/****************************************************************************** * ao_win32.c: Windows waveOut interface for MPlayer * Copyright (c) 2002 - 2004 Sascha Sommer . * * 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 #include #include #include #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 ", "" }; 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; }