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ffmpeg/libavformat/lxfdec.c

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/*
* 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 "internal.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) {
avpriv_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");
avpriv_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 = avformat_new_stream(s, NULL)))
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 = avformat_new_stream(s, NULL)))
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->sample_rate = LXF_SAMPLERATE;
st->codec->channels = lxf->channels;
avpriv_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},
};
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