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/*
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* General DV muxer/demuxer
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* Copyright (c) 2003 Roman Shaposhnik
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*
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* Many thanks to Dan Dennedy <dan@dennedy.org> for providing wealth
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* of DV technical info.
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*
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* Raw DV format
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* Copyright (c) 2002 Fabrice Bellard.
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*
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* 50 Mbps (DVCPRO50) support
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* Copyright (c) 2006 Daniel Maas <dmaas@maasdigital.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <time.h>
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#include "avformat.h"
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#include "dvdata.h"
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#include "dv.h"
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struct DVDemuxContext {
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const DVprofile* sys; /* Current DV profile. E.g.: 525/60, 625/50 */
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AVFormatContext* fctx;
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AVStream* vst;
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AVStream* ast[2];
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AVPacket audio_pkt[2];
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uint8_t audio_buf[2][8192];
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int ach;
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int frames;
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uint64_t abytes;
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};
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struct DVMuxContext {
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const DVprofile* sys; /* Current DV profile. E.g.: 525/60, 625/50 */
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int n_ast; /* Number of stereo audio streams (up to 2) */
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AVStream *ast[2]; /* Stereo audio streams */
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FifoBuffer audio_data[2]; /* Fifo for storing excessive amounts of PCM */
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int frames; /* Number of a current frame */
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time_t start_time; /* Start time of recording */
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int has_audio; /* frame under contruction has audio */
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int has_video; /* frame under contruction has video */
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uint8_t frame_buf[DV_MAX_FRAME_SIZE]; /* frame under contruction */
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};
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static const int dv_aaux_packs_dist[12][9] = {
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{ 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
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{ 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
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{ 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
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{ 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
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{ 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
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{ 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
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{ 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
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{ 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
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{ 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
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{ 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
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{ 0xff, 0xff, 0xff, 0x50, 0x51, 0x52, 0x53, 0xff, 0xff },
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{ 0x50, 0x51, 0x52, 0x53, 0xff, 0xff, 0xff, 0xff, 0xff },
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};
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static inline uint16_t dv_audio_12to16(uint16_t sample)
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{
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uint16_t shift, result;
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sample = (sample < 0x800) ? sample : sample | 0xf000;
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shift = (sample & 0xf00) >> 8;
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if (shift < 0x2 || shift > 0xd) {
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result = sample;
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} else if (shift < 0x8) {
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shift--;
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result = (sample - (256 * shift)) << shift;
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} else {
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shift = 0xe - shift;
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result = ((sample + ((256 * shift) + 1)) << shift) - 1;
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}
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return result;
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}
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static int dv_audio_frame_size(const DVprofile* sys, int frame)
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{
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return sys->audio_samples_dist[frame % (sizeof(sys->audio_samples_dist)/
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sizeof(sys->audio_samples_dist[0]))];
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}
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static int dv_write_pack(enum dv_pack_type pack_id, DVMuxContext *c, uint8_t* buf)
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{
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struct tm tc;
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time_t ct;
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int ltc_frame;
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/* Its hard to tell what SMPTE requires w.r.t. APT, but Quicktime needs it.
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* We set it based on pix_fmt value but it really should be per DV profile */
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int apt = (c->sys->pix_fmt == PIX_FMT_YUV422P ? 1 : 0);
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buf[0] = (uint8_t)pack_id;
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switch (pack_id) {
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case dv_timecode:
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ct = (time_t)(c->frames / ((float)c->sys->frame_rate /
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(float)c->sys->frame_rate_base));
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brktimegm(ct, &tc);
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/*
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* LTC drop-frame frame counter drops two frames (0 and 1) every
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* minute, unless it is exactly divisible by 10
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*/
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ltc_frame = (c->frames + 2*ct/60 - 2*ct/600) % c->sys->ltc_divisor;
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buf[1] = (0 << 7) | /* Color fame: 0 - unsync; 1 - sync mode */
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(1 << 6) | /* Drop frame timecode: 0 - nondrop; 1 - drop */
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((ltc_frame / 10) << 4) | /* Tens of frames */
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(ltc_frame % 10); /* Units of frames */
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buf[2] = (1 << 7) | /* Biphase mark polarity correction: 0 - even; 1 - odd */
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((tc.tm_sec / 10) << 4) | /* Tens of seconds */
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(tc.tm_sec % 10); /* Units of seconds */
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buf[3] = (1 << 7) | /* Binary group flag BGF0 */
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((tc.tm_min / 10) << 4) | /* Tens of minutes */
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(tc.tm_min % 10); /* Units of minutes */
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buf[4] = (1 << 7) | /* Binary group flag BGF2 */
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(1 << 6) | /* Binary group flag BGF1 */
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((tc.tm_hour / 10) << 4) | /* Tens of hours */
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(tc.tm_hour % 10); /* Units of hours */
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break;
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case dv_audio_source: /* AAUX source pack */
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buf[1] = (0 << 7) | /* locked mode */
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(1 << 6) | /* reserved -- always 1 */
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(dv_audio_frame_size(c->sys, c->frames) -
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c->sys->audio_min_samples[0]);
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/* # of samples */
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buf[2] = (0 << 7) | /* multi-stereo */
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(0 << 5) | /* #of audio channels per block: 0 -- 1 channel */
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(0 << 4) | /* pair bit: 0 -- one pair of channels */
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0; /* audio mode */
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buf[3] = (1 << 7) | /* res */
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(1 << 6) | /* multi-language flag */
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(c->sys->dsf << 5) | /* system: 60fields/50fields */
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(apt << 1);/* definition: 0 -- 25Mbps, 2 -- 50Mbps */
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buf[4] = (1 << 7) | /* emphasis: 1 -- off */
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(0 << 6) | /* emphasis time constant: 0 -- reserved */
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(0 << 3) | /* frequency: 0 -- 48Khz, 1 -- 44,1Khz, 2 -- 32Khz */
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0; /* quantization: 0 -- 16bit linear, 1 -- 12bit nonlinear */
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break;
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case dv_audio_control:
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buf[1] = (0 << 6) | /* copy protection: 0 -- unrestricted */
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(1 << 4) | /* input source: 1 -- digital input */
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(3 << 2) | /* compression: 3 -- no information */
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0; /* misc. info/SMPTE emphasis off */
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buf[2] = (1 << 7) | /* recording start point: 1 -- no */
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(1 << 6) | /* recording end point: 1 -- no */
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(1 << 3) | /* recording mode: 1 -- original */
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7;
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buf[3] = (1 << 7) | /* direction: 1 -- forward */
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0x20; /* speed */
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buf[4] = (1 << 7) | /* reserved -- always 1 */
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0x7f; /* genre category */
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break;
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case dv_audio_recdate:
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case dv_video_recdate: /* VAUX recording date */
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ct = c->start_time + (time_t)(c->frames /
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((float)c->sys->frame_rate / (float)c->sys->frame_rate_base));
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brktimegm(ct, &tc);
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buf[1] = 0xff; /* ds, tm, tens of time zone, units of time zone */
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/* 0xff is very likely to be "unknown" */
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buf[2] = (3 << 6) | /* reserved -- always 1 */
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((tc.tm_mday / 10) << 4) | /* Tens of day */
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(tc.tm_mday % 10); /* Units of day */
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buf[3] = /* we set high 4 bits to 0, shouldn't we set them to week? */
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((tc.tm_mon / 10) << 4) | /* Tens of month */
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(tc.tm_mon % 10); /* Units of month */
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buf[4] = (((tc.tm_year % 100) / 10) << 4) | /* Tens of year */
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(tc.tm_year % 10); /* Units of year */
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break;
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case dv_audio_rectime: /* AAUX recording time */
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case dv_video_rectime: /* VAUX recording time */
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ct = c->start_time + (time_t)(c->frames /
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((float)c->sys->frame_rate / (float)c->sys->frame_rate_base));
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brktimegm(ct, &tc);
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buf[1] = (3 << 6) | /* reserved -- always 1 */
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0x3f; /* tens of frame, units of frame: 0x3f - "unknown" ? */
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buf[2] = (1 << 7) | /* reserved -- always 1 */
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((tc.tm_sec / 10) << 4) | /* Tens of seconds */
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(tc.tm_sec % 10); /* Units of seconds */
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buf[3] = (1 << 7) | /* reserved -- always 1 */
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((tc.tm_min / 10) << 4) | /* Tens of minutes */
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(tc.tm_min % 10); /* Units of minutes */
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buf[4] = (3 << 6) | /* reserved -- always 1 */
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((tc.tm_hour / 10) << 4) | /* Tens of hours */
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(tc.tm_hour % 10); /* Units of hours */
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break;
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default:
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buf[1] = buf[2] = buf[3] = buf[4] = 0xff;
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}
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return 5;
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}
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static void dv_inject_audio(DVMuxContext *c, int channel, uint8_t* frame_ptr)
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{
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int i, j, d, of, size;
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size = 4 * dv_audio_frame_size(c->sys, c->frames);
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frame_ptr += channel * c->sys->difseg_size * 150 * 80;
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for (i = 0; i < c->sys->difseg_size; i++) {
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frame_ptr += 6 * 80; /* skip DIF segment header */
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for (j = 0; j < 9; j++) {
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dv_write_pack(dv_aaux_packs_dist[i][j], c, &frame_ptr[3]);
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for (d = 8; d < 80; d+=2) {
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of = c->sys->audio_shuffle[i][j] + (d - 8)/2 * c->sys->audio_stride;
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if (of*2 >= size)
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continue;
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frame_ptr[d] = fifo_peek(&c->audio_data[channel], of*2+1); // FIXME: may be we have to admit
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frame_ptr[d+1] = fifo_peek(&c->audio_data[channel], of*2); // that DV is a big endian PCM
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}
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frame_ptr += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
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}
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}
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}
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static void dv_inject_metadata(DVMuxContext *c, uint8_t* frame)
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{
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int j, k;
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uint8_t* buf;
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for (buf = frame; buf < frame + c->sys->frame_size; buf += 150 * 80) {
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/* DV subcode: 2nd and 3d DIFs */
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for (j = 80; j < 80 * 3; j += 80) {
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for (k = 6; k < 6 * 8; k += 8)
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dv_write_pack(dv_timecode, c, &buf[j+k]);
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if (((long)(buf-frame)/(c->sys->frame_size/(c->sys->difseg_size*c->sys->n_difchan))%c->sys->difseg_size) > 5) { /* FIXME: is this really needed ? */
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dv_write_pack(dv_video_recdate, c, &buf[j+14]);
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dv_write_pack(dv_video_rectime, c, &buf[j+22]);
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dv_write_pack(dv_video_recdate, c, &buf[j+38]);
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dv_write_pack(dv_video_rectime, c, &buf[j+46]);
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}
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}
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/* DV VAUX: 4th, 5th and 6th 3DIFs */
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for (j = 80*3 + 3; j < 80*6; j += 80) {
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dv_write_pack(dv_video_recdate, c, &buf[j+5*2]);
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dv_write_pack(dv_video_rectime, c, &buf[j+5*3]);
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dv_write_pack(dv_video_recdate, c, &buf[j+5*11]);
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dv_write_pack(dv_video_rectime, c, &buf[j+5*12]);
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}
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}
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}
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/*
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* This is the dumbest implementation of all -- it simply looks at
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* a fixed offset and if pack isn't there -- fails. We might want
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* to have a fallback mechanism for complete search of missing packs.
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*/
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static const uint8_t* dv_extract_pack(uint8_t* frame, enum dv_pack_type t)
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{
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int offs;
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switch (t) {
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case dv_audio_source:
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offs = (80*6 + 80*16*3 + 3);
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break;
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case dv_audio_control:
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offs = (80*6 + 80*16*4 + 3);
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break;
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case dv_video_control:
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offs = (80*5 + 48 + 5);
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break;
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default:
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return NULL;
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}
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return (frame[offs] == t ? &frame[offs] : NULL);
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}
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/*
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* There's a couple of assumptions being made here:
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* 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples.
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* We can pass them upwards when ffmpeg will be ready to deal with them.
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* 2. We don't do software emphasis.
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* 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples
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* are converted into 16bit linear ones.
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*/
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static int dv_extract_audio(uint8_t* frame, uint8_t* pcm, uint8_t* pcm2,
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const DVprofile *sys)
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{
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int size, chan, i, j, d, of, smpls, freq, quant, half_ch;
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uint16_t lc, rc;
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const uint8_t* as_pack;
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as_pack = dv_extract_pack(frame, dv_audio_source);
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if (!as_pack) /* No audio ? */
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return 0;
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smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
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freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */
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quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
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if (quant > 1)
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return -1; /* Unsupported quantization */
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size = (sys->audio_min_samples[freq] + smpls) * 4; /* 2ch, 2bytes */
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|
|
half_ch = sys->difseg_size/2;
|
|
|
|
|
|
|
|
/* for each DIF channel */
|
|
|
|
for (chan = 0; chan < sys->n_difchan; chan++) {
|
|
|
|
/* for each DIF segment */
|
|
|
|
for (i = 0; i < sys->difseg_size; i++) {
|
|
|
|
frame += 6 * 80; /* skip DIF segment header */
|
|
|
|
if (quant == 1 && i == half_ch) {
|
|
|
|
/* next stereo channel (12bit mode only) */
|
|
|
|
if (!pcm2)
|
|
|
|
break;
|
|
|
|
else
|
|
|
|
pcm = pcm2;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* for each AV sequence */
|
|
|
|
for (j = 0; j < 9; j++) {
|
|
|
|
for (d = 8; d < 80; d += 2) {
|
|
|
|
if (quant == 0) { /* 16bit quantization */
|
|
|
|
of = sys->audio_shuffle[i][j] + (d - 8)/2 * sys->audio_stride;
|
|
|
|
if (of*2 >= size)
|
|
|
|
continue;
|
|
|
|
|
|
|
|
pcm[of*2] = frame[d+1]; // FIXME: may be we have to admit
|
|
|
|
pcm[of*2+1] = frame[d]; // that DV is a big endian PCM
|
|
|
|
if (pcm[of*2+1] == 0x80 && pcm[of*2] == 0x00)
|
|
|
|
pcm[of*2+1] = 0;
|
|
|
|
} else { /* 12bit quantization */
|
|
|
|
lc = ((uint16_t)frame[d] << 4) |
|
|
|
|
((uint16_t)frame[d+2] >> 4);
|
|
|
|
rc = ((uint16_t)frame[d+1] << 4) |
|
|
|
|
((uint16_t)frame[d+2] & 0x0f);
|
|
|
|
lc = (lc == 0x800 ? 0 : dv_audio_12to16(lc));
|
|
|
|
rc = (rc == 0x800 ? 0 : dv_audio_12to16(rc));
|
|
|
|
|
|
|
|
of = sys->audio_shuffle[i%half_ch][j] + (d - 8)/3 * sys->audio_stride;
|
|
|
|
if (of*2 >= size)
|
|
|
|
continue;
|
|
|
|
|
|
|
|
pcm[of*2] = lc & 0xff; // FIXME: may be we have to admit
|
|
|
|
pcm[of*2+1] = lc >> 8; // that DV is a big endian PCM
|
|
|
|
of = sys->audio_shuffle[i%half_ch+half_ch][j] +
|
|
|
|
(d - 8)/3 * sys->audio_stride;
|
|
|
|
pcm[of*2] = rc & 0xff; // FIXME: may be we have to admit
|
|
|
|
pcm[of*2+1] = rc >> 8; // that DV is a big endian PCM
|
|
|
|
++d;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
frame += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* next stereo channel (50Mbps only) */
|
|
|
|
if(!pcm2)
|
|
|
|
break;
|
|
|
|
pcm = pcm2;
|
|
|
|
}
|
|
|
|
|
|
|
|
return size;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int dv_extract_audio_info(DVDemuxContext* c, uint8_t* frame)
|
|
|
|
{
|
|
|
|
const uint8_t* as_pack;
|
|
|
|
int freq, stype, smpls, quant, i, ach;
|
|
|
|
|
|
|
|
as_pack = dv_extract_pack(frame, dv_audio_source);
|
|
|
|
if (!as_pack || !c->sys) { /* No audio ? */
|
|
|
|
c->ach = 0;
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */
|
|
|
|
freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */
|
|
|
|
stype = (as_pack[3] & 0x1f); /* 0 - 2CH, 2 - 4CH */
|
|
|
|
quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */
|
|
|
|
|
|
|
|
/* note: ach counts PAIRS of channels (i.e. stereo channels) */
|
|
|
|
ach = (stype == 2 || (quant && (freq == 2))) ? 2 : 1;
|
|
|
|
|
|
|
|
/* Dynamic handling of the audio streams in DV */
|
|
|
|
for (i=0; i<ach; i++) {
|
|
|
|
if (!c->ast[i]) {
|
|
|
|
c->ast[i] = av_new_stream(c->fctx, 0);
|
|
|
|
if (!c->ast[i])
|
|
|
|
break;
|
|
|
|
av_set_pts_info(c->ast[i], 64, 1, 30000);
|
|
|
|
c->ast[i]->codec->codec_type = CODEC_TYPE_AUDIO;
|
|
|
|
c->ast[i]->codec->codec_id = CODEC_ID_PCM_S16LE;
|
|
|
|
|
|
|
|
av_init_packet(&c->audio_pkt[i]);
|
|
|
|
c->audio_pkt[i].size = 0;
|
|
|
|
c->audio_pkt[i].data = c->audio_buf[i];
|
|
|
|
c->audio_pkt[i].stream_index = c->ast[i]->index;
|
|
|
|
c->audio_pkt[i].flags |= PKT_FLAG_KEY;
|
|
|
|
}
|
|
|
|
c->ast[i]->codec->sample_rate = dv_audio_frequency[freq];
|
|
|
|
c->ast[i]->codec->channels = 2;
|
|
|
|
c->ast[i]->codec->bit_rate = 2 * dv_audio_frequency[freq] * 16;
|
|
|
|
c->ast[i]->start_time = 0;
|
|
|
|
}
|
|
|
|
c->ach = i;
|
|
|
|
|
|
|
|
return (c->sys->audio_min_samples[freq] + smpls) * 4; /* 2ch, 2bytes */;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int dv_extract_video_info(DVDemuxContext *c, uint8_t* frame)
|
|
|
|
{
|
|
|
|
const uint8_t* vsc_pack;
|
|
|
|
AVCodecContext* avctx;
|
|
|
|
int apt, is16_9;
|
|
|
|
int size = 0;
|
|
|
|
|
|
|
|
if (c->sys) {
|
|
|
|
avctx = c->vst->codec;
|
|
|
|
|
|
|
|
av_set_pts_info(c->vst, 64, c->sys->frame_rate_base, c->sys->frame_rate);
|
|
|
|
avctx->time_base= (AVRational){c->sys->frame_rate_base, c->sys->frame_rate};
|
|
|
|
if(!avctx->width){
|
|
|
|
avctx->width = c->sys->width;
|
|
|
|
avctx->height = c->sys->height;
|
|
|
|
}
|
|
|
|
avctx->pix_fmt = c->sys->pix_fmt;
|
|
|
|
|
|
|
|
/* finding out SAR is a little bit messy */
|
|
|
|
vsc_pack = dv_extract_pack(frame, dv_video_control);
|
|
|
|
apt = frame[4] & 0x07;
|
|
|
|
is16_9 = (vsc_pack && ((vsc_pack[2] & 0x07) == 0x02 ||
|
|
|
|
(!apt && (vsc_pack[2] & 0x07) == 0x07)));
|
|
|
|
avctx->sample_aspect_ratio = c->sys->sar[is16_9];
|
|
|
|
avctx->bit_rate = av_rescale(c->sys->frame_size * 8,
|
|
|
|
c->sys->frame_rate,
|
|
|
|
c->sys->frame_rate_base);
|
|
|
|
size = c->sys->frame_size;
|
|
|
|
}
|
|
|
|
return size;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
* The following 6 functions constitute our interface to the world
|
|
|
|
*/
|
|
|
|
|
|
|
|
int dv_assemble_frame(DVMuxContext *c, AVStream* st,
|
|
|
|
const uint8_t* data, int data_size, uint8_t** frame)
|
|
|
|
{
|
|
|
|
int i, reqasize;
|
|
|
|
|
|
|
|
*frame = &c->frame_buf[0];
|
|
|
|
reqasize = 4 * dv_audio_frame_size(c->sys, c->frames);
|
|
|
|
|
|
|
|
switch (st->codec->codec_type) {
|
|
|
|
case CODEC_TYPE_VIDEO:
|
|
|
|
/* FIXME: we have to have more sensible approach than this one */
|
|
|
|
if (c->has_video)
|
|
|
|
av_log(st->codec, AV_LOG_ERROR, "Can't process DV frame #%d. Insufficient audio data or severe sync problem.\n", c->frames);
|
|
|
|
|
|
|
|
memcpy(*frame, data, c->sys->frame_size);
|
|
|
|
c->has_video = 1;
|
|
|
|
break;
|
|
|
|
case CODEC_TYPE_AUDIO:
|
|
|
|
for (i = 0; i < c->n_ast && st != c->ast[i]; i++);
|
|
|
|
|
|
|
|
/* FIXME: we have to have more sensible approach than this one */
|
|
|
|
if (fifo_size(&c->audio_data[i], c->audio_data[i].rptr) + data_size >= 100*AVCODEC_MAX_AUDIO_FRAME_SIZE)
|
|
|
|
av_log(st->codec, AV_LOG_ERROR, "Can't process DV frame #%d. Insufficient video data or severe sync problem.\n", c->frames);
|
|
|
|
fifo_write(&c->audio_data[i], data, data_size, &c->audio_data[i].wptr);
|
|
|
|
|
|
|
|
/* Lets see if we've got enough audio for one DV frame */
|
|
|
|
c->has_audio |= ((reqasize <= fifo_size(&c->audio_data[i], c->audio_data[i].rptr)) << i);
|
|
|
|
|
|
|
|
break;
|
|
|
|
default:
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Lets see if we have enough data to construct one DV frame */
|
|
|
|
if (c->has_video == 1 && c->has_audio + 1 == 1<<c->n_ast) {
|
|
|
|
dv_inject_metadata(c, *frame);
|
|
|
|
for (i=0; i<c->n_ast; i++) {
|
|
|
|
dv_inject_audio(c, i, *frame);
|
|
|
|
fifo_drain(&c->audio_data[i], reqasize);
|
|
|
|
}
|
|
|
|
|
|
|
|
c->has_video = 0;
|
|
|
|
c->has_audio = 0;
|
|
|
|
c->frames++;
|
|
|
|
|
|
|
|
return c->sys->frame_size;
|
|
|
|
}
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
DVMuxContext* dv_init_mux(AVFormatContext* s)
|
|
|
|
{
|
|
|
|
DVMuxContext *c;
|
|
|
|
AVStream *vst = NULL;
|
|
|
|
int i;
|
|
|
|
|
|
|
|
/* we support at most 1 video and 2 audio streams */
|
|
|
|
if (s->nb_streams > 3)
|
|
|
|
return NULL;
|
|
|
|
|
|
|
|
c = av_mallocz(sizeof(DVMuxContext));
|
|
|
|
if (!c)
|
|
|
|
return NULL;
|
|
|
|
|
|
|
|
c->n_ast = 0;
|
|
|
|
c->ast[0] = c->ast[1] = NULL;
|
|
|
|
|
|
|
|
/* We have to sort out where audio and where video stream is */
|
|
|
|
for (i=0; i<s->nb_streams; i++) {
|
|
|
|
switch (s->streams[i]->codec->codec_type) {
|
|
|
|
case CODEC_TYPE_VIDEO:
|
|
|
|
vst = s->streams[i];
|
|
|
|
break;
|
|
|
|
case CODEC_TYPE_AUDIO:
|
|
|
|
c->ast[c->n_ast++] = s->streams[i];
|
|
|
|
break;
|
|
|
|
default:
|
|
|
|
goto bail_out;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Some checks -- DV format is very picky about its incoming streams */
|
|
|
|
if (!vst || vst->codec->codec_id != CODEC_ID_DVVIDEO)
|
|
|
|
goto bail_out;
|
|
|
|
for (i=0; i<c->n_ast; i++) {
|
|
|
|
if (c->ast[i] && (c->ast[i]->codec->codec_id != CODEC_ID_PCM_S16LE ||
|
|
|
|
c->ast[i]->codec->sample_rate != 48000 ||
|
|
|
|
c->ast[i]->codec->channels != 2))
|
|
|
|
goto bail_out;
|
|
|
|
}
|
|
|
|
c->sys = dv_codec_profile(vst->codec);
|
|
|
|
if (!c->sys)
|
|
|
|
goto bail_out;
|
|
|
|
|
|
|
|
if((c->n_ast > 1) && (c->sys->n_difchan < 2)) {
|
|
|
|
/* only 1 stereo pair is allowed in 25Mbps mode */
|
|
|
|
goto bail_out;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Ok, everything seems to be in working order */
|
|
|
|
c->frames = 0;
|
|
|
|
c->has_audio = 0;
|
|
|
|
c->has_video = 0;
|
|
|
|
c->start_time = (time_t)s->timestamp;
|
|
|
|
|
|
|
|
for (i=0; i<c->n_ast; i++) {
|
|
|
|
if (c->ast[i] && fifo_init(&c->audio_data[i], 100*AVCODEC_MAX_AUDIO_FRAME_SIZE) < 0) {
|
|
|
|
while (i>0) {
|
|
|
|
i--;
|
|
|
|
fifo_free(&c->audio_data[i]);
|
|
|
|
}
|
|
|
|
goto bail_out;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
return c;
|
|
|
|
|
|
|
|
bail_out:
|
|
|
|
av_free(c);
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
void dv_delete_mux(DVMuxContext *c)
|
|
|
|
{
|
|
|
|
int i;
|
|
|
|
for (i=0; i < c->n_ast; i++)
|
|
|
|
fifo_free(&c->audio_data[i]);
|
|
|
|
}
|
|
|
|
|
|
|
|
DVDemuxContext* dv_init_demux(AVFormatContext *s)
|
|
|
|
{
|
|
|
|
DVDemuxContext *c;
|
|
|
|
|
|
|
|
c = av_mallocz(sizeof(DVDemuxContext));
|
|
|
|
if (!c)
|
|
|
|
return NULL;
|
|
|
|
|
|
|
|
c->vst = av_new_stream(s, 0);
|
|
|
|
if (!c->vst) {
|
|
|
|
av_free(c);
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
c->sys = NULL;
|
|
|
|
c->fctx = s;
|
|
|
|
c->ast[0] = c->ast[1] = NULL;
|
|
|
|
c->ach = 0;
|
|
|
|
c->frames = 0;
|
|
|
|
c->abytes = 0;
|
|
|
|
|
|
|
|
c->vst->codec->codec_type = CODEC_TYPE_VIDEO;
|
|
|
|
c->vst->codec->codec_id = CODEC_ID_DVVIDEO;
|
|
|
|
c->vst->codec->bit_rate = 25000000;
|
|
|
|
c->vst->start_time = 0;
|
|
|
|
|
|
|
|
return c;
|
|
|
|
}
|
|
|
|
|
|
|
|
int dv_get_packet(DVDemuxContext *c, AVPacket *pkt)
|
|
|
|
{
|
|
|
|
int size = -1;
|
|
|
|
int i;
|
|
|
|
|
|
|
|
for (i=0; i<c->ach; i++) {
|
|
|
|
if (c->ast[i] && c->audio_pkt[i].size) {
|
|
|
|
*pkt = c->audio_pkt[i];
|
|
|
|
c->audio_pkt[i].size = 0;
|
|
|
|
size = pkt->size;
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
return size;
|
|
|
|
}
|
|
|
|
|
|
|
|
int dv_produce_packet(DVDemuxContext *c, AVPacket *pkt,
|
|
|
|
uint8_t* buf, int buf_size)
|
|
|
|
{
|
|
|
|
int size, i;
|
|
|
|
|
|
|
|
if (buf_size < DV_PROFILE_BYTES ||
|
|
|
|
!(c->sys = dv_frame_profile(buf)) ||
|
|
|
|
buf_size < c->sys->frame_size) {
|
|
|
|
return -1; /* Broken frame, or not enough data */
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Queueing audio packet */
|
|
|
|
/* FIXME: in case of no audio/bad audio we have to do something */
|
|
|
|
size = dv_extract_audio_info(c, buf);
|
|
|
|
for (i=0; i<c->ach; i++) {
|
|
|
|
c->audio_pkt[i].size = size;
|
|
|
|
c->audio_pkt[i].pts = c->abytes * 30000*8 / c->ast[i]->codec->bit_rate;
|
|
|
|
}
|
|
|
|
dv_extract_audio(buf, c->audio_buf[0], c->audio_buf[1], c->sys);
|
|
|
|
c->abytes += size;
|
|
|
|
|
|
|
|
/* Now it's time to return video packet */
|
|
|
|
size = dv_extract_video_info(c, buf);
|
|
|
|
av_init_packet(pkt);
|
|
|
|
pkt->data = buf;
|
|
|
|
pkt->size = size;
|
|
|
|
pkt->flags |= PKT_FLAG_KEY;
|
|
|
|
pkt->stream_index = c->vst->id;
|
|
|
|
pkt->pts = c->frames;
|
|
|
|
|
|
|
|
c->frames++;
|
|
|
|
|
|
|
|
return size;
|
|
|
|
}
|
|
|
|
|
|
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static int64_t dv_frame_offset(AVFormatContext *s, DVDemuxContext *c,
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int64_t timestamp, int flags)
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{
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// FIXME: sys may be wrong if last dv_read_packet() failed (buffer is junk)
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const DVprofile* sys = dv_codec_profile(c->vst->codec);
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int64_t offset;
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int64_t size = url_fsize(&s->pb);
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int64_t max_offset = ((size-1) / sys->frame_size) * sys->frame_size;
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offset = sys->frame_size * timestamp;
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if (offset > max_offset) offset = max_offset;
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else if (offset < 0) offset = 0;
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return offset;
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}
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void dv_flush_audio_packets(DVDemuxContext *c)
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|
{
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|
c->audio_pkt[0].size = c->audio_pkt[1].size = 0;
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}
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/************************************************************
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* Implementation of the easiest DV storage of all -- raw DV.
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|
************************************************************/
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typedef struct RawDVContext {
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DVDemuxContext* dv_demux;
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uint8_t buf[DV_MAX_FRAME_SIZE];
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} RawDVContext;
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static int dv_read_header(AVFormatContext *s,
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|
AVFormatParameters *ap)
|
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|
{
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|
RawDVContext *c = s->priv_data;
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c->dv_demux = dv_init_demux(s);
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if (!c->dv_demux)
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return -1;
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if (get_buffer(&s->pb, c->buf, DV_PROFILE_BYTES) <= 0 ||
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url_fseek(&s->pb, -DV_PROFILE_BYTES, SEEK_CUR) < 0)
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return AVERROR_IO;
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c->dv_demux->sys = dv_frame_profile(c->buf);
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s->bit_rate = av_rescale(c->dv_demux->sys->frame_size * 8,
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|
c->dv_demux->sys->frame_rate,
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|
|
c->dv_demux->sys->frame_rate_base);
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return 0;
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|
}
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|
static int dv_read_packet(AVFormatContext *s, AVPacket *pkt)
|
|
|
|
{
|
|
|
|
int size;
|
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|
|
RawDVContext *c = s->priv_data;
|
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|
|
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|
|
size = dv_get_packet(c->dv_demux, pkt);
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|
|
if (size < 0) {
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|
size = c->dv_demux->sys->frame_size;
|
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|
|
if (get_buffer(&s->pb, c->buf, size) <= 0)
|
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|
|
return AVERROR_IO;
|
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|
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|
|
size = dv_produce_packet(c->dv_demux, pkt, c->buf, size);
|
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|
|
}
|
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|
|
return size;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int dv_read_seek(AVFormatContext *s, int stream_index,
|
|
|
|
int64_t timestamp, int flags)
|
|
|
|
{
|
|
|
|
RawDVContext *r = s->priv_data;
|
|
|
|
DVDemuxContext *c = r->dv_demux;
|
|
|
|
int64_t offset= dv_frame_offset(s, c, timestamp, flags);
|
|
|
|
|
|
|
|
c->frames= offset / c->sys->frame_size;
|
|
|
|
if (c->ach)
|
|
|
|
c->abytes= av_rescale(c->frames,
|
|
|
|
c->ast[0]->codec->bit_rate * (int64_t)c->sys->frame_rate_base,
|
|
|
|
8*c->sys->frame_rate);
|
|
|
|
|
|
|
|
dv_flush_audio_packets(c);
|
|
|
|
return url_fseek(&s->pb, offset, SEEK_SET);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int dv_read_close(AVFormatContext *s)
|
|
|
|
{
|
|
|
|
RawDVContext *c = s->priv_data;
|
|
|
|
av_free(c->dv_demux);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
#ifdef CONFIG_MUXERS
|
|
|
|
static int dv_write_header(AVFormatContext *s)
|
|
|
|
{
|
|
|
|
s->priv_data = dv_init_mux(s);
|
|
|
|
if (!s->priv_data) {
|
|
|
|
av_log(s, AV_LOG_ERROR, "Can't initialize DV format!\n"
|
|
|
|
"Make sure that you supply exactly two streams:\n"
|
|
|
|
" video: 25fps or 29.97fps, audio: 2ch/48Khz/PCM\n"
|
|
|
|
" (50Mbps allows an optional second audio stream)\n");
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int dv_write_packet(struct AVFormatContext *s, AVPacket *pkt)
|
|
|
|
{
|
|
|
|
uint8_t* frame;
|
|
|
|
int fsize;
|
|
|
|
|
|
|
|
fsize = dv_assemble_frame((DVMuxContext *)s->priv_data, s->streams[pkt->stream_index],
|
|
|
|
pkt->data, pkt->size, &frame);
|
|
|
|
if (fsize > 0) {
|
|
|
|
put_buffer(&s->pb, frame, fsize);
|
|
|
|
put_flush_packet(&s->pb);
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
* We might end up with some extra A/V data without matching counterpart.
|
|
|
|
* E.g. video data without enough audio to write the complete frame.
|
|
|
|
* Currently we simply drop the last frame. I don't know whether this
|
|
|
|
* is the best strategy of all
|
|
|
|
*/
|
|
|
|
static int dv_write_trailer(struct AVFormatContext *s)
|
|
|
|
{
|
|
|
|
dv_delete_mux((DVMuxContext *)s->priv_data);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
#endif /* CONFIG_MUXERS */
|
|
|
|
|
|
|
|
#ifdef CONFIG_DV_DEMUXER
|
|
|
|
AVInputFormat dv_demuxer = {
|
|
|
|
"dv",
|
|
|
|
"DV video format",
|
|
|
|
sizeof(RawDVContext),
|
|
|
|
NULL,
|
|
|
|
dv_read_header,
|
|
|
|
dv_read_packet,
|
|
|
|
dv_read_close,
|
|
|
|
dv_read_seek,
|
|
|
|
.extensions = "dv,dif",
|
|
|
|
};
|
|
|
|
#endif
|
|
|
|
#ifdef CONFIG_DV_MUXER
|
|
|
|
AVOutputFormat dv_muxer = {
|
|
|
|
"dv",
|
|
|
|
"DV video format",
|
|
|
|
NULL,
|
|
|
|
"dv",
|
|
|
|
sizeof(DVMuxContext),
|
|
|
|
CODEC_ID_PCM_S16LE,
|
|
|
|
CODEC_ID_DVVIDEO,
|
|
|
|
dv_write_header,
|
|
|
|
dv_write_packet,
|
|
|
|
dv_write_trailer,
|
|
|
|
};
|
|
|
|
#endif
|