Mercurial > mplayer.hg
view libmpcodecs/ad_ffmpeg.c @ 24129:50e1e79056b6
Make terminal input work more like VO key input
The Unix version of getch2() could either return an internally buffered
key or do a second-level select() in addition to the input.c one and
then read more data. Change getch2() to always add all read keys with
mplayer_put_key() (like video output window keyboard input does) and
remove the internal select() from the Unix version. Make input.c call
mplayer_get_key() directly.
The primary motivation for this change is to make combining multiple
event sources under one select() easier. Now getch2() only needs to be
called when the corresponding fd is readable, and it will be possible to
handle events from X-based VOs with the same code.
author | uau |
---|---|
date | Sat, 25 Aug 2007 04:28:08 +0000 |
parents | 51cd49ee92d6 |
children | 352d7d9422b5 |
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#include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" #include "ad_internal.h" #include "mpbswap.h" static ad_info_t info = { "FFmpeg/libavcodec audio decoders", "ffmpeg", "Nick Kurshev", "ffmpeg.sf.net", "" }; LIBAD_EXTERN(ffmpeg) #define assert(x) #ifdef USE_LIBAVCODEC_SO #include <ffmpeg/avcodec.h> #else #include "avcodec.h" #endif extern int avcodec_inited; static int preinit(sh_audio_t *sh) { sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE; return 1; } static int init(sh_audio_t *sh_audio) { int x; AVCodecContext *lavc_context; AVCodec *lavc_codec; mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); if(!avcodec_inited){ avcodec_init(); avcodec_register_all(); avcodec_inited=1; } lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll); if(!lavc_codec){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); return 0; } lavc_context = avcodec_alloc_context(); sh_audio->context=lavc_context; if(sh_audio->wf){ lavc_context->channels = sh_audio->wf->nChannels; lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; lavc_context->block_align = sh_audio->wf->nBlockAlign; lavc_context->bits_per_sample = sh_audio->wf->wBitsPerSample; } lavc_context->request_channels = audio_output_channels; lavc_context->codec_tag = sh_audio->format; //FOURCC lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi /* alloc extra data */ if (sh_audio->wf && sh_audio->wf->cbSize > 0) { lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); lavc_context->extradata_size = sh_audio->wf->cbSize; memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX), lavc_context->extradata_size); } // for QDM2 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) { lavc_context->extradata = av_malloc(sh_audio->codecdata_len); lavc_context->extradata_size = sh_audio->codecdata_len; memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, lavc_context->extradata_size); } /* open it */ if (avcodec_open(lavc_context, lavc_codec) < 0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); return 0; } mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n"); // printf("\nFOURCC: 0x%X\n",sh_audio->format); if(sh_audio->format==0x3343414D){ // MACE 3:1 sh_audio->ds->ss_div = 2*3; // 1 samples/packet sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet } else if(sh_audio->format==0x3643414D){ // MACE 6:1 sh_audio->ds->ss_div = 2*6; // 1 samples/packet sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet } // Decode at least 1 byte: (to get header filled) x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); if(x>0) sh_audio->a_buffer_len=x; sh_audio->channels=lavc_context->channels; sh_audio->samplerate=lavc_context->sample_rate; sh_audio->i_bps=lavc_context->bit_rate/8; if(sh_audio->wf){ // If the decoder uses the wrong number of channels all is lost anyway. // sh_audio->channels=sh_audio->wf->nChannels; if (sh_audio->wf->nSamplesPerSec) sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; if (sh_audio->wf->nAvgBytesPerSec) sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; } sh_audio->samplesize=2; return 1; } static void uninit(sh_audio_t *sh) { AVCodecContext *lavc_context = sh->context; if (avcodec_close(lavc_context) < 0) mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); av_freep(&lavc_context->extradata); av_freep(&lavc_context); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { AVCodecContext *lavc_context = sh->context; switch(cmd){ case ADCTRL_RESYNC_STREAM: avcodec_flush_buffers(lavc_context); return CONTROL_TRUE; } return CONTROL_UNKNOWN; } static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) { unsigned char *start=NULL; int y,len=-1; while(len<minlen){ int len2=maxlen; double pts; int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); if(x<=0) break; // error if (pts != MP_NOPTS_VALUE) { sh_audio->pts = pts; sh_audio->pts_bytes = 0; } y=avcodec_decode_audio2(sh_audio->context,(int16_t*)buf,&len2,start,x); //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) if(len2>0){ //len=len2;break; if(len<0) len=len2; else len+=len2; buf+=len2; maxlen -= len2; sh_audio->pts_bytes += len2; } mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); } return len; }