mirror of
https://github.com/mpv-player/mpv
synced 2024-12-11 09:25:56 +00:00
05d61f2f62
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14530 b3059339-0415-0410-9bf9-f77b7e298cf2
329 lines
10 KiB
C
329 lines
10 KiB
C
/******************************************************************************
|
|
* ao_win32.c: Windows waveOut interface for MPlayer
|
|
* Copyright (c) 2002 - 2004 Sascha Sommer <saschasommer@freenet.de>.
|
|
*
|
|
* This program 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.
|
|
*
|
|
* This program 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 this program; if not, write to the Free Software
|
|
* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, 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 unsigned int buf_write_pos=0;
|
|
static int full_buffers=0;
|
|
static int buffered_bytes=0;
|
|
|
|
|
|
static 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;
|
|
if (full_buffers) {
|
|
buffered_bytes-=BUFFER_SIZE;
|
|
--full_buffers;
|
|
} else {
|
|
buffered_bytes=0;
|
|
}
|
|
}
|
|
|
|
// 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;
|
|
DWORD totalBufferSize = (BUFFER_SIZE + sizeof(WAVEHDR)) * BUFFER_COUNT;
|
|
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_S8:
|
|
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;
|
|
if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
|
|
ao_data.bps*=2;
|
|
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 = malloc(totalBufferSize);
|
|
memset(buffer,0x0,totalBufferSize);
|
|
//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_write_pos=0;
|
|
full_buffers=0;
|
|
buffered_bytes=0;
|
|
|
|
return 1;
|
|
}
|
|
|
|
// close audio device
|
|
static void uninit(int immed)
|
|
{
|
|
if(!immed)while(buffered_bytes > 0)usec_sleep(50000);
|
|
else buffered_bytes=0;
|
|
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()
|
|
{
|
|
waveOutReset(hWaveOut);
|
|
buf_write=0;
|
|
buf_write_pos=0;
|
|
full_buffers=0;
|
|
buffered_bytes=0;
|
|
}
|
|
|
|
// stop playing, keep buffers (for pause)
|
|
static void audio_pause()
|
|
{
|
|
waveOutPause(hWaveOut);
|
|
}
|
|
|
|
// resume playing, after audio_pause()
|
|
static void audio_resume()
|
|
{
|
|
waveOutRestart(hWaveOut);
|
|
}
|
|
|
|
// return: how many bytes can be played without blocking
|
|
static int get_space()
|
|
{
|
|
return BUFFER_COUNT*BUFFER_SIZE - buffered_bytes;
|
|
}
|
|
|
|
//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){
|
|
current = &waveBlocks[buf_write];
|
|
if(buffered_bytes==BUFFER_COUNT*BUFFER_SIZE) break;
|
|
//unprepare the header if it is prepared
|
|
if(current->dwFlags & WHDR_PREPARED)
|
|
waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
|
|
x=BUFFER_SIZE-buf_write_pos;
|
|
if(x>len) x=len;
|
|
memcpy(current->lpData+buf_write_pos,data+len2,x);
|
|
if(buf_write_pos==0)full_buffers++;
|
|
len2+=x; len-=x;
|
|
buffered_bytes+=x; buf_write_pos+=x;
|
|
//prepare header and write data to device
|
|
current->dwBufferLength = buf_write_pos;
|
|
waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
|
|
waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));
|
|
|
|
if(buf_write_pos>=BUFFER_SIZE){ //buffer is full find next
|
|
// block is full, find next!
|
|
buf_write=(buf_write+1)%BUFFER_COUNT;
|
|
buf_write_pos=0;
|
|
}
|
|
}
|
|
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)
|
|
{
|
|
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()
|
|
{
|
|
return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
|
|
}
|