORMAT_SI|AF_FORMAT_32BIT|AF_FORMAT_BE) -extern char* fmt2str(int format, char* str, size_t size); +#define AF_FORMAT_FLOAT_LE (AF_FORMAT_F|AF_FORMAT_32BIT|AF_FORMAT_LE) +#define AF_FORMAT_FLOAT_BE (AF_FORMAT_F|AF_FORMAT_32BIT|AF_FORMAT_BE) +#ifdef WORDS_BIGENDIAN +#define AF_FORMAT_U16_NE AF_FORMAT_U16_BE +#define AF_FORMAT_S16_NE AF_FORMAT_S16_BE +#define AF_FORMAT_U24_NE AF_FORMAT_U24_BE +#define AF_FORMAT_S24_NE AF_FORMAT_S24_BE +#define AF_FORMAT_U32_NE AF_FORMAT_U32_BE +#define AF_FORMAT_S32_NE AF_FORMAT_S32_BE +#define AF_FORMAT_FLOAT_NE AF_FORMAT_FLOAT_BE +#else +#define AF_FORMAT_U16_NE AF_FORMAT_U16_LE +#define AF_FORMAT_S16_NE AF_FORMAT_S16_LE +#define AF_FORMAT_U24_NE AF_FORMAT_U24_LE +#define AF_FORMAT_S24_NE AF_FORMAT_S24_LE +#define AF_FORMAT_U32_NE AF_FORMAT_U32_LE +#define AF_FORMAT_S32_NE AF_FORMAT_S32_LE +#define AF_FORMAT_FLOAT_NE AF_FORMAT_FLOAT_LE +#endif + +extern int af_str2fmt(char *str); +extern int af_fmt2bits(int format); +extern char* af_fmt2str(int format, char* str, int size); + +#endif /* __af_format_h__ */
--- a/libaf/af_gate.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_gate.c Mon Dec 27 17:30:15 2004 +0000 @@ -42,7 +42,7 @@ af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; - af->data->format = AF_FORMAT_F | AF_FORMAT_NE; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; // Time constant set to 0.1s
--- a/libaf/af_hrtf.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_hrtf.c Mon Dec 27 17:30:15 2004 +0000 @@ -119,7 +119,7 @@ if(af->data->nch < 5) { af->data->nch = 5; } - af->data->format = AF_FORMAT_SI | AF_FORMAT_NE; + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; return af_test_output(af, (af_data_t*)arg); case AF_CONTROL_COMMAND_LINE:
--- a/libaf/af_lavcresample.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_lavcresample.c Mon Dec 27 17:30:15 2004 +0000 @@ -52,7 +52,7 @@ af->data->nch = data->nch; if (af->data->nch > CHANS) af->data->nch = CHANS; - af->data->format = AF_FORMAT_SI | AF_FORMAT_NE; + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; g= ff_gcd(af->data->rate, data->rate); af->mul.n = af->data->rate/g;
--- a/libaf/af_mp.c Mon Dec 27 11:09:49 2004 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,98 +0,0 @@ -#include "af.h" - -/* Decodes the format from mplayer format to libaf format */ -int af_format_decode(int ifmt) -{ - int ofmt = ~0; - // Check input ifmt - switch(ifmt){ - case(AFMT_U8): - ofmt = AF_FORMAT_LE|AF_FORMAT_US; break; - case(AFMT_S8): - ofmt = AF_FORMAT_LE|AF_FORMAT_SI; break; - case(AFMT_S16_LE): - ofmt = AF_FORMAT_LE|AF_FORMAT_SI; break; - case(AFMT_S16_BE): - ofmt = AF_FORMAT_BE|AF_FORMAT_SI; break; - case(AFMT_U16_LE): - ofmt = AF_FORMAT_LE|AF_FORMAT_US; break; - case(AFMT_U16_BE): - ofmt = AF_FORMAT_BE|AF_FORMAT_US; break; - case(AFMT_S24_LE): - ofmt = AF_FORMAT_LE|AF_FORMAT_SI; break; - case(AFMT_S24_BE): - ofmt = AF_FORMAT_BE|AF_FORMAT_SI; break; - case(AFMT_U24_LE): - ofmt = AF_FORMAT_LE|AF_FORMAT_US; break; - case(AFMT_U24_BE): - ofmt = AF_FORMAT_BE|AF_FORMAT_US; break; - case(AFMT_S32_LE): - ofmt = AF_FORMAT_LE|AF_FORMAT_SI; break; - case(AFMT_S32_BE): - ofmt = AF_FORMAT_BE|AF_FORMAT_SI; break; - case(AFMT_U32_LE): - ofmt = AF_FORMAT_LE|AF_FORMAT_US; break; - case(AFMT_U32_BE): - ofmt = AF_FORMAT_BE|AF_FORMAT_US; break; - case(AFMT_IMA_ADPCM): - ofmt = AF_FORMAT_IMA_ADPCM; break; - case(AFMT_MU_LAW): - ofmt = AF_FORMAT_MU_LAW; break; - case(AFMT_A_LAW): - ofmt = AF_FORMAT_A_LAW; break; - case(AFMT_MPEG): - ofmt = AF_FORMAT_MPEG2; break; - case(AFMT_AC3): - ofmt = AF_FORMAT_AC3; break; - case(AFMT_FLOAT): - ofmt = AF_FORMAT_F | AF_FORMAT_NE; break; - default: - if ((ifmt & AFMT_AF_FLAGS) == AFMT_AF_FLAGS) { - ofmt = ifmt & ~AFMT_AF_FLAGS; - break; - } - //This can not happen .... - af_msg(AF_MSG_FATAL,"Unrecognized input audio format %i\n",ifmt); - break; - } - return ofmt; -} - -/* Encodes the format from libaf format to mplayer (OSS) format */ -int af_format_encode(void* fmtp) -{ - af_data_t* fmt=(af_data_t*) fmtp; - switch(fmt->format&AF_FORMAT_SPECIAL_MASK){ - case 0: // PCM: - if((fmt->format&AF_FORMAT_POINT_MASK)==AF_FORMAT_I){ - if((fmt->format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI){ - // signed int PCM: - switch(fmt->bps){ - case 1: return AFMT_S8; - case 2: return (fmt->format&AF_FORMAT_LE) ? AFMT_S16_LE : AFMT_S16_BE; - case 3: return (fmt->format&AF_FORMAT_LE) ? AFMT_S24_LE : AFMT_S24_BE; - case 4: return (fmt->format&AF_FORMAT_LE) ? AFMT_S32_LE : AFMT_S32_BE; - } - } else { - // unsigned int PCM: - switch(fmt->bps){ - case 1: return AFMT_U8; - case 2: return (fmt->format&AF_FORMAT_LE) ? AFMT_U16_LE : AFMT_U16_BE; - case 3: return (fmt->format&AF_FORMAT_LE) ? AFMT_U24_LE : AFMT_U24_BE; - case 4: return (fmt->format&AF_FORMAT_LE) ? AFMT_U32_LE : AFMT_U32_BE; - } - } - } else { - // float PCM: - return AFMT_FLOAT; // FIXME? - } - break; - case AF_FORMAT_MU_LAW: return AFMT_MU_LAW; - case AF_FORMAT_A_LAW: return AFMT_A_LAW; - case AF_FORMAT_MPEG2: return AFMT_MPEG; - case AF_FORMAT_AC3: return AFMT_AC3; - case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM; - } - return (fmt->format | AFMT_AF_FLAGS); -} -
--- a/libaf/af_mp.h Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_mp.h Mon Dec 27 17:30:15 2004 +0000 @@ -5,7 +5,6 @@ #include "../config.h" #include "../mp_msg.h" #include "../cpudetect.h" -#include "../libao2/afmt.h" /* Set the initialization type from mplayers cpudetect */ #ifdef AF_INIT_TYPE @@ -20,8 +19,4 @@ #define af_msg(lev, args... ) \ mp_msg(MSGT_AFILTER,(((lev)<0)?((lev)+3):(((lev)==0)?MSGL_INFO:((lev)+5))), ##args ) -/* Decodes the format from mplayer format to libaf format */ -extern int af_format_decode(int format); -extern int af_format_encode(void* fmt); - #endif /* __af_mp_h__ */
--- a/libaf/af_pan.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_pan.c Mon Dec 27 17:30:15 2004 +0000 @@ -37,7 +37,7 @@ if(!arg) return AF_ERROR; af->data->rate = ((af_data_t*)arg)->rate; - af->data->format = AF_FORMAT_F | AF_FORMAT_NE; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; af->mul.n = af->data->nch; af->mul.d = ((af_data_t*)arg)->nch;
--- a/libaf/af_resample.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_resample.c Mon Dec 27 17:30:15 2004 +0000 @@ -135,29 +135,28 @@ // Make sure this filter isn't redundant if((af->data->rate == data->rate) || (af->data->rate == 0)) return AF_DETACH; - /* If sloppy and small resampling difference (2%) */ rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate; if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) && - (data->format != (AF_FORMAT_NE | AF_FORMAT_F))) || + (data->format != (AF_FORMAT_FLOAT_NE))) || ((s->setup & RSMP_MASK) == RSMP_LIN)){ s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN; - af->data->format = AF_FORMAT_NE | AF_FORMAT_SI; + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; af_msg(AF_MSG_VERBOSE,"[resample] Using linear interpolation. \n"); } else{ /* If the input format is float or if float is explicitly selected use float, otherwise use int */ - if((data->format == (AF_FORMAT_NE | AF_FORMAT_F)) || + if((data->format == (AF_FORMAT_FLOAT_NE)) || ((s->setup & RSMP_MASK) == RSMP_FLOAT)){ s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT; - af->data->format = AF_FORMAT_NE | AF_FORMAT_F; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; } else{ s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT; - af->data->format = AF_FORMAT_NE | AF_FORMAT_SI; + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; } af_msg(AF_MSG_VERBOSE,"[resample] Using %s processing and %s frequecy"
--- a/libaf/af_sub.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_sub.c Mon Dec 27 17:30:15 2004 +0000 @@ -61,7 +61,7 @@ af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = max(s->ch+1,((af_data_t*)arg)->nch); - af->data->format = AF_FORMAT_F | AF_FORMAT_NE; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; // Design low-pass filter
--- a/libaf/af_surround.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_surround.c Mon Dec 27 17:30:15 2004 +0000 @@ -93,7 +93,7 @@ float fc; af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch*2; - af->data->format = AF_FORMAT_F | AF_FORMAT_NE; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; if (af->data->nch != 4){
--- a/libaf/af_sweep.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_sweep.c Mon Dec 27 17:30:15 2004 +0000 @@ -25,7 +25,7 @@ switch(cmd){ case AF_CONTROL_REINIT: af->data->nch = data->nch; - af->data->format = AF_FORMAT_SI | AF_FORMAT_NE; + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; af->data->rate = data->rate;
--- a/libaf/af_volnorm.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_volnorm.c Mon Dec 27 17:30:15 2004 +0000 @@ -79,11 +79,11 @@ af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; - if(((af_data_t*)arg)->format == (AF_FORMAT_SI | AF_FORMAT_NE)){ - af->data->format = AF_FORMAT_SI | AF_FORMAT_NE; + if(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){ + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; }else{ - af->data->format = AF_FORMAT_F | AF_FORMAT_NE; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; } return af_test_output(af,(af_data_t*)arg); @@ -288,14 +288,14 @@ { af_volnorm_t *s = af->setup; - if(af->data->format == (AF_FORMAT_SI | AF_FORMAT_NE)) + if(af->data->format == (AF_FORMAT_S16_NE)) { if (s->method) method2_int16(s, data); else method1_int16(s, data); } - else if(af->data->format == (AF_FORMAT_F | AF_FORMAT_NE)) + else if(af->data->format == (AF_FORMAT_FLOAT_NE)) { if (s->method) method2_float(s, data);
--- a/libaf/af_volume.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/af_volume.c Mon Dec 27 17:30:15 2004 +0000 @@ -60,8 +60,8 @@ af->data->rate = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; - if(s->fast && (((af_data_t*)arg)->format != (AF_FORMAT_F | AF_FORMAT_NE))){ - af->data->format = AF_FORMAT_SI | AF_FORMAT_NE; + if(s->fast && (((af_data_t*)arg)->format != (AF_FORMAT_FLOAT_NE))){ + af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; } else{ @@ -70,7 +70,7 @@ float t = 2.0-cos(x); s->time = 1.0 - (t - sqrt(t*t - 1)); af_msg(AF_MSG_DEBUG0,"[volume] Forgetting factor = %0.5f\n",s->time); - af->data->format = AF_FORMAT_F | AF_FORMAT_NE; + af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; } return af_test_output(af,(af_data_t*)arg); @@ -140,7 +140,7 @@ register int i = 0; // Basic operation volume control only (used on slow machines) - if(af->data->format == (AF_FORMAT_SI | AF_FORMAT_NE)){ + if(af->data->format == (AF_FORMAT_S16_NE)){ int16_t* a = (int16_t*)c->audio; // Audio data int len = c->len/2; // Number of samples for(ch = 0; ch < nch ; ch++){ @@ -154,7 +154,7 @@ } } // Machine is fast and data is floating point - else if(af->data->format == (AF_FORMAT_F | AF_FORMAT_NE)){ + else if(af->data->format == (AF_FORMAT_FLOAT_NE)){ float* a = (float*)c->audio; // Audio data int len = c->len/4; // Number of samples for(ch = 0; ch < nch ; ch++){
--- a/libaf/config.h Mon Dec 27 11:09:49 2004 +0000 +++ b/libaf/config.h Mon Dec 27 17:30:15 2004 +0000 @@ -8,6 +8,8 @@ //============================================================================= */ +#include "../config.h" // WORDS_BIGENDIAN + // Number of channels #ifndef AF_NCH #define AF_NCH 6
--- a/libao2/Makefile Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/Makefile Mon Dec 27 17:30:15 2004 +0000 @@ -2,7 +2,7 @@ LIBNAME = libao2.a -SRCS=afmt.c audio_out.c ao_mpegpes.c ao_null.c ao_pcm.c ao_plugin.c pl_delay.c pl_format.c pl_surround.c remez.c pl_resample.c pl_volume.c pl_extrastereo.c pl_volnorm.c pl_eq.c $(OPTIONAL_SRCS) +SRCS=audio_out.c ao_mpegpes.c ao_null.c ao_pcm.c ao_plugin.c pl_delay.c pl_format.c pl_surround.c remez.c pl_resample.c pl_volume.c pl_extrastereo.c pl_volnorm.c pl_eq.c $(OPTIONAL_SRCS) OBJS=$(SRCS:.c=.o)
--- a/libao2/afmt.c Mon Dec 27 11:09:49 2004 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,90 +0,0 @@ -#include <stdio.h> -#include <stdlib.h> - -#include "config.h" -#include "afmt.h" - -char *audio_out_format_name(int format) -{ - switch (format) - { - case AFMT_MU_LAW: - return("Mu-Law"); - case AFMT_A_LAW: - return("A-Law"); - case AFMT_IMA_ADPCM: - return("Ima-ADPCM"); - case AFMT_S8: - return("Signed 8-bit"); - case AFMT_U8: - return("Unsigned 8-bit"); - case AFMT_U16_LE: - return("Unsigned 16-bit (Little-Endian)"); - case AFMT_U16_BE: - return("Unsigned 16-bit (Big-Endian)"); - case AFMT_S16_LE: - return("Signed 16-bit (Little-Endian)"); - case AFMT_S16_BE: - return("Signed 16-bit (Big-Endian)"); - case AFMT_MPEG: - return("MPEG (2) audio"); - case AFMT_AC3: - return("AC3"); - case AFMT_U32_LE: - return("Unsigned 32-bit (Little-Endian)"); - case AFMT_U32_BE: - return("Unsigned 32-bit (Big-Endian)"); - case AFMT_S32_LE: - return("Signed 32-bit (Little-Endian)"); - case AFMT_S32_BE: - return("Signed 32-bit (Big-Endian)"); - case AFMT_U24_LE: - return("Unsigned 24-bit (Little-Endian)"); - case AFMT_U24_BE: - return("Unsigned 24-bit (Big-Endian)"); - case AFMT_S24_LE: - return("Signed 24-bit (Little-Endian)"); - case AFMT_S24_BE: - return("Signed 24-bit (Big-Endian)"); - case AFMT_FLOAT: - return("Floating Point"); - } - return("Unknown"); -} - -// return number of bits for 1 sample in one channel, or 8 bits for compressed -int audio_out_format_bits(int format){ - switch (format) - { - case AFMT_S16_LE: - case AFMT_S16_BE: - case AFMT_U16_LE: - case AFMT_U16_BE: - return 16;//16 bits - - case AFMT_S32_LE: - case AFMT_S32_BE: - case AFMT_U32_LE: - case AFMT_U32_BE: - case AFMT_FLOAT: - return 32; - - case AFMT_S24_LE: - case AFMT_S24_BE: - case AFMT_U24_LE: - case AFMT_U24_BE: - return 24; - - case AFMT_MU_LAW: - case AFMT_A_LAW: - case AFMT_IMA_ADPCM: - case AFMT_S8: - case AFMT_U8: - case AFMT_MPEG: - case AFMT_AC3: - default: - return 8;//default 1 byte - - } - return 8; -}
--- a/libao2/afmt.h Mon Dec 27 11:09:49 2004 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,84 +0,0 @@ - -/* Defines that AFMT_ stuff */ - -#ifdef HAVE_SYS_SOUNDCARD_H -#include <sys/soundcard.h> /* For AFMT_* on linux */ -#else -#ifdef HAVE_SOUNDCARD_H -#include <soundcard.h> /* OpenBSD have this instead of <sys/soundcard> */ -#endif -#endif - -#include "config.h" /* for native endianness */ - -/* standard, old OSS audio formats */ -#ifndef AFMT_MU_LAW -# define AFMT_MU_LAW 0x00000001 -# define AFMT_A_LAW 0x00000002 -# define AFMT_IMA_ADPCM 0x00000004 -# define AFMT_U8 0x00000008 -# define AFMT_S16_LE 0x00000010 /* Little endian signed 16*/ -# define AFMT_S16_BE 0x00000020 /* Big endian signed 16 */ -# define AFMT_S8 0x00000040 -# define AFMT_U16_LE 0x00000080 /* Little endian U16 */ -# define AFMT_U16_BE 0x00000100 /* Big endian U16 */ -#endif - -#ifndef AFMT_MPEG -# define AFMT_MPEG 0x00000200 /* MPEG (2) audio */ -#endif - -#ifndef AFMT_AC3 -# define AFMT_AC3 0x00000400 /* Dolby Digital AC3 */ -#endif - -/* 24 bit formats from the linux kernel */ -#ifndef AFMT_S24_LE - -// FreeBSD fix... -#if AFMT_S32_LE == 0x1000 - -#define AFMT_S24_LE 0x00010000 -#define AFMT_S24_BE 0x00020000 -#define AFMT_U24_LE 0x00040000 -#define AFMT_U24_BE 0x00080000 - -#else - -#define AFMT_S24_LE 0x00000800 -#define AFMT_S24_BE 0x00001000 -#define AFMT_U24_LE 0x00002000 -#define AFMT_U24_BE 0x00004000 - -#endif - -#endif - -/* 32 bit formats from the linux kernel */ -#ifndef AFMT_S32_LE -#define AFMT_S32_LE 0x00008000 -#define AFMT_S32_BE 0x00010000 -#define AFMT_U32_LE 0x00020000 -#define AFMT_U32_BE 0x00040000 -#endif - -/* native endian formats */ -#ifndef AFMT_S16_NE -# if WORDS_BIGENDIAN -# define AFMT_S16_NE AFMT_S16_BE -# define AFMT_S24_NE AFMT_S24_BE -# define AFMT_S32_NE AFMT_S32_BE -# else -# define AFMT_S16_NE AFMT_S16_LE -# define AFMT_S24_NE AFMT_S24_LE -# define AFMT_S32_NE AFMT_S32_LE -# endif -#endif - -#ifndef AFMT_FLOAT -# define AFMT_FLOAT 0x00100000 -#endif - -/* for formats that don't have a corresponding AFMT_* type, - * use the flags from libaf/af_format.h or'ed with this */ -#define AFMT_AF_FLAGS 0x70000000
--- a/libao2/ao_alsa.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_alsa.c Mon Dec 27 17:30:15 2004 +0000 @@ -37,7 +37,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = { @@ -125,7 +125,7 @@ } if(mixer_device) card = mixer_device; - if(ao_data.format == AFMT_AC3) + if(ao_data.format == AF_FORMAT_AC3) return CONTROL_TRUE; //allocate simple id @@ -275,42 +275,42 @@ switch (format) { - case AFMT_S8: + case AF_FORMAT_S8: alsa_format = SND_PCM_FORMAT_S8; break; - case AFMT_U8: + case AF_FORMAT_U8: alsa_format = SND_PCM_FORMAT_U8; break; - case AFMT_U16_LE: + case AF_FORMAT_U16_LE: alsa_format = SND_PCM_FORMAT_U16_LE; break; - case AFMT_U16_BE: + case AF_FORMAT_U16_BE: alsa_format = SND_PCM_FORMAT_U16_BE; break; #ifndef WORDS_BIGENDIAN - case AFMT_AC3: + case AF_FORMAT_AC3: #endif - case AFMT_S16_LE: + case AF_FORMAT_S16_LE: alsa_format = SND_PCM_FORMAT_S16_LE; break; #ifdef WORDS_BIGENDIAN - case AFMT_AC3: + case AF_FORMAT_AC3: #endif - case AFMT_S16_BE: + case AF_FORMAT_S16_BE: alsa_format = SND_PCM_FORMAT_S16_BE; break; - case AFMT_S32_LE: + case AF_FORMAT_S32_LE: alsa_format = SND_PCM_FORMAT_S32_LE; break; - case AFMT_S32_BE: + case AF_FORMAT_S32_BE: alsa_format = SND_PCM_FORMAT_S32_BE; break; - case AFMT_FLOAT: + case AF_FORMAT_FLOAT_LE: alsa_format = SND_PCM_FORMAT_FLOAT_LE; break; default: - alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1 + alsa_format = SND_PCM_FORMAT_MPEG2; //? default should be -1 break; } @@ -412,7 +412,7 @@ * while opening the abstract alias for the spdif subdevice * 'iec958' */ - if (format == AFMT_AC3) { + if (format == AF_FORMAT_AC3) { unsigned char s[4]; switch (channels) { @@ -590,7 +590,7 @@ "alsa-init: format %s are not supported by hardware, trying default\n", audio_out_format_name(format)); alsa_format = SND_PCM_FORMAT_S16_LE; - ao_data.format = AFMT_S16_LE; + ao_data.format = AF_FORMAT_S16_LE; ao_data.bps = channels * rate_hz * 2; }
--- a/libao2/ao_alsa5.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_alsa5.c Mon Dec 27 17:30:15 2004 +0000 @@ -13,7 +13,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "mp_msg.h" #include "help_mp.h" @@ -75,28 +75,28 @@ memset(&alsa_format, 0, sizeof(alsa_format)); switch (format) { - case AFMT_S8: + case AF_FORMAT_S8: alsa_format.format = SND_PCM_SFMT_S8; break; - case AFMT_U8: + case AF_FORMAT_U8: alsa_format.format = SND_PCM_SFMT_U8; break; - case AFMT_U16_LE: + case AF_FORMAT_U16_LE: alsa_format.format = SND_PCM_SFMT_U16_LE; break; - case AFMT_U16_BE: + case AF_FORMAT_U16_BE: alsa_format.format = SND_PCM_SFMT_U16_BE; break; #ifndef WORDS_BIGENDIAN - case AFMT_AC3: + case AF_FORMAT_AC3: #endif - case AFMT_S16_LE: + case AF_FORMAT_S16_LE: alsa_format.format = SND_PCM_SFMT_S16_LE; break; #ifdef WORDS_BIGENDIAN - case AFMT_AC3: + case AF_FORMAT_AC3: #endif - case AFMT_S16_BE: + case AF_FORMAT_S16_BE: alsa_format.format = SND_PCM_SFMT_S16_BE; break; default:
--- a/libao2/ao_arts.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_arts.c Mon Dec 27 17:30:15 2004 +0000 @@ -12,12 +12,12 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "config.h" #include "mp_msg.h" #include "help_mp.h" -#define OBTAIN_BITRATE(a) (((a != AFMT_U8) && (a != AFMT_S8)) ? 16 : 8) +#define OBTAIN_BITRATE(a) (((a != AF_FORMAT_U8) && (a != AF_FORMAT_S8)) ? 16 : 8) /* Feel free to experiment with the following values: */ #define ARTS_PACKETS 10 /* Number of audio packets */ @@ -60,12 +60,12 @@ * using mplayer's audio filters. */ switch (format) { - case AFMT_U8: - case AFMT_S8: - format = AFMT_U8; + case AF_FORMAT_U8: + case AF_FORMAT_S8: + format = AF_FORMAT_U8; break; default: - format = AFMT_S16_LE; /* artsd always expects little endian?*/ + format = AF_FORMAT_S16_LE; /* artsd always expects little endian?*/ break; } @@ -74,7 +74,7 @@ ao_data.samplerate = rate_hz; ao_data.bps = (rate_hz*channels); - if(format != AFMT_U8 && format != AFMT_S8) + if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) ao_data.bps*=2; stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer");
--- a/libao2/ao_dsound.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_dsound.c Mon Dec 27 17:30:15 2004 +0000 @@ -28,7 +28,7 @@ #define DIRECTSOUND_VERSION 0x0600 #include <dsound.h> -#include "afmt.h" +#include "libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" #include "mp_msg.h" @@ -265,7 +265,7 @@ if (SUCCEEDED(res)) { - if( (ao_data.channels == 6) && (ao_data.format!=AFMT_AC3) ) { + if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) { // reorder channels while writing to pointers. // it's this easy because buffer size and len are always // aligned to multiples of channels*bytespersample @@ -366,14 +366,14 @@ //check if the format is supported in general switch(format){ - case AFMT_AC3: - case AFMT_S24_LE: - case AFMT_S16_LE: - case AFMT_S8: + case AF_FORMAT_AC3: + case AF_FORMAT_S24_LE: + case AF_FORMAT_S16_LE: + 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)); - format=AFMT_S16_LE; + format=AF_FORMAT_S16_LE; } //fill global ao_data ao_data.channels = channels; @@ -389,7 +389,7 @@ wformat.Format.cbSize = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0; wformat.Format.nChannels = channels; wformat.Format.nSamplesPerSec = rate; - if (format == AFMT_AC3) { + if (format == AF_FORMAT_AC3) { wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; wformat.Format.wBitsPerSample = 16; wformat.Format.nBlockAlign = 4;
--- a/libao2/ao_dxr2.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_dxr2.c Mon Dec 27 17:30:15 2004 +0000 @@ -11,8 +11,7 @@ #include "audio_out.h" #include "audio_out_internal.h" - -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = @@ -158,9 +157,9 @@ // return: number of bytes played static int play(void* data,int len,int flags){ // MPEG and AC3 don't work :-( - if(ao_data.format==AFMT_MPEG) + if(ao_data.format==AF_FORMAT_MPEG2) dxr2_send_packet(data,len,0xC0,ao_data.pts); - else if(ao_data.format==AFMT_AC3) + else if(ao_data.format==AF_FORMAT_AC3) dxr2_send_packet(data,len,0x80,ao_data.pts); else { int i;
--- a/libao2/ao_esd.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_esd.c Mon Dec 27 17:30:15 2004 +0000 @@ -32,7 +32,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "config.h" #include "mp_msg.h" #include "help_mp.h" @@ -200,14 +200,14 @@ /* EsounD can play 8bit unsigned and 16bit signed native */ switch (format) { - case AFMT_S8: - case AFMT_U8: + case AF_FORMAT_S8: + case AF_FORMAT_U8: esd_fmt |= ESD_BITS8; - ao_data.format = AFMT_U8; + ao_data.format = AF_FORMAT_U8; break; default: esd_fmt |= ESD_BITS16; - ao_data.format = AFMT_S16_NE; + ao_data.format = AF_FORMAT_S16_NE; bytes_per_sample *= 2; break; }
--- a/libao2/ao_jack.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_jack.c Mon Dec 27 17:30:15 2004 +0000 @@ -14,7 +14,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "config.h" #include "mp_msg.h" @@ -146,14 +146,14 @@ } switch (format) { - case AFMT_U8: - case AFMT_S8: - format = AFMT_U8; + case AF_FORMAT_U8: + case AF_FORMAT_S8: + format = AF_FORMAT_U8; bits_per_sample = 8; m = 1; break; default: - format = AFMT_S16_LE; + format = AF_FORMAT_S16_LE; bits_per_sample = 16; m = 2; break;
--- a/libao2/ao_macosx.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_macosx.c Mon Dec 27 17:30:15 2004 +0000 @@ -47,7 +47,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = { @@ -252,7 +252,7 @@ if (ao->outputStreamBasicDescription.mFormatID == kAudioFormatLinearPCM) { uint32_t flags = ao->outputStreamBasicDescription.mFormatFlags; if (flags & kAudioFormatFlagIsFloat) { - ao_data.format = AFMT_FLOAT; + ao_data.format = AF_FORMAT_FLOAT_NE; } else { ao_msg(MSGT_AO,MSGL_WARN, "Unsupported audio output " "format %d. Please report this to the developer\n",
--- a/libao2/ao_mpegpes.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_mpegpes.c Mon Dec 27 17:30:15 2004 +0000 @@ -15,7 +15,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "mp_msg.h" #include "help_mp.h" @@ -96,14 +96,14 @@ ao_data.channels=2; ao_data.outburst=2000; switch(format){ - case AFMT_S16_LE: - case AFMT_S16_BE: - case AFMT_MPEG: - case AFMT_AC3: + case AF_FORMAT_S16_LE: + case AF_FORMAT_S16_BE: + case AF_FORMAT_MPEG2: + case AF_FORMAT_AC3: ao_data.format=format; break; default: - ao_data.format=AFMT_S16_BE; + ao_data.format=AF_FORMAT_S16_BE; } retry: @@ -174,14 +174,14 @@ // return: number of bytes played static int play(void* data,int len,int flags){ // printf("\nao_mpegpes: play(%d) freq=%d\n",len,freq_id); - if(ao_data.format==AFMT_MPEG) + if(ao_data.format==AF_FORMAT_MPEG2) send_pes_packet(data,len,0x1C0,ao_data.pts); else { int i; unsigned short *s=data; // if(len>2000) len=2000; // printf("ao_mpegpes: len=%d \n",len); - if(ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_AC3) + if(ao_data.format==AF_FORMAT_S16_LE || ao_data.format==AF_FORMAT_AC3) for(i=0;i<len/2;i++) s[i]=(s[i]>>8)|(s[i]<<8); // le<->be send_lpcm_packet(data,len,0xA0,ao_data.pts,freq_id); }
--- a/libao2/ao_nas.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_nas.c Mon Dec 27 17:30:15 2004 +0000 @@ -37,7 +37,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #define NAS_FRAG_SIZE 4096 @@ -312,22 +312,22 @@ static unsigned int nas_aformat_to_auformat(unsigned int *format) { switch (*format) { - case AFMT_U8: + case AF_FORMAT_U8: return AuFormatLinearUnsigned8; - case AFMT_S8: + case AF_FORMAT_S8: return AuFormatLinearSigned8; - case AFMT_U16_LE: + case AF_FORMAT_U16_LE: return AuFormatLinearUnsigned16LSB; - case AFMT_U16_BE: + case AF_FORMAT_U16_BE: return AuFormatLinearUnsigned16MSB; - case AFMT_S16_LE: + case AF_FORMAT_S16_LE: return AuFormatLinearSigned16LSB; - case AFMT_S16_BE: + case AF_FORMAT_S16_BE: return AuFormatLinearSigned16MSB; - case AFMT_MU_LAW: + case AF_FORMAT_MU_LAW: return AuFormatULAW8; default: - *format=AFMT_S16_NE; + *format=AF_FORMAT_S16_NE; return nas_aformat_to_auformat(format); } }
--- a/libao2/ao_null.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_null.c Mon Dec 27 17:30:15 2004 +0000 @@ -2,7 +2,7 @@ #include <stdlib.h> #include <sys/time.h> -#include "afmt.h" +#include "libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" @@ -55,7 +55,7 @@ ao_data.samplerate=rate; ao_data.format=format; ao_data.bps=channels*rate; - if (format != AFMT_U8 && format != AFMT_S8) + if (format != AF_FORMAT_U8 && format != AF_FORMAT_S8) ao_data.bps*=2; buffer=0; gettimeofday(&last_tv, 0);
--- a/libao2/ao_oss.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_oss.c Mon Dec 27 17:30:15 2004 +0000 @@ -9,14 +9,21 @@ #include <fcntl.h> #include <errno.h> #include <string.h> -//#include <sys/soundcard.h> #include "config.h" #include "mp_msg.h" #include "mixer.h" #include "help_mp.h" -#include "afmt.h" +#ifdef HAVE_SYS_SOUNDCARD_H +#include <sys/soundcard.h> +#else +#ifdef HAVE_SOUNDCARD_H +#include <soundcard.h> +#endif +#endif + +#include "../libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" @@ -33,6 +40,86 @@ LIBAO_EXTERN(oss) +static int format2oss(int format) +{ + switch(format) + { + case AF_FORMAT_U8: return AFMT_U8; + case AF_FORMAT_S8: return AFMT_S8; + case AF_FORMAT_U16_LE: return AFMT_U16_LE; + case AF_FORMAT_U16_BE: return AFMT_U16_BE; + case AF_FORMAT_S16_LE: return AFMT_S16_LE; + case AF_FORMAT_S16_BE: return AFMT_S16_BE; +#ifdef AFMT_S24_LE + case AF_FORMAT_U24_LE: return AFMT_U24_LE; + case AF_FORMAT_U24_BE: return AFMT_U24_BE; + case AF_FORMAT_S24_LE: return AFMT_S24_LE; + case AF_FORMAT_S24_BE: return AFMT_S24_BE; +#endif +#ifdef AFMT_S32_LE + case AF_FORMAT_U32_LE: return AFMT_U32_LE; + case AF_FORMAT_U32_BE: return AFMT_U32_BE; + case AF_FORMAT_S32_LE: return AFMT_S32_LE; + case AF_FORMAT_S32_BE: return AFMT_S32_BE; +#endif +#ifdef AFMT_FLOAT + case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT; +#endif + // SPECIALS + case AF_FORMAT_MU_LAW: return AFMT_MU_LAW; + case AF_FORMAT_A_LAW: return AFMT_A_LAW; + case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM; +#ifdef AFMT_MPEG + case AF_FORMAT_MPEG2: return AFMT_MPEG; +#endif +#ifdef AFMT_AC3 + case AF_FORMAT_AC3: return AFMT_AC3; +#endif + } + printf("Unknown format: %x\n", format); + return -1; +} + +static int oss2format(int format) +{ + switch(format) + { + case AFMT_U8: return AF_FORMAT_U8; + case AFMT_S8: return AF_FORMAT_S8; + case AFMT_U16_LE: return AF_FORMAT_U16_LE; + case AFMT_U16_BE: return AF_FORMAT_U16_BE; + case AFMT_S16_LE: return AF_FORMAT_S16_LE; + case AFMT_S16_BE: return AF_FORMAT_S16_BE; +#ifdef AFMT_S24_LE + case AFMT_U24_LE: return AF_FORMAT_U24_LE; + case AFMT_U24_BE: return AF_FORMAT_U24_BE; + case AFMT_S24_LE: return AF_FORMAT_S24_LE; + case AFMT_S24_BE: return AF_FORMAT_S24_BE; +#endif +#ifdef AFMT_S32_LE + case AFMT_U32_LE: return AF_FORMAT_U32_LE; + case AFMT_U32_BE: return AF_FORMAT_U32_BE; + case AFMT_S32_LE: return AF_FORMAT_S32_LE; + case AFMT_S32_BE: return AF_FORMAT_S32_BE; +#endif +#ifdef AFMT_FLOAT + case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE; +#endif + // SPECIALS + case AFMT_MU_LAW: return AF_FORMAT_MU_LAW; + case AFMT_A_LAW: return AF_FORMAT_A_LAW; + case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM; +#ifdef AFMT_MPEG + case AFMT_MPEG: return AF_FORMAT_MPEG2; +#endif +#ifdef AFMT_AC3 + case AFMT_AC3: return AF_FORMAT_AC3; +#endif + } + printf("Unknown format: %x\n", format); + return -1; +} + static char *dsp=PATH_DEV_DSP; static audio_buf_info zz; static int audio_fd=-1; @@ -57,7 +144,7 @@ ao_control_vol_t *vol = (ao_control_vol_t *)arg; int fd, v, devs; - if(ao_data.format == AFMT_AC3) + if(ao_data.format == AF_FORMAT_AC3) return CONTROL_TRUE; if ((fd = open(oss_mixer_device, O_RDONLY)) > 0) @@ -95,9 +182,10 @@ // return: 1=success 0=fail 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, +// audio_out_format_name(format)); if (ao_subdevice) dsp = ao_subdevice; @@ -160,32 +248,39 @@ fcntl(audio_fd, F_SETFD, FD_CLOEXEC); #endif - if(format == AFMT_AC3) { + if(format == AF_FORMAT_AC3) { ao_data.samplerate=rate; ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); } ac3_retry: ao_data.format=format; - if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format)<0 || - ao_data.format != format) if(format == AFMT_AC3){ + oss_format=format2oss(format); + if (oss_format == -1) return 0; + if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 || + oss_format != format2oss(format)) if(format == AF_FORMAT_AC3){ mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSetAC3, dsp); #ifdef WORDS_BIGENDIAN - format=AFMT_S16_BE; + oss_format=AFMT_S16_BE; + format=AF_FORMAT_S16_BE; #else - format=AFMT_S16_LE; + oss_format=AFMT_S16_LE; + format=AF_FORMAT_S16_LE; #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)); +// 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(ao_data.format!=format) + 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)); #endif + + ao_data.format = oss2format(oss_format); + if (ao_data.format == -1) return 0; ao_data.channels = channels; - if(format != AFMT_AC3) { + if(format != AF_FORMAT_AC3) { // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it if (ao_data.channels > 2) { if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 || @@ -253,7 +348,7 @@ } ao_data.bps=ao_data.channels; - if(ao_data.format != AFMT_U8 && ao_data.format != AFMT_S8) + if(ao_data.format != AF_FORMAT_U8 && ao_data.format != AF_FORMAT_S8) ao_data.bps*=2; ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down @@ -280,6 +375,7 @@ // stop playing and empty buffers (for seeking/pause) static void reset(){ + int oss_format; uninit(1); audio_fd=open(dsp, O_WRONLY); if(audio_fd < 0){ @@ -291,8 +387,9 @@ fcntl(audio_fd, F_SETFD, FD_CLOEXEC); #endif - ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format); - if(ao_data.format != AFMT_AC3) { + oss_format = format2oss(ao_data.format); + ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format); + if(ao_data.format != AF_FORMAT_AC3) { if (ao_data.channels > 2) ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels); else {
--- a/libao2/ao_pcm.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_pcm.c Mon Dec 27 17:30:15 2004 +0000 @@ -5,7 +5,7 @@ #include <string.h> #include "bswap.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" #include "mp_msg.h" @@ -89,12 +89,12 @@ bits=8; switch(format){ - case AFMT_S8: - format=AFMT_U8; - case AFMT_U8: + case AF_FORMAT_S8: + format=AF_FORMAT_U8; + case AF_FORMAT_U8: break; default: - format=AFMT_S16_LE; + format=AF_FORMAT_S16_LE; bits=16; break; } @@ -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", audio_out_format_name(format)); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo); fp = fopen(ao_outputfilename, "wb");
--- a/libao2/ao_plugin.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_plugin.c Mon Dec 27 17:30:15 2004 +0000 @@ -6,7 +6,7 @@ #include "mp_msg.h" #include "help_mp.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" @@ -140,7 +140,7 @@ ao_plugin_local_data.format=format; ao_plugin_local_data.channels=channels; - ao_plugin_local_data.bpm=audio_out_format_bits(format); + ao_plugin_local_data.bpm=af_fmt2bits(format); ao_plugin_data.rate=rate; ao_plugin_data.channels=channels; @@ -168,7 +168,7 @@ // Calculate bps ao_plugin_local_data.bps=(float)(ao_plugin_data.rate * ao_plugin_data.channels); - ao_plugin_local_data.bps*=audio_out_format_bits(ao_plugin_data.format)/8; + ao_plugin_local_data.bps*=af_fmt2bits(ao_plugin_data.format)/8; // This should never happen but check anyway if(NULL==ao_plugin_local_data.driver)
--- a/libao2/ao_polyp.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_polyp.c Mon Dec 27 17:30:15 2004 +0000 @@ -6,7 +6,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "config.h" #include "mp_msg.h" @@ -91,16 +91,16 @@ ss.rate = rate_hz; switch (format) { - case AFMT_U8: + case AF_FORMAT_U8: ss.format = PA_SAMPLE_U8; break; - case AFMT_S16_LE: + case AF_FORMAT_S16_LE: ss.format = PA_SAMPLE_S16LE; break; - case AFMT_S16_BE: + case AF_FORMAT_S16_BE: ss.format = PA_SAMPLE_S16BE; break; - case AFMT_FLOAT: + case AF_FORMAT_FLOAT_NE: ss.format = PA_SAMPLE_FLOAT32; break; default:
--- a/libao2/ao_sdl.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_sdl.c Mon Dec 27 17:30:15 2004 +0000 @@ -20,7 +20,7 @@ #include "audio_out.h" #include "audio_out_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include <SDL.h> #include "osdep/timer.h" @@ -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", audio_out_format_name(format)); if(ao_subdevice) { setenv("SDL_AUDIODRIVER", ao_subdevice, 1); @@ -193,32 +193,32 @@ ao_data.format=format; ao_data.bps=channels*rate; - if(format != AFMT_U8 && format != AFMT_S8) + if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) ao_data.bps*=2; /* The desired audio format (see SDL_AudioSpec) */ switch(format) { - case AFMT_U8: + case AF_FORMAT_U8: aspec.format = AUDIO_U8; break; - case AFMT_S16_LE: + case AF_FORMAT_S16_LE: aspec.format = AUDIO_S16LSB; break; - case AFMT_S16_BE: + case AF_FORMAT_S16_BE: aspec.format = AUDIO_S16MSB; break; - case AFMT_S8: + case AF_FORMAT_S8: aspec.format = AUDIO_S8; break; - case AFMT_U16_LE: + case AF_FORMAT_U16_LE: aspec.format = AUDIO_U16LSB; break; - case AFMT_U16_BE: + case AF_FORMAT_U16_BE: aspec.format = AUDIO_U16MSB; break; default: aspec.format = AUDIO_S16LSB; - ao_data.format = AFMT_S16_LE; + ao_data.format = AF_FORMAT_S16_LE; mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, format); } @@ -256,22 +256,22 @@ switch(obtained.format) { case AUDIO_U8 : - ao_data.format = AFMT_U8; + ao_data.format = AF_FORMAT_U8; break; case AUDIO_S16LSB : - ao_data.format = AFMT_S16_LE; + ao_data.format = AF_FORMAT_S16_LE; break; case AUDIO_S16MSB : - ao_data.format = AFMT_S16_BE; + ao_data.format = AF_FORMAT_S16_BE; break; case AUDIO_S8 : - ao_data.format = AFMT_S8; + ao_data.format = AF_FORMAT_S8; break; case AUDIO_U16LSB : - ao_data.format = AFMT_U16_LE; + ao_data.format = AF_FORMAT_U16_LE; break; case AUDIO_U16MSB : - ao_data.format = AFMT_U16_BE; + ao_data.format = AF_FORMAT_U16_BE; break; default: mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, obtained.format);
--- a/libao2/ao_win32.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/ao_win32.c Mon Dec 27 17:30:15 2004 +0000 @@ -23,7 +23,7 @@ #include <windows.h> #include <mmsystem.h> -#include "afmt.h" +#include "libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" #include "mp_msg.h" @@ -149,21 +149,21 @@ int i; switch(format){ - case AFMT_AC3: - case AFMT_S24_LE: - case AFMT_S16_LE: - case AFMT_S8: + case AF_FORMAT_AC3: + case AF_FORMAT_S24_LE: + case AF_FORMAT_S16_LE: + 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)); - format=AFMT_S16_LE; + 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; - if(format != AFMT_U8 && format != AFMT_S8) + if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) ao_data.bps*=2; if(ao_data.buffersize==-1) { @@ -179,7 +179,7 @@ wformat.Format.cbSize = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0; wformat.Format.nChannels = channels; wformat.Format.nSamplesPerSec = rate; - if(format == AFMT_AC3) + if(format == AF_FORMAT_AC3) { wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; wformat.Format.wBitsPerSample = 16; @@ -208,7 +208,7 @@ mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n"); ao_data.channels = wformat.Format.nChannels = 2; ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100; - ao_data.format = AFMT_S16_LE; + ao_data.format = AF_FORMAT_S16_LE; ao_data.bps=ao_data.channels * ao_data.samplerate*2; wformat.Format.wBitsPerSample=16; wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
--- a/libao2/audio_out.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/audio_out.c Mon Dec 27 17:30:15 2004 +0000 @@ -4,7 +4,6 @@ #include "config.h" #include "audio_out.h" -#include "afmt.h" #include "mp_msg.h" #include "help_mp.h" @@ -187,6 +186,7 @@ audio_out_plugin.control(AOCONTROL_SET_PLUGIN_DRIVER,audio_out); audio_out=&audio_out_plugin; } +// if(audio_out->control(AOCONTROL_QUERY_FORMAT, (int)format) == CONTROL_TRUE) if(audio_out->init(rate,channels,format,flags)) return audio_out; // success! }
--- a/libao2/audio_plugin.h Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/audio_plugin.h Mon Dec 27 17:30:15 2004 +0000 @@ -45,7 +45,7 @@ // Configuration defaults #define CFG_DEFAULTS { \ NULL, \ - AFMT_S16_LE, \ + AF_FORMAT_S16_LE, \ 0, \ 48000, \ 101, \
--- a/libao2/pl_delay.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/pl_delay.c Mon Dec 27 17:30:15 2004 +0000 @@ -12,7 +12,7 @@ #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = { @@ -65,7 +65,7 @@ // Tell ao_plugin how much this plugin adds to the overall time delay time_delay=-1*(float)ao_plugin_cfg.pl_delay_len/((float)pl_delay.channels*(float)pl_delay.rate); - if(pl_delay.format != AFMT_U8 && pl_delay.format != AFMT_S8) + if(pl_delay.format != AF_FORMAT_U8 && pl_delay.format != AF_FORMAT_S8) time_delay/=2; ao_plugin_data.delay_fix+=time_delay;
--- a/libao2/pl_eq.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/pl_eq.c Mon Dec 27 17:30:15 2004 +0000 @@ -24,7 +24,7 @@ #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "eq.h" static ao_info_t info = @@ -122,7 +122,7 @@ float F[KM] = CF; // Check input format - if(ao_plugin_data.format != AFMT_S16_NE){ + if(ao_plugin_data.format != AF_FORMAT_S16_NE){ fprintf(stderr,"[pl_eq] Input audio format not yet supported. \n"); return 0; }
--- a/libao2/pl_extrastereo.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/pl_extrastereo.c Mon Dec 27 17:30:15 2004 +0000 @@ -19,7 +19,7 @@ #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = { "Extra stereo plugin", @@ -57,7 +57,7 @@ // return: 1=success 0=fail static int init(){ switch(ao_plugin_data.format){ - case(AFMT_S16_NE): + case(AF_FORMAT_S16_NE): break; default: fprintf(stderr,"[pl_extrastereo] Audio format not yet suported \n"); @@ -87,7 +87,7 @@ static int play(){ switch(pl_extrastereo.format){ - case(AFMT_S16_NE): { + case(AF_FORMAT_S16_NE): { int16_t* data=(int16_t*)ao_plugin_data.data; int len=ao_plugin_data.len / 2; // 16 bits samples
--- a/libao2/pl_format.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/pl_format.c Mon Dec 27 17:30:15 2004 +0000 @@ -16,7 +16,7 @@ #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = { @@ -75,29 +75,30 @@ // open & setup audio device // return: 1=success 0=fail static int init(){ + char buf1[128], buf2[128]; // Sheck input format switch(ao_plugin_data.format){ - case(AFMT_U8): + case(AF_FORMAT_U8): pl_format.in=LE|B08|US; break; - case(AFMT_S8): + case(AF_FORMAT_S8): pl_format.in=LE|B08|SI; break; - case(AFMT_S16_LE): + case(AF_FORMAT_S16_LE): pl_format.in=LE|B16|SI; break; - case(AFMT_S16_BE): + case(AF_FORMAT_S16_BE): pl_format.in=BE|B16|SI; break; - case(AFMT_U16_LE): + case(AF_FORMAT_U16_LE): pl_format.in=LE|B16|US; break; - case(AFMT_U16_BE): + case(AF_FORMAT_U16_BE): pl_format.in=BE|B16|US; break; - case(AFMT_S32_LE): + case(AF_FORMAT_S32_LE): pl_format.in=LE|B32|SI; break; - case(AFMT_S32_BE): + case(AF_FORMAT_S32_BE): pl_format.in=BE|B32|SI; break; - case(AFMT_IMA_ADPCM): - case(AFMT_MU_LAW): - case(AFMT_A_LAW): - case(AFMT_MPEG): - case(AFMT_AC3): + case(AF_FORMAT_IMA_ADPCM): + case(AF_FORMAT_MU_LAW): + case(AF_FORMAT_A_LAW): + case(AF_FORMAT_MPEG2): + case(AF_FORMAT_AC3): printf("[pl_format] Input audio format not yet suported \n"); return 0; default: @@ -106,27 +107,27 @@ } // Sheck output format switch(ao_plugin_cfg.pl_format_type){ - case(AFMT_U8): + case(AF_FORMAT_U8): pl_format.out=LE|B08|US; break; - case(AFMT_S8): + case(AF_FORMAT_S8): pl_format.out=LE|B08|SI; break; - case(AFMT_S16_LE): + case(AF_FORMAT_S16_LE): pl_format.out=LE|B16|SI; break; - case(AFMT_S16_BE): + case(AF_FORMAT_S16_BE): pl_format.out=BE|B16|SI; break; - case(AFMT_U16_LE): + case(AF_FORMAT_U16_LE): pl_format.out=LE|B16|US; break; - case(AFMT_U16_BE): + case(AF_FORMAT_U16_BE): pl_format.out=BE|B16|US; break; - case(AFMT_S32_LE): + case(AF_FORMAT_S32_LE): pl_format.out=LE|B32|SI; break; - case(AFMT_S32_BE): + case(AF_FORMAT_S32_BE): pl_format.out=BE|B32|SI; break; - case(AFMT_IMA_ADPCM): - case(AFMT_MU_LAW): - case(AFMT_A_LAW): - case(AFMT_MPEG): - case(AFMT_AC3): + case(AF_FORMAT_IMA_ADPCM): + case(AF_FORMAT_MU_LAW): + case(AF_FORMAT_A_LAW): + case(AF_FORMAT_MPEG2): + case(AF_FORMAT_AC3): printf("[pl_format] Output audio format not yet suported \n"); return 0; default: @@ -136,8 +137,8 @@ // Tell the world what we are up to printf("[pl_format] Input format: %s, output format: %s \n", - audio_out_format_name(ao_plugin_data.format), - audio_out_format_name(ao_plugin_cfg.pl_format_type)); + af_fmt2str(ao_plugin_data.format, &buf1, 128), + af_fmt2str(ao_plugin_cfg.pl_format_type, &buf2, 128)); // We are changing the format ao_plugin_data.format=ao_plugin_cfg.pl_format_type;
--- a/libao2/pl_resample.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/pl_resample.c Mon Dec 27 17:30:15 2004 +0000 @@ -23,7 +23,7 @@ #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = { @@ -119,7 +119,7 @@ pl_resample.up=UP; // Sheck input format - if(ao_plugin_data.format != AFMT_S16_NE){ + if(ao_plugin_data.format != AF_FORMAT_S16_NE){ fprintf(stderr,"[pl_resample] Input audio format not yet suported. \n"); return 0; }
--- a/libao2/pl_surround.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/pl_surround.c Mon Dec 27 17:30:15 2004 +0000 @@ -44,7 +44,7 @@ #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" #include "remez.h" #include "firfilter.c" @@ -108,8 +108,8 @@ pl_surround.passthrough = 1; return 1; } - if (ao_plugin_data.format != AFMT_S16_NE) { - fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_NE audio format, using passthrough mode\n"); + if (ao_plugin_data.format != AF_FORMAT_S16_NE) { + fprintf(stderr, "pl_surround: I'm dumb and can only handle AF_FORMAT_S16_NE audio format, using passthrough mode\n"); pl_surround.passthrough = 1; return 1; }
--- a/libao2/pl_volnorm.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/pl_volnorm.c Mon Dec 27 17:30:15 2004 +0000 @@ -30,7 +30,7 @@ #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = { "Volume normalizer", @@ -116,7 +116,7 @@ // return: 1=success 0=fail static int init(){ switch(ao_plugin_data.format){ - case(AFMT_S16_NE): + case(AF_FORMAT_S16_NE): break; default: fprintf(stderr,"[pl_volnorm] Audio format not yet supported.\n"); @@ -142,7 +142,7 @@ int i; mul = MUL_INIT; switch(ao_plugin_data.format) { - case(AFMT_S16_NE): + case(AF_FORMAT_S16_NE): #if AVG==1 lastavg = MID_S16; #elif AVG==2 @@ -165,7 +165,7 @@ static int play(){ switch(pl_volnorm.format){ - case(AFMT_S16_NE): { + case(AF_FORMAT_S16_NE): { #define CLAMP(x,m,M) do { if ((x)<(m)) (x) = (m); else if ((x)>(M)) (x) = (M); } while(0) int16_t* data=(int16_t*)ao_plugin_data.data;
--- a/libao2/pl_volume.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libao2/pl_volume.c Mon Dec 27 17:30:15 2004 +0000 @@ -20,7 +20,7 @@ #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" -#include "afmt.h" +#include "libaf/af_format.h" static ao_info_t info = { @@ -80,8 +80,8 @@ static int init(){ // Sanity sheck this plugin supports AFMT_U8 and AFMT_S16_LE switch(ao_plugin_data.format){ - case(AFMT_U8): - case(AFMT_S16_NE): + case(AF_FORMAT_U8): + case(AF_FORMAT_S16_NE): break; default: fprintf(stderr,"[pl_volume] Audio format not yet suported \n"); @@ -117,7 +117,7 @@ vol=(vol*vol*vol)>>12; // Change the volume. switch(pl_volume.format){ - case(AFMT_U8):{ + case(AF_FORMAT_U8):{ register uint8_t* data=(uint8_t*)ao_plugin_data.data; register int x; for(i=0;i<ao_plugin_data.len;i++){ @@ -135,7 +135,7 @@ } break; } - case(AFMT_S16_NE):{ + case(AF_FORMAT_S16_NE):{ register int len=ao_plugin_data.len>>1; register int16_t* data=(int16_t*)ao_plugin_data.data; register int x;
--- a/libmpcodecs/ad_dvdpcm.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpcodecs/ad_dvdpcm.c Mon Dec 27 17:30:15 2004 +0000 @@ -6,7 +6,6 @@ #include "mp_msg.h" #include "help_mp.h" #include "ad_internal.h" -#include "../libaf/af_format.h" static ad_info_t info = { @@ -35,26 +34,25 @@ } switch ((h >> 6) & 3) { case 0: - sh->sample_format = AFMT_S16_BE; + sh->sample_format = AF_FORMAT_S16_BE; sh->samplesize = 2; break; case 1: mp_msg(MSGT_DECAUDIO, MSGL_INFO, MSGTR_SamplesWanted); sh->i_bps = sh->channels * sh->samplerate * 5 / 2; case 2: - sh->sample_format = AFMT_AF_FLAGS | AF_FORMAT_I | - AF_FORMAT_BE | AF_FORMAT_SI; + sh->sample_format = AF_FORMAT_S24_BE; sh->samplesize = 3; break; default: - sh->sample_format = AFMT_S16_BE; + sh->sample_format = AF_FORMAT_S16_BE; sh->samplesize = 2; } } else { // use defaults: sh->channels=2; sh->samplerate=48000; - sh->sample_format = AFMT_S16_BE; + sh->sample_format = AF_FORMAT_S16_BE; sh->samplesize = 2; } if (!sh->i_bps)
--- a/libmpcodecs/ad_hwac3.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpcodecs/ad_hwac3.c Mon Dec 27 17:30:15 2004 +0000 @@ -105,7 +105,7 @@ sh->audio_in_minsize = 8192; sh->channels = 2; sh->samplesize = 2; - sh->sample_format = AFMT_AC3; + sh->sample_format = AF_FORMAT_AC3; return 1; }
--- a/libmpcodecs/ad_internal.h Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpcodecs/ad_internal.h Mon Dec 27 17:30:15 2004 +0000 @@ -1,6 +1,6 @@ #include "codec-cfg.h" -#include "../libao2/afmt.h" +#include "../libaf/af_format.h" #include "stream.h" #include "demuxer.h"
--- a/libmpcodecs/ad_pcm.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpcodecs/ad_pcm.c Mon Dec 27 17:30:15 2004 +0000 @@ -24,31 +24,31 @@ sh_audio->channels=h->nChannels; sh_audio->samplerate=h->nSamplesPerSec; sh_audio->samplesize=(h->wBitsPerSample+7)/8; - sh_audio->sample_format=AFMT_S16_LE; // default + sh_audio->sample_format=AF_FORMAT_S16_LE; // default switch(sh_audio->format){ /* hardware formats: */ case 0x0: case 0x1: // Microsoft PCM switch (sh_audio->samplesize) { - case 1: sh_audio->sample_format=AFMT_U8; break; - case 2: sh_audio->sample_format=AFMT_S16_LE; break; - case 3: sh_audio->sample_format=AFMT_S24_LE; break; - case 4: sh_audio->sample_format=AFMT_S32_LE; break; + case 1: sh_audio->sample_format=AF_FORMAT_U8; break; + case 2: sh_audio->sample_format=AF_FORMAT_S16_LE; break; + case 3: sh_audio->sample_format=AF_FORMAT_S24_LE; break; + case 4: sh_audio->sample_format=AF_FORMAT_S32_LE; break; } break; - case 0x6: sh_audio->sample_format=AFMT_A_LAW;break; - case 0x7: sh_audio->sample_format=AFMT_MU_LAW;break; - case 0x11: sh_audio->sample_format=AFMT_IMA_ADPCM;break; - case 0x50: sh_audio->sample_format=AFMT_MPEG;break; + case 0x6: sh_audio->sample_format=AF_FORMAT_A_LAW;break; + case 0x7: sh_audio->sample_format=AF_FORMAT_MU_LAW;break; + case 0x11: sh_audio->sample_format=AF_FORMAT_IMA_ADPCM;break; + case 0x50: sh_audio->sample_format=AF_FORMAT_MPEG2;break; /* case 0x2000: sh_audio->sample_format=AFMT_AC3; */ case 0x20776172: // 'raw ' - sh_audio->sample_format=AFMT_S16_BE; - if(sh_audio->samplesize==1) sh_audio->sample_format=AFMT_U8; + sh_audio->sample_format=AF_FORMAT_S16_BE; + if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_U8; break; case 0x736F7774: // 'twos' - sh_audio->sample_format=AFMT_S16_BE; + sh_audio->sample_format=AF_FORMAT_S16_BE; // intended fall-through case 0x74776F73: // 'swot' - if(sh_audio->samplesize==1) sh_audio->sample_format=AFMT_S8; + if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_S8; // Uncomment this if twos audio is broken for you // (typically with movies made on sgi machines) // This is just a workaround, the real bug is elsewhere @@ -58,10 +58,10 @@ #endif break; case 0x32336c66: // 'fl32', bigendian float32 - sh_audio->sample_format=AFMT_AF_FLAGS | AF_FORMAT_BE | AF_FORMAT_F; + sh_audio->sample_format=AF_FORMAT_FLOAT_BE; sh_audio->samplesize=4; break; - default: if(sh_audio->samplesize!=2) sh_audio->sample_format=AFMT_U8; + default: if(sh_audio->samplesize!=2) sh_audio->sample_format=AF_FORMAT_U8; } return 1; }
--- a/libmpcodecs/ad_ra1428.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpcodecs/ad_ra1428.c Mon Dec 27 17:30:15 2004 +0000 @@ -25,7 +25,7 @@ sh->samplerate=sh->wf->nSamplesPerSec; sh->samplesize=sh->wf->wBitsPerSample/8; sh->channels=sh->wf->nChannels; - sh->sample_format=AFMT_S16_LE; + sh->sample_format=AF_FORMAT_S16_LE; switch (sh->format) { case mmioFOURCC('1','4','_','4'):
--- a/libmpcodecs/ad_sample.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpcodecs/ad_sample.c Mon Dec 27 17:30:15 2004 +0000 @@ -47,7 +47,7 @@ sh->samplesize=2; // bytes (not bits!) per sample per channel sh->channels=2; // number of channels sh->samplerate=44100; // samplerate - sh->sample_format=AFMT_S16_LE; // sample format, see libao2/afmt.h + sh->sample_format=AF_FORMAT_S16_LE; // sample format, see libao2/afmt.h sh->i_bps=64000/8; // input data rate (compressed bytes per second) // Note: if you have VBR or unknown input rate, set it to some common or
--- a/libmpcodecs/dec_audio.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpcodecs/dec_audio.c Mon Dec 27 17:30:15 2004 +0000 @@ -14,7 +14,7 @@ #include "dec_audio.h" #include "ad.h" -#include "../libao2/afmt.h" +#include "../libaf/af_format.h" #include "../libaf/af.h" @@ -267,13 +267,15 @@ // input format: same as codec's output format: afs->input.rate = in_samplerate; afs->input.nch = in_channels; - afs->input.format = af_format_decode(in_format); +// afs->input.format = af_format_decode(in_format); + afs->input.format = in_format; afs->input.bps = in_bps; // output format: same as ao driver's input format (if missing, fallback to input) afs->output.rate = *out_samplerate ? *out_samplerate : afs->input.rate; afs->output.nch = *out_channels ? *out_channels : afs->input.nch; - afs->output.format = *out_format ? af_format_decode(*out_format) : afs->input.format; +// afs->output.format = *out_format ? af_format_decode(*out_format) : afs->input.format; + afs->output.format = *out_format ? *out_format : afs->input.format; afs->output.bps = out_bps ? out_bps : afs->input.bps; // filter config: @@ -291,11 +293,12 @@ *out_samplerate=afs->output.rate; *out_channels=afs->output.nch; - *out_format=af_format_encode((void*)(&afs->output)); +// *out_format=af_format_encode((void*)(&afs->output)); + *out_format=afs->output.format; mp_msg(MSGT_DECAUDIO, MSGL_INFO, "AF_pre: af format: %d bps, %d ch, %d hz, %s\n", afs->output.bps, afs->output.nch, afs->output.rate, - fmt2str(afs->output.format,strbuf,200)); + af_fmt2str(afs->output.format,strbuf,200)); sh_audio->afilter=(void*)afs; return 1; @@ -315,13 +318,15 @@ // input format: same as codec's output format: afs->input.rate = in_samplerate; afs->input.nch = in_channels; - afs->input.format = af_format_decode(in_format); +// afs->input.format = af_format_decode(in_format); + afs->input.format = in_format; afs->input.bps = in_bps; // output format: same as ao driver's input format (if missing, fallback to input) afs->output.rate = out_samplerate ? out_samplerate : afs->input.rate; afs->output.nch = out_channels ? out_channels : afs->input.nch; - afs->output.format = af_format_decode(out_format ? out_format : afs->input.format); +// afs->output.format = af_format_decode(out_format ? out_format : afs->input.format); + afs->output.format = out_format ? out_format : afs->input.format; afs->output.bps = out_bps ? out_bps : afs->input.bps; // filter config: @@ -404,7 +409,8 @@ afd.len=declen; afd.rate=sh_audio->samplerate; afd.nch=sh_audio->channels; - afd.format=af_format_decode(sh_audio->sample_format); +// afd.format=af_format_decode(sh_audio->sample_format); + afd.format=sh_audio->sample_format; afd.bps=sh_audio->samplesize; //pafd=&afd; // printf("\nAF: %d --> ",declen);
--- a/libmpdemux/demuxer.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpdemux/demuxer.c Mon Dec 27 17:30:15 2004 +0000 @@ -18,7 +18,7 @@ #include "stheader.h" #include "mf.h" -#include "../libao2/afmt.h" +#include "../libaf/af_format.h" #include "../libvo/fastmemcpy.h" // Should be set to 1 by demux module if ids it passes to new_sh_audio and @@ -91,7 +91,7 @@ sh = demuxer->a_streams[id]; // set some defaults sh->samplesize=2; - sh->sample_format=AFMT_S16_NE; + sh->sample_format=AF_FORMAT_S16_NE; sh->audio_out_minsize=8192;/* default size, maybe not enough for Win32/ACM*/ if (identify && !demux_aid_vid_mismatch) mp_msg(MSGT_GLOBAL, MSGL_INFO, "ID_AUDIO_ID=%d\n", id);
--- a/libmpdemux/tv.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpdemux/tv.c Mon Dec 27 17:30:15 2004 +0000 @@ -29,8 +29,7 @@ #include "demuxer.h" #include "stheader.h" -#include "../libao2/afmt.h" -#include "../libao2/audio_out.h" +#include "../libaf/af_format.h" #include "../libvo/img_format.h" #include "../libvo/fastmemcpy.h" @@ -516,6 +515,7 @@ { int audio_format; int sh_audio_format; + char buf[128]; /* yeah, audio is present */ @@ -527,24 +527,24 @@ switch(audio_format) { - case AFMT_U8: - case AFMT_S8: - case AFMT_U16_LE: - case AFMT_U16_BE: - case AFMT_S16_LE: - case AFMT_S16_BE: - case AFMT_S32_LE: - case AFMT_S32_BE: + case AF_FORMAT_U8: + case AF_FORMAT_S8: + case AF_FORMAT_U16_LE: + case AF_FORMAT_U16_BE: + case AF_FORMAT_S16_LE: + case AF_FORMAT_S16_BE: + case AF_FORMAT_S32_LE: + case AF_FORMAT_S32_BE: sh_audio_format = 0x1; /* PCM */ break; - case AFMT_IMA_ADPCM: - case AFMT_MU_LAW: - case AFMT_A_LAW: - case AFMT_MPEG: - case AFMT_AC3: + case AF_FORMAT_IMA_ADPCM: + case AF_FORMAT_MU_LAW: + case AF_FORMAT_A_LAW: + case AF_FORMAT_MPEG2: + case AF_FORMAT_AC3: default: mp_msg(MSGT_TV, MSGL_ERR, "Audio type '%s (%x)' unsupported!\n", - audio_out_format_name(audio_format), audio_format); + af_fmt2str(audio_format, &buf, 128), audio_format); goto no_audio; }
--- a/libmpdemux/tvi_bsdbt848.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpdemux/tvi_bsdbt848.c Mon Dec 27 17:30:15 2004 +0000 @@ -61,6 +61,7 @@ #endif #endif +#include "../libaf/af_format.h" #include "../libvo/img_format.h" #include "tv.h" @@ -246,7 +247,7 @@ case TVI_CONTROL_AUD_GET_FORMAT: { - (int)*(void **)arg = AFMT_S16_LE; + (int)*(void **)arg = AF_FORMAT_S16_LE; return(TVI_CONTROL_TRUE); } case TVI_CONTROL_AUD_GET_CHANNELS:
--- a/libmpdemux/tvi_v4l.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpdemux/tvi_v4l.c Mon Dec 27 17:30:15 2004 +0000 @@ -41,7 +41,7 @@ #endif #include "mp_msg.h" -#include "../libao2/afmt.h" +#include "../libaf/af_format.h" #include "../libvo/img_format.h" #include "../libvo/fastmemcpy.h" #include "../libvo/videodev_mjpeg.h" @@ -1285,7 +1285,7 @@ /* ========== AUDIO controls =========== */ case TVI_CONTROL_AUD_GET_FORMAT: { - *(int *)arg = AFMT_S16_LE; + *(int *)arg = AF_FORMAT_S16_LE; return(TVI_CONTROL_TRUE); } case TVI_CONTROL_AUD_GET_CHANNELS:
--- a/libmpdemux/tvi_v4l2.c Mon Dec 27 11:09:49 2004 +0000 +++ b/libmpdemux/tvi_v4l2.c Mon Dec 27 17:30:15 2004 +0000 @@ -43,7 +43,7 @@ #include "videodev2.h" #include "../mp_msg.h" #include "../libvo/img_format.h" -#include "../libao2/afmt.h" +#include "../libaf/af_format.h" #include "tv.h" #include "audio_in.h" @@ -731,7 +731,7 @@ } return TVI_CONTROL_TRUE; case TVI_CONTROL_AUD_GET_FORMAT: - *(int *)arg = AFMT_S16_LE; + *(int *)arg = AF_FORMAT_S16_LE; mp_msg(MSGT_TV, MSGL_V, "%s: get audio format: %d\n", info.short_name, *(int *)arg); return TVI_CONTROL_TRUE;
--- a/mencoder.c Mon Dec 27 11:09:49 2004 +0000 +++ b/mencoder.c Mon Dec 27 17:30:15 2004 +0000 @@ -54,7 +54,7 @@ #include "libvo/video_out.h" -#include "libao2/afmt.h" +#include "libaf/af_format.h" #include "libmpcodecs/mp_image.h" #include "libmpcodecs/dec_audio.h" @@ -809,7 +809,7 @@ sh_audio->samplerate, sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize, mux_a->wf->nSamplesPerSec, mux_a->wf->nChannels, - (mux_a->wf->wBitsPerSample==8)? AFMT_U8:AFMT_S16_LE, + (mux_a->wf->wBitsPerSample==8)? AF_FORMAT_U8:AF_FORMAT_S16_LE, mux_a->wf->wBitsPerSample/8, 16384, mux_a->wf->nAvgBytesPerSec)){ mp_msg(MSGT_CPLAYER,MSGL_ERR,MSGTR_NoMatchingFilter); @@ -842,11 +842,7 @@ sh_audio->samplerate, sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize, mux_a->wf->nSamplesPerSec, mux_a->wf->nChannels, -#ifdef WORDS_BIGENDIAN - AFMT_S16_BE, 2, -#else - AFMT_S16_LE, 2, -#endif + AF_FORMAT_S16_NE, 2, 4608, mux_a->h.dwRate*mux_a->wf->nChannels*2)){ mp_msg(MSGT_CPLAYER,MSGL_ERR,MSGTR_NoMatchingFilter); } @@ -983,7 +979,7 @@ sh_audio->samplerate, sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize, mux_a->wf->nSamplesPerSec, mux_a->wf->nChannels, - AFMT_S16_NE, 2, + AF_FORMAT_S16_NE, 2, mux_a->h.dwSuggestedBufferSize, mux_a->h.dwSuggestedBufferSize*2)) { mp_msg(MSGT_CPLAYER, MSGL_ERR, MSGTR_NoMatchingFilter); @@ -1048,7 +1044,7 @@ sh_audio->samplerate, sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize, mux_a->wf->nSamplesPerSec, mux_a->wf->nChannels, - AFMT_S16_NE, 2, + AF_FORMAT_S16_NE, 2, mux_a->h.dwSuggestedBufferSize, mux_a->h.dwSuggestedBufferSize*2)) { mp_msg(MSGT_CPLAYER, MSGL_ERR, "Couldn't find matching filter / ao format!\n");
--- a/mplayer.c Mon Dec 27 11:09:49 2004 +0000 +++ b/mplayer.c Mon Dec 27 17:30:15 2004 +0000 @@ -942,7 +942,7 @@ result = init_audio_filters(sh_audio, new_srate, 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, */ + af_fmt2bits(ao_data->format) / 8, /* ao_data.bps, */ ao_data->outburst * 4, ao_data->buffersize); mixer.afilter = sh_audio->afilter; return result; @@ -2094,12 +2094,13 @@ 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)){ + af_fmt2bits(ao_data.format)/8)){ mp_msg(MSGT_CPLAYER,MSGL_ERR,MSGTR_AudioFilterChainPreinitError); } else { + char buf[128]; 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)); + af_fmt2str(ao_data.format,&buf,128)); } #endif current_module="ao2_init"; @@ -2114,12 +2115,13 @@ sh_audio=d_audio->sh=NULL; // -> nosound } else { // SUCCESS: + char buf[128]; inited_flags|=INITED_AO; 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 ); + af_fmt2str(ao_data.format, &buf, 128), + af_fmt2bits(ao_data.format)/8 ); mp_msg(MSGT_CPLAYER,MSGL_V,"AO: Description: %s\nAO: Author: %s\n", audio_out->info->name, audio_out->info->author); if(strlen(audio_out->info->comment) > 0)