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mpv/dec_audio.c

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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 "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/DirectShow/DS_AudioDec.h"
static void* ds_adec=NULL;
#endif
#ifdef HAVE_OGGVORBIS
/* XXX is math.h really needed? - atmos */
#include <math.h>
#include <vorbis/codec.h>
typedef struct ov_struct_st {
ogg_sync_state oy; /* sync and verify incoming physical bitstream */
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
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 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>
static struct mad_stream mad_stream;
static struct mad_frame mad_frame;
static struct mad_synth mad_synth;
// 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)
{
int delta = (unsigned char*)ms->next_frame - (unsigned char *)sh_audio->a_in_buffer;
if(delta != 0) {
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)
{
printf("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)
{
printf("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;
#if 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 */
sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame
#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->channels=1;
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;
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_FOX61ADPCM:
sh_audio->audio_out_minsize=FOX61_ADPCM_SAMPLES_PER_BLOCK * 4;
sh_audio->ds->ss_div=FOX61_ADPCM_SAMPLES_PER_BLOCK;
sh_audio->ds->ss_mul=FOX61_ADPCM_BLOCK_SIZE;
break;
case AFM_FOX62ADPCM:
sh_audio->audio_out_minsize=FOX62_ADPCM_SAMPLES_PER_BLOCK * 4;
sh_audio->ds->ss_div=FOX62_ADPCM_SAMPLES_PER_BLOCK;
sh_audio->ds->ss_mul=FOX62_ADPCM_BLOCK_SIZE;
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:
printf(__FILE__ ":%d:mad: setting minimum outputsize\n", __LINE__);
sh_audio->audio_out_minsize=4608;
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;
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;
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:
sh_audio->channels=a52_printinfo(sh_audio);
if(audio_output_channels<sh_audio->channels)
sh_audio->channels=audio_output_channels;
// channels setup:
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)){
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=1;
sh_audio->samplesize=4; // 1*4*(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_FOX61ADPCM:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps=FOX61_ADPCM_BLOCK_SIZE*
(sh_audio->channels*sh_audio->samplerate) / FOX61_ADPCM_SAMPLES_PER_BLOCK;
break;
case AFM_FOX62ADPCM:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps=FOX62_ADPCM_BLOCK_SIZE*
(sh_audio->channels*sh_audio->samplerate) / FOX62_ADPCM_SAMPLES_PER_BLOCK;
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;
// printf("[\n");
sh_audio->a_buffer_len=MP3_DecodeFrame(sh_audio->a_buffer,-1);
// printf("]\n");
sh_audio->channels=2; // hack
sh_audio->samplerate=MP3_samplerate;
sh_audio->i_bps=MP3_bitrate*(1000/8);
break;
}
#ifdef HAVE_OGGVORBIS
case AFM_VORBIS: {
// OggVorbis Audio:
#if 0 /* just here for reference - atmos */
ogg_sync_state oy; /* sync and verify incoming physical bitstream */
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
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 */
#else
/* nix, nada, rien, nothing, nem, n<>x */
#endif
uint32_t hdrsizes[3];/* stores vorbis header sizes from AVI audio header,
maybe use ogg_uint32_t */
//int i;
int ret;
char *buffer;
ogg_packet hdr;
//ov_struct_t *s=&sh_audio->ov;
sh_audio->ov=malloc(sizeof(ov_struct_t));
//s=&sh_audio->ov;
vorbis_info_init(&sh_audio->ov->vi);
vorbis_comment_init(&sh_audio->ov->vc);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: cbsize: %i\n", sh_audio->wf->cbSize);
memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX), 3*sizeof(uint32_t));
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: Read header sizes: initial: %i comment: %i codebook: %i\n", hdrsizes[0], hdrsizes[1], hdrsizes[2]);
/*for(i=12; i <= 40; i+=2) { // header bruteforce :)
memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+i, 3*sizeof(uint32_t));
printf("OggVorbis: Read header sizes (%i): %ld %ld %ld\n", i, hdrsizes[0], hdrsizes[1], hdrsizes[2]);
}*/
/* read headers */ // FIXME disable sound on errors here, we absolutely need this headers! - atmos
hdr.packet=NULL;
hdr.b_o_s = 1; /* beginning of stream for first packet */
hdr.bytes = hdrsizes[0];
hdr.packet = realloc(hdr.packet,hdr.bytes);
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t),hdr.bytes);
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: initial (identification) header broken!\n");
hdr.b_o_s = 0;
hdr.bytes = hdrsizes[1];
hdr.packet = realloc(hdr.packet,hdr.bytes);
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0],hdr.bytes);
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: comment header broken!\n");
hdr.bytes = hdrsizes[2];
hdr.packet = realloc(hdr.packet,hdr.bytes);
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0]+hdrsizes[1],hdr.bytes);
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n");
hdr.bytes=0;
hdr.packet = realloc(hdr.packet,hdr.bytes); /* free */
/* done with the headers */
/* Throw the comments plus a few lines about the bitstream we're
decoding */
{
char **ptr=sh_audio->ov->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",sh_audio->ov->vi.channels,sh_audio->ov->vi.rate,sh_audio->ov->vi.bitrate_nominal/1000, (sh_audio->ov->vi.bitrate_lower!=sh_audio->ov->vi.bitrate_nominal)||(sh_audio->ov->vi.bitrate_upper!=sh_audio->ov->vi.bitrate_nominal)?'V':'C');
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",sh_audio->ov->vc.vendor);
}
sh_audio->channels=sh_audio->ov->vi.channels;
sh_audio->samplerate=sh_audio->ov->vi.rate;
sh_audio->i_bps=sh_audio->ov->vi.bitrate_nominal/8;
// printf("[\n");
// sh_audio->a_buffer_len=sh_audio->audio_out_minsize;///ov->vi.channels;
// printf("]\n");
/* OK, got and parsed all three headers. Initialize the Vorbis
packet->PCM decoder. */
vorbis_synthesis_init(&sh_audio->ov->vd,&sh_audio->ov->vi); /* central decode state */
vorbis_block_init(&sh_audio->ov->vd,&sh_audio->ov->vb); /* local state for most of the decode
so multiple block decodes can
proceed in parallel. We could init
multiple vorbis_block structures
for vd here */
//printf("OggVorbis: synthesis and block init done.\n");
ogg_sync_init(&sh_audio->ov->oy); /* Now we can read pages */
while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) {
if(ret == -1)
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n");
else
if(ret == 0) {
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: need more data to verify page, reading more data.\n");
/* submit a a_buffer_len block to libvorbis' Ogg layer */
buffer=ogg_sync_buffer(&sh_audio->ov->oy,256);
ogg_sync_wrote(&sh_audio->ov->oy,demux_read_data(sh_audio->ds,buffer,256));
}
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: successfull.\n");
ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og); /* we can ignore any errors here
as they'll also become apparent
at packetout */
/* Get the serial number and set up the rest of decode. */
/* serialno first; use it to set up a logical stream */
ogg_stream_init(&sh_audio->ov->os,ogg_page_serialno(&sh_audio->ov->og));
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");
break;
}
#endif
#ifdef USE_LIBMAD
case AFM_MAD:
{
printf("%s %s %s (%s)\n", mad_version, mad_copyright, mad_author, mad_build);
printf(__FILE__ ":%d:mad: initialising\n", __LINE__);
mad_frame_init(&mad_frame);
mad_stream_init(&mad_stream);
printf(__FILE__ ":%d:mad: preparing buffer\n", __LINE__);
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)
{
printf(__FILE__ ":%d:mad: post processing buffer\n", __LINE__);
mad_postprocess_buffer(sh_audio, &mad_stream);
}
else
{
printf(__FILE__ ":%d:mad: frame decoding failed\n", __LINE__);
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;
printf(__FILE__ ":%d:mad: continuing\n", __LINE__);
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: { // OggVorbis
/* note: good minlen would be 4k or 8k IMHO - atmos */
int ret;
char *buffer;
int bytes;
int samples;
float **pcm;
//ogg_int16_t convbuffer[4096];
// int convsize;
int readlen=1024;
len=0;
// convsize=minlen/sh_audio->ov->vi.channels;
while(len < minlen) { /* double loop allows for break in inner loop */
while(len < minlen) { /* without aborting the outer loop - atmos */
ret=ogg_stream_packetout(&sh_audio->ov->os,&sh_audio->ov->op);
if(ret==0) {
int xxx=0;
//printf("OggVorbis: Packetout: need more data, paging!\n");
while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) {
if(ret == -1)
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n");
else
if(ret == 0) {
//printf("OggVorbis: Pageout: need more data to verify page, reading more data.\n");
/* submit a readlen k block to libvorbis' Ogg layer */
buffer=ogg_sync_buffer(&sh_audio->ov->oy,readlen);
bytes=demux_read_data(sh_audio->ds,buffer,readlen);
xxx+=bytes;
ogg_sync_wrote(&sh_audio->ov->oy,bytes);
if(bytes==0)
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: 0Bytes written, possible End of Stream\n");
}
}
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[sync: %d ]\n",xxx);
//printf("OggVorbis: Pageout: successfull, pagin in.\n");
if(ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og)<0)
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pagein failed!\n");
break;
} else if(ret<0) {
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Packetout: missing or corrupt data, skipping packet!\n");
break;
} else {
/* we have a packet. Decode it */
if(vorbis_synthesis(&sh_audio->ov->vb,&sh_audio->ov->op)==0) /* test for success! */
vorbis_synthesis_blockin(&sh_audio->ov->vd,&sh_audio->ov->vb);
/* **pcm is a multichannel float vector. In stereo, for
example, pcm[0] is left, and pcm[1] is right. samples is
the size of each channel. Convert the float values
(-1.<=range<=1.) to whatever PCM format and write it out */
while((samples=vorbis_synthesis_pcmout(&sh_audio->ov->vd,&pcm))>0){
int i,j;
int clipflag=0;
int convsize=(maxlen-len)/(2*sh_audio->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<sh_audio->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+=sh_audio->ov->vi.channels;
}
}
if(clipflag)
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(sh_audio->ov->vd.sequence));
//fwrite(convbuffer,2*sh_audio->ov->vi.channels,bout,stderr); //dump pcm to file for debugging
//memcpy(buf+len,convbuffer,2*sh_audio->ov->vi.channels*bout);
len+=2*sh_audio->ov->vi.channels*bout;
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples);
vorbis_synthesis_read(&sh_audio->ov->vd,bout); /* tell libvorbis how
many samples we
actually consumed */
}
} // from else, packetout ok
} // while len
} // outer while len
if(ogg_page_eos(&sh_audio->ov->og))
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: End of Stream reached!\n"); // FIXME clearup decoder, notify mplayer - atmos
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[len: %d ]\n",len);
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_FOX61ADPCM:
{ unsigned char ibuf[FOX61_ADPCM_BLOCK_SIZE]; // bytes / stereo frame
if (demux_read_data(sh_audio->ds, ibuf, FOX61_ADPCM_BLOCK_SIZE) !=
FOX61_ADPCM_BLOCK_SIZE)
break; // EOF
len=2*fox61_adpcm_decode_block((unsigned short*)buf,ibuf);
break;
}
case AFM_FOX62ADPCM:
{ unsigned char ibuf[FOX62_ADPCM_BLOCK_SIZE * 2]; // bytes / stereo frame
if (demux_read_data(sh_audio->ds, ibuf,
FOX62_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) !=
FOX62_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels)
break; // EOF
len = 2 * fox62_adpcm_decode_block(
(unsigned short*)buf,ibuf, sh_audio->wf->nChannels);
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
{
printf(__FILE__ ":%d:mad: frame decoding failed (error: %d)\n", __LINE__,
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 HAVE_OGGVORBIS
case AFM_VORBIS:
//printf("OggVorbis: resetting stream.\n");
ogg_sync_reset(&sh_audio->ov->oy);
ogg_stream_reset(&sh_audio->ov->os);
break;
#endif
#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
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
}
}
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