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af6f08ce6a
mpv/libmpdemux/demux_audio.c
reimar af2988cbce Finally rename the STREAM_SEEK define to MP_STREAM_SEEK, there are just too many 
name clashes, in particular with Windows headers (which define STREAM_SEEK as an enum type).


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@29962 b3059339-0415-0410-9bf9-f77b7e298cf2
2009-11-22 15:18:21 +00:00

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/*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"
#include <stdlib.h>
#include <stdio.h>
#include "stream/stream.h"
#include "demuxer.h"
#include "stheader.h"
#include "genres.h"
#include "mp3_hdr.h"
#include "libavutil/intreadwrite.h"
#include <string.h>
#define MP3 1
#define WAV 2
#define fLaC 3
#define HDR_SIZE 4
typedef struct da_priv {
int frmt;
double next_pts;
} da_priv_t;
//! rather arbitrary value for maximum length of wav-format headers
#define MAX_WAVHDR_LEN (1 * 1024 * 1024)
//! how many valid frames in a row we need before accepting as valid MP3
#define MIN_MP3_HDRS 12
//! 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 mpa_spf;
int mpa_layer;
int mpa_br;
int cons_hdrs; // if this reaches MIN_MP3_HDRS we accept as MP3 file
struct mp3_hdr *next;
} mp3_hdr_t;
void print_wave_header(WAVEFORMATEX *h, int verbose_level);
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 mpa_spf frame size as specified by the header
* \param mpa_layer layer type ("version") as specified by the header (*)
* \param mpa_br bitrate 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.
*
* parameters marked by (*) must be the same for all headers in the same chain
*/
static mp3_hdr_t *add_mp3_hdr(mp3_hdr_t **list, off_t st_pos,
int mp3_chans, int mp3_freq, int mpa_spf,
int mpa_layer, int mpa_br, 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) || ((*list)->mpa_layer != mpa_layer) ) {
// wasn't valid!
tmp = (*list)->next;
free(*list);
*list = tmp;
} else {
(*list)->cons_hdrs++;
(*list)->next_frame_pos = st_pos + mp3_flen;
(*list)->mpa_spf = mpa_spf;
(*list)->mpa_br = mpa_br;
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
// find right position to insert to keep sorting
while (*list && (*list)->next_frame_pos <= st_pos + mp3_flen)
list = &((*list)->next);
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->mpa_spf = mpa_spf;
tmp->mpa_layer = mpa_layer;
tmp->mpa_br = mpa_br;
tmp->cons_hdrs = 1;
tmp->next = *list;
*list = tmp;
}
return NULL;
}
#if 0 /* this code is a mess, clean it up before reenabling */
#define FLAC_SIGNATURE_SIZE 4
#define FLAC_STREAMINFO_SIZE 34
#define FLAC_SEEKPOINT_SIZE 18
enum {
FLAC_STREAMINFO = 0,
FLAC_PADDING,
FLAC_APPLICATION,
FLAC_SEEKTABLE,
FLAC_VORBIS_COMMENT,
FLAC_CUESHEET
} flac_preamble_t;
static void
get_flac_metadata (demuxer_t* demuxer)
{
uint8_t preamble[4];
unsigned int blk_len;
stream_t *s = demuxer->stream;
/* file is qualified; skip over the signature bytes in the stream */
stream_seek (s, 4);
/* loop through the metadata blocks; use a do-while construct since there
* will always be 1 metadata block */
do {
int r;
r = stream_read (s, (char *) preamble, FLAC_SIGNATURE_SIZE);
if (r != FLAC_SIGNATURE_SIZE)
return;
blk_len = AV_RB24(preamble + 1);
switch (preamble[0] & 0x7F)
{
case FLAC_VORBIS_COMMENT:
{
/* For a description of the format please have a look at */
/* http://www.xiph.org/vorbis/doc/v-comment.html */
uint32_t length, comment_list_len;
char comments[blk_len];
uint8_t *ptr = comments;
char *comment;
int cn;
char c;
if (stream_read (s, comments, blk_len) == blk_len)
{
length = AV_RL32(ptr);
ptr += 4 + length;
comment_list_len = AV_RL32(ptr);
ptr += 4;
cn = 0;
for (; cn < comment_list_len; cn++)
{
length = AV_RL32(ptr);
ptr += 4;
comment = ptr;
if (&comment[length] < comments || &comment[length] >= &comments[blk_len])
return;
c = comment[length];
comment[length] = 0;
if (!strncasecmp ("TITLE=", comment, 6) && (length - 6 > 0))
demux_info_add (demuxer, "Title", comment + 6);
else if (!strncasecmp ("ARTIST=", comment, 7) && (length - 7 > 0))
demux_info_add (demuxer, "Artist", comment + 7);
else if (!strncasecmp ("ALBUM=", comment, 6) && (length - 6 > 0))
demux_info_add (demuxer, "Album", comment + 6);
else if (!strncasecmp ("DATE=", comment, 5) && (length - 5 > 0))
demux_info_add (demuxer, "Year", comment + 5);
else if (!strncasecmp ("GENRE=", comment, 6) && (length - 6 > 0))
demux_info_add (demuxer, "Genre", comment + 6);
else if (!strncasecmp ("Comment=", comment, 8) && (length - 8 > 0))
demux_info_add (demuxer, "Comment", comment + 8);
else if (!strncasecmp ("TRACKNUMBER=", comment, 12)
&& (length - 12 > 0))
{
char buf[31];
buf[30] = '\0';
sprintf (buf, "%d", atoi (comment + 12));
demux_info_add(demuxer, "Track", buf);
}
comment[length] = c;
ptr += length;
}
}
break;
}
case FLAC_STREAMINFO:
case FLAC_PADDING:
case FLAC_APPLICATION:
case FLAC_SEEKTABLE:
case FLAC_CUESHEET:
default:
/* 6-127 are presently reserved */
stream_skip (s, blk_len);
break;
}
} while ((preamble[0] & 0x80) == 0);
}
#endif
static 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;
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;
s = demuxer->stream;
stream_read(s, hdr, HDR_SIZE);
while(n < 30000 && !s->eof) {
int mp3_freq, mp3_chans, mp3_flen, mpa_layer, mpa_spf, mpa_br;
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,
&mpa_spf, &mpa_layer, &mpa_br)) > 0) {
mp3_found = add_mp3_hdr(&mp3_hdrs, st_pos, mp3_chans, mp3_freq,
mpa_spf, mpa_layer, mpa_br, mp3_flen);
if (mp3_found) {
frmt = MP3;
break;
}
} else if( hdr[0] == 'f' && hdr[1] == 'L' && hdr[2] == 'a' && hdr[3] == 'C' ) {
frmt = fLaC;
if (!mp3_hdrs || mp3_hdrs->cons_hdrs < 3)
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 = (mp3_found->mpa_layer < 3 ? 0x50 : 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 = mp3_found->mpa_spf;
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->nAvgBytesPerSec = mp3_found->mpa_br * (1000 / 8);
sh_audio->wf->nBlockAlign = mp3_found->mpa_spf;
sh_audio->wf->wBitsPerSample = 16;
sh_audio->wf->cbSize = 0;
sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
free(mp3_found);
mp3_found = NULL;
if(s->end_pos && (s->flags & MP_STREAM_SEEK) == MP_STREAM_SEEK) {
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);
l = 16;
}
if(l > MAX_WAVHDR_LEN) {
mp_msg(MSGT_DEMUX,MSGL_ERR,"[demux_audio] Bad wav header length: too long (%d)!!!\n",l);
l = 16;
}
sh_audio->wf = w = 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 = stream_read_word_le(s);
sh_audio->samplesize = (w->wBitsPerSample + 7) / 8;
w->cbSize = 0;
sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
l -= 16;
if (l >= 2) {
w->cbSize = stream_read_word_le(s);
l -= 2;
if (l < w->cbSize) {
mp_msg(MSGT_DEMUX,MSGL_ERR,"[demux_audio] truncated extradata (%d < %d)\n",
l,w->cbSize);
w->cbSize = l;
}
stream_read(s,(char*)((char*)(w)+sizeof(WAVEFORMATEX)),w->cbSize);
l -= w->cbSize;
}
if( mp_msg_test(MSGT_DEMUX,MSGL_V) ) print_wave_header(w, MSGL_V);
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 (!s->eof && chunk_type != mmioFOURCC('d', 'a', 't', 'a'));
demuxer->movi_start = stream_tell(s);
demuxer->movi_end = chunk_size ? demuxer->movi_start + chunk_size : 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;
memset(buf, 0, sizeof(buf));
stream_read(s, buf, sizeof(buf));
for (i = 0; i < sizeof(buf) - 5; 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) - 4;
demuxer->movi_end = s->end_pos;
if (demuxer->movi_end > demuxer->movi_start) {
// try to find out approx. bitrate
int64_t size = demuxer->movi_end - demuxer->movi_start;
int64_t num_samples = 0;
int32_t srate = 0;
stream_skip(s, 14);
stream_read(s, (char *)&srate, 3);
srate = be2me_32(srate) >> 12;
stream_read(s, (char *)&num_samples, 5);
num_samples = (be2me_64(num_samples) >> 24) & 0xfffffffffULL;
if (num_samples && srate)
sh_audio->i_bps = size * srate / num_samples;
}
if (sh_audio->i_bps < 1) // guess value to prevent crash
sh_audio->i_bps = 64 * 1024;
// get_flac_metadata (demuxer);
break;
}
priv = malloc(sizeof(da_priv_t));
priv->frmt = frmt;
priv->next_pts = 0;
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 DEMUXER_TYPE_AUDIO;
}
static int demux_audio_fill_buffer(demuxer_t *demux, demux_stream_t *ds) {
int l;
demux_packet_t* dp;
sh_audio_t* sh_audio = ds->sh;
da_priv_t* priv = demux->priv;
double this_pts = priv->next_pts;
stream_t* s = demux->stream;
if(s->eof)
return 0;
switch(priv->frmt) {
case MP3 :
while(1) {
uint8_t hdr[4];
stream_read(s,hdr,4);
if (s->eof)
return 0;
l = mp_decode_mp3_header(hdr);
if(l < 0) {
if (demux->movi_end && stream_tell(s) >= demux->movi_end)
return 0; // might be ID3 tag, i.e. EOF
stream_skip(s,-3);
} else {
dp = new_demux_packet(l);
memcpy(dp->buffer,hdr,4);
if (stream_read(s,dp->buffer + 4,l-4) != l-4)
{
free_demux_packet(dp);
return 0;
}
priv->next_pts += sh_audio->audio.dwScale/(double)sh_audio->samplerate;
break;
}
} break;
case WAV : {
unsigned align = sh_audio->wf->nBlockAlign;
l = sh_audio->wf->nAvgBytesPerSec;
if (l <= 0) l = 65536;
if (demux->movi_end && l > demux->movi_end - stream_tell(s)) {
// do not read beyond end, there might be junk after data chunk
l = demux->movi_end - stream_tell(s);
if (l <= 0) return 0;
}
if (align)
l = (l + align - 1) / align * align;
dp = new_demux_packet(l);
l = stream_read(s,dp->buffer,l);
priv->next_pts += l/(double)sh_audio->i_bps;
break;
}
case fLaC: {
l = 65535;
dp = new_demux_packet(l);
l = stream_read(s,dp->buffer,l);
/* FLAC is not a constant-bitrate codec. These values will be wrong. */
priv->next_pts += l/(double)sh_audio->i_bps;
break;
}
default:
mp_msg(MSGT_DEMUXER,MSGL_WARN,MSGTR_MPDEMUX_AUDIO_UnknownFormat,priv->frmt);
return 0;
}
resize_demux_packet(dp, l);
dp->pts = this_pts;
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/sh->audio.dwScale;
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->next_pts += sh->audio.dwScale/(double)sh->samplerate;
nf--;
}
}
static void demux_audio_seek(demuxer_t *demuxer,float rel_seek_secs,float audio_delay,int flags){
sh_audio_t* sh_audio;
stream_t* s;
int64_t 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 & SEEK_FACTOR)) {
len = (flags & SEEK_ABSOLUTE) ? rel_seek_secs - priv->next_pts : rel_seek_secs;
if(len < 0) {
stream_seek(s,demuxer->movi_start);
len = priv->next_pts + len;
priv->next_pts = 0;
}
if(len > 0)
high_res_mp3_seek(demuxer,len);
return;
}
base = flags&SEEK_ABSOLUTE ? demuxer->movi_start : stream_tell(s);
if(flags&SEEK_FACTOR)
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;
} else if(pos < demuxer->movi_start)
pos = demuxer->movi_start;
priv->next_pts = (pos-demuxer->movi_start)/(double)sh_audio->i_bps;
switch(priv->frmt) {
case WAV:
pos -= (pos - demuxer->movi_start) %
(sh_audio->wf->nBlockAlign ? sh_audio->wf->nBlockAlign :
(sh_audio->channels * sh_audio->samplesize));
break;
}
stream_seek(s,pos);
}
static void demux_close_audio(demuxer_t* demuxer) {
da_priv_t* priv = demuxer->priv;
if(!priv)
return;
free(priv);
}
static int demux_audio_control(demuxer_t *demuxer,int cmd, void *arg){
sh_audio_t *sh_audio=demuxer->audio->sh;
int audio_length = sh_audio->i_bps ? demuxer->movi_end / sh_audio->i_bps : 0;
da_priv_t* priv = demuxer->priv;
switch(cmd) {
case DEMUXER_CTRL_GET_TIME_LENGTH:
if (audio_length<=0) return DEMUXER_CTRL_DONTKNOW;
*((double *)arg)=(double)audio_length;
return DEMUXER_CTRL_GUESS;
case DEMUXER_CTRL_GET_PERCENT_POS:
if (audio_length<=0)
return DEMUXER_CTRL_DONTKNOW;
*((int *)arg)=(int)( (priv->next_pts*100) / audio_length);
return DEMUXER_CTRL_OK;
default:
return DEMUXER_CTRL_NOTIMPL;
}
}
const demuxer_desc_t demuxer_desc_audio = {
"Audio demuxer",
"audio",
"Audio only",
"?",
"Audio only files",
DEMUXER_TYPE_AUDIO,
0, // unsafe autodetect
demux_audio_open,
demux_audio_fill_buffer,
NULL,
demux_close_audio,
demux_audio_seek,
demux_audio_control
};
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