Mercurial > mplayer.hg
view libao2/ao_dxr2.c @ 33671:a460339acfdf
Fix volume and balance bug.
Changing the volume changed the balance as well,
because the calculation for the balance was wrong.
Additionally, use macro FFMAX() and replace
identical code by a call to existing code.
author | ib |
---|---|
date | Tue, 28 Jun 2011 17:52:30 +0000 |
parents | 2821d9499621 |
children |
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/* * DXR2 audio output driver * * 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 <math.h> #include <stdio.h> #include <stdlib.h> #include <sys/ioctl.h> #include <inttypes.h> #include <dxr2ioctl.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" #include "libavutil/common.h" #include "mpbswap.h" #include "audio_out.h" #include "audio_out_internal.h" #include "libaf/af_format.h" #include "libmpdemux/mpeg_packetizer.h" #include "libvo/vo_dxr2.h" #include "libvo/video_out.h" /* only for vo_pts */ static const ao_info_t info = { "DXR2 audio output", "dxr2", "Tobias Diedrich <ranma+mplayer@tdiedrich.de>", "" }; LIBAO_EXTERN(dxr2) static int volume=19; static int last_freq_id = -1; // to set/get/query special features/parameters static int control(int cmd,void *arg){ switch(cmd){ case AOCONTROL_GET_VOLUME: if(dxr2_fd > 0) { ao_control_vol_t* vol = (ao_control_vol_t*)arg; vol->left = vol->right = volume * 19.0 / 100.0; return CONTROL_OK; } return CONTROL_ERROR; case AOCONTROL_SET_VOLUME: if(dxr2_fd > 0) { dxr2_oneArg_t v; float diff; ao_control_vol_t* vol = (ao_control_vol_t*)arg; // We need this trick because the volume stepping is often too small diff = ((vol->left+vol->right) / 2 - (volume*19.0/100.0)) * 19.0 / 100.0; v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff)); if(v.arg > 19) v.arg = 19; if(v.arg < 0) v.arg = 0; if(v.arg != volume) { volume = v.arg; if( ioctl(dxr2_fd,DXR2_IOC_SET_AUDIO_VOLUME,&v) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_SetVolFailed,volume); return CONTROL_ERROR; } } return CONTROL_OK; } return CONTROL_ERROR; } return CONTROL_UNKNOWN; } static int freq=0; static int freq_id=0; // open & setup audio device // return: 1=success 0=fail static int init(int rate,int channels,int format,int flags){ if(dxr2_fd <= 0) return 0; last_freq_id = -1; ao_data.outburst=2048; ao_data.samplerate=rate; ao_data.channels=channels; ao_data.buffersize=2048; ao_data.bps=rate*4; ao_data.format=format; freq=rate; switch(rate){ case 48000: freq_id=DXR2_AUDIO_FREQ_48; break; case 96000: freq_id=DXR2_AUDIO_FREQ_96; break; case 44100: freq_id=DXR2_AUDIO_FREQ_441; break; case 32000: freq_id=DXR2_AUDIO_FREQ_32; break; case 22050: freq_id=DXR2_AUDIO_FREQ_2205; break; #ifdef DXR2_AUDIO_FREQ_24 // This is not yet in the dxr2 driver CVS // you can get the patch at // http://www.tdiedrich.de/~ranma/patches/dxr2.pcm1723.20020513 case 24000: freq_id=DXR2_AUDIO_FREQ_24; break; case 64000: freq_id=DXR2_AUDIO_FREQ_64; break; case 88200: freq_id=DXR2_AUDIO_FREQ_882; break; #endif default: mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_UnsupSamplerate,rate); return 0; } return 1; } // close audio device static void uninit(int immed){ } // stop playing and empty buffers (for seeking/pause) static void reset(void){ } // stop playing, keep buffers (for pause) static void audio_pause(void) { // for now, just call reset(); reset(); } // resume playing, after audio_pause() static void audio_resume(void) { } // return: how many bytes can be played without blocking static int get_space(void){ float x=(float)(vo_pts-ao_data.pts)/90000.0; int y; if(x<=0) return 0; y=freq*4*x;y/=ao_data.outburst;y*=ao_data.outburst; if(y>32768) y=32768; return y; } static void dxr2_send_lpcm_packet(unsigned char* data,int len,int id,unsigned int timestamp,int freq_id) { if(dxr2_fd < 0) { mp_msg(MSGT_AO,MSGL_ERR,"DXR2 fd is not valid\n"); return; } if(last_freq_id != freq_id) { ioctl(dxr2_fd, DXR2_IOC_SET_AUDIO_SAMPLE_FREQUENCY, &freq_id); last_freq_id = freq_id; } send_mpeg_lpcm_packet (data, len, id, timestamp, freq_id, write_dxr2); } // 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){ // MPEG and AC3 don't work :-( if(ao_data.format==AF_FORMAT_MPEG2) send_mpeg_ps_packet (data, len, 0xC0, ao_data.pts, 2, write_dxr2); else if(AF_FORMAT_IS_AC3(ao_data.format)) send_mpeg_ps_packet (data, len, 0x80, ao_data.pts, 2, write_dxr2); else { int i; //unsigned short *s=data; uint16_t *s=data; #if !HAVE_BIGENDIAN for(i=0;i<len/2;i++) s[i] = bswap_16(s[i]); #endif dxr2_send_lpcm_packet(data,len,0xA0,ao_data.pts-10000,freq_id); } return len; } // return: delay in seconds between first and last sample in buffer static float get_delay(void){ return 0.0; }