/* * LXF demuxer * Copyright (c) 2010 Tomas Härdin * * 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 "libavutil/intreadwrite.h" #include "avformat.h" #include "riff.h" #define LXF_PACKET_HEADER_SIZE 60 #define LXF_HEADER_DATA_SIZE 120 #define LXF_IDENT "LEITCH\0" #define LXF_IDENT_LENGTH 8 #define LXF_SAMPLERATE 48000 #define LXF_MAX_AUDIO_PACKET (8008*15*4) ///< 15-channel 32-bit NTSC audio frame static const AVCodecTag lxf_tags[] = { { CODEC_ID_MJPEG, 0 }, { CODEC_ID_MPEG1VIDEO, 1 }, { CODEC_ID_MPEG2VIDEO, 2 }, //MpMl, 4:2:0 { CODEC_ID_MPEG2VIDEO, 3 }, //MpPl, 4:2:2 { CODEC_ID_DVVIDEO, 4 }, //DV25 { CODEC_ID_DVVIDEO, 5 }, //DVCPRO { CODEC_ID_DVVIDEO, 6 }, //DVCPRO50 { CODEC_ID_RAWVIDEO, 7 }, //PIX_FMT_ARGB, where alpha is used for chroma keying { CODEC_ID_RAWVIDEO, 8 }, //16-bit chroma key { CODEC_ID_MPEG2VIDEO, 9 }, //4:2:2 CBP ("Constrained Bytes per Gop") { CODEC_ID_NONE, 0 }, }; typedef struct { int channels; ///< number of audio channels. zero means no audio uint8_t temp[LXF_MAX_AUDIO_PACKET]; ///< temp buffer for de-planarizing the audio data int frame_number; ///< current video frame } LXFDemuxContext; static int lxf_probe(AVProbeData *p) { if (!memcmp(p->buf, LXF_IDENT, LXF_IDENT_LENGTH)) return AVPROBE_SCORE_MAX; return 0; } /** * Verify the checksum of an LXF packet header * * @param[in] header the packet header to check * @return zero if the checksum is OK, non-zero otherwise */ static int check_checksum(const uint8_t *header) { int x; uint32_t sum = 0; for (x = 0; x < LXF_PACKET_HEADER_SIZE; x += 4) sum += AV_RL32(&header[x]); return sum; } /** * Read input until we find the next ident. If found, copy it to the header buffer * * @param[out] header where to copy the ident to * @return 0 if an ident was found, < 0 on I/O error */ static int sync(AVFormatContext *s, uint8_t *header) { uint8_t buf[LXF_IDENT_LENGTH]; int ret; if ((ret = avio_read(s->pb, buf, LXF_IDENT_LENGTH)) != LXF_IDENT_LENGTH) return ret < 0 ? ret : AVERROR_EOF; while (memcmp(buf, LXF_IDENT, LXF_IDENT_LENGTH)) { if (url_feof(s->pb)) return AVERROR_EOF; memmove(buf, &buf[1], LXF_IDENT_LENGTH-1); buf[LXF_IDENT_LENGTH-1] = avio_r8(s->pb); } memcpy(header, LXF_IDENT, LXF_IDENT_LENGTH); return 0; } /** * Read and checksum the next packet header * * @param[out] header the read packet header * @param[out] format context dependent format information * @return the size of the payload following the header or < 0 on failure */ static int get_packet_header(AVFormatContext *s, uint8_t *header, uint32_t *format) { AVIOContext *pb = s->pb; int track_size, samples, ret; AVStream *st; //find and read the ident if ((ret = sync(s, header)) < 0) return ret; //read the rest of the packet header if ((ret = avio_read(pb, header + LXF_IDENT_LENGTH, LXF_PACKET_HEADER_SIZE - LXF_IDENT_LENGTH)) != LXF_PACKET_HEADER_SIZE - LXF_IDENT_LENGTH) { return ret < 0 ? ret : AVERROR_EOF; } if (check_checksum(header)) av_log(s, AV_LOG_ERROR, "checksum error\n"); *format = AV_RL32(&header[32]); ret = AV_RL32(&header[36]); //type switch (AV_RL32(&header[16])) { case 0: //video //skip VBI data and metadata avio_skip(pb, (int64_t)(uint32_t)AV_RL32(&header[44]) + (int64_t)(uint32_t)AV_RL32(&header[52])); break; case 1: //audio if (!(st = s->streams[1])) { av_log(s, AV_LOG_INFO, "got audio packet, but no audio stream present\n"); break; } //set codec based on specified audio bitdepth //we only support tightly packed 16-, 20-, 24- and 32-bit PCM at the moment *format = AV_RL32(&header[40]); st->codec->bits_per_coded_sample = (*format >> 6) & 0x3F; if (st->codec->bits_per_coded_sample != (*format & 0x3F)) { av_log(s, AV_LOG_WARNING, "only tightly packed PCM currently supported\n"); return AVERROR_PATCHWELCOME; } switch (st->codec->bits_per_coded_sample) { case 16: st->codec->codec_id = CODEC_ID_PCM_S16LE; break; case 20: st->codec->codec_id = CODEC_ID_PCM_LXF; break; case 24: st->codec->codec_id = CODEC_ID_PCM_S24LE; break; case 32: st->codec->codec_id = CODEC_ID_PCM_S32LE; break; default: av_log(s, AV_LOG_WARNING, "only 16-, 20-, 24- and 32-bit PCM currently supported\n"); return AVERROR_PATCHWELCOME; } track_size = AV_RL32(&header[48]); samples = track_size * 8 / st->codec->bits_per_coded_sample; //use audio packet size to determine video standard //for NTSC we have one 8008-sample audio frame per five video frames if (samples == LXF_SAMPLERATE * 5005 / 30000) { av_set_pts_info(s->streams[0], 64, 1001, 30000); } else { //assume PAL, but warn if we don't have 1920 samples if (samples != LXF_SAMPLERATE / 25) av_log(s, AV_LOG_WARNING, "video doesn't seem to be PAL or NTSC. guessing PAL\n"); av_set_pts_info(s->streams[0], 64, 1, 25); } //TODO: warning if track mask != (1 << channels) - 1? ret = av_popcount(AV_RL32(&header[44])) * track_size; break; default: break; } return ret; } static int lxf_read_header(AVFormatContext *s, AVFormatParameters *ap) { LXFDemuxContext *lxf = s->priv_data; AVIOContext *pb = s->pb; uint8_t header[LXF_PACKET_HEADER_SIZE], header_data[LXF_HEADER_DATA_SIZE]; int ret; AVStream *st; uint32_t format, video_params, disk_params; uint16_t record_date, expiration_date; if ((ret = get_packet_header(s, header, &format)) < 0) return ret; if (ret != LXF_HEADER_DATA_SIZE) { av_log(s, AV_LOG_ERROR, "expected %d B size header, got %d\n", LXF_HEADER_DATA_SIZE, ret); return AVERROR_INVALIDDATA; } if ((ret = avio_read(pb, header_data, LXF_HEADER_DATA_SIZE)) != LXF_HEADER_DATA_SIZE) return ret < 0 ? ret : AVERROR_EOF; if (!(st = av_new_stream(s, 0))) return AVERROR(ENOMEM); st->duration = AV_RL32(&header_data[32]); video_params = AV_RL32(&header_data[40]); record_date = AV_RL16(&header_data[56]); expiration_date = AV_RL16(&header_data[58]); disk_params = AV_RL32(&header_data[116]); st->codec->codec_type = AVMEDIA_TYPE_VIDEO; st->codec->bit_rate = 1000000 * ((video_params >> 14) & 0xFF); st->codec->codec_tag = video_params & 0xF; st->codec->codec_id = ff_codec_get_id(lxf_tags, st->codec->codec_tag); av_log(s, AV_LOG_DEBUG, "record: %x = %i-%02i-%02i\n", record_date, 1900 + (record_date & 0x7F), (record_date >> 7) & 0xF, (record_date >> 11) & 0x1F); av_log(s, AV_LOG_DEBUG, "expire: %x = %i-%02i-%02i\n", expiration_date, 1900 + (expiration_date & 0x7F), (expiration_date >> 7) & 0xF, (expiration_date >> 11) & 0x1F); if ((video_params >> 22) & 1) av_log(s, AV_LOG_WARNING, "VBI data not yet supported\n"); if ((lxf->channels = (disk_params >> 2) & 0xF)) { if (!(st = av_new_stream(s, 1))) return AVERROR(ENOMEM); st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->sample_rate = LXF_SAMPLERATE; st->codec->channels = lxf->channels; av_set_pts_info(st, 64, 1, st->codec->sample_rate); } if (format == 1) { //skip extended field data avio_skip(s->pb, (uint32_t)AV_RL32(&header[40])); } return 0; } /** * De-planerize the PCM data in lxf->temp * FIXME: remove this once support for planar audio is added to libavcodec * * @param[out] out where to write the de-planerized data to * @param[in] bytes the total size of the PCM data */ static void deplanarize(LXFDemuxContext *lxf, AVStream *ast, uint8_t *out, int bytes) { int x, y, z, i, bytes_per_sample = ast->codec->bits_per_coded_sample >> 3; for (z = i = 0; z < lxf->channels; z++) for (y = 0; y < bytes / bytes_per_sample / lxf->channels; y++) for (x = 0; x < bytes_per_sample; x++, i++) out[x + bytes_per_sample*(z + y*lxf->channels)] = lxf->temp[i]; } static int lxf_read_packet(AVFormatContext *s, AVPacket *pkt) { LXFDemuxContext *lxf = s->priv_data; AVIOContext *pb = s->pb; uint8_t header[LXF_PACKET_HEADER_SIZE], *buf; AVStream *ast = NULL; uint32_t stream, format; int ret, ret2; if ((ret = get_packet_header(s, header, &format)) < 0) return ret; stream = AV_RL32(&header[16]); if (stream > 1) { av_log(s, AV_LOG_WARNING, "got packet with illegal stream index %u\n", stream); return AVERROR(EAGAIN); } if (stream == 1 && !(ast = s->streams[1])) { av_log(s, AV_LOG_ERROR, "got audio packet without having an audio stream\n"); return AVERROR_INVALIDDATA; } //make sure the data fits in the de-planerization buffer if (ast && ret > LXF_MAX_AUDIO_PACKET) { av_log(s, AV_LOG_ERROR, "audio packet too large (%i > %i)\n", ret, LXF_MAX_AUDIO_PACKET); return AVERROR_INVALIDDATA; } if ((ret2 = av_new_packet(pkt, ret)) < 0) return ret2; //read non-20-bit audio data into lxf->temp so we can deplanarize it buf = ast && ast->codec->codec_id != CODEC_ID_PCM_LXF ? lxf->temp : pkt->data; if ((ret2 = avio_read(pb, buf, ret)) != ret) { av_free_packet(pkt); return ret2 < 0 ? ret2 : AVERROR_EOF; } pkt->stream_index = stream; if (ast) { if(ast->codec->codec_id != CODEC_ID_PCM_LXF) deplanarize(lxf, ast, pkt->data, ret); } else { //picture type (0 = closed I, 1 = open I, 2 = P, 3 = B) if (((format >> 22) & 0x3) < 2) pkt->flags |= AV_PKT_FLAG_KEY; pkt->dts = lxf->frame_number++; } return ret; } AVInputFormat ff_lxf_demuxer = { .name = "lxf", .long_name = NULL_IF_CONFIG_SMALL("VR native stream format (LXF)"), .priv_data_size = sizeof(LXFDemuxContext), .read_probe = lxf_probe, .read_header = lxf_read_header, .read_packet = lxf_read_packet, .codec_tag = (const AVCodecTag* const []){lxf_tags, 0}, };