libvo, libao: remove useless video and audio output drivers

Some of these have only limited use, and some of these have no use at
all. Remove them. They make maintainance harder and nobody needs them.
It's possible that many of the removed drivers were very useful a dozen
of years ago, but now it's 2012.

Note that some of these could be added back, in case they were more
useful than I thought. But right now, they are just a burden.

Reason for removal for each module:
    vo_3dfx, vo_dfbmga, vo_dxr3, vo_ivtv, vo_mga, vo_s3fb,
    vo_tdfxfb, vo_xmga, vo_tdfx_vid:
        All of these are for very specific and outdated hardware. Some
        of them require non-standard kernel drivers or do direct HW
        access.
    vo_dga: the most crappy and ancient way to get fast output on X.
    vo_aa: there's vo_caca for the same purpose.
    vo_ggi: this never lived, and is entirely useless.
    vo_mpegpes: for DVB cards, I can't test this and it's crappy.
    vo_fbdev, vo_fbdev2: there's vo_directfb2
    vo_bl: what is this even? But it's neither important, nor alive.
    vo_svga, vo_vesa: you want to use this? You can't be serious.
    vo_wii: I can't test this, and who the hell uses this?
    vo_xvr100: some Sun thing.
    vo_xover: only useful in connection with xvr100.
    ao_nas: still alive, but I doubt it has any meaning today.
    ao_sun: Sun.
    ao_win32: use ao_dsound or ao_portaudio instead.
    ao_ivtv: removed along vo_ivtv.

Also get rid of anything SDL related. SDL 1.x is total crap for video
output, and will be replaced with SDL 2.x soon (perhaps), so if you
want to use SDL, write output drivers for SDL 2.x.

Additionally, I accidentally damaged Sun support, which made me
completely remove Sun/Solaris support. Nobody cares about this anyway.

Some left overs from previous commits removing modules were cleaned up.

1 files changed, 0 insertions, 326 deletions

diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c
deleted file mode 100644
index 55ed17b457..0000000000
--- a/libao2/ao_win32.c
+++ /dev/null
@@ -1,326 +0,0 @@
-/*
- * Windows waveOut interface
- *
- * Copyright (c) 2002 - 2004 Sascha Sommer <saschasommer@freenet.de>
- *
- * 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 <windows.h>
-#include <mmsystem.h>
-
-#include "config.h"
-#include "libaf/af_format.h"
-#include "audio_out.h"
-#include "audio_out_internal.h"
-#include "mp_msg.h"
-#include "libvo/fastmemcpy.h"
-#include "osdep/timer.h"
-
-#define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
-#define WAVE_FORMAT_EXTENSIBLE      0xFFFE
-
-static const  GUID KSDATAFORMAT_SUBTYPE_PCM = {
-	0x1,0x0000,0x0010,{0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}
-};
-
-typedef struct {
-  WAVEFORMATEX  Format;
-  union {
-    WORD  wValidBitsPerSample;
-    WORD  wSamplesPerBlock;
-    WORD  wReserved;
-  } Samples;
-  DWORD  dwChannelMask;
-  GUID  SubFormat;
-} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
-
-#define SPEAKER_FRONT_LEFT              0x1
-#define SPEAKER_FRONT_RIGHT             0x2
-#define SPEAKER_FRONT_CENTER            0x4
-#define SPEAKER_LOW_FREQUENCY           0x8
-#define SPEAKER_BACK_LEFT               0x10
-#define SPEAKER_BACK_RIGHT              0x20
-#define SPEAKER_FRONT_LEFT_OF_CENTER    0x40
-#define SPEAKER_FRONT_RIGHT_OF_CENTER   0x80
-#define SPEAKER_BACK_CENTER             0x100
-#define SPEAKER_SIDE_LEFT               0x200
-#define SPEAKER_SIDE_RIGHT              0x400
-#define SPEAKER_TOP_CENTER              0x800
-#define SPEAKER_TOP_FRONT_LEFT          0x1000
-#define SPEAKER_TOP_FRONT_CENTER        0x2000
-#define SPEAKER_TOP_FRONT_RIGHT         0x4000
-#define SPEAKER_TOP_BACK_LEFT           0x8000
-#define SPEAKER_TOP_BACK_CENTER         0x10000
-#define SPEAKER_TOP_BACK_RIGHT          0x20000
-
-static const int channel_mask[] = {
-  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_LOW_FREQUENCY,
-  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_LOW_FREQUENCY,
-  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_CENTER  | SPEAKER_LOW_FREQUENCY,
-  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT     | SPEAKER_LOW_FREQUENCY
-};
-
-
-
-#define SAMPLESIZE   1024
-#define BUFFER_SIZE  4096
-#define BUFFER_COUNT 16
-
-
-static WAVEHDR*     waveBlocks;         //pointer to our ringbuffer memory
-static HWAVEOUT     hWaveOut;           //handle to the waveout device
-static unsigned int buf_write=0;
-static volatile int buf_read=0;
-
-
-static const ao_info_t info =
-{
-	"Windows waveOut audio output",
-	"win32",
-	"Sascha Sommer <saschasommer@freenet.de>",
-	""
-};
-
-LIBAO_EXTERN(win32)
-
-static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,
-    DWORD dwParam1,DWORD dwParam2)
-{
-	if(uMsg != WOM_DONE)
-        return;
-	buf_read = (buf_read + 1) % BUFFER_COUNT;
-}
-
-// to set/get/query special features/parameters
-static int control(int cmd,void *arg)
-{
-	DWORD volume;
-	switch (cmd)
-	{
-		case AOCONTROL_GET_VOLUME:
-		{
-			ao_control_vol_t* vol = (ao_control_vol_t*)arg;
-			waveOutGetVolume(hWaveOut,&volume);
-			vol->left = (float)(LOWORD(volume)/655.35);
-			vol->right = (float)(HIWORD(volume)/655.35);
-			mp_msg(MSGT_AO, MSGL_DBG2,"ao_win32: volume left:%f volume right:%f\n",vol->left,vol->right);
-			return CONTROL_OK;
-		}
-		case AOCONTROL_SET_VOLUME:
-		{
-			ao_control_vol_t* vol = (ao_control_vol_t*)arg;
-			volume = MAKELONG(vol->left*655.35,vol->right*655.35);
-			waveOutSetVolume(hWaveOut,volume);
-			return CONTROL_OK;
-		}
-	}
-    return -1;
-}
-
-// open & setup audio device
-// return: 1=success 0=fail
-static int init(int rate,int channels,int format,int flags)
-{
-	WAVEFORMATEXTENSIBLE wformat;
-	MMRESULT result;
-	unsigned char* buffer;
-	int i;
-
-	if (AF_FORMAT_IS_AC3(format))
-		format = AF_FORMAT_AC3_NE;
-	switch(format){
-		case AF_FORMAT_AC3_NE:
-		case AF_FORMAT_S24_LE:
-		case AF_FORMAT_S16_LE:
-		case AF_FORMAT_U8:
-			break;
-		default:
-			mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
-			format=AF_FORMAT_S16_LE;
-	}
-
-	// FIXME multichannel mode is buggy
-	if(channels > 2)
-		channels = 2;
-
-	//fill global ao_data
-	ao_data.channels=channels;
-	ao_data.samplerate=rate;
-	ao_data.format=format;
-	ao_data.bps=channels*rate;
-	if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
-	  ao_data.bps*=2;
-	ao_data.outburst = BUFFER_SIZE;
-	if(ao_data.buffersize==-1)
-	{
-		ao_data.buffersize=af_fmt2bits(format)/8;
-        ao_data.buffersize*= channels;
-		ao_data.buffersize*= SAMPLESIZE;
-	}
-	mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format));
-    mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
-
-	//fill waveformatex
-    ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE));
-    wformat.Format.cbSize          = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
-    wformat.Format.nChannels       = channels;
-    wformat.Format.nSamplesPerSec  = rate;
-    if(AF_FORMAT_IS_AC3(format))
-    {
-        wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
-        wformat.Format.wBitsPerSample  = 16;
-        wformat.Format.nBlockAlign     = 4;
-    }
-    else
-    {
-        wformat.Format.wFormatTag      = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM;
-        wformat.Format.wBitsPerSample  = af_fmt2bits(format);
-        wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
-    }
-	if(channels>2)
-	{
-        wformat.dwChannelMask = channel_mask[channels-3];
-        wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
-	    wformat.Samples.wValidBitsPerSample=af_fmt2bits(format);
-    }
-
-    wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
-
-    //open sound device
-    //WAVE_MAPPER always points to the default wave device on the system
-    result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
-	if(result == WAVERR_BADFORMAT)
-	{
-		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 = AF_FORMAT_S16_LE;
-		ao_data.bps=ao_data.channels * ao_data.samplerate*2;
-	    wformat.Format.wBitsPerSample=16;
-        wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
-		wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
-        wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
-		ao_data.buffersize=(wformat.Format.wBitsPerSample>>3)*wformat.Format.nChannels*SAMPLESIZE;
-        result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
-	}
-	if(result != MMSYSERR_NOERROR)
-	{
-		mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: unable to open wave mapper device (result=%i)\n",result);
-		return 0;
-    }
-	//allocate buffer memory as one big block
-	buffer = calloc(BUFFER_COUNT, BUFFER_SIZE + sizeof(WAVEHDR));
-    //and setup pointers to each buffer
-    waveBlocks = (WAVEHDR*)buffer;
-    buffer += sizeof(WAVEHDR) * BUFFER_COUNT;
-    for(i = 0; i < BUFFER_COUNT; i++) {
-        waveBlocks[i].lpData = buffer;
-        buffer += BUFFER_SIZE;
-    }
-    buf_write=0;
-    buf_read=0;
-
-    return 1;
-}
-
-// close audio device
-static void uninit(int immed)
-{
-    if(!immed)
-	usec_sleep(get_delay() * 1000 * 1000);
-    else
-	waveOutReset(hWaveOut);
-    while (waveOutClose(hWaveOut) == WAVERR_STILLPLAYING) usec_sleep(0);
-	mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n");
-    free(waveBlocks);
-	mp_msg(MSGT_AO, MSGL_V,"buffer memory freed\n");
-}
-
-// stop playing and empty buffers (for seeking/pause)
-static void reset(void)
-{
-   	waveOutReset(hWaveOut);
-	buf_write=0;
-	buf_read=0;
-}
-
-// stop playing, keep buffers (for pause)
-static void audio_pause(void)
-{
-    waveOutPause(hWaveOut);
-}
-
-// resume playing, after audio_pause()
-static void audio_resume(void)
-{
-	waveOutRestart(hWaveOut);
-}
-
-// return: how many bytes can be played without blocking
-static int get_space(void)
-{
-    int free = buf_read - buf_write - 1;
-    if (free < 0) free += BUFFER_COUNT;
-    return free * BUFFER_SIZE;
-}
-
-//writes data into buffer, based on ringbuffer code in ao_sdl.c
-static int write_waveOutBuffer(unsigned char* data,int len){
-  WAVEHDR* current;
-  int len2=0;
-  int x;
-  while(len>0){
-    int buf_next = (buf_write + 1) % BUFFER_COUNT;
-    current = &waveBlocks[buf_write];
-    if(buf_next == buf_read) break;
-    //unprepare the header if it is prepared
-	if(current->dwFlags & WHDR_PREPARED)
-           waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
-	x=BUFFER_SIZE;
-    if(x>len) x=len;
-    fast_memcpy(current->lpData,data+len2,x);
-    len2+=x; len-=x;
-	//prepare header and write data to device
-	current->dwBufferLength = x;
-	waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
-	waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));
-
-       buf_write = buf_next;
-  }
-  return len2;
-}
-
-// 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)
-{
-	if (!(flags & AOPLAY_FINAL_CHUNK))
-	len = (len/ao_data.outburst)*ao_data.outburst;
-	return write_waveOutBuffer(data,len);
-}
-
-// return: delay in seconds between first and last sample in buffer
-static float get_delay(void)
-{
-	int used = buf_write - buf_read;
-	if (used < 0) used += BUFFER_COUNT;
-	return (float)(used * BUFFER_SIZE + ao_data.buffersize)/(float)ao_data.bps;
-}