Mercurial > mplayer.hg
view libao2/ao_win32.c @ 25372:9850c8b67ccc
Always enable largefile support by default
The largefile configure option was disabled by default, but the
enabled-by-default dvdread and dvdcss options force in on (dvdnav
code also tries to enable it, but sets the variable too late so that
is has no effect). Change configure to enable largefile support
independently of other options unless explicitly disabled.
author | uau |
---|---|
date | Fri, 14 Dec 2007 22:02:48 +0000 |
parents | a124f3abc1ec |
children | 0f42fb42843c |
line wrap: on
line source
/****************************************************************************** * 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., 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 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; } // 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; 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(void) { 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(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) { 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; fast_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) { 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) { return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps; }