avformat: add MIDI Sample Dump Standard demuxer

Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2017-01-20 17:01:31 +01:00
parent d5d474aea5
commit 7f9978b0bd
6 changed files with 170 additions and 1 deletions

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@ -14,6 +14,7 @@ version <next>:
- Apple Pixlet decoder
- QDMC audio decoder
- NewTek SpeedHQ decoder
- MIDI Sample Dump Standard demuxer
version 3.2:
- libopenmpt demuxer

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@ -387,6 +387,7 @@ library:
@tab Audio format used on the PS3.
@item Mirillis FIC video @tab @tab X
@tab No cursor rendering.
@item MIDI Sample Dump Standard @tab @tab X
@item MIME multipart JPEG @tab X @tab
@item MSN TCP webcam @tab @tab X
@tab Used by MSN Messenger webcam streams.

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@ -436,6 +436,7 @@ OBJS-$(CONFIG_SAP_MUXER) += sapenc.o
OBJS-$(CONFIG_SBG_DEMUXER) += sbgdec.o
OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o
OBJS-$(CONFIG_SDR2_DEMUXER) += sdr2.o
OBJS-$(CONFIG_SDS_DEMUXER) += sdsdec.o
OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o
OBJS-$(CONFIG_SEGMENT_MUXER) += segment.o
OBJS-$(CONFIG_SHORTEN_DEMUXER) += shortendec.o rawdec.o

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@ -275,6 +275,7 @@ void av_register_all(void)
REGISTER_DEMUXER (SBG, sbg);
REGISTER_DEMUXER (SDP, sdp);
REGISTER_DEMUXER (SDR2, sdr2);
REGISTER_DEMUXER (SDS, sds);
#if CONFIG_RTPDEC
ff_register_rtp_dynamic_payload_handlers();
ff_register_rdt_dynamic_payload_handlers();

165
libavformat/sdsdec.c Normal file
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@ -0,0 +1,165 @@
/*
* MIDI Sample Dump Standard format demuxer
* Copyright (c) 2017 Paul B Mahol
*
* 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"
typedef struct SDSContext {
uint8_t data[120];
int bit_depth;
int size;
void (*read_block)(const uint8_t *src, uint32_t *dst);
} SDSContext;
static int sds_probe(AVProbeData *p)
{
if (AV_RB32(p->buf) == 0xF07E0001 && p->buf[20] == 0xF7 &&
p->buf[6] >= 8 && p->buf[6] <= 28)
return AVPROBE_SCORE_EXTENSION;
return 0;
}
static void byte2_read(const uint8_t *src, uint32_t *dst)
{
int i;
for (i = 0; i < 120; i += 2) {
unsigned sample = (src[i + 0] << 25) + (src[i + 1] << 18);
dst[i / 2] = sample;
}
}
static void byte3_read(const uint8_t *src, uint32_t *dst)
{
int i;
for (i = 0; i < 120; i += 3) {
unsigned sample;
sample = (src[i + 0] << 25) | (src[i + 1] << 18) | (src[i + 2] << 11);
dst[i / 3] = sample;
}
}
static void byte4_read(const uint8_t *src, uint32_t *dst)
{
int i;
for (i = 0; i < 120; i += 4) {
unsigned sample;
sample = (src[i + 0] << 25) | (src[i + 1] << 18) | (src[i + 2] << 11) | (src[i + 3] << 4);
dst[i / 4] = sample;
}
}
#define SDS_3BYTE_TO_INT_DECODE(x) (((x) & 0x7F) | (((x) & 0x7F00) >> 1) | (((x) & 0x7F0000) >> 2))
static int sds_read_header(AVFormatContext *ctx)
{
SDSContext *s = ctx->priv_data;
unsigned sample_period;
AVIOContext *pb = ctx->pb;
AVStream *st;
st = avformat_new_stream(ctx, NULL);
if (!st)
return AVERROR(ENOMEM);
avio_skip(pb, 4);
avio_skip(pb, 2);
s->bit_depth = avio_r8(pb);
if (s->bit_depth < 8 || s->bit_depth > 28)
return AVERROR_INVALIDDATA;
if (s->bit_depth < 14) {
s->read_block = byte2_read;
s->size = 60 * 4;
} else if (s->bit_depth < 21) {
s->read_block = byte3_read;
s->size = 40 * 4;
} else {
s->read_block = byte4_read;
s->size = 30 * 4;
}
st->codecpar->codec_id = AV_CODEC_ID_PCM_U32LE;
sample_period = avio_rl24(pb);
sample_period = SDS_3BYTE_TO_INT_DECODE(sample_period);
avio_skip(pb, 11);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->channels = 1;
st->codecpar->sample_rate = sample_period ? 1000000000 / sample_period : 16000;
st->duration = (avio_size(pb) - 21) / (127) * s->size / 4;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
return 0;
}
static int sds_read_packet(AVFormatContext *ctx, AVPacket *pkt)
{
SDSContext *s = ctx->priv_data;
AVIOContext *pb = ctx->pb;
int64_t pos;
int ret;
if (avio_feof(pb))
return AVERROR_EOF;
pos = avio_tell(pb);
if (avio_rb16(pb) != 0xF07E)
return AVERROR_INVALIDDATA;
avio_skip(pb, 3);
ret = av_new_packet(pkt, s->size);
if (ret < 0)
return ret;
ret = avio_read(pb, s->data, 120);
s->read_block(s->data, (uint32_t *)pkt->data);
avio_skip(pb, 1); // checksum
if (avio_r8(pb) != 0xF7)
return AVERROR_INVALIDDATA;
pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
pkt->stream_index = 0;
pkt->pos = pos;
return ret;
}
AVInputFormat ff_sds_demuxer = {
.name = "sds",
.long_name = NULL_IF_CONFIG_SMALL("MIDI Sample Dump Standard"),
.priv_data_size = sizeof(SDSContext),
.read_probe = sds_probe,
.read_header = sds_read_header,
.read_packet = sds_read_packet,
.extensions = "sds",
.flags = AVFMT_GENERIC_INDEX,
};

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@ -32,7 +32,7 @@
// Major bumping may affect Ticket5467, 5421, 5451(compatibility with Chromium)
// Also please add any ticket numbers that you believe might be affected here
#define LIBAVFORMAT_VERSION_MAJOR 57
#define LIBAVFORMAT_VERSION_MINOR 62
#define LIBAVFORMAT_VERSION_MINOR 63
#define LIBAVFORMAT_VERSION_MICRO 100
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \