Mercurial > mplayer.hg
view liba52/test.c @ 24900:9079c9745ff9
A/V sync: take audio filter buffers into account
Substract the delay caused by filter buffering when calculating
currently playing audio position. This matters for af_scaletempo which
buffers significant and varying amounts of data. For other current
filters the effect is normally insignificant.
Instead of the old time-based filter delay field (which was ignored)
this version stores the per-filter delay in units of bytes input read
without corresponding output. This allows the current scaletempo
behavior where other filters before and after it can see the same
nominal samplerate even though the real duration of the data varies;
in this case the other filters can not know the delay they're causing
in terms of real time.
author | uau |
---|---|
date | Thu, 01 Nov 2007 06:52:50 +0000 |
parents | 4e7d8679d6d8 |
children | 531116b7693d |
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// liba52 sample by A'rpi/ESP-team // reads ac3 stream form stdin, decodes and downmix to s16 stereo pcm and // writes it to stdout. resulting stream playbackable with sox: // play -c 2 -r 48000 out.sw //#define TIMING //needs Pentium or newer #include <stdio.h> #include <stdlib.h> #include <inttypes.h> #include <string.h> #include "a52.h" #include "mm_accel.h" #include "../cpudetect.h" static sample_t * samples; static a52_state_t state; static uint8_t buf[3840]; static int buf_size=0; static int16_t out_buf[6*256*6]; void mp_msg_c( int x, const char *format, ... ) // stub for cpudetect.c { } #ifdef TIMING static inline long long rdtsc() { long long l; asm volatile( "rdtsc\n\t" : "=A" (l) ); // printf("%d\n", int(l/1000)); return l; } #define STARTTIMING t=rdtsc(); #define ENDTIMING sum+=rdtsc()-t; t=rdtsc(); #else #define STARTTIMING ; #define ENDTIMING ; #endif int main(){ int accel=0; int sample_rate=0; int bit_rate=0; #ifdef TIMING long long t, sum=0, min=256*256*256*64; #endif FILE *temp= stdout; stdout= stderr; //EVIL HACK FIXME GetCpuCaps(&gCpuCaps); stdout= temp; // gCpuCaps.hasMMX=0; // gCpuCaps.hasSSE=0; if(gCpuCaps.hasMMX) accel |= MM_ACCEL_X86_MMX; if(gCpuCaps.hasMMX2) accel |= MM_ACCEL_X86_MMXEXT; if(gCpuCaps.hasSSE) accel |= MM_ACCEL_X86_SSE; if(gCpuCaps.has3DNow) accel |= MM_ACCEL_X86_3DNOW; // if(gCpuCaps.has3DNowExt) accel |= MM_ACCEL_X86_3DNOWEXT; samples = a52_init (accel); if (samples == NULL) { fprintf (stderr, "A52 init failed\n"); return 1; } while(1){ int length,i; int16_t *s16; sample_t level=1, bias=384; int flags=0; int channels=0; while(buf_size<7){ int c=getchar(); if(c<0) goto eof; buf[buf_size++]=c; } STARTTIMING length = a52_syncinfo (buf, &flags, &sample_rate, &bit_rate); ENDTIMING if(!length){ // bad file => resync memcpy(buf,buf+1,6); --buf_size; continue; } fprintf(stderr,"sync. %d bytes 0x%X %d Hz %d kbit\n",length,flags,sample_rate,bit_rate); while(buf_size<length){ buf[buf_size++]=getchar(); } buf_size=0; // decode: flags=A52_STEREO; //A52_STEREO; //A52_DOLBY; //A52_STEREO; // A52_DOLBY // A52_2F2R // A52_3F2R | A52_LFE channels=2; flags |= A52_ADJUST_LEVEL; STARTTIMING if (a52_frame (&state, buf, &flags, &level, bias)) { fprintf(stderr,"error at decoding\n"); continue; } ENDTIMING // a52_dynrng (&state, NULL, NULL); // disable dynamic range compensation STARTTIMING a52_resample_init(accel,flags,channels); s16 = out_buf; for (i = 0; i < 6; i++) { if (a52_block (&state, samples)) { fprintf(stderr,"error at sampling\n"); break; } // float->int + channels interleaving: s16+=a52_resample(samples,s16); ENDTIMING } #ifdef TIMING if(sum<min) min=sum; sum=0; #endif fwrite(out_buf,6*256*2*channels,1,stdout); } eof: #ifdef TIMING fprintf(stderr, "%4.4fk cycles ",min/1000.0); sum=0; #endif }