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Diffstat (limited to 'libavformat/lxfdec.c')
-rw-r--r--libavformat/lxfdec.c61
1 files changed, 15 insertions, 46 deletions
diff --git a/libavformat/lxfdec.c b/libavformat/lxfdec.c
index f29b773d6e..aa31738e0c 100644
--- a/libavformat/lxfdec.c
+++ b/libavformat/lxfdec.c
@@ -2,20 +2,20 @@
* LXF demuxer
* Copyright (c) 2010 Tomas Härdin
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -29,7 +29,6 @@
#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[] = {
{ AV_CODEC_ID_MJPEG, 0 },
@@ -47,7 +46,6 @@ static const AVCodecTag lxf_tags[] = {
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
uint32_t video_format, packet_type, extended_size;
} LXFDemuxContext;
@@ -92,7 +90,7 @@ static int sync(AVFormatContext *s, uint8_t *header)
return ret < 0 ? ret : AVERROR_EOF;
while (memcmp(buf, LXF_IDENT, LXF_IDENT_LENGTH)) {
- if (s->pb->eof_reached)
+ if (url_feof(s->pb))
return AVERROR_EOF;
memmove(buf, &buf[1], LXF_IDENT_LENGTH-1);
@@ -140,8 +138,9 @@ static int get_packet_header(AVFormatContext *s)
}
//read the rest of the packet header
- ret = avio_read(pb, header + (p - header), header_size - (p - header));
- if (ret != header_size - (p - header))
+ if ((ret = avio_read(pb, header + (p - header),
+ header_size - (p - header))) !=
+ header_size - (p - header))
return ret < 0 ? ret : AVERROR_EOF;
if (check_checksum(header, header_size))
@@ -162,7 +161,7 @@ static int get_packet_header(AVFormatContext *s)
break;
case 1:
//audio
- if (!(st = s->streams[1])) {
+ if (!s->streams || !(st = s->streams[1])) {
av_log(s, AV_LOG_INFO, "got audio packet, but no audio stream present\n");
break;
}
@@ -183,10 +182,10 @@ static int get_packet_header(AVFormatContext *s)
}
switch (st->codec->bits_per_coded_sample) {
- case 16: st->codec->codec_id = AV_CODEC_ID_PCM_S16LE; break;
+ case 16: st->codec->codec_id = AV_CODEC_ID_PCM_S16LE_PLANAR; break;
case 20: st->codec->codec_id = AV_CODEC_ID_PCM_LXF; break;
- case 24: st->codec->codec_id = AV_CODEC_ID_PCM_S24LE; break;
- case 32: st->codec->codec_id = AV_CODEC_ID_PCM_S32LE; break;
+ case 24: st->codec->codec_id = AV_CODEC_ID_PCM_S24LE_PLANAR; break;
+ case 32: st->codec->codec_id = AV_CODEC_ID_PCM_S32LE_PLANAR; break;
default:
av_log(s, AV_LOG_WARNING,
"only 16-, 20-, 24- and 32-bit PCM currently supported\n");
@@ -258,6 +257,7 @@ static int lxf_read_header(AVFormatContext *s)
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);
+ st->need_parsing = AVSTREAM_PARSE_HEADERS;
av_log(s, AV_LOG_DEBUG, "record: %x = %i-%02i-%02i\n",
record_date, 1900 + (record_date & 0x7F), (record_date >> 7) & 0xF,
@@ -286,28 +286,10 @@ static int lxf_read_header(AVFormatContext *s)
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 *buf;
AVStream *ast = NULL;
uint32_t stream;
int ret, ret2;
@@ -327,30 +309,17 @@ static int lxf_read_packet(AVFormatContext *s, AVPacket *pkt)
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 != AV_CODEC_ID_PCM_LXF ? lxf->temp : pkt->data;
-
- if ((ret2 = avio_read(pb, buf, ret)) != ret) {
+ if ((ret2 = avio_read(pb, pkt->data, ret)) != ret) {
av_free_packet(pkt);
return ret2 < 0 ? ret2 : AVERROR_EOF;
}
pkt->stream_index = stream;
- if (ast) {
- if(ast->codec->codec_id != AV_CODEC_ID_PCM_LXF)
- deplanarize(lxf, ast, pkt->data, ret);
- } else {
+ if (!ast) {
//picture type (0 = closed I, 1 = open I, 2 = P, 3 = B)
if (((lxf->video_format >> 22) & 0x3) < 2)
pkt->flags |= AV_PKT_FLAG_KEY;
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-07 04:04:30 +0000