mirror of
https://github.com/mpv-player/mpv
synced 2024-12-28 06:03:45 +01:00
a655984828
ao_null accepts float input, but the code calculating ao_data.bps only checked for 1-byte formats and used samplesize 2 for everything else. Because ao_null uses the bps value in its timing calculations this effectively made "playback" advance at half the correct speed. Fixed by calculating samplesize with af_fmt2bits() instead. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@25223 b3059339-0415-0410-9bf9-f77b7e298cf2
114 lines
2.4 KiB
C
114 lines
2.4 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <sys/time.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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static ao_info_t info =
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{
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"Null audio output",
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"null",
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"Tobias Diedrich <ranma+mplayer@tdiedrich.de>",
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""
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};
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LIBAO_EXTERN(null)
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struct timeval last_tv;
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int buffer;
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static void drain(void){
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struct timeval now_tv;
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int temp, temp2;
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gettimeofday(&now_tv, 0);
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temp = now_tv.tv_sec - last_tv.tv_sec;
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temp *= ao_data.bps;
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temp2 = now_tv.tv_usec - last_tv.tv_usec;
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temp2 /= 1000;
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temp2 *= ao_data.bps;
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temp2 /= 1000;
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temp += temp2;
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buffer-=temp;
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if (buffer<0) buffer=0;
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if(temp>0) last_tv = now_tv;//mplayer is fast
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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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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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int samplesize = af_fmt2bits(format) / 8;
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ao_data.outburst = 256 * channels * samplesize;
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// A "buffer" for about 0.2 seconds of audio
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ao_data.buffersize = (int)(rate * 0.2 / 256 + 1) * ao_data.outburst;
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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*samplesize;
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buffer=0;
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gettimeofday(&last_tv, 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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// stop playing and empty buffers (for seeking/pause)
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static void reset(void){
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buffer=0;
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause(void)
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{
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// for now, just call reset();
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reset();
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}
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// resume playing, after audio_pause()
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static void audio_resume(void)
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{
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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(void){
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drain();
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return ao_data.buffersize - buffer;
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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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int maxbursts = (ao_data.buffersize - buffer) / ao_data.outburst;
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int playbursts = len / ao_data.outburst;
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int bursts = playbursts > maxbursts ? maxbursts : playbursts;
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buffer += bursts * ao_data.outburst;
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return bursts * ao_data.outburst;
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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(void){
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drain();
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return (float) buffer / (float) ao_data.bps;
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}
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