view libao2/ao_win32.c @ 37003:2d8157b3b386

Fix the index passed to PutImage() for item type itVPotmeter. Unlike the position for drawing the button (from bottom to top), the index for the phase image has to be in the usual range, i.e. 0 for 0% and numphases - 1 for 100%, so that the phase image can be designed straightforwardly. Fix the parameter although PutImage() currently doesn't work for item type itVPotmeter at all.

/*
 * Windows waveOut interface
 *
 * Copyright (c) 2002 - 2004 Sascha Sommer <[e-mail]>
 *
 * 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_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 <[e-mail]>",
	""
};

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 = LOWORD(volume)/655.35;
			vol->right = 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;
	}

	//fill global ao_data
	ao_data.channels=channels;
	ao_data.samplerate=rate;
	ao_data.format=format;
	ao_data.bps=channels*rate;
	ao_data.bps*=af_fmt2bits(format)/8;
	if(ao_data.buffersize==-1)
	{
		ao_data.buffersize=af_fmt2bits(format)/8;
        ao_data.buffersize*= channels;
		ao_data.buffersize*= SAMPLESIZE;
	}
	ao_data.outburst = ao_data.buffersize;
	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;
    wformat.Format.wBitsPerSample  = af_fmt2bits(format);
    if(AF_FORMAT_IS_AC3(format))
    {
        wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
        wformat.Format.nBlockAlign     = 4;
    }
    else
    {
        wformat.Format.wFormatTag      = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM;
        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, ao_data.buffersize + 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 += ao_data.buffersize;
    }
    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 * ao_data.buffersize;
}

//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=ao_data.buffersize;
    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 + 1) * ao_data.buffersize)/(float)ao_data.bps;
}