Mercurial > mplayer.hg
view libmpcodecs/ad_hwac3.c @ 30874:78c91b9f7d3d
Fix compilation after FFmpeg r22569 at least for some architectures.
author | cehoyos |
---|---|
date | Wed, 17 Mar 2010 10:46:18 +0000 |
parents | cc27da5d7286 |
children | 025d6c8eebb6 |
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/* * 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 "help_mp.h" #include "mpbswap.h" #include "libavutil/common.h" #include "libavutil/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; int fsize; int amode; 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; }