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avformat: add MIDI Sample Dump Standard demuxer
Signed-off-by: Paul B Mahol <onemda@gmail.com>
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@ -14,6 +14,7 @@ version <next>:
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- Apple Pixlet decoder
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- QDMC audio decoder
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- NewTek SpeedHQ decoder
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- MIDI Sample Dump Standard demuxer
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version 3.2:
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- libopenmpt demuxer
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@ -387,6 +387,7 @@ library:
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@tab Audio format used on the PS3.
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@item Mirillis FIC video @tab @tab X
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@tab No cursor rendering.
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@item MIDI Sample Dump Standard @tab @tab X
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@item MIME multipart JPEG @tab X @tab
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@item MSN TCP webcam @tab @tab X
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@tab Used by MSN Messenger webcam streams.
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@ -436,6 +436,7 @@ OBJS-$(CONFIG_SAP_MUXER) += sapenc.o
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OBJS-$(CONFIG_SBG_DEMUXER) += sbgdec.o
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OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o
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OBJS-$(CONFIG_SDR2_DEMUXER) += sdr2.o
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OBJS-$(CONFIG_SDS_DEMUXER) += sdsdec.o
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OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o
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OBJS-$(CONFIG_SEGMENT_MUXER) += segment.o
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OBJS-$(CONFIG_SHORTEN_DEMUXER) += shortendec.o rawdec.o
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@ -275,6 +275,7 @@ void av_register_all(void)
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REGISTER_DEMUXER (SBG, sbg);
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REGISTER_DEMUXER (SDP, sdp);
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REGISTER_DEMUXER (SDR2, sdr2);
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REGISTER_DEMUXER (SDS, sds);
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#if CONFIG_RTPDEC
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ff_register_rtp_dynamic_payload_handlers();
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ff_register_rdt_dynamic_payload_handlers();
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165
libavformat/sdsdec.c
Normal file
165
libavformat/sdsdec.c
Normal file
@ -0,0 +1,165 @@
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/*
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* MIDI Sample Dump Standard format demuxer
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* Copyright (c) 2017 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "internal.h"
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typedef struct SDSContext {
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uint8_t data[120];
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int bit_depth;
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int size;
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void (*read_block)(const uint8_t *src, uint32_t *dst);
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} SDSContext;
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static int sds_probe(AVProbeData *p)
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{
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if (AV_RB32(p->buf) == 0xF07E0001 && p->buf[20] == 0xF7 &&
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p->buf[6] >= 8 && p->buf[6] <= 28)
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return AVPROBE_SCORE_EXTENSION;
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return 0;
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}
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static void byte2_read(const uint8_t *src, uint32_t *dst)
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{
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int i;
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for (i = 0; i < 120; i += 2) {
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unsigned sample = (src[i + 0] << 25) + (src[i + 1] << 18);
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dst[i / 2] = sample;
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}
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}
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static void byte3_read(const uint8_t *src, uint32_t *dst)
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{
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int i;
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for (i = 0; i < 120; i += 3) {
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unsigned sample;
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sample = (src[i + 0] << 25) | (src[i + 1] << 18) | (src[i + 2] << 11);
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dst[i / 3] = sample;
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}
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}
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static void byte4_read(const uint8_t *src, uint32_t *dst)
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{
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int i;
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for (i = 0; i < 120; i += 4) {
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unsigned sample;
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sample = (src[i + 0] << 25) | (src[i + 1] << 18) | (src[i + 2] << 11) | (src[i + 3] << 4);
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dst[i / 4] = sample;
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}
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}
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#define SDS_3BYTE_TO_INT_DECODE(x) (((x) & 0x7F) | (((x) & 0x7F00) >> 1) | (((x) & 0x7F0000) >> 2))
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static int sds_read_header(AVFormatContext *ctx)
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{
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SDSContext *s = ctx->priv_data;
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unsigned sample_period;
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AVIOContext *pb = ctx->pb;
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AVStream *st;
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st = avformat_new_stream(ctx, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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avio_skip(pb, 4);
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avio_skip(pb, 2);
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s->bit_depth = avio_r8(pb);
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if (s->bit_depth < 8 || s->bit_depth > 28)
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return AVERROR_INVALIDDATA;
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if (s->bit_depth < 14) {
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s->read_block = byte2_read;
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s->size = 60 * 4;
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} else if (s->bit_depth < 21) {
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s->read_block = byte3_read;
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s->size = 40 * 4;
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} else {
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s->read_block = byte4_read;
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s->size = 30 * 4;
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}
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st->codecpar->codec_id = AV_CODEC_ID_PCM_U32LE;
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sample_period = avio_rl24(pb);
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sample_period = SDS_3BYTE_TO_INT_DECODE(sample_period);
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avio_skip(pb, 11);
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->channels = 1;
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st->codecpar->sample_rate = sample_period ? 1000000000 / sample_period : 16000;
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st->duration = (avio_size(pb) - 21) / (127) * s->size / 4;
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avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
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return 0;
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}
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static int sds_read_packet(AVFormatContext *ctx, AVPacket *pkt)
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{
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SDSContext *s = ctx->priv_data;
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AVIOContext *pb = ctx->pb;
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int64_t pos;
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int ret;
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if (avio_feof(pb))
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return AVERROR_EOF;
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pos = avio_tell(pb);
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if (avio_rb16(pb) != 0xF07E)
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return AVERROR_INVALIDDATA;
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avio_skip(pb, 3);
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ret = av_new_packet(pkt, s->size);
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if (ret < 0)
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return ret;
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ret = avio_read(pb, s->data, 120);
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s->read_block(s->data, (uint32_t *)pkt->data);
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avio_skip(pb, 1); // checksum
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if (avio_r8(pb) != 0xF7)
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return AVERROR_INVALIDDATA;
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pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
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pkt->stream_index = 0;
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pkt->pos = pos;
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return ret;
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}
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AVInputFormat ff_sds_demuxer = {
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.name = "sds",
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.long_name = NULL_IF_CONFIG_SMALL("MIDI Sample Dump Standard"),
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.priv_data_size = sizeof(SDSContext),
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.read_probe = sds_probe,
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.read_header = sds_read_header,
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.read_packet = sds_read_packet,
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.extensions = "sds",
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.flags = AVFMT_GENERIC_INDEX,
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};
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@ -32,7 +32,7 @@
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// Major bumping may affect Ticket5467, 5421, 5451(compatibility with Chromium)
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// Also please add any ticket numbers that you believe might be affected here
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#define LIBAVFORMAT_VERSION_MAJOR 57
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#define LIBAVFORMAT_VERSION_MINOR 62
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#define LIBAVFORMAT_VERSION_MINOR 63
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#define LIBAVFORMAT_VERSION_MICRO 100
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#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
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