# HG changeset patch # User alex # Date 1104169792 0 # Node ID 7ea3c86f32950cb70fa2c7410244a4fc232fedb2 # Parent 474a40917f734e06c84917a8cfc5cf92a156163f 10l diff -r 474a40917f73 -r 7ea3c86f3295 libao2/ao_sun.c --- a/libao2/ao_sun.c Mon Dec 27 17:32:54 2004 +0000 +++ b/libao2/ao_sun.c Mon Dec 27 17:49:52 2004 +0000 @@ -25,7 +25,7 @@ #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" @@ -69,22 +69,22 @@ // convert an OSS audio format specification into a sun audio encoding -static int oss2sunfmt(int oss_format) +static int af2sunfmt(int format) { - switch (oss_format){ - case AFMT_MU_LAW: + switch (format){ + case AF_FORMAT_MU_LAW: return AUDIO_ENCODING_ULAW; - case AFMT_A_LAW: + case AF_FORMAT_A_LAW: return AUDIO_ENCODING_ALAW; - case AFMT_S16_BE: - case AFMT_S16_LE: + case AF_FORMAT_S16_BE: + case AF_FORMAT_S16_LE: return AUDIO_ENCODING_LINEAR; #ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1... - case AFMT_U8: + case AF_FORMAT_U8: return AUDIO_ENCODING_LINEAR8; #endif #ifdef AUDIO_ENCODING_DVI // Missing on NetBSD... - case AFMT_IMA_ADPCM: + case AF_FORMAT_IMA_ADPCM: return AUDIO_ENCODING_DVI; #endif default: @@ -465,20 +465,6 @@ enable_sample_timing = realtime_samplecounter_available(audio_dev); } -#define AF_FILTER_TEST 0 -#if AF_FILTER_TEST - /* test code to force use of the audio filter modules */ - { - char *s; - if (s = getenv("AF_RATE")) - rate = atoi(s); - if (s = getenv("AF_CHANNELS")) - channels = atoi(s); - if (s = getenv("AF_BITS")) - format = atoi(s) == 16 ? AFMT_S16_NE : AFMT_U8; - } -#endif - // printf("ao2: %d Hz %d chans %s [0x%X]\n", // rate,channels,audio_out_format_name(format),format); @@ -495,7 +481,7 @@ AUDIO_INITINFO(&info); info.play.encoding = oss2sunfmt(ao_data.format = format); info.play.precision = - (format==AFMT_S16_LE || format==AFMT_S16_BE + (format==AF_FORMAT_S16_LE || format==AF_FORMAT_S16_BE ? AUDIO_PRECISION_16 : AUDIO_PRECISION_8); info.play.channels = ao_data.channels = channels; @@ -578,7 +564,7 @@ ao_data.buffersize=0; #ifdef HAVE_AUDIO_SELECT data = malloc(ao_data.outburst); - memset(data, format==AFMT_U8 ? 0x80 : 0, ao_data.outburst); + memset(data, format==AF_FORMAT_U8 ? 0x80 : 0, ao_data.outburst); while(ao_data.buffersize<0x40000){ fd_set rfds; struct timeval tv; @@ -641,7 +627,7 @@ AUDIO_INITINFO(&info); info.play.encoding = oss2sunfmt(ao_data.format); info.play.precision = - (ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_S16_BE + (ao_data.format==AF_FORMAT_S16_LE || ao_data.format==AF_FORMAT_S16_BE ? AUDIO_PRECISION_16 : AUDIO_PRECISION_8); info.play.channels = ao_data.channels; @@ -710,9 +696,9 @@ // return: number of bytes played static int play(void* data,int len,int flags){ #if WORDS_BIGENDIAN - int native_endian = AFMT_S16_BE; + int native_endian = AF_FORMAT_S16_BE; #else - int native_endian = AFMT_S16_LE; + int native_endian = AF_FORMAT_S16_LE; #endif if (len < ao_data.outburst) return 0; @@ -720,7 +706,7 @@ len *= ao_data.outburst; /* 16-bit format using the 'wrong' byteorder? swap words */ - if ((ao_data.format == AFMT_S16_LE || ao_data.format == AFMT_S16_BE) + if ((ao_data.format == AF_FORMAT_S16_LE || ao_data.format == AF_FORMAT_S16_BE) && ao_data.format != native_endian) { static void *swab_buf; static int swab_len; @@ -734,7 +720,7 @@ } swab(data, swab_buf, len); data = swab_buf; - } else if (ao_data.format == AFMT_U8 && convert_u8_s8) { + } else if (ao_data.format == AF_FORMAT_U8 && convert_u8_s8) { int i; unsigned char *p = data;