build: remove IRIX support

2 files changed, 0 insertions, 304 deletions

diff --git a/libao2/ao_sgi.c b/libao2/ao_sgi.c
deleted file mode 100644
index 492c8ff3ba..0000000000
--- a/libao2/ao_sgi.c
+++ /dev/null
@@ -1,301 +0,0 @@
-/*
- * SGI/IRIX audio output driver
- *
- * copyright (c) 2001 oliver.schoenbrunner@jku.at
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <errno.h>
-#include <dmedia/audio.h>
-
-#include "audio_out.h"
-#include "audio_out_internal.h"
-#include "mp_msg.h"
-#include "libaf/af_format.h"
-
-static const ao_info_t info =
-{
-	"sgi audio output",
-	"sgi",
-	"Oliver Schoenbrunner",
-	""
-};
-
-LIBAO_EXTERN(sgi)
-
-
-static ALconfig	ao_config;
-static ALport	ao_port;
-static int sample_rate;
-static int queue_size;
-static int bytes_per_frame;
-
-/**
- * \param   [in/out]  format
- * \param   [out]     width
- *
- * \return  the closest matching SGI AL sample format
- *
- * \note    width is set to required per-channel sample width
- *          format is updated to match the SGI AL sample format
- */
-static int fmt2sgial(int *format, int *width) {
-  int smpfmt = AL_SAMPFMT_TWOSCOMP;
-
-  /* SGI AL only supports float and signed integers in native
-   * endianness. If this is something else, we must rely on the audio
-   * filter to convert it to a compatible format. */
-
-  /* 24-bit audio is supported, but only with 32-bit alignment.
-   * mplayer's 24-bit format is packed, unfortunately.
-   * So we must upgrade 24-bit requests to 32 bits. Then we drop the
-   * lowest 8 bits during playback. */
-
-  switch(*format) {
-  case AF_FORMAT_U8:
-  case AF_FORMAT_S8:
-    *width = AL_SAMPLE_8;
-    *format = AF_FORMAT_S8;
-    break;
-
-  case AF_FORMAT_U16_LE:
-  case AF_FORMAT_U16_BE:
-  case AF_FORMAT_S16_LE:
-  case AF_FORMAT_S16_BE:
-    *width = AL_SAMPLE_16;
-    *format = AF_FORMAT_S16_NE;
-    break;
-
-  case AF_FORMAT_U24_LE:
-  case AF_FORMAT_U24_BE:
-  case AF_FORMAT_S24_LE:
-  case AF_FORMAT_S24_BE:
-  case AF_FORMAT_U32_LE:
-  case AF_FORMAT_U32_BE:
-  case AF_FORMAT_S32_LE:
-  case AF_FORMAT_S32_BE:
-    *width = AL_SAMPLE_24;
-    *format = AF_FORMAT_S32_NE;
-    break;
-
-  case AF_FORMAT_FLOAT_LE:
-  case AF_FORMAT_FLOAT_BE:
-    *width = 4;
-    *format = AF_FORMAT_FLOAT_NE;
-    smpfmt = AL_SAMPFMT_FLOAT;
-    break;
-
-  default:
-    *width = AL_SAMPLE_16;
-    *format = AF_FORMAT_S16_NE;
-    break;
-
-  }
-
-  return smpfmt;
-}
-
-// to set/get/query special features/parameters
-static int control(int cmd, void *arg){
-
-  mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] control.\n");
-
-  return CONTROL_UNKNOWN;
-}
-
-// open & setup audio device
-// return: 1=success 0=fail
-static int init(int rate, int channels, int format, int flags) {
-
-  int smpwidth, smpfmt;
-  int rv = AL_DEFAULT_OUTPUT;
-
-  smpfmt = fmt2sgial(&format, &smpwidth);
-
-  mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
-
-  { /* from /usr/share/src/dmedia/audio/setrate.c */
-
-    double frate, realrate;
-    ALpv x[2];
-
-    if(ao_subdevice) {
-      rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE);
-      if (!rv) {
-	mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] play: invalid device.\n");
-	return 0;
-      }
-    }
-
-    frate = rate;
-
-    x[0].param = AL_RATE;
-    x[0].value.ll = alDoubleToFixed(rate);
-    x[1].param = AL_MASTER_CLOCK;
-    x[1].value.i = AL_CRYSTAL_MCLK_TYPE;
-
-    if (alSetParams(rv,x, 2)<0) {
-      mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: setparams failed: %s\nCould not set desired samplerate.\n", alGetErrorString(oserror()));
-    }
-
-    if (x[0].sizeOut < 0) {
-      mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: AL_RATE was not accepted on the given resource.\n");
-    }
-
-    if (alGetParams(rv,x, 1)<0) {
-      mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: getparams failed: %s\n", alGetErrorString(oserror()));
-    }
-
-    realrate = alFixedToDouble(x[0].value.ll);
-    if (frate != realrate) {
-      mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: samplerate is now %f (desired rate is %f)\n", realrate, frate);
-    }
-    sample_rate = (int)realrate;
-  }
-
-  bytes_per_frame = channels * smpwidth;
-
-  ao_data.samplerate = sample_rate;
-  ao_data.channels = channels;
-  ao_data.format = format;
-  ao_data.bps = sample_rate * bytes_per_frame;
-  ao_data.buffersize=131072;
-  ao_data.outburst = ao_data.buffersize/16;
-
-  ao_config = alNewConfig();
-
-  if (!ao_config) {
-    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
-    return 0;
-  }
-
-  if(alSetChannels(ao_config, channels) < 0 ||
-     alSetWidth(ao_config, smpwidth) < 0 ||
-     alSetSampFmt(ao_config, smpfmt) < 0 ||
-     alSetQueueSize(ao_config, sample_rate) < 0 ||
-     alSetDevice(ao_config, rv) < 0) {
-    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
-    return 0;
-  }
-
-  ao_port = alOpenPort("mplayer", "w", ao_config);
-
-  if (!ao_port) {
-    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: Unable to open audio channel: %s\n", alGetErrorString(oserror()));
-    return 0;
-  }
-
-  // printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
-  queue_size = alGetQueueSize(ao_config);
-  return 1;
-
-}
-
-// close audio device
-static void uninit(int immed) {
-
-  /* TODO: samplerate should be set back to the value before mplayer was started! */
-
-  mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] uninit: ...\n");
-
-  if (ao_config) {
-    alFreeConfig(ao_config);
-    ao_config = NULL;
-  }
-
-  if (ao_port) {
-    if (!immed)
-    while(alGetFilled(ao_port) > 0) sginap(1);
-    alClosePort(ao_port);
-    ao_port = NULL;
-  }
-
-}
-
-// stop playing and empty buffers (for seeking/pause)
-static void reset(void) {
-
-  mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] reset: ...\n");
-
-  alDiscardFrames(ao_port, queue_size);
-}
-
-// stop playing, keep buffers (for pause)
-static void audio_pause(void) {
-
-  mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] audio_pause: ...\n");
-
-}
-
-// resume playing, after audio_pause()
-static void audio_resume(void) {
-
-  mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] audio_resume: ...\n");
-
-}
-
-// return: how many bytes can be played without blocking
-static int get_space(void) {
-
-  // printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_data.outburst);
-  // printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
-
-  return alGetFillable(ao_port) * bytes_per_frame;
-
-}
-
-
-// plays 'len' bytes of 'data'
-// it should round it down to outburst*n
-// return: number of bytes played
-static int play(void* data, int len, int flags) {
-
-  /* Always process data in quadword-aligned chunks (64-bits). */
-  const int plen = len / (sizeof(uint64_t) * bytes_per_frame);
-  const int framecount = plen * sizeof(uint64_t);
-
-  // printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config);
-  // printf("channels %d\n", ao_data.channels);
-
-  if(ao_data.format == AF_FORMAT_S32_NE) {
-    /* The zen of this is explained in fmt2sgial() */
-    int32_t *smpls = data;
-    const int32_t *smple = smpls + (framecount * ao_data.channels);
-    while(smpls < smple)
-      *smpls++ >>= 8;
-  }
-
-  alWriteFrames(ao_port, data, framecount);
-
-  return framecount * bytes_per_frame;
-
-}
-
-// return: delay in seconds between first and last sample in buffer
-static float get_delay(void){
-
-  // printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize);
-
-  // return  (float)queue_size/((float)sample_rate);
-  const int outstanding = alGetFilled(ao_port);
-  return (float)((outstanding < 0) ? queue_size : outstanding) /
-    ((float)sample_rate);
-}
diff --git a/libao2/audio_out.c b/libao2/audio_out.c
index d9a81b93b6..5dcb10c027 100644
--- a/libao2/audio_out.c
+++ b/libao2/audio_out.c
@@ -86,9 +86,6 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #ifdef CONFIG_PORTAUDIO
     &audio_out_portaudio,
 #endif
-#ifdef CONFIG_SGI_AUDIO
-    &audio_out_sgi,
-#endif
 #ifdef CONFIG_SUN_AUDIO
     &audio_out_sun,
 #endif