mirror of https://github.com/mpv-player/mpv
549 lines
14 KiB
C
549 lines
14 KiB
C
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// Reference: DOCS/tech/hwac3.txt !!!!!
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/* DTS code based on "ac3/decode_dts.c" and "ac3/conversion.c" from "ogle 0.9"
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(see http://www.dtek.chalmers.se/~dvd/)
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*/
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#define _XOPEN_SOURCE 600
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <unistd.h>
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#include "config.h"
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#include "mp_msg.h"
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#include "help_mp.h"
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#include "mpbswap.h"
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#include "ad_internal.h"
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#ifdef CONFIG_LIBA52_INTERNAL
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#include "liba52/a52.h"
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#else
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#include <a52dec/a52.h>
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#endif
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static int isdts = -1;
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static ad_info_t info =
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{
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"AC3/DTS pass-through S/PDIF",
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"hwac3",
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"Nick Kurshev/Peter Schüller",
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"???",
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""
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};
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LIBAD_EXTERN(hwac3)
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static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate);
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static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf);
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static int ac3dts_fillbuff(sh_audio_t *sh_audio)
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{
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int length = 0;
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int flags = 0;
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int sample_rate = 0;
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int bit_rate = 0;
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sh_audio->a_in_buffer_len = 0;
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/* sync frame:*/
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while(1)
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{
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// Original code DTS has a 10 bytes header.
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// Now max 12 bytes for 14 bits DTS header.
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while(sh_audio->a_in_buffer_len < 12)
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{
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int c = demux_getc(sh_audio->ds);
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if(c<0)
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return -1; /* EOF*/
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sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++] = c;
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}
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if (sh_audio->format == 0x2001)
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{
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length = dts_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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if(length >= 12)
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{
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if(isdts != 1)
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{
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mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to DTS, %d bps, %d Hz\n", bit_rate, sample_rate);
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isdts = 1;
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}
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break;
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}
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}
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else
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{
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length = a52_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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if(length >= 7 && length <= 3840)
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{
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if(isdts != 0)
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{
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mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to AC3, %d bps, %d Hz\n", bit_rate, sample_rate);
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isdts = 0;
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}
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break; /* we're done.*/
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}
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}
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/* bad file => resync*/
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memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer + 1, 11);
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--sh_audio->a_in_buffer_len;
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}
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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);
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sh_audio->samplerate = sample_rate;
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sh_audio->i_bps = bit_rate / 8;
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demux_read_data(sh_audio->ds, sh_audio->a_in_buffer + 12, length - 12);
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sh_audio->a_in_buffer_len = length;
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// TODO: is DTS also checksummed?
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#ifdef CONFIG_LIBA52_INTERNAL
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if(isdts == 0 && crc16_block(sh_audio->a_in_buffer + 2, length - 2) != 0)
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mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "a52: CRC check failed! \n");
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#endif
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return length;
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}
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static int preinit(sh_audio_t *sh)
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{
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/* Dolby AC3 audio: */
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sh->audio_out_minsize = 128 * 32 * 2 * 2; // DTS seems to need more than AC3
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sh->audio_in_minsize = 8192;
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sh->channels = 2;
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sh->samplesize = 2;
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sh->sample_format = AF_FORMAT_AC3;
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return 1;
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}
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static int init(sh_audio_t *sh_audio)
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{
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/* Dolby AC3 passthrough:*/
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a52_state_t *a52_state = a52_init(0);
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if(a52_state == NULL)
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{
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "A52 init failed\n");
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return 0;
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}
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if(ac3dts_fillbuff(sh_audio) < 0)
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{
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a52_free(a52_state);
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "AC3/DTS sync failed\n");
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return 0;
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}
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sh_audio->context = a52_state;
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return 1;
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}
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static void uninit(sh_audio_t *sh)
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{
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a52_free(sh->context);
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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{
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switch(cmd)
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{
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case ADCTRL_RESYNC_STREAM:
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case ADCTRL_SKIP_FRAME:
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ac3dts_fillbuff(sh);
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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{
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int len = sh_audio->a_in_buffer_len;
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if(len <= 0)
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if((len = ac3dts_fillbuff(sh_audio)) <= 0)
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return len; /*EOF*/
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sh_audio->a_in_buffer_len = 0;
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if(isdts == 1)
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{
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return decode_audio_dts(sh_audio->a_in_buffer, len, buf);
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}
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else if(isdts == 0)
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{
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uint16_t *buf16 = (uint16_t *)buf;
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buf16[0] = 0xF872; // iec 61937 syncword 1
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buf16[1] = 0x4E1F; // iec 61937 syncword 2
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buf16[2] = 0x0001; // data-type ac3
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buf16[2] |= (sh_audio->a_in_buffer[5] & 0x7) << 8; // bsmod
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buf16[3] = len << 3; // number of bits in payload
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#if HAVE_BIGENDIAN
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memcpy(buf + 8, sh_audio->a_in_buffer, len);
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#else
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swab(sh_audio->a_in_buffer, buf + 8, len);
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if (len & 1) {
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buf[8+len-1] = 0;
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buf[8+len] = sh_audio->a_in_buffer[len-1];
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len++;
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}
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#endif
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memset(buf + 8 + len, 0, 6144 - 8 - len);
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return 6144;
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}
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else
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return -1;
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}
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static const int DTS_SAMPLEFREQS[16] =
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{
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0,
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8000,
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16000,
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32000,
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64000,
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128000,
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11025,
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22050,
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44100,
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88200,
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176400,
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12000,
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24000,
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48000,
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96000,
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192000
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};
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static const int DTS_BITRATES[30] =
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{
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32000,
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56000,
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64000,
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96000,
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112000,
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128000,
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192000,
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224000,
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256000,
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320000,
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384000,
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448000,
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512000,
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576000,
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640000,
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768000,
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896000,
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1024000,
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1152000,
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1280000,
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1344000,
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1408000,
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1411200,
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1472000,
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1536000,
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1920000,
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2048000,
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3072000,
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3840000,
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4096000
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};
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static int dts_decode_header(uint8_t *indata_ptr, int *rate, int *nblks, int *sfreq)
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{
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int ftype;
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int surp;
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int unknown_bit;
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int fsize;
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int amode;
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int word_mode;
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int le_mode;
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unsigned int first4bytes = indata_ptr[0] << 24 | indata_ptr[1] << 16
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| indata_ptr[2] << 8 | indata_ptr[3];
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switch(first4bytes)
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{
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/* 14 bits LE */
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case 0xff1f00e8:
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/* Also make sure frame type is 1. */
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if ((indata_ptr[4]&0xf0) != 0xf0 || indata_ptr[5] != 0x07)
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return -1;
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word_mode = 0;
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le_mode = 1;
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break;
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/* 14 bits BE */
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case 0x1fffe800:
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/* Also make sure frame type is 1. */
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if (indata_ptr[4] != 0x07 || (indata_ptr[5]&0xf0) != 0xf0)
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return -1;
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word_mode = 0;
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le_mode = 0;
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break;
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/* 16 bits LE */
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case 0xfe7f0180:
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word_mode = 1;
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le_mode = 1;
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break;
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/* 16 bits BE */
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case 0x7ffe8001:
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word_mode = 1;
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le_mode = 0;
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break;
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default:
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return -1;
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}
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if(word_mode)
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{
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/* First bit after first 32 bits:
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Frame type ( 1: Normal frame; 0: Termination frame ) */
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ftype = indata_ptr[4+le_mode] >> 7;
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if(ftype != 1)
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{
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Termination frames not handled, REPORT BUG\n");
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return -1;
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}
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/* Next 5 bits: Surplus Sample Count V SURP 5 bits */
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surp = indata_ptr[4+le_mode] >> 2 & 0x1f;
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/* Number of surplus samples */
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surp = (surp + 1) % 32;
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/* One unknown bit, crc? */
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unknown_bit = indata_ptr[4+le_mode] >> 1 & 0x01;
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/* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
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*nblks = (indata_ptr[4+le_mode] & 0x01) << 6 | indata_ptr[5-le_mode] >> 2;
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/* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
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encoded in the current frame per channel. */
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++(*nblks);
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/* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
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(ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
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FSIZE defines the byte size of the current audio frame. */
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fsize = (indata_ptr[5-le_mode] & 0x03) << 12 | indata_ptr[6+le_mode] << 4
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| indata_ptr[7-le_mode] >> 4;
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++fsize;
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/* Audio Channel Arrangement ACC AMODE 6 bits */
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amode = (indata_ptr[7-le_mode] & 0x0f) << 2 | indata_ptr[8+le_mode] >> 6;
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/* Source Sampling rate ACC SFREQ 4 bits */
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*sfreq = indata_ptr[8+le_mode] >> 2 & 0x0f;
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/* Transmission Bit Rate ACC RATE 5 bits */
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*rate = (indata_ptr[8+le_mode] & 0x03) << 3
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| (indata_ptr[9-le_mode] >> 5 & 0x07);
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}
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else
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{
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/* in the case judgement, we assure this */
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ftype = 1;
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surp = 0;
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/* 14 bits support, every 2 bytes, & 0x3fff, got used 14 bits */
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/* Bits usage:
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32 bits: Sync code (28 + 4) 1th and 2th word, 4 bits in 3th word
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1 bits: Frame type 1 bits in 3th word
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5 bits: SURP 5 bits in 3th word
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1 bits: crc? 1 bits in 3th word
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7 bits: NBLKS 3 bits in 3th word, 4 bits in 4th word
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14 bits: FSIZE 10 bits in 4th word, 4 bits in 5th word
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in 14 bits mode, FSIZE = FSIZE*8/14*2
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6 bits: AMODE 6 bits in 5th word
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4 bits: SFREQ 4 bits in 5th word
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5 bits: RATE 5 bits in 6th word
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total bits: 75 bits */
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/* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
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*nblks = (indata_ptr[5-le_mode] & 0x07) << 4
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| (indata_ptr[6+le_mode] & 0x3f) >> 2;
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/* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
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encoded in the current frame per channel. */
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++(*nblks);
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/* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
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(ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
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FSIZE defines the byte size of the current audio frame. */
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fsize = (indata_ptr[6+le_mode] & 0x03) << 12 | indata_ptr[7-le_mode] << 4
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| (indata_ptr[8+le_mode] & 0x3f) >> 2;
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++fsize;
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fsize = fsize * 8 / 14 * 2;
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/* Audio Channel Arrangement ACC AMODE 6 bits */
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amode = (indata_ptr[8+le_mode] & 0x03) << 4
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| (indata_ptr[9-le_mode] & 0xf0) >> 4;
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/* Source Sampling rate ACC SFREQ 4 bits */
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*sfreq = indata_ptr[9-le_mode] & 0x0f;
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/* Transmission Bit Rate ACC RATE 5 bits */
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*rate = (indata_ptr[10+le_mode] & 0x3f) >> 1;
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}
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#if 0
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if(*sfreq != 13)
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{
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Only 48kHz supported, REPORT BUG\n");
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return -1;
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}
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#endif
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if((fsize > 8192) || (fsize < 96))
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{
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: fsize: %d invalid, REPORT BUG\n", fsize);
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return -1;
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}
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if(*nblks != 8 &&
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*nblks != 16 &&
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*nblks != 32 &&
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*nblks != 64 &&
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*nblks != 128 &&
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ftype == 1)
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{
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: nblks %d not valid for normal frame, REPORT BUG\n", *nblks);
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return -1;
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}
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return fsize;
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}
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static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate)
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{
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int nblks;
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int fsize;
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int rate;
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int sfreq;
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fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
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if(fsize >= 0)
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{
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if(rate >= 0 && rate <= 29)
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*bit_rate = DTS_BITRATES[rate];
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else
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*bit_rate = 0;
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if(sfreq >= 1 && sfreq <= 15)
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*sample_rate = DTS_SAMPLEFREQS[sfreq];
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else
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*sample_rate = 0;
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}
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return fsize;
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}
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static int convert_14bits_to_16bits(const unsigned char *src,
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unsigned char *dest,
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int len,
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int is_le)
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{
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uint16_t *p = (uint16_t *)dest;
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uint16_t buf = 0;
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int spacebits = 16;
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if (len <= 0) return 0;
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while (len > 0) {
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uint16_t v;
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if (len == 1)
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v = is_le ? src[0] : src[0] << 8;
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else
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v = is_le ? src[1] << 8 | src[0] : src[0] << 8 | src[1];
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v <<= 2;
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src += 2;
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len -= 2;
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buf |= v >> (16 - spacebits);
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spacebits -= 14;
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if (spacebits < 0) {
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*p++ = buf;
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spacebits += 16;
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buf = v << (spacebits - 2);
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}
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}
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*p++ = buf;
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return (unsigned char *)p - dest;
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}
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static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf)
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{
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int nblks;
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int fsize;
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int rate;
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int sfreq;
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int nr_samples;
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int convert_16bits = 0;
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uint16_t *buf16 = (uint16_t *)buf;
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fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
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if(fsize < 0)
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return -1;
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nr_samples = nblks * 32;
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buf16[0] = 0xf872; /* iec 61937 */
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buf16[1] = 0x4e1f; /* syncword */
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switch(nr_samples)
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{
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case 512:
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buf16[2] = 0x000b; /* DTS-1 (512-sample bursts) */
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break;
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case 1024:
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buf16[2] = 0x000c; /* DTS-2 (1024-sample bursts) */
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break;
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case 2048:
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buf16[2] = 0x000d; /* DTS-3 (2048-sample bursts) */
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break;
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default:
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mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: %d-sample bursts not supported\n", nr_samples);
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buf16[2] = 0x0000;
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break;
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}
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if(fsize + 8 > nr_samples * 2 * 2)
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{
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// dts wav (14bits LE) match this condition, one way to passthrough
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// is not add iec 61937 header, decoders will notice the dts header
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// and identify the dts stream. Another way here is convert
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// the stream from 14 bits to 16 bits.
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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;
|
|
}
|