Mercurial > mplayer.hg
view libao2/ao_null.c @ 36149:f69d9caa3d84
Split init_vo and call only the necessary part from get_buffer.
This unbreaks mencoder which would run into a libavcodec
assert when trying to open the encoder from inside the
open function of the decoder.
author | reimar |
---|---|
date | Tue, 07 May 2013 20:39:14 +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; }