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author | Tomas Härdin <tomas.hardin@codemill.se> | 2010-10-01 11:08:24 +0000 |
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committer | Tomas Härdin <tomas.hardin@codemill.se> | 2010-10-01 11:08:24 +0000 |
commit | 8878e3b21b5fe836e636881d11bc7e3a3f9364fd (patch) | |
tree | 6669696d0edb12ebe5a2bc65cab810b5e0bc493f /libavformat/lxfdec.c | |
parent | fbeabfca2f648bd83fa1969c3346d288a158d857 (diff) | |
download | ffmpeg-8878e3b21b5fe836e636881d11bc7e3a3f9364fd.tar.gz |
Add demuxer for LXF (Leitch/Harris' VR native stream format)
Originally committed as revision 25281 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/lxfdec.c')
-rw-r--r-- | libavformat/lxfdec.c | 348 |
1 files changed, 348 insertions, 0 deletions
diff --git a/libavformat/lxfdec.c b/libavformat/lxfdec.c new file mode 100644 index 0000000000..2575f36914 --- /dev/null +++ b/libavformat/lxfdec.c @@ -0,0 +1,348 @@ +/* + * 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 = get_buffer(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] = get_byte(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) +{ + ByteIOContext *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 = get_buffer(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 + url_fskip(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; + ByteIOContext *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 = get_buffer(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_NOMEM; + + 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_NOMEM; + + 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 + url_fskip(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; + ByteIOContext *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 = get_buffer(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 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}, +}; + |