RepoMirrors
/
mpv
mirror of https://github.com/mpv-player/mpv
1
0
Fork 0
Code Issues Releases Wiki Activity
mpv/libmpcodecs/ad_hwac3.c

575 lines
15 KiB
C
Raw Blame History

/*
* DTS code based on "ac3/decode_dts.c" and "ac3/conversion.c" from "ogle 0.9"
* (see http://www.dtek.chalmers.se/~dvd/)
* Reference: DOCS/tech/hwac3.txt !!!!!
*
* 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.
*/
#define _XOPEN_SOURCE 600
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include "config.h"
#include "mp_msg.h"
#include "mpbswap.h"
#include "libavutil/common.h"
#include "ffmpeg_files/intreadwrite.h"
#include "ad_internal.h"
static int isdts = -1;
static const ad_info_t info =
{
"AC3/DTS pass-through S/PDIF",
"hwac3",
"Nick Kurshev/Peter Schüller",
"???",
""
};
LIBAD_EXTERN(hwac3)
static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate);
static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf);
static int a52_syncinfo (uint8_t *buf, int *sample_rate, int *bit_rate)
{
static const uint16_t rate[] = { 32, 40, 48, 56, 64, 80, 96, 112,
128, 160, 192, 224, 256, 320, 384, 448,
512, 576, 640};
int frmsizecod;
int bitrate;
int half;
if (buf[0] != 0x0b || buf[1] != 0x77) /* syncword */
return 0;
if (buf[5] >= 0x60) /* bsid >= 12 */
return 0;
half = buf[5] >> 3;
half = FFMAX(half - 8, 0);
frmsizecod = buf[4] & 63;
if (frmsizecod >= 38)
return 0;
bitrate = rate[frmsizecod >> 1];
*bit_rate = (bitrate * 1000) >> half;
switch (buf[4] & 0xc0) {
case 0:
*sample_rate = 48000 >> half;
return 4 * bitrate;
case 0x40:
*sample_rate = 44100 >> half;
return 2 * (320 * bitrate / 147 + (frmsizecod & 1));
case 0x80:
*sample_rate = 32000 >> half;
return 6 * bitrate;
default:
return 0;
}
}
static int ac3dts_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)
{
// Original code DTS has a 10 bytes header.
// Now max 12 bytes for 14 bits DTS header.
while(sh_audio->a_in_buffer_len < 12)
{
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;
}
if (sh_audio->format == 0x2001)
{
length = dts_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
if(length >= 12)
{
if(isdts != 1)
{
mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to DTS, %d bps, %d Hz\n", bit_rate, sample_rate);
isdts = 1;
}
break;
}
}
else
{
length = a52_syncinfo(sh_audio->a_in_buffer, &sample_rate, &bit_rate);
if(length >= 7 && length <= 3840)
{
if(isdts != 0)
{
mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to AC3, %d bps, %d Hz\n", bit_rate, sample_rate);
isdts = 0;
}
break; /* we're done.*/
}
}
/* bad file => resync*/
memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer + 1, 11);
--sh_audio->a_in_buffer_len;
}
mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "ac3dts: %s len=%d flags=0x%X %d Hz %d bit/s\n", isdts == 1 ? "DTS" : isdts == 0 ? "AC3" : "unknown", 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 + 12, length - 12);
sh_audio->a_in_buffer_len = length;
return length;
}
static int preinit(sh_audio_t *sh)
{
/* Dolby AC3 audio: */
sh->audio_out_minsize = 128 * 32 * 2 * 2; // DTS seems to need more than AC3
sh->audio_in_minsize = 8192;
sh->channels = 2;
sh->samplesize = 2;
sh->sample_format = AF_FORMAT_AC3_BE;
// HACK for DTS where useless swapping can't easily be removed
if (sh->format == 0x2001)
sh->sample_format = AF_FORMAT_AC3_NE;
return 1;
}
static int init(sh_audio_t *sh_audio)
{
/* Dolby AC3 passthrough:*/
if(ac3dts_fillbuff(sh_audio) < 0)
{
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "AC3/DTS sync failed\n");
return 0;
}
return 1;
}
static void uninit(sh_audio_t *sh)
{
}
static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
switch(cmd)
{
case ADCTRL_RESYNC_STREAM:
case ADCTRL_SKIP_FRAME:
ac3dts_fillbuff(sh);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
int len = sh_audio->a_in_buffer_len;
if(len <= 0)
if((len = ac3dts_fillbuff(sh_audio)) <= 0)
return len; /*EOF*/
sh_audio->a_in_buffer_len = 0;
if(isdts == 1)
{
return decode_audio_dts(sh_audio->a_in_buffer, len, buf);
}
else if(isdts == 0)
{
AV_WB16(buf, 0xF872); // iec 61937 syncword 1
AV_WB16(buf + 2, 0x4E1F); // iec 61937 syncword 2
buf[4] = sh_audio->a_in_buffer[5] & 0x7; // bsmod
buf[5] = 0x01; // data-type ac3
AV_WB16(buf + 6, len << 3); // number of bits in payload
memcpy(buf + 8, sh_audio->a_in_buffer, len);
memset(buf + 8 + len, 0, 6144 - 8 - len);
return 6144;
}
else
return -1;
}
static const int DTS_SAMPLEFREQS[16] =
{
0,
8000,
16000,
32000,
64000,
128000,
11025,
22050,
44100,
88200,
176400,
12000,
24000,
48000,
96000,
192000
};
static const int DTS_BITRATES[30] =
{
32000,
56000,
64000,
96000,
112000,
128000,
192000,
224000,
256000,
320000,
384000,
448000,
512000,
576000,
640000,
768000,
896000,
1024000,
1152000,
1280000,
1344000,
1408000,
1411200,
1472000,
1536000,
1920000,
2048000,
3072000,
3840000,
4096000
};
static int dts_decode_header(uint8_t *indata_ptr, int *rate, int *nblks, int *sfreq)
{
int ftype;
int surp;
int unknown_bit av_unused;
int fsize;
int amode av_unused;
int word_mode;
int le_mode;
unsigned int first4bytes = indata_ptr[0] << 24 | indata_ptr[1] << 16
| indata_ptr[2] << 8 | indata_ptr[3];
switch(first4bytes)
{
/* 14 bits LE */
case 0xff1f00e8:
/* Also make sure frame type is 1. */
if ((indata_ptr[4]&0xf0) != 0xf0 || indata_ptr[5] != 0x07)
return -1;
word_mode = 0;
le_mode = 1;
break;
/* 14 bits BE */
case 0x1fffe800:
/* Also make sure frame type is 1. */
if (indata_ptr[4] != 0x07 || (indata_ptr[5]&0xf0) != 0xf0)
return -1;
word_mode = 0;
le_mode = 0;
break;
/* 16 bits LE */
case 0xfe7f0180:
word_mode = 1;
le_mode = 1;
break;
/* 16 bits BE */
case 0x7ffe8001:
word_mode = 1;
le_mode = 0;
break;
default:
return -1;
}
if(word_mode)
{
/* First bit after first 32 bits:
Frame type ( 1: Normal frame; 0: Termination frame ) */
ftype = indata_ptr[4+le_mode] >> 7;
if(ftype != 1)
{
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Termination frames not handled, REPORT BUG\n");
return -1;
}
/* Next 5 bits: Surplus Sample Count V SURP 5 bits */
surp = indata_ptr[4+le_mode] >> 2 & 0x1f;
/* Number of surplus samples */
surp = (surp + 1) % 32;
/* One unknown bit, crc? */
unknown_bit = indata_ptr[4+le_mode] >> 1 & 0x01;
/* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
*nblks = (indata_ptr[4+le_mode] & 0x01) << 6 | indata_ptr[5-le_mode] >> 2;
/* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
encoded in the current frame per channel. */
++(*nblks);
/* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
(ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
FSIZE defines the byte size of the current audio frame. */
fsize = (indata_ptr[5-le_mode] & 0x03) << 12 | indata_ptr[6+le_mode] << 4
| indata_ptr[7-le_mode] >> 4;
++fsize;
/* Audio Channel Arrangement ACC AMODE 6 bits */
amode = (indata_ptr[7-le_mode] & 0x0f) << 2 | indata_ptr[8+le_mode] >> 6;
/* Source Sampling rate ACC SFREQ 4 bits */
*sfreq = indata_ptr[8+le_mode] >> 2 & 0x0f;
/* Transmission Bit Rate ACC RATE 5 bits */
*rate = (indata_ptr[8+le_mode] & 0x03) << 3
| (indata_ptr[9-le_mode] >> 5 & 0x07);
}
else
{
/* in the case judgement, we assure this */
ftype = 1;
surp = 0;
/* 14 bits support, every 2 bytes, & 0x3fff, got used 14 bits */
/* Bits usage:
32 bits: Sync code (28 + 4) 1th and 2th word, 4 bits in 3th word
1 bits: Frame type 1 bits in 3th word
5 bits: SURP 5 bits in 3th word
1 bits: crc? 1 bits in 3th word
7 bits: NBLKS 3 bits in 3th word, 4 bits in 4th word
14 bits: FSIZE 10 bits in 4th word, 4 bits in 5th word
in 14 bits mode, FSIZE = FSIZE*8/14*2
6 bits: AMODE 6 bits in 5th word
4 bits: SFREQ 4 bits in 5th word
5 bits: RATE 5 bits in 6th word
total bits: 75 bits */
/* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
*nblks = (indata_ptr[5-le_mode] & 0x07) << 4
| (indata_ptr[6+le_mode] & 0x3f) >> 2;
/* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
encoded in the current frame per channel. */
++(*nblks);
/* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
(ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
FSIZE defines the byte size of the current audio frame. */
fsize = (indata_ptr[6+le_mode] & 0x03) << 12 | indata_ptr[7-le_mode] << 4
| (indata_ptr[8+le_mode] & 0x3f) >> 2;
++fsize;
fsize = fsize * 8 / 14 * 2;
/* Audio Channel Arrangement ACC AMODE 6 bits */
amode = (indata_ptr[8+le_mode] & 0x03) << 4
| (indata_ptr[9-le_mode] & 0xf0) >> 4;
/* Source Sampling rate ACC SFREQ 4 bits */
*sfreq = indata_ptr[9-le_mode] & 0x0f;
/* Transmission Bit Rate ACC RATE 5 bits */
*rate = (indata_ptr[10+le_mode] & 0x3f) >> 1;
}
#if 0
if(*sfreq != 13)
{
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Only 48kHz supported, REPORT BUG\n");
return -1;
}
#endif
if((fsize > 8192) || (fsize < 96))
{
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: fsize: %d invalid, REPORT BUG\n", fsize);
return -1;
}
if(*nblks != 8 &&
*nblks != 16 &&
*nblks != 32 &&
*nblks != 64 &&
*nblks != 128 &&
ftype == 1)
{
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: nblks %d not valid for normal frame, REPORT BUG\n", *nblks);
return -1;
}
return fsize;
}
static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate)
{
int nblks;
int fsize;
int rate;
int sfreq;
fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
if(fsize >= 0)
{
if(rate >= 0 && rate <= 29)
*bit_rate = DTS_BITRATES[rate];
else
*bit_rate = 0;
if(sfreq >= 1 && sfreq <= 15)
*sample_rate = DTS_SAMPLEFREQS[sfreq];
else
*sample_rate = 0;
}
return fsize;
}
static int convert_14bits_to_16bits(const unsigned char *src,
unsigned char *dest,
int len,
int is_le)
{
uint16_t *p = (uint16_t *)dest;
uint16_t buf = 0;
int spacebits = 16;
if (len <= 0) return 0;
while (len > 0) {
uint16_t v;
if (len == 1)
v = is_le ? src[0] : src[0] << 8;
else
v = is_le ? src[1] << 8 | src[0] : src[0] << 8 | src[1];
v <<= 2;
src += 2;
len -= 2;
buf |= v >> (16 - spacebits);
spacebits -= 14;
if (spacebits < 0) {
*p++ = buf;
spacebits += 16;
buf = v << (spacebits - 2);
}
}
*p++ = buf;
return (unsigned char *)p - dest;
}
static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf)
{
int nblks;
int fsize;
int rate;
int sfreq;
int nr_samples;
int convert_16bits = 0;
uint16_t *buf16 = (uint16_t *)buf;
fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
if(fsize < 0)
return -1;
nr_samples = nblks * 32;
buf16[0] = 0xf872; /* iec 61937 */
buf16[1] = 0x4e1f; /* syncword */
switch(nr_samples)
{
case 512:
buf16[2] = 0x000b; /* DTS-1 (512-sample bursts) */
break;
case 1024:
buf16[2] = 0x000c; /* DTS-2 (1024-sample bursts) */
break;
case 2048:
buf16[2] = 0x000d; /* DTS-3 (2048-sample bursts) */
break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: %d-sample bursts not supported\n", nr_samples);
buf16[2] = 0x0000;
break;
}
if(fsize + 8 > nr_samples * 2 * 2)
{
// dts wav (14bits LE) match this condition, one way to passthrough
// is not add iec 61937 header, decoders will notice the dts header
// and identify the dts stream. Another way here is convert
// the stream from 14 bits to 16 bits.
if ((indata_ptr[0] == 0xff || indata_ptr[0] == 0x1f)
&& fsize * 14 / 16 + 8 <= nr_samples * 2 * 2) {
// The input stream is 14 bits, we can shrink it to 16 bits
// to save space for add the 61937 header
fsize = convert_14bits_to_16bits(indata_ptr,
&buf[8],
fsize,
indata_ptr[0] == 0xff /* is LE */
);
mp_msg(MSGT_DECAUDIO, MSGL_DBG3, "DTS: shrink 14 bits stream to "
"16 bits %02x%02x%02x%02x => %02x%02x%02x%02x, new size %d.\n",
indata_ptr[0], indata_ptr[1], indata_ptr[2], indata_ptr[3],
buf[8], buf[9], buf[10], buf[11], fsize);
convert_16bits = 1;
}
else
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: more data than fits\n");
}
buf16[3] = fsize << 3;
if (!convert_16bits) {
#if HAVE_BIGENDIAN
/* BE stream */
if (indata_ptr[0] == 0x1f || indata_ptr[0] == 0x7f)
#else
/* LE stream */
if (indata_ptr[0] == 0xff || indata_ptr[0] == 0xfe)
#endif
memcpy(&buf[8], indata_ptr, fsize);
else
{
swab(indata_ptr, &buf[8], fsize);
if (fsize & 1) {
buf[8+fsize-1] = 0;
buf[8+fsize] = indata_ptr[fsize-1];
fsize++;
}
}
}
memset(&buf[fsize + 8], 0, nr_samples * 2 * 2 - (fsize + 8));
return nr_samples * 2 * 2;
}
Reference in New Issue View Git Blame Copy Permalink