Mercurial > mplayer.hg
view libao2/ao_null.c @ 34634:0ef7177a063b
Fix linking after FFmpeg merge.
Protocols that are available only when librtmp is present
are no longer disabled in the code, so we have to remove
them in our configure.
author | iive |
---|---|
date | Tue, 14 Feb 2012 16:50:42 +0000 |
parents | ba55ceb04748 |
children |
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/* * null 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 <stdio.h> #include <stdlib.h> #include <sys/time.h> #include "config.h" #include "libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" static const ao_info_t info = { "Null audio output", "null", "Tobias Diedrich <ranma+mplayer@tdiedrich.de>", "" }; LIBAO_EXTERN(null) static struct timeval last_tv; static int buffer; static void drain(void){ struct timeval now_tv; int temp, temp2; gettimeofday(&now_tv, 0); temp = now_tv.tv_sec - last_tv.tv_sec; temp *= ao_data.bps; temp2 = now_tv.tv_usec - last_tv.tv_usec; temp2 /= 1000; temp2 *= ao_data.bps; temp2 /= 1000; temp += temp2; buffer-=temp; if (buffer<0) buffer=0; if(temp>0) last_tv = now_tv;//mplayer is fast } // to set/get/query special features/parameters static int control(int cmd,void *arg){ return -1; } // open & setup audio device // return: 1=success 0=fail static int init(int rate,int channels,int format,int flags){ int samplesize = af_fmt2bits(format) / 8; ao_data.outburst = 256 * channels * samplesize; // A "buffer" for about 0.2 seconds of audio ao_data.buffersize = (int)(rate * 0.2 / 256 + 1) * ao_data.outburst; ao_data.channels=channels; ao_data.samplerate=rate; ao_data.format=format; ao_data.bps=channels*rate*samplesize; buffer=0; gettimeofday(&last_tv, 0); return 1; } // close audio device static void uninit(int immed){ } // stop playing and empty buffers (for seeking/pause) static void reset(void){ buffer=0; } // 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){ drain(); return ao_data.buffersize - buffer; } // 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){ int maxbursts = (ao_data.buffersize - buffer) / ao_data.outburst; int playbursts = len / ao_data.outburst; int bursts = playbursts > maxbursts ? maxbursts : playbursts; buffer += bursts * ao_data.outburst; return bursts * ao_data.outburst; } // return: delay in seconds between first and last sample in buffer static float get_delay(void){ drain(); return (float) buffer / (float) ao_data.bps; }