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mpv/libao2/ao_win32.c

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/*
* 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;
switch(format){
case AF_FORMAT_AC3:
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(format == AF_FORMAT_AC3)
{
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);
waveOutReset(hWaveOut);
waveOutClose(hWaveOut);
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;
}
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