Mercurial > mplayer.hg
view dec_audio.c @ 5190:59df6b778d78
Beta AAC decoding support, seeking totally broken yet, add philipps mpeg4 video in qt to ffmpeg4 although it's still buggy in decoding
| author | atmos4 |
|---|---|
| date | Mon, 18 Mar 2002 23:30:04 +0000 |
| parents | 9841a86d66f9 |
| children | 2ca5a9bfaa98 |
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#define USE_G72X //#define USE_LIBAC3 #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" extern int verbose; // defined in mplayer.c #include "stream.h" #include "demuxer.h" #include "codec-cfg.h" #include "stheader.h" #include "dec_audio.h" #include "roqav.h" //========================================================================== #include "libao2/afmt.h" #include "dll_init.h" #include "mp3lib/mp3.h" #ifdef USE_LIBAC3 #include "libac3/ac3.h" #endif #include "liba52/a52.h" #include "liba52/mm_accel.h" static sample_t * a52_samples; static a52_state_t a52_state; static uint32_t a52_accel=0; static uint32_t a52_flags=0; #ifdef USE_G72X #include "g72x/g72x.h" static G72x_DATA g72x_data; #endif #include "alaw.h" #include "xa/xa_gsm.h" #include "ac3-iec958.h" #include "adpcm.h" #include "cpudetect.h" /* used for ac3surround decoder - set using -channels option */ int audio_output_channels = 2; #ifdef USE_FAKE_MONO int fakemono=0; #endif #ifdef USE_DIRECTSHOW #include "loader/dshow/DS_AudioDecoder.h" static DS_AudioDecoder* ds_adec=NULL; #endif #ifdef HAVE_OGGVORBIS /* XXX is math.h really needed? - atmos */ #include <math.h> #include <vorbis/codec.h> // This struct is also defined in demux_ogg.c => common header ? typedef struct ov_struct_st { vorbis_info vi; /* struct that stores all the static vorbis bitstream settings */ vorbis_comment vc; /* struct that stores all the bitstream user comments */ vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ vorbis_block vb; /* local working space for packet->PCM decode */ } ov_struct_t; #endif #ifdef HAVE_FAAD #include <faad.h> static faacDecHandle faac_hdec; static faacDecFrameInfo faac_finfo; static int faac_bytesconsumed = 0; static unsigned char *faac_buffer; /* configure maximum supported channels, * * this is theoretically max. 64 chans */ #define FAAD_MAX_CHANNELS 6 #define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS) #endif #ifdef USE_LIBAVCODEC #ifdef USE_LIBAVCODEC_SO #include <libffmpeg/avcodec.h> #else #include "libavcodec/avcodec.h" #endif static AVCodec *lavc_codec=NULL; static AVCodecContext lavc_context; extern int avcodec_inited; #endif #ifdef USE_LIBMAD #include <mad.h> #define MAD_SINGLE_BUFFER_SIZE 8192 #define MAD_TOTAL_BUFFER_SIZE ((MAD_SINGLE_BUFFER_SIZE)*3) static struct mad_stream mad_stream; static struct mad_frame mad_frame; static struct mad_synth mad_synth; static char* mad_in_buffer = 0; /* base pointer of buffer */ // ensure buffer is filled with some data static void mad_prepare_buffer(sh_audio_t* sh_audio, struct mad_stream* ms, int length) { if(sh_audio->a_in_buffer_len < length) { int len = demux_read_data(sh_audio->ds, sh_audio->a_in_buffer+sh_audio->a_in_buffer_len, length-sh_audio->a_in_buffer_len); sh_audio->a_in_buffer_len += len; // printf("mad_prepare_buffer: read %d bytes\n", len); } } static void mad_postprocess_buffer(sh_audio_t* sh_audio, struct mad_stream* ms) { /* rotate buffer while possible, in order to reduce the overhead of endless memcpy */ int delta = (unsigned char*)ms->next_frame - (unsigned char *)sh_audio->a_in_buffer; if((unsigned long)(sh_audio->a_in_buffer) - (unsigned long)mad_in_buffer < (MAD_TOTAL_BUFFER_SIZE - MAD_SINGLE_BUFFER_SIZE - delta)) { sh_audio->a_in_buffer += delta; sh_audio->a_in_buffer_len -= delta; } else { sh_audio->a_in_buffer = mad_in_buffer; sh_audio->a_in_buffer_len -= delta; memcpy(sh_audio->a_in_buffer, ms->next_frame, sh_audio->a_in_buffer_len); } } static inline signed short mad_scale(mad_fixed_t sample) { /* round */ sample += (1L << (MAD_F_FRACBITS - 16)); /* clip */ if (sample >= MAD_F_ONE) sample = MAD_F_ONE - 1; else if (sample < -MAD_F_ONE) sample = -MAD_F_ONE; /* quantize */ return sample >> (MAD_F_FRACBITS + 1 - 16); } static void mad_sync(sh_audio_t* sh_audio, struct mad_stream* ms) { int len; #if 1 int skipped = 0; // printf("buffer len: %d\n", sh_audio->a_in_buffer_len); while(sh_audio->a_in_buffer_len - skipped) { len = mp_decode_mp3_header(sh_audio->a_in_buffer+skipped); if (len != -1) { // printf("Frame len=%d\n", len); break; } else skipped++; } if (skipped) { mp_msg(MSGT_DECAUDIO, MSGL_INFO, "mad: audio synced, skipped bytes: %d\n", skipped); // ms->skiplen += skipped; // printf("skiplen: %d (skipped: %d)\n", ms->skiplen, skipped); // if (sh_audio->a_in_buffer_len - skipped < MAD_BUFFER_GUARD) // printf("Mad reports: too small buffer\n"); // mad_stream_buffer(ms, sh_audio->a_in_buffer+skipped, sh_audio->a_in_buffer_len-skipped); // mad_prepare_buffer(sh_audio, ms, sh_audio->a_in_buffer_len-skipped); /* move frame to the beginning of the buffer and fill up to a_in_buffer_size */ sh_audio->a_in_buffer_len -= skipped; memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer+skipped, sh_audio->a_in_buffer_len); mad_prepare_buffer(sh_audio, ms, sh_audio->a_in_buffer_size); mad_stream_buffer(ms, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); // printf("bufflen: %d\n", sh_audio->a_in_buffer_len); // len = mp_decode_mp3_header(sh_audio->a_in_buffer); // printf("len: %d\n", len); ms->md_len = len; } #else len = mad_stream_sync(&ms); if (len == -1) { mp_msg(MSGT_DECVIDEO, MSGL_ERR, "Mad sync failed\n"); } #endif } static void mad_print_error(struct mad_stream *mad_stream) { printf("error (0x%x): ", mad_stream->error); switch(mad_stream->error) { case MAD_ERROR_BUFLEN: printf("buffer too small"); break; case MAD_ERROR_BUFPTR: printf("invalid buffer pointer"); break; case MAD_ERROR_NOMEM: printf("not enought memory"); break; case MAD_ERROR_LOSTSYNC: printf("lost sync"); break; case MAD_ERROR_BADLAYER: printf("bad layer"); break; case MAD_ERROR_BADBITRATE: printf("bad bitrate"); break; case MAD_ERROR_BADSAMPLERATE: printf("bad samplerate"); break; case MAD_ERROR_BADEMPHASIS: printf("bad emphasis"); break; case MAD_ERROR_BADCRC: printf("bad crc"); break; case MAD_ERROR_BADBITALLOC: printf("forbidden bit alloc val"); break; case MAD_ERROR_BADSCALEFACTOR: printf("bad scalefactor index"); break; case MAD_ERROR_BADFRAMELEN: printf("bad frame length"); break; case MAD_ERROR_BADBIGVALUES: printf("bad bigvalues count"); break; case MAD_ERROR_BADBLOCKTYPE: printf("reserved blocktype"); break; case MAD_ERROR_BADSCFSI: printf("bad scalefactor selinfo"); break; case MAD_ERROR_BADDATAPTR: printf("bad maindatabegin ptr"); break; case MAD_ERROR_BADPART3LEN: printf("bad audio data len"); break; case MAD_ERROR_BADHUFFTABLE: printf("bad huffman table sel"); break; case MAD_ERROR_BADHUFFDATA: printf("huffman data overrun"); break; case MAD_ERROR_BADSTEREO: printf("incomp. blocktype for JS"); break; default: printf("unknown error"); } printf("\n"); } #endif static int a52_fillbuff(sh_audio_t *sh_audio){ int length=0; int flags=0; int sample_rate=0; int bit_rate=0; sh_audio->a_in_buffer_len=0; // sync frame: while(1){ while(sh_audio->a_in_buffer_len<7){ int c=demux_getc(sh_audio->ds); if(c<0) return -1; // EOF sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c; } length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); if(length>=7 && length<=3840) break; // we're done. // bad file => resync memcpy(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,6); --sh_audio->a_in_buffer_len; } mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate); sh_audio->samplerate=sample_rate; sh_audio->i_bps=bit_rate/8; demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+7,length-7); if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0) mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n"); return length; } // returns: number of available channels static int a52_printinfo(sh_audio_t *sh_audio){ int flags, sample_rate, bit_rate; char* mode="unknown"; int channels=0; a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); switch(flags&A52_CHANNEL_MASK){ case A52_CHANNEL: mode="channel"; channels=2; break; case A52_MONO: mode="mono"; channels=1; break; case A52_STEREO: mode="stereo"; channels=2; break; case A52_3F: mode="3f";channels=3;break; case A52_2F1R: mode="2f+1r";channels=3;break; case A52_3F1R: mode="3f+1r";channels=4;break; case A52_2F2R: mode="2f+2r";channels=4;break; case A52_3F2R: mode="3f+2r";channels=5;break; case A52_CHANNEL1: mode="channel1"; channels=2; break; case A52_CHANNEL2: mode="channel2"; channels=2; break; case A52_DOLBY: mode="dolby"; channels=2; break; } mp_msg(MSGT_DECAUDIO,MSGL_INFO,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n", channels, (flags&A52_LFE)?1:0, mode, (flags&A52_LFE)?"+lfe":"", sample_rate, bit_rate*0.001f); return (flags&A52_LFE) ? (channels+1) : channels; } int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen); static sh_audio_t* dec_audio_sh=NULL; #ifdef USE_LIBAC3 // AC3 decoder buffer callback: static void ac3_fill_buffer(uint8_t **start,uint8_t **end){ int len=ds_get_packet(dec_audio_sh->ds,start); //printf("<ac3:%d>\n",len); if(len<0) *start = *end = NULL; else *end = *start + len; } #endif // MP3 decoder buffer callback: int mplayer_audio_read(char *buf,int size){ int len; len=demux_read_data(dec_audio_sh->ds,buf,size); return len; } int init_audio(sh_audio_t *sh_audio){ int driver=sh_audio->codec->driver; sh_audio->samplesize=2; #ifdef WORDS_BIGENDIAN sh_audio->sample_format=AFMT_S16_BE; #else sh_audio->sample_format=AFMT_S16_LE; #endif sh_audio->samplerate=0; //sh_audio->pcm_bswap=0; sh_audio->o_bps=0; sh_audio->a_buffer_size=0; sh_audio->a_buffer=NULL; sh_audio->a_in_buffer_len=0; // setup required min. in/out buffer size: sh_audio->audio_out_minsize=8192;// default size, maybe not enough for Win32/ACM switch(driver){ case AFM_ACM: #ifndef USE_WIN32DLL mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoACMSupport); driver=0; #else // Win32 ACM audio codec: if(init_acm_audio_codec(sh_audio)){ sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; sh_audio->channels=sh_audio->o_wf.nChannels; sh_audio->samplerate=sh_audio->o_wf.nSamplesPerSec; // if(sh_audio->audio_out_minsize>16384) sh_audio->audio_out_minsize=16384; // sh_audio->a_buffer_size=sh_audio->audio_out_minsize; // if(sh_audio->a_buffer_size<sh_audio->audio_out_minsize+MAX_OUTBURST) // sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; } else { mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_ACMiniterror); driver=0; } #endif break; case AFM_DSHOW: #ifndef USE_DIRECTSHOW mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoDShowAudio); driver=0; #else // Win32 DShow audio codec: // printf("DShow_audio: channs=%d rate=%d\n",sh_audio->channels,sh_audio->samplerate); if(!(ds_adec=DS_AudioDecoder_Open(sh_audio->codec->dll,&sh_audio->codec->guid,sh_audio->wf))){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingDLLcodec,sh_audio->codec->dll); driver=0; } else { sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->audio_in_minsize=2*sh_audio->wf->nBlockAlign; if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192; sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize; sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size); sh_audio->a_in_buffer_len=0; sh_audio->audio_out_minsize=16384; } #endif break; case AFM_VORBIS: #ifndef HAVE_OGGVORBIS mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoOggVorbis); driver=0; #else /* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */ // Is there always 1024 samples/frame ? ***** Albeu sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame #endif break; case AFM_AAC: // AAC (MPEG2 Audio, MPEG4 Audio) #ifndef HAVE_FAAD mp_msg(MSGT_DECAUDIO,MSGL_ERR,"Error: Cannot decode AAC data, because MPlayer was compiled without FAAD support\n"/*MSGTR_NoFAAD*/); driver=0; #else mp_msg(MSGT_DECAUDIO,MSGL_V,"Using FAAD to decode AAC content!\n"/*MSGTR_UseFAAD*/); // Samples per frame * channels per frame, this might not work with >2 chan AAC, need test samples! ::atmos sh_audio->audio_out_minsize=2048*2; #endif break; case AFM_PCM: case AFM_DVDPCM: case AFM_ALAW: // PCM, aLaw sh_audio->audio_out_minsize=2048; break; case AFM_AC3: case AFM_A52: // Dolby AC3 audio: // however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame sh_audio->audio_out_minsize=audio_output_channels*2*256*6; break; case AFM_HWAC3: // Dolby AC3 audio: sh_audio->audio_out_minsize=4*256*6; // sh_audio->sample_format = AFMT_AC3; // sh_audio->sample_format = AFMT_S16_LE; sh_audio->channels=2; break; case AFM_GSM: // MS-GSM audio codec: sh_audio->audio_out_minsize=4*320; break; case AFM_IMAADPCM: sh_audio->audio_out_minsize=4096; sh_audio->ds->ss_div=IMA_ADPCM_SAMPLES_PER_BLOCK; sh_audio->ds->ss_mul=IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels; break; case AFM_MSADPCM: sh_audio->audio_out_minsize=sh_audio->wf->nBlockAlign * 8; sh_audio->ds->ss_div = MS_ADPCM_SAMPLES_PER_BLOCK; sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign; break; case AFM_DK4ADPCM: sh_audio->audio_out_minsize=DK4_ADPCM_SAMPLES_PER_BLOCK * 4; sh_audio->ds->ss_div=DK4_ADPCM_SAMPLES_PER_BLOCK; sh_audio->ds->ss_mul=sh_audio->wf->nBlockAlign; break; case AFM_DK3ADPCM: sh_audio->audio_out_minsize=DK3_ADPCM_SAMPLES_PER_BLOCK * 4; sh_audio->ds->ss_div=DK3_ADPCM_SAMPLES_PER_BLOCK; sh_audio->ds->ss_mul=DK3_ADPCM_BLOCK_SIZE; break; case AFM_ROQAUDIO: // minsize was stored in wf->nBlockAlign by the RoQ demuxer sh_audio->audio_out_minsize=sh_audio->wf->nBlockAlign; sh_audio->ds->ss_div=DK3_ADPCM_SAMPLES_PER_BLOCK; sh_audio->ds->ss_mul=DK3_ADPCM_BLOCK_SIZE; sh_audio->context = roq_decode_audio_init(); break; case AFM_MPEG: // MPEG Audio: sh_audio->audio_out_minsize=4608; break; #ifdef USE_G72X case AFM_G72X: // g72x_reader_init(&g72x_data,G723_16_BITS_PER_SAMPLE); g72x_reader_init(&g72x_data,G723_24_BITS_PER_SAMPLE); // g72x_reader_init(&g72x_data,G721_32_BITS_PER_SAMPLE); // g72x_reader_init(&g72x_data,G721_40_BITS_PER_SAMPLE); sh_audio->audio_out_minsize=g72x_data.samplesperblock*4; break; #endif case AFM_FFMPEG: #ifndef USE_LIBAVCODEC mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoLAVCsupport); return 0; #else // FFmpeg Audio: sh_audio->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE; break; #endif #ifdef USE_LIBMAD case AFM_MAD: mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: setting minimum outputsize\n"); sh_audio->audio_out_minsize=4608; if(sh_audio->audio_in_minsize<MAD_SINGLE_BUFFER_SIZE) sh_audio->audio_in_minsize=MAD_SINGLE_BUFFER_SIZE; sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize; mad_in_buffer = sh_audio->a_in_buffer = malloc(MAD_TOTAL_BUFFER_SIZE); sh_audio->a_in_buffer_len=0; break; #endif } if(!driver) return 0; // allocate audio out buffer: sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; // worst case calc. mp_msg(MSGT_DECAUDIO,MSGL_V,"dec_audio: Allocating %d + %d = %d bytes for output buffer\n", sh_audio->audio_out_minsize,MAX_OUTBURST,sh_audio->a_buffer_size); sh_audio->a_buffer=malloc(sh_audio->a_buffer_size); if(!sh_audio->a_buffer){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_CantAllocAudioBuf); return 0; } memset(sh_audio->a_buffer,0,sh_audio->a_buffer_size); sh_audio->a_buffer_len=0; switch(driver){ #ifdef USE_WIN32DLL case AFM_ACM: { int ret=acm_decode_audio(sh_audio,sh_audio->a_buffer,4096,sh_audio->a_buffer_size); if(ret<0){ mp_msg(MSGT_DECAUDIO,MSGL_INFO,"ACM decoding error: %d\n",ret); driver=0; } sh_audio->a_buffer_len=ret; break; } #endif case AFM_PCM: { // AVI PCM Audio: WAVEFORMATEX *h=sh_audio->wf; sh_audio->i_bps=h->nAvgBytesPerSec; sh_audio->channels=h->nChannels; sh_audio->samplerate=h->nSamplesPerSec; sh_audio->samplesize=(h->wBitsPerSample+7)/8; switch(sh_audio->format){ // hardware formats: case 0x6: sh_audio->sample_format=AFMT_A_LAW;break; case 0x7: sh_audio->sample_format=AFMT_MU_LAW;break; case 0x11: sh_audio->sample_format=AFMT_IMA_ADPCM;break; case 0x50: sh_audio->sample_format=AFMT_MPEG;break; case 0x736F7774: sh_audio->sample_format=AFMT_S16_LE;sh_audio->codec->driver=AFM_DVDPCM;break; // case 0x2000: sh_audio->sample_format=AFMT_AC3; default: sh_audio->sample_format=(sh_audio->samplesize==2)?AFMT_S16_LE:AFMT_U8; } break; } case AFM_DVDPCM: { // DVD PCM Audio: sh_audio->channels=2; sh_audio->samplerate=48000; sh_audio->i_bps=2*2*48000; // sh_audio->pcm_bswap=1; break; } case AFM_AC3: { #ifndef USE_LIBAC3 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"WARNING: libac3 support is disabled. (hint: upgrade codecs.conf)\n"); driver=0; #else // Dolby AC3 audio: dec_audio_sh=sh_audio; // save sh_audio for the callback: ac3_config.fill_buffer_callback = ac3_fill_buffer; ac3_config.num_output_ch = audio_output_channels; ac3_config.flags = 0; if(gCpuCaps.hasMMX){ ac3_config.flags |= AC3_MMX_ENABLE; } if(gCpuCaps.has3DNow){ ac3_config.flags |= AC3_3DNOW_ENABLE; } ac3_init(); sh_audio->ac3_frame = ac3_decode_frame(); if(sh_audio->ac3_frame){ ac3_frame_t* fr=(ac3_frame_t*)sh_audio->ac3_frame; sh_audio->samplerate=fr->sampling_rate; sh_audio->channels=ac3_config.num_output_ch; // 1 frame: 6*256 samples 1 sec: sh_audio->samplerate samples //sh_audio->i_bps=fr->frame_size*fr->sampling_rate/(6*256); sh_audio->i_bps=fr->bit_rate*(1000/8); } else { driver=0; // bad frame -> disable audio } #endif break; } case AFM_A52: { sample_t level=1, bias=384; int flags=0; // Dolby AC3 audio: if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE; if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX; if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT; if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW; if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT; a52_samples=a52_init (a52_accel); if (a52_samples == NULL) { mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n"); driver=0;break; } sh_audio->a_in_buffer_size=3840; sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size); sh_audio->a_in_buffer_len=0; if(a52_fillbuff(sh_audio)<0){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n"); driver=0;break; } // 'a52 cannot upmix' hotfix: a52_printinfo(sh_audio); // if(audio_output_channels<sh_audio->channels) // sh_audio->channels=audio_output_channels; // channels setup: sh_audio->channels=audio_output_channels; while(sh_audio->channels>0){ switch(sh_audio->channels){ case 1: a52_flags=A52_MONO; break; // case 2: a52_flags=A52_STEREO; break; case 2: a52_flags=A52_DOLBY; break; // case 3: a52_flags=A52_3F; break; case 3: a52_flags=A52_2F1R; break; case 4: a52_flags=A52_2F2R; break; // 2+2 case 5: a52_flags=A52_3F2R; break; case 6: a52_flags=A52_3F2R|A52_LFE; break; // 5.1 } // test: flags=a52_flags|A52_ADJUST_LEVEL; mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags); if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n"); driver=0;break; } mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags); // frame decoded, let's init resampler: if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break; --sh_audio->channels; // try to decrease no. of channels } if(sh_audio->channels<=0){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n"); driver=0;break; } break; } case AFM_HWAC3: { // Dolby AC3 passthrough: a52_samples=a52_init (a52_accel); if (a52_samples == NULL) { mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n"); driver=0;break; } sh_audio->a_in_buffer_size=3840; sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size); sh_audio->a_in_buffer_len=0; if(a52_fillbuff(sh_audio)<0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n"); driver=0;break; } //sh_audio->samplerate=ai.samplerate; // SET by a52_fillbuff() //sh_audio->samplesize=ai.framesize; //sh_audio->i_bps=ai.bitrate*(1000/8); // SET by a52_fillbuff() //sh_audio->ac3_frame=malloc(6144); //sh_audio->o_bps=sh_audio->i_bps; // XXX FIXME!!! XXX // o_bps is calculated from samplesize*channels*samplerate // a single ac3 frame is always translated to 6144 byte packet. (zero padding) sh_audio->channels=2; sh_audio->samplesize=2; // 2*2*(6*256) = 6144 (very TRICKY!) break; } case AFM_ALAW: { // aLaw audio codec: sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps=sh_audio->channels*sh_audio->samplerate; break; } #ifdef USE_G72X case AFM_G72X: { // GSM 723 audio codec: sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps=(sh_audio->samplerate/g72x_data.samplesperblock)*g72x_data.blocksize; break; } #endif #ifdef USE_LIBAVCODEC case AFM_FFMPEG: { int x; 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; } memset(&lavc_context, 0, sizeof(lavc_context)); /* 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"); // 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; #if 1 sh_audio->channels=lavc_context.channels; sh_audio->samplerate=lavc_context.sample_rate; sh_audio->i_bps=lavc_context.bit_rate/8; #else sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; #endif break; } #endif case AFM_GSM: { // MS-GSM audio codec: GSM_Init(); sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; // decodes 65 byte -> 320 short // 1 sec: sh_audio->channels*sh_audio->samplerate samples // 1 frame: 320 samples sh_audio->i_bps=65*(sh_audio->channels*sh_audio->samplerate)/320; // 1:10 break; } case AFM_IMAADPCM: // IMA-ADPCM 4:1 audio codec: sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; // decodes 34 byte -> 64 short sh_audio->i_bps=IMA_ADPCM_BLOCK_SIZE*(sh_audio->channels*sh_audio->samplerate)/IMA_ADPCM_SAMPLES_PER_BLOCK; // 1:4 break; case AFM_MSADPCM: sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps = sh_audio->wf->nBlockAlign * (sh_audio->channels*sh_audio->samplerate) / MS_ADPCM_SAMPLES_PER_BLOCK; break; case AFM_DK4ADPCM: sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps = sh_audio->wf->nBlockAlign * (sh_audio->channels*sh_audio->samplerate) / DK4_ADPCM_SAMPLES_PER_BLOCK; break; case AFM_DK3ADPCM: sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps=DK3_ADPCM_BLOCK_SIZE* (sh_audio->channels*sh_audio->samplerate) / DK3_ADPCM_SAMPLES_PER_BLOCK; break; case AFM_ROQAUDIO: sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; sh_audio->i_bps = (sh_audio->channels * 22050) / 2; break; case AFM_MPEG: { // MPEG Audio: dec_audio_sh=sh_audio; // save sh_audio for the callback: #ifdef USE_FAKE_MONO MP3_Init(fakemono); #else MP3_Init(); #endif MP3_samplerate=MP3_channels=0; sh_audio->a_buffer_len=MP3_DecodeFrame(sh_audio->a_buffer,-1); sh_audio->channels=2; // hack sh_audio->samplerate=MP3_samplerate; sh_audio->i_bps=MP3_bitrate*(1000/8); MP3_PrintHeader(); break; } #ifdef HAVE_OGGVORBIS case AFM_VORBIS: { ogg_packet op; vorbis_comment vc; struct ov_struct_st *ov; /// Init the decoder with the 3 header packets ov = (struct ov_struct_st*)malloc(sizeof(struct ov_struct_st)); vorbis_info_init(&ov->vi); vorbis_comment_init(&vc); op.bytes = ds_get_packet(sh_audio->ds,&op.packet); op.b_o_s = 1; /// Header if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: initial (identification) header broken!\n"); driver = 0; free(ov); break; } op.bytes = ds_get_packet(sh_audio->ds,&op.packet); op.b_o_s = 0; /// Comments if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: comment header broken!\n"); driver = 0; free(ov); break; } op.bytes = ds_get_packet(sh_audio->ds,&op.packet); //// Codebook if(vorbis_synthesis_headerin(&ov->vi,&vc,&op)<0) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n"); driver = 0; free(ov); break; } else { /// Print the infos char **ptr=vc.user_comments; while(*ptr){ mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr); ++ptr; } mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel, %ldHz, %ldkbit/s %cBR\n",ov->vi.channels,ov->vi.rate,ov->vi.bitrate_nominal/1000, (ov->vi.bitrate_lower!=ov->vi.bitrate_nominal)||(ov->vi.bitrate_upper!=ov->vi.bitrate_nominal)?'V':'C'); mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",vc.vendor); } // Setup the decoder sh_audio->channels=ov->vi.channels; sh_audio->samplerate=ov->vi.rate; sh_audio->i_bps=ov->vi.bitrate_nominal/8; sh_audio->context = ov; /// Finish the decoder init vorbis_synthesis_init(&ov->vd,&ov->vi); vorbis_block_init(&ov->vd,&ov->vb); mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n"); } break; #endif #ifdef HAVE_FAAD case AFM_AAC: { unsigned long faac_samplerate, faac_channels; faacDecConfigurationPtr faac_conf; faac_hdec = faacDecOpen(); #if 0 /* Set the default object type and samplerate */ /* This is useful for RAW AAC files */ faac_conf = faacDecGetCurrentConfiguration(faac_hdec); if(sh_audio->samplerate) faac_conf->defSampleRate = sh_audio->samplerate; /* XXX: is outputFormat samplesize of compressed data or samplesize of * decoded data, maybe upsampled? Also, FAAD support FLOAT output, * how do we handle that (FAAD_FMT_FLOAT)? ::atmos */ if(sh_audio->samplesize) switch(sh_audio->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 if(faac_buffer == NULL) faac_buffer = (unsigned char*)malloc(FAAD_BUFFLEN); memset(faac_buffer, 0, FAAD_BUFFLEN); demux_read_data(sh_audio->ds, faac_buffer, FAAD_BUFFLEN); /* init the codec */ if((faac_bytesconsumed = faacDecInit(faac_hdec, faac_buffer, &faac_samplerate, &faac_channels)) < 0) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup! faacDecClose(faac_hdec); free(faac_buffer); faac_buffer = NULL; driver = 0; } else { mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", faac_bytesconsumed); // XXX: remove or move to debug! mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels); sh_audio->channels = faac_channels; sh_audio->samplerate = faac_samplerate; sh_audio->i_bps = 128*1000/8; // XXX: HACK!!! There's currently no way to get bitrate from libfaad2! ::atmos } } break; #endif #ifdef USE_LIBMAD case AFM_MAD: { printf("%s %s %s (%s)\n", mad_version, mad_copyright, mad_author, mad_build); mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: initialising\n"); mad_frame_init(&mad_frame); mad_stream_init(&mad_stream); mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: preparing buffer\n"); mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); mad_stream_buffer(&mad_stream, (unsigned char*)(sh_audio->a_in_buffer), sh_audio->a_in_buffer_len); // mad_stream_sync(&mad_stream); mad_sync(sh_audio, &mad_stream); mad_synth_init(&mad_synth); if(mad_frame_decode(&mad_frame, &mad_stream) == 0) { mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: post processing buffer\n"); mad_postprocess_buffer(sh_audio, &mad_stream); } else { mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: frame decoding failed\n"); mad_print_error(&mad_stream); } switch (mad_frame.header.mode) { case MAD_MODE_SINGLE_CHANNEL: sh_audio->channels=1; break; case MAD_MODE_DUAL_CHANNEL: case MAD_MODE_JOINT_STEREO: case MAD_MODE_STEREO: sh_audio->channels=2; break; default: mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "mad: unknown number of channels\n"); } mp_msg(MSGT_DECAUDIO, MSGL_HINT, "mad: channels: %d (mad channel mode: %d)\n", sh_audio->channels, mad_frame.header.mode); /* var. name changed in 0.13.0 (beta) (libmad/CHANGES) -- alex */ #if (MAD_VERSION_MAJOR >= 0) && (MAD_VERSION_MINOR >= 13) sh_audio->samplerate=mad_frame.header.samplerate; #else sh_audio->samplerate=mad_frame.header.sfreq; #endif sh_audio->i_bps=mad_frame.header.bitrate; mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: continuing\n"); break; } #endif } if(!sh_audio->channels || !sh_audio->samplerate){ mp_msg(MSGT_DECAUDIO,MSGL_WARN,MSGTR_UnknownAudio); driver=0; } if(!driver){ if(sh_audio->a_buffer) free(sh_audio->a_buffer); sh_audio->a_buffer=NULL; return 0; } if(!sh_audio->o_bps) sh_audio->o_bps=sh_audio->channels*sh_audio->samplerate*sh_audio->samplesize; return driver; } // Audio decoding: // Decode a single frame (mp3,acm etc) or 'minlen' bytes (pcm/alaw etc) // buffer length is 'maxlen' bytes, it shouldn't be exceeded... int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){ int len=-1; switch(sh_audio->codec->driver){ #ifdef USE_LIBAVCODEC case AFM_FFMPEG: { unsigned char *start=NULL; int y; while(len<minlen){ int len2=0; int x=ds_get_packet(sh_audio->ds,&start); if(x<=0) break; // error y=avcodec_decode_audio(&lavc_context,(INT16*)buf,&len2,start,x); 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; } mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); } } break; #endif case AFM_MPEG: // MPEG layer 2 or 3 len=MP3_DecodeFrame(buf,-1); // len=MP3_DecodeFrame(buf,3); break; #ifdef HAVE_OGGVORBIS case AFM_VORBIS: { // Vorbis int samples; float **pcm; ogg_packet op; char* np; struct ov_struct_st *ov = sh_audio->context; len = 0; op.b_o_s = op.e_o_s = 0; while(len < minlen) { op.bytes = ds_get_packet(sh_audio->ds,&op.packet); if(!op.packet) break; if(vorbis_synthesis(&ov->vb,&op)==0) /* test for success! */ vorbis_synthesis_blockin(&ov->vd,&ov->vb); while((samples=vorbis_synthesis_pcmout(&ov->vd,&pcm))>0){ int i,j; int clipflag=0; int convsize=(maxlen-len)/(2*ov->vi.channels); // max size! int bout=(samples<convsize?samples:convsize); if(bout<=0) break; /* convert floats to 16 bit signed ints (host order) and interleave */ for(i=0;i<ov->vi.channels;i++){ ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]); ogg_int16_t *ptr=convbuffer+i; float *mono=pcm[i]; for(j=0;j<bout;j++){ #if 1 int val=mono[j]*32767.f; #else /* optional dither */ int val=mono[j]*32767.f+drand48()-0.5f; #endif /* might as well guard against clipping */ if(val>32767){ val=32767; clipflag=1; } if(val<-32768){ val=-32768; clipflag=1; } *ptr=val; ptr+=ov->vi.channels; } } if(clipflag) mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(ov->vd.sequence)); len+=2*ov->vi.channels*bout; mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples); vorbis_synthesis_read(&ov->vd,bout); /* tell libvorbis how many samples we actually consumed */ } } } break; #endif #ifdef HAVE_FAAD case AFM_AAC: { int /*i,*/ k, j = 0; void *faac_sample_buffer; len = 0; while(len < minlen) { /* update buffer */ if (faac_bytesconsumed > 0) { for (k = 0; k < (FAAD_BUFFLEN - faac_bytesconsumed); k++) faac_buffer[k] = faac_buffer[k + faac_bytesconsumed]; demux_read_data(sh_audio->ds, faac_buffer + (FAAD_BUFFLEN) - faac_bytesconsumed, faac_bytesconsumed); faac_bytesconsumed = 0; } /*for (i = 0; i < 16; i++) printf ("%02X ", faac_buffer[i]); printf ("\n");*/ do { faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, faac_buffer+j); /* update buffer index after faacDecDecode */ faac_bytesconsumed += faac_finfo.bytesconsumed; if(faac_finfo.error > 0) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Trying to resync!\n"); j++; } else break; } while(j < FAAD_BUFFLEN); 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 { mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%dBytes)!\n", faac_finfo.samples*faac_finfo.channels); memcpy(buf+len,faac_sample_buffer, faac_finfo.samples*faac_finfo.channels); len += faac_finfo.samples*faac_finfo.channels; } } } break; #endif case AFM_PCM: // AVI PCM len=demux_read_data(sh_audio->ds,buf,minlen); break; case AFM_DVDPCM: // DVD PCM { int j; len=demux_read_data(sh_audio->ds,buf,minlen); //if(i&1){ printf("Warning! pcm_audio_size&1 !=0 (%d)\n",i);i&=~1; } // swap endian: for(j=0;j<len;j+=2){ char x=buf[j]; buf[j]=buf[j+1]; buf[j+1]=x; } break; } case AFM_ALAW: // aLaw decoder { int l=demux_read_data(sh_audio->ds,buf,minlen/2); unsigned short *d=(unsigned short *) buf; unsigned char *s=buf; len=2*l; if(sh_audio->format==6){ // aLaw while(l>0){ --l; d[l]=alaw2short[s[l]]; } } else { // uLaw while(l>0){ --l; d[l]=ulaw2short[s[l]]; } } break; } case AFM_GSM: // MS-GSM decoder { unsigned char ibuf[65]; // 65 bytes / frame if(demux_read_data(sh_audio->ds,ibuf,65)!=65) break; // EOF XA_MSGSM_Decoder(ibuf,(unsigned short *) buf); // decodes 65 byte -> 320 short // XA_GSM_Decoder(buf,(unsigned short *) &sh_audio->a_buffer[sh_audio->a_buffer_len]); // decodes 33 byte -> 160 short len=2*320; break; } #ifdef USE_G72X case AFM_G72X: // GSM 723 decoder { if(demux_read_data(sh_audio->ds,g72x_data.block, g72x_data.blocksize)!=g72x_data.blocksize) break; // EOF g72x_decode_block(&g72x_data); len=2*g72x_data.samplesperblock; memcpy(buf,g72x_data.samples,len); break; } #endif case AFM_IMAADPCM: { unsigned char ibuf[IMA_ADPCM_BLOCK_SIZE * 2]; // bytes / stereo frame if (demux_read_data(sh_audio->ds, ibuf, IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) != IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) break; // EOF len=2*ima_adpcm_decode_block((unsigned short*)buf,ibuf, sh_audio->wf->nChannels); break; } case AFM_MSADPCM: { static unsigned char *ibuf = NULL; if (!ibuf) ibuf = (unsigned char *)malloc (sh_audio->wf->nBlockAlign * sh_audio->wf->nChannels); if (demux_read_data(sh_audio->ds, ibuf, sh_audio->wf->nBlockAlign) != sh_audio->wf->nBlockAlign) break; // EOF len= 2 * ms_adpcm_decode_block( (unsigned short*)buf,ibuf, sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign); break; } case AFM_DK4ADPCM: { static unsigned char *ibuf = NULL; if (!ibuf) ibuf = (unsigned char *)malloc(sh_audio->wf->nBlockAlign); if (demux_read_data(sh_audio->ds, ibuf, sh_audio->wf->nBlockAlign) != sh_audio->wf->nBlockAlign) break; // EOF len=2*dk4_adpcm_decode_block((unsigned short*)buf,ibuf, sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign); break; } case AFM_DK3ADPCM: { unsigned char ibuf[DK3_ADPCM_BLOCK_SIZE * 2]; // bytes / stereo frame if (demux_read_data(sh_audio->ds, ibuf, DK3_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) != DK3_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) break; // EOF len = 2 * dk3_adpcm_decode_block( (unsigned short*)buf,ibuf); break; } case AFM_ROQAUDIO: { static unsigned char *ibuf = NULL; unsigned char header_data[6]; int read_len; if (!ibuf) ibuf = (unsigned char *)malloc(sh_audio->audio_out_minsize / 2); // figure out how much data to read if (demux_read_data(sh_audio->ds, header_data, 6) != 6) break; // EOF read_len = (header_data[5] << 24) | (header_data[4] << 16) | (header_data[3] << 8) | header_data[2]; read_len += 2; // 16-bit arguments if (demux_read_data(sh_audio->ds, ibuf, read_len) != read_len) break; len = 2 * roq_decode_audio((unsigned short *)buf, ibuf, read_len, sh_audio->channels, sh_audio->context); break; } #ifdef USE_LIBAC3 case AFM_AC3: // AC3 decoder //printf("{1:%d}",avi_header.idx_pos);fflush(stdout); if(!sh_audio->ac3_frame) sh_audio->ac3_frame=ac3_decode_frame(); //printf("{2:%d}",avi_header.idx_pos);fflush(stdout); if(sh_audio->ac3_frame){ len = 256 * 6 *sh_audio->channels*sh_audio->samplesize; memcpy(buf,((ac3_frame_t*)sh_audio->ac3_frame)->audio_data,len); sh_audio->ac3_frame=NULL; } //printf("{3:%d}",avi_header.idx_pos);fflush(stdout); break; #endif case AFM_A52: { // AC3 decoder sample_t level=1, bias=384; int flags=a52_flags|A52_ADJUST_LEVEL; int i; if(!sh_audio->a_in_buffer_len) if(a52_fillbuff(sh_audio)<0) break; // EOF sh_audio->a_in_buffer_len=0; if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n"); break; } // a52_dynrng (&state, NULL, NULL); // disable dynamic range compensation // frame decoded, let's resample: //a52_resample_init(a52_accel,flags,sh_audio->channels); len=0; for (i = 0; i < 6; i++) { if (a52_block (&a52_state, a52_samples)){ mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n"); break; } len+=2*a52_resample(a52_samples,&buf[len]); } // printf("len = %d \n",len); // 6144 on all vobs I tried so far... (5.1 and 2.0) ::atmos break; } case AFM_HWAC3: // AC3 through SPDIF if(!sh_audio->a_in_buffer_len) if((len=a52_fillbuff(sh_audio))<0) break; //EOF sh_audio->a_in_buffer_len=0; len = ac3_iec958_build_burst(len, 0x01, 1, sh_audio->a_in_buffer, buf); // len = 6144 = 4*(6*256) break; #ifdef USE_WIN32DLL case AFM_ACM: // len=sh_audio->audio_out_minsize; // optimal decoded fragment size // if(len<minlen) len=minlen; else // if(len>maxlen) len=maxlen; // len=acm_decode_audio(sh_audio,buf,len); len=acm_decode_audio(sh_audio,buf,minlen,maxlen); break; #endif #ifdef USE_DIRECTSHOW case AFM_DSHOW: // DirectShow { int size_in=0; int size_out=0; int srcsize=DS_AudioDecoder_GetSrcSize(ds_adec, maxlen); mp_msg(MSGT_DECAUDIO,MSGL_DBG3,"DShow says: srcsize=%d (buffsize=%d) out_size=%d\n",srcsize,sh_audio->a_in_buffer_size,maxlen); if(srcsize>sh_audio->a_in_buffer_size) srcsize=sh_audio->a_in_buffer_size; // !!!!!! if(sh_audio->a_in_buffer_len<srcsize){ sh_audio->a_in_buffer_len+= demux_read_data(sh_audio->ds,&sh_audio->a_in_buffer[sh_audio->a_in_buffer_len], srcsize-sh_audio->a_in_buffer_len); } DS_AudioDecoder_Convert(ds_adec, sh_audio->a_in_buffer,sh_audio->a_in_buffer_len, buf,maxlen, &size_in,&size_out); mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"DShow: audio %d -> %d converted (in_buf_len=%d of %d) %d\n",size_in,size_out,sh_audio->a_in_buffer_len,sh_audio->a_in_buffer_size,ds_tell_pts(sh_audio->ds)); if(size_in>=sh_audio->a_in_buffer_len){ sh_audio->a_in_buffer_len=0; } else { sh_audio->a_in_buffer_len-=size_in; memcpy(sh_audio->a_in_buffer,&sh_audio->a_in_buffer[size_in],sh_audio->a_in_buffer_len); } len=size_out; break; } #endif #ifdef USE_LIBMAD case AFM_MAD: { mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); // mad_stream_sync(&mad_stream); mad_sync(sh_audio, &mad_stream); if(mad_frame_decode(&mad_frame, &mad_stream) == 0) { mad_synth_frame(&mad_synth, &mad_frame); mad_postprocess_buffer(sh_audio, &mad_stream); /* and fill buffer */ { int i; int end_size = mad_synth.pcm.length; signed short* samples = (signed short*)buf; if(end_size > maxlen/4) end_size=maxlen/4; for(i=0; i<mad_synth.pcm.length; ++i) { *samples++ = mad_scale(mad_synth.pcm.samples[0][i]); *samples++ = mad_scale(mad_synth.pcm.samples[0][i]); // *buf++ = mad_scale(mad_synth.pcm.sampAles[1][i]); } len = end_size*4; } } else { mp_msg(MSGT_DECVIDEO, MSGL_ERR, "mad: frame decoding failed (error: %d)\n", mad_stream.error); mad_print_error(&mad_stream); } break; } #endif } return len; } void resync_audio_stream(sh_audio_t *sh_audio){ switch(sh_audio->codec->driver){ case AFM_MPEG: MP3_DecodeFrame(NULL,-2); // resync MP3_DecodeFrame(NULL,-2); // resync MP3_DecodeFrame(NULL,-2); // resync break; #ifdef USE_LIBAC3 case AFM_AC3: ac3_bitstream_reset(); // reset AC3 bitstream buffer // if(verbose){ printf("Resyncing AC3 audio...");fflush(stdout);} sh_audio->ac3_frame=ac3_decode_frame(); // resync // if(verbose) printf(" OK!\n"); break; #endif #ifdef HAVE_FAAD case AFM_AAC: //if(faac_buffer != NULL) faac_bytesconsumed = 0; memset(faac_buffer, 0, FAAD_BUFFLEN); //demux_read_data(sh_audio->ds, faac_buffer, FAAD_BUFFLEN); break; #endif case AFM_A52: case AFM_ACM: case AFM_DSHOW: case AFM_HWAC3: sh_audio->a_in_buffer_len=0; // reset ACM/DShow audio buffer break; #ifdef USE_LIBMAD case AFM_MAD: mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); // mad_stream_sync(&mad_stream); mad_sync(sh_audio, &mad_stream); mad_postprocess_buffer(sh_audio, &mad_stream); break; #endif } } void skip_audio_frame(sh_audio_t *sh_audio){ switch(sh_audio->codec->driver){ case AFM_MPEG: MP3_DecodeFrame(NULL,-2);break; // skip MPEG frame #ifdef USE_LIBAC3 case AFM_AC3: sh_audio->ac3_frame=ac3_decode_frame();break; // skip AC3 frame #endif case AFM_HWAC3: case AFM_A52: a52_fillbuff(sh_audio);break; // skip AC3 frame case AFM_ACM: case AFM_DSHOW: { int skip=sh_audio->wf->nBlockAlign; if(skip<16){ skip=(sh_audio->wf->nAvgBytesPerSec/16)&(~7); if(skip<16) skip=16; } demux_read_data(sh_audio->ds,NULL,skip); break; } case AFM_PCM: case AFM_DVDPCM: case AFM_ALAW: { int skip=sh_audio->i_bps/16; skip=skip&(~3); demux_read_data(sh_audio->ds,NULL,skip); break; } #ifdef USE_LIBMAD case AFM_MAD: { mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); mad_stream_skip(&mad_stream, 2); // mad_stream_sync(&mad_stream); mad_sync(sh_audio, &mad_stream); mad_postprocess_buffer(sh_audio, &mad_stream); break; } #endif default: ds_fill_buffer(sh_audio->ds); // skip PCM frame } }