Mercurial > mplayer.hg
view libmpcodecs/ad_libdv.c @ 26355:a8fbc0224b81
Remove Win32 linker option for netstream. Other winsock using code does not
need it, it should be set from configure and the reason why it was set in
the first place has been lost in the mists of time.
author | diego |
---|---|
date | Fri, 11 Apr 2008 07:37:27 +0000 |
parents | 71b3e04d0555 |
children | 0f1b5b68af32 |
line wrap: on
line source
#include <stdio.h> #include <stdlib.h> #include <string.h> #include <sys/types.h> #include <unistd.h> #include <math.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" #include "img_format.h" #include <libdv/dv.h> #include <libdv/dv_types.h> #include "stream/stream.h" #include "libmpdemux/demuxer.h" #include "libmpdemux/stheader.h" #include "ad_internal.h" static ad_info_t info = { "Raw DV Audio Decoder", "libdv", "Alexander Neundorf <neundorf@kde.org>", "http://libdv.sf.net", "" }; LIBAD_EXTERN(libdv) // defined in vd_libdv.c: dv_decoder_t* init_global_rawdv_decoder(void); static int preinit(sh_audio_t *sh_audio) { sh_audio->audio_out_minsize=4*DV_AUDIO_MAX_SAMPLES*2; return 1; } static int16_t *audioBuffers[4]={NULL,NULL,NULL,NULL}; static int init(sh_audio_t *sh) { int i; WAVEFORMATEX *h=sh->wf; if(!h) return 0; sh->i_bps=h->nAvgBytesPerSec; sh->channels=h->nChannels; sh->samplerate=h->nSamplesPerSec; sh->samplesize=(h->wBitsPerSample+7)/8; sh->context=init_global_rawdv_decoder(); for (i=0; i < 4; i++) audioBuffers[i] = malloc(2*DV_AUDIO_MAX_SAMPLES); return 1; } static void uninit(sh_audio_t *sh_audio) { int i; for (i=0; i < 4; i++) free(audioBuffers[i]); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { // TODO!!! return CONTROL_UNKNOWN; } static int decode_audio(sh_audio_t *audio, unsigned char *buf, int minlen, int maxlen) { int len=0; dv_decoder_t* decoder=audio->context; //global_rawdv_decoder; unsigned char* dv_audio_frame=NULL; int xx=ds_get_packet(audio->ds,&dv_audio_frame); if(xx<=0 || !dv_audio_frame) return 0; // EOF? dv_parse_header(decoder, dv_audio_frame); if(xx!=decoder->frame_size) mp_msg(MSGT_GLOBAL,MSGL_WARN,MSGTR_MPCODECS_AudioFramesizeDiffers, xx, decoder->frame_size); if (dv_decode_full_audio(decoder, dv_audio_frame,(int16_t**) audioBuffers)) { /* Interleave the audio into a single buffer */ int i=0; int16_t *bufP=(int16_t*)buf; // printf("samples=%d/%d chans=%d mem=%d \n",decoder->audio->samples_this_frame,DV_AUDIO_MAX_SAMPLES, // decoder->audio->num_channels, decoder->audio->samples_this_frame*decoder->audio->num_channels*2); // return (44100/30)*4; for (i=0; i < decoder->audio->samples_this_frame; i++) { int ch; for (ch=0; ch < decoder->audio->num_channels; ch++) bufP[len++] = audioBuffers[ch][i]; } } return len*2; }