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https://git.ffmpeg.org/ffmpeg.git
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b800327f4c
This commit does for AVInputFormat what commit59c9dc82f4
did for AVOutputFormat: It adds a new type FFInputFormat, moves all the internals of AVInputFormat to it and adds a now reduced AVInputFormat as first member. This does not affect/improve extensibility of both public or private fields for demuxers (it is still a mess due to lavd). This is possible since50f34172e0
(which removed the last usage of an internal field of AVInputFormat in fftools). (Hint: tools/probetest.c accesses the internals of FFInputFormat as well, but given that it is a testing tool this is not considered a problem.) Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
222 lines
7.0 KiB
C
222 lines
7.0 KiB
C
/*
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* DSD Stream File (DSF) demuxer
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* Copyright (c) 2014 Peter Ross
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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/channel_layout.h"
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "demux.h"
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#include "internal.h"
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#include "id3v2.h"
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typedef struct {
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uint64_t data_end;
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uint64_t audio_size;
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uint64_t data_size;
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} DSFContext;
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static int dsf_probe(const AVProbeData *p)
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{
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if (p->buf_size < 12 || memcmp(p->buf, "DSD ", 4) || AV_RL64(p->buf + 4) != 28)
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return 0;
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return AVPROBE_SCORE_MAX;
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}
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static const AVChannelLayout dsf_channel_layout[] = {
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{ .order = AV_CHANNEL_ORDER_UNSPEC },
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AV_CHANNEL_LAYOUT_MONO,
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AV_CHANNEL_LAYOUT_STEREO,
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AV_CHANNEL_LAYOUT_SURROUND,
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AV_CHANNEL_LAYOUT_QUAD,
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AV_CHANNEL_LAYOUT_4POINT0,
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AV_CHANNEL_LAYOUT_5POINT0_BACK,
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AV_CHANNEL_LAYOUT_5POINT1_BACK,
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};
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static void read_id3(AVFormatContext *s, uint64_t id3pos)
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{
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ID3v2ExtraMeta *id3v2_extra_meta;
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if (avio_seek(s->pb, id3pos, SEEK_SET) < 0)
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return;
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ff_id3v2_read(s, ID3v2_DEFAULT_MAGIC, &id3v2_extra_meta, 0);
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if (id3v2_extra_meta) {
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ff_id3v2_parse_apic(s, id3v2_extra_meta);
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ff_id3v2_parse_chapters(s, id3v2_extra_meta);
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}
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ff_id3v2_free_extra_meta(&id3v2_extra_meta);
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}
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static int dsf_read_header(AVFormatContext *s)
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{
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DSFContext *dsf = s->priv_data;
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AVIOContext *pb = s->pb;
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AVStream *st;
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uint64_t id3pos;
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unsigned int channel_type;
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int channels;
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avio_skip(pb, 4);
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if (avio_rl64(pb) != 28)
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return AVERROR_INVALIDDATA;
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/* create primary stream before any id3 coverart streams */
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st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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avio_skip(pb, 8);
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id3pos = avio_rl64(pb);
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if (pb->seekable & AVIO_SEEKABLE_NORMAL) {
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read_id3(s, id3pos);
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avio_seek(pb, 28, SEEK_SET);
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}
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/* fmt chunk */
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if (avio_rl32(pb) != MKTAG('f', 'm', 't', ' ') || avio_rl64(pb) != 52)
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return AVERROR_INVALIDDATA;
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if (avio_rl32(pb) != 1) {
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avpriv_request_sample(s, "unknown format version");
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return AVERROR_INVALIDDATA;
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}
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if (avio_rl32(pb)) {
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avpriv_request_sample(s, "unknown format id");
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return AVERROR_INVALIDDATA;
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}
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channel_type = avio_rl32(pb);
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if (channel_type < FF_ARRAY_ELEMS(dsf_channel_layout))
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st->codecpar->ch_layout = dsf_channel_layout[channel_type];
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if (!st->codecpar->ch_layout.nb_channels)
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avpriv_request_sample(s, "channel type %i", channel_type);
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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channels = avio_rl32(pb);
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if (!st->codecpar->ch_layout.nb_channels) {
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st->codecpar->ch_layout.nb_channels = channels;
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} else if (channels != st->codecpar->ch_layout.nb_channels) {
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av_log(s, AV_LOG_ERROR, "Channel count mismatch\n");
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return AVERROR(EINVAL);
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}
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st->codecpar->sample_rate = avio_rl32(pb) / 8;
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if (st->codecpar->ch_layout.nb_channels <= 0)
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return AVERROR_INVALIDDATA;
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switch(avio_rl32(pb)) {
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case 1: st->codecpar->codec_id = AV_CODEC_ID_DSD_LSBF_PLANAR; break;
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case 8: st->codecpar->codec_id = AV_CODEC_ID_DSD_MSBF_PLANAR; break;
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default:
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avpriv_request_sample(s, "unknown most significant bit");
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return AVERROR_INVALIDDATA;
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}
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dsf->audio_size = avio_rl64(pb) / 8 * st->codecpar->ch_layout.nb_channels;
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st->codecpar->block_align = avio_rl32(pb);
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if (st->codecpar->block_align > INT_MAX / st->codecpar->ch_layout.nb_channels ||
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st->codecpar->block_align <= 0) {
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avpriv_request_sample(s, "block_align invalid");
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return AVERROR_INVALIDDATA;
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}
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st->codecpar->block_align *= st->codecpar->ch_layout.nb_channels;
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st->codecpar->bit_rate = st->codecpar->ch_layout.nb_channels * 8LL * st->codecpar->sample_rate;
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avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
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avio_skip(pb, 4);
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/* data chunk */
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dsf->data_end = avio_tell(pb);
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if (avio_rl32(pb) != MKTAG('d', 'a', 't', 'a'))
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return AVERROR_INVALIDDATA;
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dsf->data_size = avio_rl64(pb) - 12;
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dsf->data_end += dsf->data_size + 12;
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return 0;
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}
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static int dsf_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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FFFormatContext *const si = ffformatcontext(s);
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DSFContext *dsf = s->priv_data;
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AVIOContext *pb = s->pb;
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AVStream *st = s->streams[0];
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int64_t pos = avio_tell(pb);
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int channels = st->codecpar->ch_layout.nb_channels;
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int ret;
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if (pos >= dsf->data_end)
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return AVERROR_EOF;
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if (dsf->data_size > dsf->audio_size) {
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int last_packet = pos == (dsf->data_end - st->codecpar->block_align);
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if (last_packet) {
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int64_t data_pos = pos - si->data_offset;
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int64_t packet_size = dsf->audio_size - data_pos;
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int64_t skip_size = dsf->data_size - data_pos - packet_size;
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uint8_t *dst;
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int ch, ret;
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if (packet_size <= 0 || skip_size <= 0)
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return AVERROR_INVALIDDATA;
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if ((ret = av_new_packet(pkt, packet_size)) < 0)
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return ret;
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dst = pkt->data;
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for (ch = 0; ch < st->codecpar->ch_layout.nb_channels; ch++) {
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ret = avio_read(pb, dst, packet_size / st->codecpar->ch_layout.nb_channels);
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if (ret < packet_size / st->codecpar->ch_layout.nb_channels)
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return AVERROR_EOF;
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dst += ret;
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avio_skip(pb, skip_size / st->codecpar->ch_layout.nb_channels);
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}
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pkt->pos = pos;
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pkt->stream_index = 0;
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pkt->pts = (pos - si->data_offset) / channels;
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pkt->duration = packet_size / channels;
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return 0;
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}
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}
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ret = av_get_packet(pb, pkt, FFMIN(dsf->data_end - pos, st->codecpar->block_align));
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if (ret < 0)
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return ret;
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pkt->stream_index = 0;
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pkt->pts = (pos - si->data_offset) / channels;
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pkt->duration = st->codecpar->block_align / channels;
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return 0;
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}
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const FFInputFormat ff_dsf_demuxer = {
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.p.name = "dsf",
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.p.long_name = NULL_IF_CONFIG_SMALL("DSD Stream File (DSF)"),
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.p.flags = AVFMT_GENERIC_INDEX | AVFMT_NO_BYTE_SEEK,
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.priv_data_size = sizeof(DSFContext),
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.read_probe = dsf_probe,
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.read_header = dsf_read_header,
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.read_packet = dsf_read_packet,
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};
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