Mercurial > mplayer.hg
view libmpcodecs/ad_faad.c @ 35605:71f19d352561
Fix bug with wrong video window size after end of fullscreen playback.
(This is a regression introduced in r35572.)
The normal size the window should have (media size or configured size)
depends on the current state of playback. In order to influence the
size, add a boolean parameter. This usually is True unless playback has
stopped.
author | ib |
---|---|
date | Wed, 19 Dec 2012 13:58:06 +0000 |
parents | a93891202051 |
children |
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/* * MPlayer AAC decoder using libfaad2 * * Copyright (C) 2002 Felix Buenemann <atmosfear at users.sourceforge.net> * * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <faad.h> #include "config.h" #include "mp_msg.h" #include "ad_internal.h" #include "dec_audio.h" #include "libaf/reorder_ch.h" static const ad_info_t info = { "AAC (MPEG2/4 Advanced Audio Coding)", "faad", "Felix Buenemann", "faad2", "uses libfaad2" }; LIBAD_EXTERN(faad) /* configure maximum supported channels, * * this is theoretically max. 64 chans */ #define FAAD_MAX_CHANNELS 8 #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=8192*FAAD_MAX_CHANNELS; sh->audio_in_minsize=FAAD_BUFFLEN; return 1; } static int aac_probe(unsigned char *buffer, int len) { int i = 0, pos = 0; mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: %d bytes\n", len); while(i <= len-4) { if( ((buffer[i] == 0xff) && ((buffer[i+1] & 0xf6) == 0xf0)) || (buffer[i] == 'A' && buffer[i+1] == 'D' && buffer[i+2] == 'I' && buffer[i+3] == 'F') ) { pos = i; break; } mp_msg(MSGT_DECAUDIO,MSGL_V, "AUDIO PAYLOAD: %x %x %x %x\n", buffer[i], buffer[i+1], buffer[i+2], buffer[i+3]); i++; } mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: ret %d\n", pos); return pos; } static int init(sh_audio_t *sh) { unsigned long faac_samplerate; unsigned char faac_channels; int faac_init, pos = 0; faac_hdec = faacDecOpen(); // If we don't get the ES descriptor, try manual config if(!sh->codecdata_len && sh->wf) { sh->codecdata_len = sh->wf->cbSize; sh->codecdata = malloc(sh->codecdata_len); memcpy(sh->codecdata, sh->wf+1, sh->codecdata_len); mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n"); } if(!sh->codecdata_len || sh->format == mmioFOURCC('M', 'P', '4', 'L')) { 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 (audio_output_channels <= 2) faac_conf->downMatrix = 1; switch(sh->samplesize){ case 1: // 8Bit mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); default: sh->samplesize=2; 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); sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size); if (!sh->a_in_buffer_len) { // faad init will crash with 0 buffer length mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Could not get audio data!\n"); return 0; } /* external faad does not have latm lookup support */ faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, sh->a_in_buffer_len, &faac_samplerate, &faac_channels); if (faac_init < 0) { pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len); if(pos) { sh->a_in_buffer_len -= pos; memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len); 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); pos = 0; } /* init the codec */ faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, sh->a_in_buffer_len, &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 faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec); if (audio_output_channels <= 2) { faac_conf->downMatrix = 1; faacDecSetConfiguration(faac_hdec, faac_conf); } /*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: %ldHz channels: %d\n", faac_samplerate, faac_channels); // 8 channels is aac channel order #7. sh->channels = faac_channels == 7 ? 8 : faac_channels; if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1; sh->samplerate = faac_samplerate; sh->samplesize=2; //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 aac_sync(sh_audio_t *sh) { int pos = 0; // do not probe LATM, faad does that if(!sh->codecdata_len && sh->format != mmioFOURCC('M', 'P', '4', 'L')) { 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); } pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len); if(pos) { sh->a_in_buffer_len -= pos; memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len); mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos); } } return pos; } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { switch(cmd) { case ADCTRL_RESYNC_STREAM: aac_sync(sh); return CONTROL_TRUE; #if 0 case ADCTRL_SKIP_FRAME: return CONTROL_TRUE; #endif } return CONTROL_UNKNOWN; } #define MAX_FAAD_ERRORS 10 static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) { int len = 0, last_dec_len = 1, errors = 0; // int j = 0; void *faac_sample_buffer; while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) { /* 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, sh->a_in_buffer_len); /* 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; memmove(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: error: %s, trying to resync!\n", faacDecGetErrorMessage(faac_finfo.error)); if (sh->a_in_buffer_len <= 0) { errors = MAX_FAAD_ERRORS; break; } sh->a_in_buffer_len--; memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len); aac_sync(sh); errors++; } else break; } while(errors < MAX_FAAD_ERRORS); } else { // packetized (.mp4) aac stream: unsigned char* bufptr=NULL; double pts; int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts); if(buflen<=0) break; if (pts != MP_NOPTS_VALUE) { sh->pts = pts; sh->pts_bytes = 0; } faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen); } //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]); 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 (%ld Bytes)!\n", sh->samplesize*faac_finfo.samples); if (sh->channels >= 5) reorder_channel_copy_nch(faac_sample_buffer, AF_CHANNEL_LAYOUT_AAC_DEFAULT, buf+len, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, sh->channels, faac_finfo.samples, sh->samplesize); else memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples); last_dec_len = sh->samplesize*faac_finfo.samples; len += last_dec_len; sh->pts_bytes += last_dec_len; //printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed); } } return len; }