mpv/libmpdemux/demux_audio.c

535 lines
16 KiB
C

#include "config.h"
#include "../mp_msg.h"
#include <stdlib.h>
#include <stdio.h>
#include "stream.h"
#include "demuxer.h"
#include "stheader.h"
#include "genres.h"
#include "mp3_hdr.h"
#include <string.h>
#ifdef MP_DEBUG
#include <assert.h>
#endif
#define MP3 1
#define WAV 2
#define fLaC 3
#define HDR_SIZE 4
typedef struct da_priv {
int frmt;
float last_pts;
} da_priv_t;
// how many valid frames in a row we need before accepting as valid MP3
#define MIN_MP3_HDRS 5
//! Used to describe a potential (chain of) MP3 headers we found
typedef struct mp3_hdr {
off_t frame_pos; // start of first frame in this "chain" of headers
off_t next_frame_pos; // here we expect the next header with same parameters
int mp3_chans;
int mp3_freq;
int cons_hdrs; // if this reaches MIN_MP3_HDRS we accept as MP3 file
struct mp3_hdr *next;
} mp3_hdr_t;
extern void free_sh_audio(sh_audio_t* sh);
extern void resync_audio_stream(sh_audio_t *sh_audio);
extern void print_wave_header(WAVEFORMATEX *h);
int hr_mp3_seek = 0;
/**
* \brief free a list of MP3 header descriptions
* \param list pointer to the head-of-list pointer
*/
static void free_mp3_hdrs(mp3_hdr_t **list) {
mp3_hdr_t *tmp;
while (*list) {
tmp = (*list)->next;
free(*list);
*list = tmp;
}
}
/**
* \brief add another potential MP3 header to our list
* If it fits into an existing chain this one is expanded otherwise
* a new one is created.
* All entries that expected a MP3 header before the current position
* are discarded.
* The list is expected to be and will be kept sorted by next_frame_pos
* and when those are equal by frame_pos.
* \param list pointer to the head-of-list pointer
* \param st_pos stream position where the described header starts
* \param mp3_chans number of channels as specified by the header
* \param mp3_freq sampling frequency as specified by the header
* \param mp3_flen length of the frame as specified by the header
* \return If non-null the current file is accepted as MP3 and the
* mp3_hdr struct describing the valid chain is returned. Must be
* freed independent of the list.
*/
static mp3_hdr_t *add_mp3_hdr(mp3_hdr_t **list, off_t st_pos,
int mp3_chans, int mp3_freq, int mp3_flen) {
mp3_hdr_t *tmp;
int in_list = 0;
while (*list && (*list)->next_frame_pos <= st_pos) {
if (((*list)->next_frame_pos < st_pos) || ((*list)->mp3_chans != mp3_chans)
|| ((*list)->mp3_freq != mp3_freq)) { // wasn't valid!
tmp = (*list)->next;
free(*list);
*list = tmp;
} else {
(*list)->cons_hdrs++;
(*list)->next_frame_pos = st_pos + mp3_flen;
if ((*list)->cons_hdrs >= MIN_MP3_HDRS) {
// copy the valid entry, so that the list can be easily freed
tmp = malloc(sizeof(mp3_hdr_t));
memcpy(tmp, *list, sizeof(mp3_hdr_t));
tmp->next = NULL;
return tmp;
}
in_list = 1;
list = &((*list)->next);
}
}
if (!in_list) { // does not belong into an existing chain, insert
tmp = malloc(sizeof(mp3_hdr_t));
tmp->frame_pos = st_pos;
tmp->next_frame_pos = st_pos + mp3_flen;
tmp->mp3_chans = mp3_chans;
tmp->mp3_freq = mp3_freq;
tmp->cons_hdrs = 1;
tmp->next = *list;
*list = tmp;
}
return NULL;
}
int demux_audio_open(demuxer_t* demuxer) {
stream_t *s;
sh_audio_t* sh_audio;
uint8_t hdr[HDR_SIZE];
int frmt = 0, n = 0, step, mp3_freq, mp3_chans, mp3_flen;
off_t st_pos = 0, next_frame_pos = 0;
// mp3_hdrs list is sorted first by next_frame_pos and then by frame_pos
mp3_hdr_t *mp3_hdrs = NULL, *mp3_found = NULL;
da_priv_t* priv;
#ifdef MP_DEBUG
assert(demuxer != NULL);
assert(demuxer->stream != NULL);
#endif
s = demuxer->stream;
stream_read(s, hdr, HDR_SIZE);
while(n < 30000 && !s->eof) {
st_pos = stream_tell(s) - HDR_SIZE;
step = 1;
if( hdr[0] == 'R' && hdr[1] == 'I' && hdr[2] == 'F' && hdr[3] == 'F' ) {
stream_skip(s,4);
if(s->eof)
break;
stream_read(s,hdr,4);
if(s->eof)
break;
if(hdr[0] != 'W' || hdr[1] != 'A' || hdr[2] != 'V' || hdr[3] != 'E' )
stream_skip(s,-8);
else
// We found wav header. Now we can have 'fmt ' or a mp3 header
// empty the buffer
step = 4;
} else if( hdr[0] == 'I' && hdr[1] == 'D' && hdr[2] == '3' && (hdr[3] >= 2)) {
int len;
stream_skip(s,2);
stream_read(s,hdr,4);
len = (hdr[0]<<21) | (hdr[1]<<14) | (hdr[2]<<7) | hdr[3];
stream_skip(s,len);
step = 4;
} else if( hdr[0] == 'f' && hdr[1] == 'm' && hdr[2] == 't' && hdr[3] == ' ' ) {
frmt = WAV;
break;
} else if((mp3_flen = mp_get_mp3_header(hdr,&mp3_chans,&mp3_freq)) > 0) {
mp3_found = add_mp3_hdr(&mp3_hdrs, st_pos, mp3_chans, mp3_freq, mp3_flen);
if (mp3_found) {
frmt = MP3;
break;
}
} else if( hdr[0] == 'f' && hdr[1] == 'L' && hdr[2] == 'a' && hdr[3] == 'C' ) {
frmt = fLaC;
stream_skip(s,-4);
break;
}
// Add here some other audio format detection
if(step < HDR_SIZE)
memmove(hdr,&hdr[step],HDR_SIZE-step);
stream_read(s, &hdr[HDR_SIZE - step], step);
n++;
}
free_mp3_hdrs(&mp3_hdrs);
if(!frmt)
return 0;
sh_audio = new_sh_audio(demuxer,0);
switch(frmt) {
case MP3:
sh_audio->format = 0x55;
demuxer->movi_start = mp3_found->frame_pos;
next_frame_pos = mp3_found->next_frame_pos;
sh_audio->audio.dwSampleSize= 0;
sh_audio->audio.dwScale = 1152;
sh_audio->audio.dwRate = mp3_found->mp3_freq;
sh_audio->wf = malloc(sizeof(WAVEFORMATEX));
sh_audio->wf->wFormatTag = sh_audio->format;
sh_audio->wf->nChannels = mp3_found->mp3_chans;
sh_audio->wf->nSamplesPerSec = mp3_found->mp3_freq;
sh_audio->wf->nBlockAlign = 1152;
sh_audio->wf->wBitsPerSample = 16;
sh_audio->wf->cbSize = 0;
free(mp3_found);
mp3_found = NULL;
if(s->end_pos) {
char tag[4];
stream_seek(s,s->end_pos-128);
stream_read(s,tag,3);
tag[3] = '\0';
if(strcmp(tag,"TAG"))
demuxer->movi_end = s->end_pos;
else {
char buf[31];
uint8_t g;
demuxer->movi_end = stream_tell(s)-3;
stream_read(s,buf,30);
buf[30] = '\0';
demux_info_add(demuxer,"Title",buf);
stream_read(s,buf,30);
buf[30] = '\0';
demux_info_add(demuxer,"Artist",buf);
stream_read(s,buf,30);
buf[30] = '\0';
demux_info_add(demuxer,"Album",buf);
stream_read(s,buf,4);
buf[4] = '\0';
demux_info_add(demuxer,"Year",buf);
stream_read(s,buf,30);
buf[30] = '\0';
demux_info_add(demuxer,"Comment",buf);
if(buf[28] == 0 && buf[29] != 0) {
uint8_t trk = (uint8_t)buf[29];
sprintf(buf,"%d",trk);
demux_info_add(demuxer,"Track",buf);
}
g = stream_read_char(s);
demux_info_add(demuxer,"Genre",genres[g]);
}
}
break;
case WAV: {
unsigned int chunk_type;
unsigned int chunk_size;
WAVEFORMATEX* w;
int l;
l = stream_read_dword_le(s);
if(l < 16) {
mp_msg(MSGT_DEMUX,MSGL_ERR,"[demux_audio] Bad wav header length: too short (%d)!!!\n",l);
free_sh_audio(sh_audio);
return 0;
}
sh_audio->wf = w = (WAVEFORMATEX*)malloc(l > sizeof(WAVEFORMATEX) ? l : sizeof(WAVEFORMATEX));
w->wFormatTag = sh_audio->format = stream_read_word_le(s);
w->nChannels = sh_audio->channels = stream_read_word_le(s);
w->nSamplesPerSec = sh_audio->samplerate = stream_read_dword_le(s);
w->nAvgBytesPerSec = stream_read_dword_le(s);
w->nBlockAlign = stream_read_word_le(s);
w->wBitsPerSample = sh_audio->samplesize = stream_read_word_le(s);
w->cbSize = 0;
l -= 16;
if (l > 0) {
w->cbSize = stream_read_word_le(s);
l -= 2;
if (w->cbSize > 0) {
if (l < w->cbSize) {
mp_msg(MSGT_DEMUX,MSGL_ERR,"[demux_audio] truncated extradata (%d < %d)\n",
l,w->cbSize);
stream_read(s,(char*)((char*)(w)+sizeof(WAVEFORMATEX)),l);
l = 0;
} else {
stream_read(s,(char*)((char*)(w)+sizeof(WAVEFORMATEX)),w->cbSize);
l -= w->cbSize;
}
}
}
if(verbose>0) print_wave_header(w);
if(l)
stream_skip(s,l);
do
{
chunk_type = stream_read_fourcc(demuxer->stream);
chunk_size = stream_read_dword_le(demuxer->stream);
if (chunk_type != mmioFOURCC('d', 'a', 't', 'a'))
stream_skip(demuxer->stream, chunk_size);
} while (chunk_type != mmioFOURCC('d', 'a', 't', 'a'));
demuxer->movi_start = stream_tell(s);
demuxer->movi_end = s->end_pos;
// printf("wav: %X .. %X\n",(int)demuxer->movi_start,(int)demuxer->movi_end);
// Check if it contains dts audio
if((w->wFormatTag == 0x01) && (w->nChannels == 2) && (w->nSamplesPerSec == 44100)) {
unsigned char buf[16384]; // vlc uses 16384*4 (4 dts frames)
unsigned int i;
stream_read(s, buf, sizeof(buf));
for (i = 0; i < sizeof(buf); i += 2) {
// DTS, 14 bit, LE
if((buf[i] == 0xff) && (buf[i+1] == 0x1f) && (buf[i+2] == 0x00) &&
(buf[i+3] == 0xe8) && ((buf[i+4] & 0xfe) == 0xf0) && (buf[i+5] == 0x07)) {
sh_audio->format = 0x2001;
mp_msg(MSGT_DEMUX,MSGL_V,"[demux_audio] DTS audio in wav, 14 bit, LE\n");
break;
}
// DTS, 14 bit, BE
if((buf[i] == 0x1f) && (buf[i+1] == 0xff) && (buf[i+2] == 0xe8) &&
(buf[i+3] == 0x00) && (buf[i+4] == 0x07) && ((buf[i+5] & 0xfe) == 0xf0)) {
sh_audio->format = 0x2001;
mp_msg(MSGT_DEMUX,MSGL_V,"[demux_audio] DTS audio in wav, 14 bit, BE\n");
break;
}
// DTS, 16 bit, BE
if((buf[i] == 0x7f) && (buf[i+1] == 0xfe) && (buf[i+2] == 0x80) &&
(buf[i+3] == 0x01)) {
sh_audio->format = 0x2001;
mp_msg(MSGT_DEMUX,MSGL_V,"[demux_audio] DTS audio in wav, 16 bit, BE\n");
break;
}
// DTS, 16 bit, LE
if((buf[i] == 0xfe) && (buf[i+1] == 0x7f) && (buf[i+2] == 0x01) &&
(buf[i+3] == 0x80)) {
sh_audio->format = 0x2001;
mp_msg(MSGT_DEMUX,MSGL_V,"[demux_audio] DTS audio in wav, 16 bit, LE\n");
break;
}
}
if (sh_audio->format == 0x2001)
mp_msg(MSGT_DEMUX,MSGL_DBG2,"[demux_audio] DTS sync offset = %u\n", i);
}
stream_seek(s,demuxer->movi_start);
} break;
case fLaC:
sh_audio->format = mmioFOURCC('f', 'L', 'a', 'C');
demuxer->movi_start = stream_tell(s);
demuxer->movi_end = s->end_pos;
break;
}
priv = (da_priv_t*)malloc(sizeof(da_priv_t));
priv->frmt = frmt;
priv->last_pts = -1;
demuxer->priv = priv;
demuxer->audio->id = 0;
demuxer->audio->sh = sh_audio;
sh_audio->ds = demuxer->audio;
sh_audio->samplerate = sh_audio->audio.dwRate;
if(stream_tell(s) != demuxer->movi_start)
{
mp_msg(MSGT_DEMUX, MSGL_V, "demux_audio: seeking from 0x%X to start pos 0x%X\n",
(int)stream_tell(s), (int)demuxer->movi_start);
stream_seek(s,demuxer->movi_start);
if (stream_tell(s) != demuxer->movi_start) {
mp_msg(MSGT_DEMUX, MSGL_V, "demux_audio: seeking failed, now at 0x%X!\n",
(int)stream_tell(s));
if (next_frame_pos) {
mp_msg(MSGT_DEMUX, MSGL_V, "demux_audio: seeking to 0x%X instead\n",
(int)next_frame_pos);
stream_seek(s, next_frame_pos);
}
}
}
mp_msg(MSGT_DEMUX,MSGL_V,"demux_audio: audio data 0x%X - 0x%X \n",(int)demuxer->movi_start,(int)demuxer->movi_end);
return 1;
}
int demux_audio_fill_buffer(demux_stream_t *ds) {
int l;
demux_packet_t* dp;
sh_audio_t* sh_audio;
demuxer_t* demux;
da_priv_t* priv;
stream_t* s;
#ifdef MP_DEBUG
assert(ds != NULL);
assert(ds->sh != NULL);
assert(ds->demuxer != NULL);
#endif
sh_audio = ds->sh;
demux = ds->demuxer;
priv = demux->priv;
s = demux->stream;
if(s->eof || (demux->movi_end && stream_tell(s) >= demux->movi_end) )
return 0;
switch(priv->frmt) {
case MP3 :
while(1) {
uint8_t hdr[4];
stream_read(s,hdr,4);
if (s->eof || (demux->movi_end && stream_tell(s) >= demux->movi_end))
return 0;
l = mp_decode_mp3_header(hdr);
if(l < 0) {
stream_skip(s,-3);
} else {
dp = new_demux_packet(l);
memcpy(dp->buffer,hdr,4);
stream_read(s,dp->buffer + 4,l-4);
priv->last_pts = priv->last_pts < 0 ? 0 : priv->last_pts + 1152/(float)sh_audio->samplerate; // FIXME: 1152->576 if MPEG-2
break;
}
} break;
case WAV : {
l = sh_audio->wf->nAvgBytesPerSec;
dp = new_demux_packet(l);
l = stream_read(s,dp->buffer,l);
priv->last_pts = priv->last_pts < 0 ? 0 : priv->last_pts + l/(float)sh_audio->i_bps;
break;
}
case fLaC: {
l = 65535;
dp = new_demux_packet(l);
l = stream_read(s,dp->buffer,l);
priv->last_pts = priv->last_pts < 0 ? 0 : priv->last_pts + l/(float)sh_audio->i_bps;
break;
}
default:
printf("Audio demuxer : unknown format %d\n",priv->frmt);
return 0;
}
resize_demux_packet(dp, l);
ds->pts = priv->last_pts - (ds_tell_pts(demux->audio) -
sh_audio->a_in_buffer_len)/(float)sh_audio->i_bps;
ds_add_packet(ds, dp);
return 1;
}
static void high_res_mp3_seek(demuxer_t *demuxer,float time) {
uint8_t hdr[4];
int len,nf;
da_priv_t* priv = demuxer->priv;
sh_audio_t* sh = (sh_audio_t*)demuxer->audio->sh;
nf = time*sh->samplerate/1152;
while(nf > 0) {
stream_read(demuxer->stream,hdr,4);
len = mp_decode_mp3_header(hdr);
if(len < 0) {
stream_skip(demuxer->stream,-3);
continue;
}
stream_skip(demuxer->stream,len-4);
priv->last_pts += 1152/(float)sh->samplerate;
nf--;
}
}
void demux_audio_seek(demuxer_t *demuxer,float rel_seek_secs,int flags){
sh_audio_t* sh_audio;
stream_t* s;
int base,pos;
float len;
da_priv_t* priv;
if(!(sh_audio = demuxer->audio->sh))
return;
s = demuxer->stream;
priv = demuxer->priv;
if(priv->frmt == MP3 && hr_mp3_seek && !(flags & 2)) {
len = (flags & 1) ? rel_seek_secs - priv->last_pts : rel_seek_secs;
if(len < 0) {
stream_seek(s,demuxer->movi_start);
len = priv->last_pts + len;
priv->last_pts = 0;
}
if(len > 0)
high_res_mp3_seek(demuxer,len);
sh_audio->delay = priv->last_pts - (ds_tell_pts(demuxer->audio)-sh_audio->a_in_buffer_len)/(float)sh_audio->i_bps;
resync_audio_stream(sh_audio);
return;
}
base = flags&1 ? demuxer->movi_start : stream_tell(s);
if(flags&2)
pos = base + ((demuxer->movi_end - demuxer->movi_start)*rel_seek_secs);
else
pos = base + (rel_seek_secs*sh_audio->i_bps);
if(demuxer->movi_end && pos >= demuxer->movi_end) {
pos = demuxer->movi_end;
//sh_audio->delay = (stream_tell(s) - demuxer->movi_start)/(float)sh_audio->i_bps;
//return;
} else if(pos < demuxer->movi_start)
pos = demuxer->movi_start;
priv->last_pts = (pos-demuxer->movi_start)/(float)sh_audio->i_bps;
sh_audio->delay = priv->last_pts - (ds_tell_pts(demuxer->audio)-sh_audio->a_in_buffer_len)/(float)sh_audio->i_bps;
switch(priv->frmt) {
case WAV:
pos -= (pos % (sh_audio->channels * sh_audio->samplesize) );
// We need to decrease the pts by one step to make it the "last one"
priv->last_pts -= sh_audio->wf->nAvgBytesPerSec/(float)sh_audio->i_bps;
break;
}
stream_seek(s,pos);
resync_audio_stream(sh_audio);
}
void demux_close_audio(demuxer_t* demuxer) {
da_priv_t* priv = demuxer->priv;
if(!priv)
return;
free(priv);
}
int demux_audio_control(demuxer_t *demuxer,int cmd, void *arg){
sh_audio_t *sh_audio=demuxer->audio->sh;
int audio_length = demuxer->movi_end / sh_audio->i_bps;
da_priv_t* priv = demuxer->priv;
switch(cmd) {
case DEMUXER_CTRL_GET_TIME_LENGTH:
if (audio_length<=0) return DEMUXER_CTRL_DONTKNOW;
*((unsigned long *)arg)=(unsigned long)audio_length;
return DEMUXER_CTRL_GUESS;
case DEMUXER_CTRL_GET_PERCENT_POS:
if (audio_length<=0)
return DEMUXER_CTRL_DONTKNOW;
*((int *)arg)=(int)( (priv->last_pts*100) / audio_length);
return DEMUXER_CTRL_OK;
default:
return DEMUXER_CTRL_NOTIMPL;
}
}