Mercurial > mplayer.hg
view libmpcodecs/ad_hwac3.c @ 23510:a6c619ee9d30
Teletext support for tv:// (v4l and v4l2 only)
modified patch from Otvos Attila oattila at chello dot hu
Module uses zvbi library for all low-level VBI operations (like I/O with vbi
device, converting vbi pages into usefull vbi_page stuctures, rendering them
into RGB32 images).
All teletext related stuff (except properties, slave commands and rendering
osd in text mode or RGB32 rendered teletext pages in spu mode) is implemented
in tvi_vbi.c
New properties:
teletext_page - switching between pages
teletext_mode - switch between on/off/opaque/transparent modes
teletext_format - (currently read-only) allows to get format info
(black/white,gray,text)
teletext_half_page - trivial zooming (displaying top/bottom half of teletext
page)
New slave commands:
teletext_add_dec - user interface for jumping to any page by editing page number
interactively
teletext_go_link - goes though links, specified on current page
author | voroshil |
---|---|
date | Sun, 10 Jun 2007 00:06:12 +0000 |
parents | 7c0af53fcd4a |
children | a81e246e3b38 |
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// Reference: DOCS/tech/hwac3.txt !!!!! /* DTS code based on "ac3/decode_dts.c" and "ac3/conversion.c" from "ogle 0.9" (see http://www.dtek.chalmers.se/~dvd/) */ #include <stdio.h> #include <stdlib.h> #include <string.h> #include <unistd.h> #include "config.h" #ifdef USE_LIBA52 #include "mp_msg.h" #include "help_mp.h" #include "ad_internal.h" #include "liba52/a52.h" static int isdts = -1; static 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 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) { // DTS has a 10 byte header while(sh_audio->a_in_buffer_len < 10) { 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 = dts_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); if(length >= 10) { 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; } length = a52_syncinfo(sh_audio->a_in_buffer, &flags, &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, 9); --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 + 10, length - 10); sh_audio->a_in_buffer_len = length; // TODO: is DTS also checksummed? if(isdts == 0 && crc16_block(sh_audio->a_in_buffer + 2, length - 2) != 0) mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "a52: CRC check failed! \n"); 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; return 1; } static int init(sh_audio_t *sh_audio) { /* Dolby AC3 passthrough:*/ a52_state_t *a52_state = a52_init(0); if(a52_state == NULL) { mp_msg(MSGT_DECAUDIO, MSGL_ERR, "A52 init failed\n"); return 0; } 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) { buf[0] = 0x72; buf[1] = 0xF8; buf[2] = 0x1F; buf[3] = 0x4E; buf[4] = 0x01; //(length) ? data_type : 0; /* & 0x1F; */ buf[5] = 0x00; buf[6] = (len << 3) & 0xFF; buf[7] = (len >> 5) & 0xFF; #ifdef WORDS_BIGENDIAN memcpy(buf + 8, sh_audio->a_in_buffer, len); // untested #else swab(sh_audio->a_in_buffer, buf + 8, len); #endif memset(buf + 8 + len, 0, 6144 - 8 - len); return 6144; } else return -1; } static int DTS_SAMPLEFREQS[16] = { 0, 8000, 16000, 32000, 64000, 128000, 11025, 22050, 44100, 88200, 176400, 12000, 24000, 48000, 96000, 192000 }; static 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; int fsize; int amode; if(((indata_ptr[0] << 24) | (indata_ptr[1] << 16) | (indata_ptr[2] << 8) | (indata_ptr[3])) != 0x7ffe8001) return -1; ftype = indata_ptr[4] >> 7; surp = (indata_ptr[4] >> 2) & 0x1f; surp = (surp + 1) % 32; unknown_bit = (indata_ptr[4] >> 1) & 0x01; *nblks = (indata_ptr[4] & 0x01) << 6 | (indata_ptr[5] >> 2); *nblks = *nblks + 1; fsize = (indata_ptr[5] & 0x03) << 12 | (indata_ptr[6] << 4) | (indata_ptr[7] >> 4); fsize = fsize + 1; amode = (indata_ptr[7] & 0x0f) << 2 | (indata_ptr[8] >> 6); *sfreq = (indata_ptr[8] >> 2) & 0x0f; *rate = (indata_ptr[8] & 0x03) << 3 | ((indata_ptr[9] >> 5) & 0x07); if(ftype != 1) { mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Termination frames not handled, REPORT BUG\n"); return -1; } if(*sfreq != 13) { mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Only 48kHz supported, REPORT BUG\n"); return -1; } 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 decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf) { int nblks; int fsize; int rate; int sfreq; int burst_len; int nr_samples; fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq); if(fsize < 0) return -1; burst_len = fsize * 8; nr_samples = nblks * 32; buf[0] = 0x72; buf[1] = 0xf8; /* iec 61937 */ buf[2] = 0x1f; buf[3] = 0x4e; /* syncword */ switch(nr_samples) { case 512: buf[4] = 0x0b; /* DTS-1 (512-sample bursts) */ break; case 1024: buf[4] = 0x0c; /* DTS-2 (1024-sample bursts) */ break; case 2048: buf[4] = 0x0d; /* DTS-3 (2048-sample bursts) */ break; default: mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: %d-sample bursts not supported\n", nr_samples); buf[4] = 0x00; break; } buf[5] = 0; /* ?? */ buf[6] = (burst_len) & 0xff; buf[7] = (burst_len >> 8) & 0xff; if(fsize + 8 > nr_samples * 2 * 2) { mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: more data than fits\n"); } #ifdef WORDS_BIGENDIAN memcpy(&buf[8], indata_ptr, fsize); // untested #else //TODO if fzise is odd, swab doesn't copy the last byte swab(indata_ptr, &buf[8], fsize); if (fsize & 1) buf[8+fsize] = indata_ptr[fsize]; #endif memset(&buf[fsize + 8], 0, nr_samples * 2 * 2 - (fsize + 8)); return nr_samples * 2 * 2; } #endif