mirror of
https://github.com/mpv-player/mpv
synced 2024-12-27 01:22:30 +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
/******************************************************************************
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* ao_win32.c: Windows waveOut interface for MPlayer
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* Copyright (c) 2002 - 2004 Sascha Sommer <saschasommer@freenet.de>.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
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*
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*****************************************************************************/
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#include <stdio.h>
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#include <stdlib.h>
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#include <windows.h>
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#include <mmsystem.h>
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#include "config.h"
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#include "libaf/af_format.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "mp_msg.h"
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#include "libvo/fastmemcpy.h"
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#include "osdep/timer.h"
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#define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
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#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
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static const GUID KSDATAFORMAT_SUBTYPE_PCM = {
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0x1,0x0000,0x0010,{0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}
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};
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typedef struct {
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WAVEFORMATEX Format;
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union {
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WORD wValidBitsPerSample;
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WORD wSamplesPerBlock;
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WORD wReserved;
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} Samples;
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DWORD dwChannelMask;
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GUID SubFormat;
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} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
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#define SPEAKER_FRONT_LEFT 0x1
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#define SPEAKER_FRONT_RIGHT 0x2
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#define SPEAKER_FRONT_CENTER 0x4
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#define SPEAKER_LOW_FREQUENCY 0x8
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#define SPEAKER_BACK_LEFT 0x10
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#define SPEAKER_BACK_RIGHT 0x20
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#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
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#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
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#define SPEAKER_BACK_CENTER 0x100
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#define SPEAKER_SIDE_LEFT 0x200
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#define SPEAKER_SIDE_RIGHT 0x400
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#define SPEAKER_TOP_CENTER 0x800
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#define SPEAKER_TOP_FRONT_LEFT 0x1000
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#define SPEAKER_TOP_FRONT_CENTER 0x2000
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#define SPEAKER_TOP_FRONT_RIGHT 0x4000
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#define SPEAKER_TOP_BACK_LEFT 0x8000
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#define SPEAKER_TOP_BACK_CENTER 0x10000
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#define SPEAKER_TOP_BACK_RIGHT 0x20000
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static const int channel_mask[] = {
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SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
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SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
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SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_CENTER | SPEAKER_LOW_FREQUENCY,
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SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY
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};
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#define SAMPLESIZE 1024
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#define BUFFER_SIZE 4096
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#define BUFFER_COUNT 16
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static WAVEHDR* waveBlocks; //pointer to our ringbuffer memory
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static HWAVEOUT hWaveOut; //handle to the waveout device
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static unsigned int buf_write=0;
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static unsigned int buf_write_pos=0;
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static int full_buffers=0;
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static int buffered_bytes=0;
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static ao_info_t info =
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{
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"Windows waveOut audio output",
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"win32",
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"Sascha Sommer <saschasommer@freenet.de>",
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""
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};
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LIBAO_EXTERN(win32)
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static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,
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DWORD dwParam1,DWORD dwParam2)
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{
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if(uMsg != WOM_DONE)
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return;
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if (full_buffers) {
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buffered_bytes-=BUFFER_SIZE;
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--full_buffers;
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} else {
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buffered_bytes=0;
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}
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}
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// to set/get/query special features/parameters
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static int control(int cmd,void *arg)
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{
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DWORD volume;
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switch (cmd)
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{
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case AOCONTROL_GET_VOLUME:
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{
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ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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waveOutGetVolume(hWaveOut,&volume);
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vol->left = (float)(LOWORD(volume)/655.35);
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vol->right = (float)(HIWORD(volume)/655.35);
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mp_msg(MSGT_AO, MSGL_DBG2,"ao_win32: volume left:%f volume right:%f\n",vol->left,vol->right);
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return CONTROL_OK;
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}
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case AOCONTROL_SET_VOLUME:
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{
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ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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volume = MAKELONG(vol->left*655.35,vol->right*655.35);
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waveOutSetVolume(hWaveOut,volume);
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return CONTROL_OK;
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}
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}
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return -1;
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}
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(int rate,int channels,int format,int flags)
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{
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WAVEFORMATEXTENSIBLE wformat;
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DWORD totalBufferSize = (BUFFER_SIZE + sizeof(WAVEHDR)) * BUFFER_COUNT;
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MMRESULT result;
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unsigned char* buffer;
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int i;
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switch(format){
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case AF_FORMAT_AC3:
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case AF_FORMAT_S24_LE:
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case AF_FORMAT_S16_LE:
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case AF_FORMAT_S8:
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break;
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default:
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mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
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format=AF_FORMAT_S16_LE;
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}
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//fill global ao_data
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ao_data.channels=channels;
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ao_data.samplerate=rate;
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ao_data.format=format;
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ao_data.bps=channels*rate;
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if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
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ao_data.bps*=2;
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if(ao_data.buffersize==-1)
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{
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ao_data.buffersize=af_fmt2bits(format)/8;
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ao_data.buffersize*= channels;
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ao_data.buffersize*= SAMPLESIZE;
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}
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mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format));
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mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
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//fill waveformatex
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ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE));
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wformat.Format.cbSize = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
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wformat.Format.nChannels = channels;
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wformat.Format.nSamplesPerSec = rate;
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if(format == AF_FORMAT_AC3)
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{
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wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
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wformat.Format.wBitsPerSample = 16;
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wformat.Format.nBlockAlign = 4;
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}
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else
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{
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wformat.Format.wFormatTag = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM;
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wformat.Format.wBitsPerSample = af_fmt2bits(format);
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wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
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}
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if(channels>2)
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{
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wformat.dwChannelMask = channel_mask[channels-3];
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wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
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wformat.Samples.wValidBitsPerSample=af_fmt2bits(format);
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}
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wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
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//open sound device
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//WAVE_MAPPER always points to the default wave device on the system
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result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
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if(result == WAVERR_BADFORMAT)
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{
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mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n");
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ao_data.channels = wformat.Format.nChannels = 2;
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ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100;
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ao_data.format = AF_FORMAT_S16_LE;
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ao_data.bps=ao_data.channels * ao_data.samplerate*2;
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wformat.Format.wBitsPerSample=16;
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wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
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wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
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wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
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ao_data.buffersize=(wformat.Format.wBitsPerSample>>3)*wformat.Format.nChannels*SAMPLESIZE;
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result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
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}
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if(result != MMSYSERR_NOERROR)
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{
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mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: unable to open wave mapper device (result=%i)\n",result);
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return 0;
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}
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//allocate buffer memory as one big block
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buffer = malloc(totalBufferSize);
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memset(buffer,0x0,totalBufferSize);
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//and setup pointers to each buffer
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waveBlocks = (WAVEHDR*)buffer;
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buffer += sizeof(WAVEHDR) * BUFFER_COUNT;
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for(i = 0; i < BUFFER_COUNT; i++) {
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waveBlocks[i].lpData = buffer;
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buffer += BUFFER_SIZE;
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}
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buf_write=0;
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buf_write_pos=0;
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full_buffers=0;
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buffered_bytes=0;
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return 1;
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}
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// close audio device
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static void uninit(int immed)
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{
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if(!immed)while(buffered_bytes > 0)usec_sleep(50000);
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else buffered_bytes=0;
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waveOutReset(hWaveOut);
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waveOutClose(hWaveOut);
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mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n");
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free(waveBlocks);
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mp_msg(MSGT_AO, MSGL_V,"buffer memory freed\n");
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}
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// stop playing and empty buffers (for seeking/pause)
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static void reset()
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{
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waveOutReset(hWaveOut);
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buf_write=0;
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buf_write_pos=0;
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full_buffers=0;
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buffered_bytes=0;
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause()
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{
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waveOutPause(hWaveOut);
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}
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// resume playing, after audio_pause()
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static void audio_resume()
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{
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waveOutRestart(hWaveOut);
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}
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// return: how many bytes can be played without blocking
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static int get_space()
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{
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return BUFFER_COUNT*BUFFER_SIZE - buffered_bytes;
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}
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//writes data into buffer, based on ringbuffer code in ao_sdl.c
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static int write_waveOutBuffer(unsigned char* data,int len){
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WAVEHDR* current;
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int len2=0;
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int x;
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while(len>0){
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current = &waveBlocks[buf_write];
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if(buffered_bytes==BUFFER_COUNT*BUFFER_SIZE) break;
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//unprepare the header if it is prepared
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if(current->dwFlags & WHDR_PREPARED)
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waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
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x=BUFFER_SIZE-buf_write_pos;
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if(x>len) x=len;
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memcpy(current->lpData+buf_write_pos,data+len2,x);
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if(buf_write_pos==0)full_buffers++;
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len2+=x; len-=x;
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buffered_bytes+=x; buf_write_pos+=x;
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//prepare header and write data to device
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current->dwBufferLength = buf_write_pos;
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waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
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waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));
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if(buf_write_pos>=BUFFER_SIZE){ //buffer is full find next
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// block is full, find next!
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buf_write=(buf_write+1)%BUFFER_COUNT;
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buf_write_pos=0;
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}
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}
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return len2;
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}
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// plays 'len' bytes of 'data'
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// it should round it down to outburst*n
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// return: number of bytes played
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static int play(void* data,int len,int flags)
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{
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len = (len/ao_data.outburst)*ao_data.outburst;
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return write_waveOutBuffer(data,len);
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}
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// return: delay in seconds between first and last sample in buffer
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static float get_delay()
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{
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return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
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}
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