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-rw-r--r--libavformat/dss.c77
1 files changed, 66 insertions, 11 deletions
diff --git a/libavformat/dss.c b/libavformat/dss.c
index f7d0ead1c2..ead0ee00dc 100644
--- a/libavformat/dss.c
+++ b/libavformat/dss.c
@@ -2,20 +2,20 @@
* Digital Speech Standard (DSS) demuxer
* Copyright (c) 2014 Oleksij Rempel <linux@rempel-privat.de>
*
- * 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
*/
@@ -54,6 +54,8 @@ typedef struct DSSDemuxContext {
int swap;
int dss_sp_swap_byte;
int8_t *dss_sp_buf;
+
+ int packet_size;
} DSSDemuxContext;
static int dss_probe(AVProbeData *p)
@@ -78,7 +80,8 @@ static int dss_read_metadata_date(AVFormatContext *s, unsigned int offset,
if (ret < DSS_TIME_SIZE)
return ret < 0 ? ret : AVERROR_EOF;
- sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec);
+ if (sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec) != 6)
+ return AVERROR_INVALIDDATA;
/* We deal with a two-digit year here, so set the default date to 2000
* and hope it will never be used in the next century. */
snprintf(datetime, sizeof(datetime), "%.4d-%.2d-%.2dT%.2d:%.2d:%.2d",
@@ -142,7 +145,7 @@ static int dss_read_header(AVFormatContext *s)
if (ctx->audio_codec == DSS_ACODEC_DSS_SP) {
st->codec->codec_id = AV_CODEC_ID_DSS_SP;
- st->codec->sample_rate = 12000;
+ st->codec->sample_rate = 11025;
} else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
st->codec->codec_id = AV_CODEC_ID_G723_1;
st->codec->sample_rate = 8000;
@@ -209,13 +212,13 @@ static void dss_sp_byte_swap(DSSDemuxContext *ctx,
static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
{
DSSDemuxContext *ctx = s->priv_data;
+ AVStream *st = s->streams[0];
int read_size, ret, offset = 0, buff_offset = 0;
+ int64_t pos = avio_tell(s->pb);
if (ctx->counter == 0)
dss_skip_audio_header(s, pkt);
- pkt->pos = avio_tell(s->pb);
-
if (ctx->swap) {
read_size = DSS_FRAME_SIZE - 2;
buff_offset = 3;
@@ -223,13 +226,16 @@ static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
read_size = DSS_FRAME_SIZE;
ctx->counter -= read_size;
+ ctx->packet_size = DSS_FRAME_SIZE - 1;
ret = av_new_packet(pkt, DSS_FRAME_SIZE);
if (ret < 0)
return ret;
- pkt->duration = 0;
+ pkt->duration = 264;
+ pkt->pos = pos;
pkt->stream_index = 0;
+ s->bit_rate = 8LL * ctx->packet_size * st->codec->sample_rate * 512 / (506 * pkt->duration);
if (ctx->counter < 0) {
int size2 = ctx->counter + read_size;
@@ -250,6 +256,11 @@ static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
dss_sp_byte_swap(ctx, pkt->data, ctx->dss_sp_buf);
+ if (ctx->dss_sp_swap_byte < 0) {
+ ret = AVERROR(EAGAIN);
+ goto error_eof;
+ }
+
if (pkt->data[0] == 0xff)
return AVERROR_INVALIDDATA;
@@ -263,12 +274,13 @@ error_eof:
static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
{
DSSDemuxContext *ctx = s->priv_data;
+ AVStream *st = s->streams[0];
int size, byte, ret, offset;
+ int64_t pos = avio_tell(s->pb);
if (ctx->counter == 0)
dss_skip_audio_header(s, pkt);
- pkt->pos = avio_tell(s->pb);
/* We make one byte-step here. Don't forget to add offset. */
byte = avio_r8(s->pb);
if (byte == 0xff)
@@ -276,15 +288,18 @@ static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
size = frame_size[byte & 3];
+ ctx->packet_size = size;
ctx->counter -= size;
ret = av_new_packet(pkt, size);
if (ret < 0)
return ret;
+ pkt->pos = pos;
pkt->data[0] = byte;
offset = 1;
pkt->duration = 240;
+ s->bit_rate = 8LL * size * st->codec->sample_rate * 512 / (506 * pkt->duration);
pkt->stream_index = 0;
@@ -325,11 +340,50 @@ static int dss_read_close(AVFormatContext *s)
{
DSSDemuxContext *ctx = s->priv_data;
- av_free(ctx->dss_sp_buf);
+ av_freep(&ctx->dss_sp_buf);
+
+ return 0;
+}
+
+static int dss_read_seek(AVFormatContext *s, int stream_index,
+ int64_t timestamp, int flags)
+{
+ DSSDemuxContext *ctx = s->priv_data;
+ int64_t ret, seekto;
+ uint8_t header[DSS_AUDIO_BLOCK_HEADER_SIZE];
+ int offset;
+
+ if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
+ seekto = timestamp / 264 * 41 / 506 * 512;
+ else
+ seekto = timestamp / 240 * ctx->packet_size / 506 * 512;
+
+ if (seekto < 0)
+ seekto = 0;
+
+ seekto += DSS_HEADER_SIZE;
+
+ ret = avio_seek(s->pb, seekto, SEEK_SET);
+ if (ret < 0)
+ return ret;
+ avio_read(s->pb, header, DSS_AUDIO_BLOCK_HEADER_SIZE);
+ ctx->swap = !!(header[0] & 0x80);
+ offset = 2*header[1] + 2*ctx->swap;
+ if (offset < DSS_AUDIO_BLOCK_HEADER_SIZE)
+ return AVERROR_INVALIDDATA;
+ if (offset == DSS_AUDIO_BLOCK_HEADER_SIZE) {
+ ctx->counter = 0;
+ offset = avio_skip(s->pb, -DSS_AUDIO_BLOCK_HEADER_SIZE);
+ } else {
+ ctx->counter = DSS_BLOCK_SIZE - offset;
+ offset = avio_skip(s->pb, offset - DSS_AUDIO_BLOCK_HEADER_SIZE);
+ }
+ ctx->dss_sp_swap_byte = -1;
return 0;
}
+
AVInputFormat ff_dss_demuxer = {
.name = "dss",
.long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"),
@@ -338,5 +392,6 @@ AVInputFormat ff_dss_demuxer = {
.read_header = dss_read_header,
.read_packet = dss_read_packet,
.read_close = dss_read_close,
+ .read_seek = dss_read_seek,
.extensions = "dss"
};
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-12 03:48:54 +0000