/* * General DV demuxer * Copyright (c) 2003 Roman Shaposhnik * * Many thanks to Dan Dennedy <dan@dennedy.org> for providing wealth * of DV technical info. * * Raw DV format * Copyright (c) 2002 Fabrice Bellard * * 50 Mbps (DVCPRO50) and 100 Mbps (DVCPRO HD) support * Copyright (c) 2006 Daniel Maas <dmaas@maasdigital.com> * Funded by BBC Research & Development * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "config_components.h" #include <time.h>#include "avformat.h"#include "demux.h"#include "internal.h"#include "libavcodec/dv_profile.h"#include "libavcodec/dv.h"#include "libavutil/channel_layout.h"#include "libavutil/intreadwrite.h"#include "libavutil/mathematics.h"#include "libavutil/mem.h"#include "libavutil/timecode.h"#include "dv.h"#include "libavutil/avassert.h" #if CONFIG_DV_DEMUXER // Must be kept in sync with AVPackettypedef struct DVPacket {    int64_t  pts;    uint8_t *data;    int      size;    int      stream_index;    int      flags;    int64_t  pos;    int64_t  duration;     int sample_rate;    int last_sample_rate;} DVPacket; struct DVDemuxContext {    const AVDVProfile*  sys;    /* Current DV profile. E.g.: 525/60, 625/50 */    AVFormatContext*  fctx;    AVStream*         vst;    AVStream*         ast[4];    struct DVPacket   audio_pkt[4];    uint8_t           audio_buf[4][8192];    int               ach;    int               frames;     int64_t           next_pts_video;    int64_t           next_pts_audio;}; static inline uint16_t dv_audio_12to16(uint16_t sample){    uint16_t shift, result;     sample = (sample < 0x800) ? sample : sample | 0xf000;    shift  = (sample & 0xf00) >> 8;     if (shift < 0x2 || shift > 0xd) {        result = sample;    } else if (shift < 0x8) {        shift--;        result = (sample - (256 * shift)) << shift;    } else {        shift  = 0xe - shift;        result = ((sample + ((256 * shift) + 1)) << shift) - 1;    }     return result;} static const uint8_t *dv_extract_pack(const uint8_t *frame, enum DVPackType t){    int offs;    int c;     for (c = 0; c < 10; c++) {        switch (t) {        case DV_AUDIO_SOURCE:            if (c&1)    offs = (80 * 6 + 80 * 16 * 0 + 3 + c*12000);            else        offs = (80 * 6 + 80 * 16 * 3 + 3 + c*12000);            break;        case DV_AUDIO_CONTROL:            if (c&1)    offs = (80 * 6 + 80 * 16 * 1 + 3 + c*12000);            else        offs = (80 * 6 + 80 * 16 * 4 + 3 + c*12000);            break;        case DV_VIDEO_CONTROL:            if (c&1)    offs = (80 * 3 + 8      + c*12000);            else        offs = (80 * 5 + 48 + 5 + c*12000);            break;        case DV_TIMECODE:            offs = (80*1 + 3 + 3);            break;        default:            return NULL;        }        if (frame[offs] == t)            break;    }     return frame[offs] == t ? &frame[offs] : NULL;} static const int dv_audio_frequency[3] = {    48000, 44100, 32000,}; /* * There's a couple of assumptions being made here: * 1. By default we silence erroneous (0x8000/16-bit 0x800/12-bit) audio samples. *    We can pass them upwards when libavcodec will be ready to deal with them. * 2. We don't do software emphasis. * 3. Audio is always returned as 16-bit linear samples: 12-bit nonlinear samples *    are converted into 16-bit linear ones. */static int dv_extract_audio(const uint8_t *frame, uint8_t **ppcm,                            const AVDVProfile *sys){    int size, chan, i, j, d, of, smpls, freq, quant, half_ch;    uint16_t lc, rc;    const uint8_t *as_pack;    uint8_t *pcm, ipcm;     as_pack = dv_extract_pack(frame, DV_AUDIO_SOURCE);    if (!as_pack)    /* No audio ? */        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 - 32kHz */    quant = as_pack[4]      & 0x07; /* 0 - 16-bit linear, 1 - 12-bit nonlinear */     if (quant > 1)        return -1;  /* unsupported quantization */     if (freq >= FF_ARRAY_ELEMS(dv_audio_frequency))        return AVERROR_INVALIDDATA;     size    = (sys->audio_min_samples[freq] + smpls) * 4; /* 2ch, 2bytes */    half_ch = sys->difseg_size / 2;     /* We work with 720p frames split in half, thus even frames have     * channels 0,1 and odd 2,3. */    ipcm = (sys->height == 720 && !(frame[1] & 0x0C)) ? 2 : 0;     if (ipcm + sys->n_difchan > (quant == 1 ? 2 : 4)) {        av_log(NULL, AV_LOG_ERROR, "too many dv pcm frames\n");        return AVERROR_INVALIDDATA;    }     /* for each DIF channel */    for (chan = 0; chan < sys->n_difchan; chan++) {        av_assert0(ipcm<4);        pcm = ppcm[ipcm++];        if (!pcm)            break;         /* 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 (12-bit mode only) */                av_assert0(ipcm<4);                pcm = ppcm[ipcm++];                if (!pcm)                    break;            }             /* for each AV sequence */            for (j = 0; j < 9; j++) {                for (d = 8; d < 80; d += 2) {                    if (quant == 0) {  /* 16-bit quantization */                        of = sys->audio_shuffle[i][j] +                             (d - 8) / 2 * sys->audio_stride;                        if (of * 2 >= size)                            continue;                         /* FIXME: maybe we have to admit that DV is a                         * big-endian PCM */                        pcm[of * 2]     = frame[d + 1];                        pcm[of * 2 + 1] = frame[d];                         if (pcm[of * 2 + 1] == 0x80 && pcm[of * 2] == 0x00)                            pcm[of * 2 + 1] = 0;                    } else {           /* 12-bit 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;                         /* FIXME: maybe we have to admit that DV is a                         * big-endian PCM */                        pcm[of * 2]     = lc & 0xff;                        pcm[of * 2 + 1] = lc >> 8;                        of = sys->audio_shuffle[i % half_ch + half_ch][j] +                             (d - 8) / 3 * sys->audio_stride;                        /* FIXME: maybe we have to admit that DV is a                         * big-endian PCM */                        pcm[of * 2]     = rc & 0xff;                        pcm[of * 2 + 1] = rc >> 8;                        ++d;                    }                }                 frame += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */            }        }    }     return size;} static int dv_extract_audio_info(DVDemuxContext *c, const uint8_t *frame){    const uint8_t *as_pack;    int freq, stype, smpls, quant, i, ach, sr;     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 - 32kHz */    stype = as_pack[3]      & 0x1f; /* 0 - 2CH, 2 - 4CH, 3 - 8CH */    quant = as_pack[4]      & 0x07; /* 0 - 16-bit linear, 1 - 12-bit nonlinear */     if (freq >= FF_ARRAY_ELEMS(dv_audio_frequency)) {        av_log(c->fctx, AV_LOG_ERROR,               "Unrecognized audio sample rate index (%d)\n", freq);        return 0;    }    sr = dv_audio_frequency[freq];     if (stype > 3) {        av_log(c->fctx, AV_LOG_ERROR, "stype %d is invalid\n", stype);        c->ach = 0;        return 0;    }     /* note: ach counts PAIRS of channels (i.e. stereo channels) */    ach = ((int[4]) { 1, 0, 2, 4 })[stype];    if (ach == 1 && quant && freq == 2)        ach = 2;     /* Dynamic handling of the audio streams in DV */    c->ach = 0;    for (i = 0; i < ach; i++) {        if (!c->ast[i]) {            c->ast[i] = avformat_new_stream(c->fctx, NULL);            if (!c->ast[i])                return AVERROR(ENOMEM);             avpriv_set_pts_info(c->ast[i], 64, 1, DV_TIMESCALE_AUDIO);            c->ast[i]->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;            c->ast[i]->codecpar->codec_id   = AV_CODEC_ID_PCM_S16LE;            c->ast[i]->codecpar->ch_layout  = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;            c->ast[i]->start_time           = 0;            c->ast[i]->codecpar->bit_rate   = 2 * sr * 16;             c->ast[i]->codecpar->sample_rate = sr;            c->audio_pkt[i].last_sample_rate = sr;             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       |= AV_PKT_FLAG_KEY;            c->audio_pkt[i].pts          = AV_NOPTS_VALUE;            c->audio_pkt[i].duration     = 0;            c->audio_pkt[i].pos          = -1;        }         c->audio_pkt[i].sample_rate = sr;    }    c->ach = ach;     return (c->sys->audio_min_samples[freq] + smpls) * 4; /* 2ch, 2bytes */} static int dv_extract_video_info(DVDemuxContext *c, const uint8_t *frame){    const uint8_t *vsc_pack;    AVCodecParameters *par;    int apt, is16_9;     par = c->vst->codecpar;     c->vst->avg_frame_rate = av_inv_q(c->vst->time_base);     /* 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)));    c->vst->sample_aspect_ratio = c->sys->sar[is16_9];    par->bit_rate = av_rescale_q(c->sys->frame_size,                                   (AVRational) { 8, 1 },                                   c->sys->time_base);    return c->sys->frame_size;} static int dv_extract_timecode(DVDemuxContext* c, const uint8_t* frame, char *tc){    const uint8_t *tc_pack;     // For PAL systems, drop frame bit is replaced by an arbitrary    // bit so its value should not be considered. Drop frame timecode    // is only relevant for NTSC systems.    int prevent_df = c->sys->ltc_divisor == 25 || c->sys->ltc_divisor == 50;     tc_pack = dv_extract_pack(frame, DV_TIMECODE);    if (!tc_pack)        return 0;    av_timecode_make_smpte_tc_string2(tc, av_inv_q(c->sys->time_base), AV_RB32(tc_pack + 1), prevent_df, 1);    return 1;} /* The following 3 functions constitute our interface to the world */ static int dv_init_demux(AVFormatContext *s, DVDemuxContext *c){    c->vst = avformat_new_stream(s, NULL);    if (!c->vst)        return AVERROR(ENOMEM);     c->fctx                   = s;    c->vst->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;    c->vst->codecpar->codec_id   = AV_CODEC_ID_DVVIDEO;    c->vst->start_time        = 0;     avpriv_set_pts_info(c->vst, 64, 1, DV_TIMESCALE_VIDEO);     /* Audio streams are added later as they are encountered. */    s->ctx_flags |= AVFMTCTX_NOHEADER;     return 0;} DVDemuxContext *avpriv_dv_init_demux(AVFormatContext *s){    DVDemuxContext *c;     c = av_mallocz(sizeof(DVDemuxContext));    if (!c)        return NULL;     if (dv_init_demux(s, c)) {        av_free(c);        return NULL;    }     return c;} int avpriv_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) {            DVPacket *dpkt = &c->audio_pkt[i];             pkt->size         = dpkt->size;            pkt->data         = dpkt->data;            pkt->stream_index = dpkt->stream_index;            pkt->flags        = dpkt->flags;            pkt->pts          = dpkt->pts;            pkt->duration     = dpkt->duration;            pkt->pos          = dpkt->pos;             dpkt->size = 0;            size       = pkt->size;             if (dpkt->last_sample_rate != dpkt->sample_rate) {                int ret = ff_add_param_change(pkt, 0, 0, dpkt->sample_rate, 0, 0);                if (ret < 0)                    return ret;                dpkt->last_sample_rate = dpkt->sample_rate;            }             break;        }    }     return size;} int avpriv_dv_produce_packet(DVDemuxContext *c, AVPacket *pkt,                             uint8_t *buf, int buf_size, int64_t pos){    int64_t pts, duration;    int size, i;    uint8_t *ppcm[5] = { 0 };     if (buf_size < DV_PROFILE_BYTES ||        !(c->sys = av_dv_frame_profile(c->sys, buf, buf_size)) ||        buf_size < c->sys->frame_size) {        return AVERROR_INVALIDDATA;    }     /* Queueing audio packet */    /* FIXME: in case of no audio/bad audio we have to do something */    size = dv_extract_audio_info(c, buf);    if (size < 0)        return size;     if (c->ach) {        int64_t next_pts_video = av_rescale_q(c->next_pts_video, c->vst->time_base,                                              c->ast[0]->time_base);         duration = av_rescale_q(size / 4,                                (AVRational){ 1, c->audio_pkt[0].sample_rate },                                c->ast[0]->time_base);         // if audio timestamps are more than one frame away from video,        // assume desync happened (e.g. due to dropped audio frames) and        // resynchronize        pts = (FFABS(next_pts_video - c->next_pts_audio) >= duration) ?              next_pts_video : c->next_pts_audio;         c->next_pts_audio = pts + duration;    }     for (i = 0; i < c->ach; i++) {        DVPacket *dpkt = &c->audio_pkt[i];         dpkt->pos      = pos;        dpkt->size     = size;        dpkt->pts      = pts;        dpkt->duration = duration;         ppcm[i] = c->audio_buf[i];    }    if (c->ach)        dv_extract_audio(buf, ppcm, c->sys);     /* We work with 720p frames split in half, thus even frames have     * channels 0,1 and odd 2,3. */    if (c->sys->height == 720) {        if (buf[1] & 0x0C) {            c->audio_pkt[2].size = c->audio_pkt[3].size = 0;        } else {            c->audio_pkt[0].size = c->audio_pkt[1].size = 0;        }    }     /* return the video packet, if the caller wants it */    if (pkt) {        size = dv_extract_video_info(c, buf);         pkt->data         = buf;        pkt->pos          = pos;        pkt->size         = size;        pkt->flags       |= AV_PKT_FLAG_KEY;        pkt->stream_index = c->vst->index;        pkt->pts          = c->next_pts_video;        pkt->duration     = av_rescale_q(1, c->sys->time_base, c->vst->time_base);         c->next_pts_video += pkt->duration;    }     c->frames++;     return size;} 

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