Mercurial > mplayer.hg
view libmpcodecs/ad_faad.c @ 6959:f7d8c02536e4
by 'lr' - <hephooey@fastmail.fm>
author | jaf |
---|---|
date | Sat, 10 Aug 2002 18:56:45 +0000 |
parents | a32870fd043b |
children | 7672615cc811 |
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/* ad_faad.c - MPlayer AAC decoder using libfaad2 * This file is part of MPlayer, see http://mplayerhq.hu/ for info. * (c)2002 by Felix Buenemann <atmosfear at users.sourceforge.net> * File licensed under the GPL, see http://www.fsf.org/ for more info. */ #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "ad_internal.h" #ifdef HAVE_FAAD static ad_info_t info = { "AAC (MPEG2/4 Advanced Audio Coding)", "libfaad2", AFM_AAC, "Felix Buenemann", "faad2", "Under development!" }; LIBAD_EXTERN(faad) #include <faad.h> /* configure maximum supported channels, * * this is theoretically max. 64 chans */ #define FAAD_MAX_CHANNELS 6 #define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS) //#define AAC_DUMP_COMPRESSED static faacDecHandle faac_hdec; static faacDecFrameInfo faac_finfo; static int preinit(sh_audio_t *sh) { sh->audio_out_minsize=2048*FAAD_MAX_CHANNELS; sh->audio_in_minsize=FAAD_BUFFLEN; return 1; } static int init(sh_audio_t *sh) { unsigned long faac_samplerate, faac_channels; int faac_init; faac_hdec = faacDecOpen(); // If we don't get the ES descriptor, try manual config if(!sh->codecdata_len) { #if 1 faacDecConfigurationPtr faac_conf; /* Set the default object type and samplerate */ /* This is useful for RAW AAC files */ faac_conf = faacDecGetCurrentConfiguration(faac_hdec); if(sh->samplerate) faac_conf->defSampleRate = sh->samplerate; /* XXX: FAAD support FLOAT output, how do we handle * that (FAAD_FMT_FLOAT)? ::atmos */ if(sh->samplesize) switch(sh->samplesize){ case 1: // 8Bit mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); default: case 2: // 16Bit faac_conf->outputFormat = FAAD_FMT_16BIT; break; case 3: // 24Bit faac_conf->outputFormat = FAAD_FMT_24BIT; break; case 4: // 32Bit faac_conf->outputFormat = FAAD_FMT_32BIT; break; } //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available. faacDecSetConfiguration(faac_hdec, faac_conf); #endif sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size); /* init the codec */ faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, &faac_samplerate, &faac_channels); sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi } else { // We have ES DS in codecdata /*int i; for(i = 0; i < sh_audio->codecdata_len; i++) printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/ faac_init = faacDecInit2(faac_hdec, sh->codecdata, sh->codecdata_len, &faac_samplerate, &faac_channels); } if(faac_init < 0) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup! faacDecClose(faac_hdec); // XXX: free a_in_buffer here or in uninit? return 0; } else { mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug! mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels); sh->channels = faac_channels; sh->samplerate = faac_samplerate; //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate; if(!sh->i_bps) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n"); sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos } else mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000); } return 1; } static void uninit(sh_audio_t *sh) { mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n"); faacDecClose(faac_hdec); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { switch(cmd) { #if 0 case ADCTRL_RESYNC_STREAM: return CONTROL_TRUE; case ADCTRL_SKIP_FRAME: return CONTROL_TRUE; #endif } return CONTROL_UNKNOWN; } static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) { int j = 0, len = 0; void *faac_sample_buffer; while(len < minlen) { /* update buffer for raw aac streams: */ if(!sh->codecdata_len) if(sh->a_in_buffer_len < sh->a_in_buffer_size){ sh->a_in_buffer_len += demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], sh->a_in_buffer_size - sh->a_in_buffer_len); } #ifdef DUMP_AAC_COMPRESSED {int i; for (i = 0; i < 16; i++) printf ("%02X ", sh->a_in_buffer[i]); printf ("\n");} #endif if(!sh->codecdata_len){ // raw aac stream: do { faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer+j); /* update buffer index after faacDecDecode */ if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) { sh->a_in_buffer_len=0; } else { sh->a_in_buffer_len-=faac_finfo.bytesconsumed; memcpy(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len); } if(faac_finfo.error > 0) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Trying to resync!\n"); j++; } else break; } while(j < FAAD_BUFFLEN); } else { // packetized (.mp4) aac stream: unsigned char* bufptr=NULL; int buflen=ds_get_packet(sh->ds, &bufptr); if(buflen<=0) break; faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr); // printf("FAAC decoded %d of %d (err: %d) \n",faac_finfo.bytesconsumed,buflen,faac_finfo.error); } if(faac_finfo.error > 0) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n", faacDecGetErrorMessage(faac_finfo.error)); } else if (faac_finfo.samples == 0) { mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n"); } else { /* XXX: samples already multiplied by channels! */ mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%d Bytes)!\n", sh->samplesize*faac_finfo.samples); memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples); len += sh->samplesize*faac_finfo.samples; //printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed); } } return len; } #endif /* !HAVE_FAAD */