mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
272 lines
7.7 KiB
C
272 lines
7.7 KiB
C
/*
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* Limitless Audio Format demuxer
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* Copyright (c) 2022 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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#define MAX_STREAMS 4096
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typedef struct StreamParams {
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AVChannelLayout layout;
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float horizontal;
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float vertical;
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int lfe;
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int stored;
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} StreamParams;
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typedef struct LAFContext {
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uint8_t *data;
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unsigned nb_stored;
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unsigned stored_index;
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unsigned index;
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unsigned bpp;
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StreamParams p[MAX_STREAMS];
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int header_len;
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uint8_t header[(MAX_STREAMS + 7) / 8];
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} LAFContext;
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static int laf_probe(const AVProbeData *p)
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{
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if (memcmp(p->buf, "LIMITLESS", 9))
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return 0;
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if (memcmp(p->buf + 9, "HEAD", 4))
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return 0;
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return AVPROBE_SCORE_MAX;
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}
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static int laf_read_header(AVFormatContext *ctx)
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{
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LAFContext *s = ctx->priv_data;
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AVIOContext *pb = ctx->pb;
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unsigned st_count, mode;
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unsigned sample_rate;
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int64_t duration;
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int codec_id;
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int quality;
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int bpp;
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avio_skip(pb, 9);
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if (avio_rb32(pb) != MKBETAG('H','E','A','D'))
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return AVERROR_INVALIDDATA;
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quality = avio_r8(pb);
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if (quality > 3)
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return AVERROR_INVALIDDATA;
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mode = avio_r8(pb);
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if (mode > 1)
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return AVERROR_INVALIDDATA;
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st_count = avio_rl32(pb);
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if (st_count == 0 || st_count > MAX_STREAMS)
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return AVERROR_INVALIDDATA;
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for (int i = 0; i < st_count; i++) {
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StreamParams *stp = &s->p[i];
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stp->vertical = av_int2float(avio_rl32(pb));
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stp->horizontal = av_int2float(avio_rl32(pb));
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stp->lfe = avio_r8(pb);
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if (stp->lfe) {
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stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_LOW_FREQUENCY));
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} else if (stp->vertical == 0.f &&
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stp->horizontal == 0.f) {
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stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_CENTER));
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} else if (stp->vertical == 0.f &&
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stp->horizontal == -30.f) {
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stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_LEFT));
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} else if (stp->vertical == 0.f &&
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stp->horizontal == 30.f) {
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stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_RIGHT));
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} else if (stp->vertical == 0.f &&
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stp->horizontal == -110.f) {
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stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_SIDE_LEFT));
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} else if (stp->vertical == 0.f &&
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stp->horizontal == 110.f) {
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stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_SIDE_RIGHT));
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} else {
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stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
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}
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}
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sample_rate = avio_rl32(pb);
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duration = avio_rl64(pb) / st_count;
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switch (quality) {
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case 0:
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codec_id = AV_CODEC_ID_PCM_U8;
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bpp = 1;
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break;
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case 1:
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codec_id = AV_CODEC_ID_PCM_S16LE;
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bpp = 2;
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break;
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case 2:
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codec_id = AV_CODEC_ID_PCM_F32LE;
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bpp = 4;
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break;
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case 3:
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codec_id = AV_CODEC_ID_PCM_S24LE;
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bpp = 3;
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break;
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}
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s->index = 0;
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s->stored_index = 0;
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s->bpp = bpp;
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if ((int64_t)bpp * st_count * (int64_t)sample_rate >= INT32_MAX)
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return AVERROR_INVALIDDATA;
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s->data = av_calloc(st_count * sample_rate, bpp);
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if (!s->data)
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return AVERROR(ENOMEM);
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for (int st = 0; st < st_count; st++) {
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StreamParams *stp = &s->p[st];
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AVCodecParameters *par;
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AVStream *st = avformat_new_stream(ctx, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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par = st->codecpar;
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par->codec_id = codec_id;
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par->codec_type = AVMEDIA_TYPE_AUDIO;
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par->ch_layout.nb_channels = 1;
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par->ch_layout = stp->layout;
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par->sample_rate = sample_rate;
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st->duration = duration;
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avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
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}
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s->header_len = (ctx->nb_streams + 7) / 8;
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return 0;
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}
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static int laf_read_packet(AVFormatContext *ctx, AVPacket *pkt)
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{
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AVIOContext *pb = ctx->pb;
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LAFContext *s = ctx->priv_data;
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AVStream *st = ctx->streams[0];
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const int bpp = s->bpp;
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StreamParams *stp;
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int64_t pos;
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int ret;
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pos = avio_tell(pb);
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again:
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if (avio_feof(pb))
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return AVERROR_EOF;
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if (s->index >= ctx->nb_streams) {
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int cur_st = 0, st_count = 0, st_index = 0;
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avio_read(pb, s->header, s->header_len);
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for (int i = 0; i < s->header_len; i++) {
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uint8_t val = s->header[i];
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for (int j = 0; j < 8 && cur_st < ctx->nb_streams; j++, cur_st++) {
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StreamParams *stp = &s->p[st_index];
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stp->stored = 0;
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if (val & 1) {
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stp->stored = 1;
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st_count++;
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}
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val >>= 1;
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st_index++;
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}
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}
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s->index = s->stored_index = 0;
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s->nb_stored = st_count;
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if (!st_count)
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return AVERROR_INVALIDDATA;
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ret = avio_read(pb, s->data, st_count * st->codecpar->sample_rate * bpp);
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if (ret < 0)
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return ret;
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}
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st = ctx->streams[s->index];
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stp = &s->p[s->index];
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while (!stp->stored) {
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s->index++;
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if (s->index >= ctx->nb_streams)
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goto again;
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stp = &s->p[s->index];
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}
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st = ctx->streams[s->index];
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ret = av_new_packet(pkt, st->codecpar->sample_rate * bpp);
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if (ret < 0)
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return ret;
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switch (bpp) {
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case 1:
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for (int n = 0; n < st->codecpar->sample_rate; n++)
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pkt->data[n] = s->data[n * s->nb_stored + s->stored_index];
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break;
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case 2:
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for (int n = 0; n < st->codecpar->sample_rate; n++)
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AV_WN16(pkt->data + n * 2, AV_RN16(s->data + n * s->nb_stored * 2 + s->stored_index * 2));
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break;
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case 3:
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for (int n = 0; n < st->codecpar->sample_rate; n++)
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AV_WL24(pkt->data + n * 3, AV_RL24(s->data + n * s->nb_stored * 3 + s->stored_index * 3));
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break;
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case 4:
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for (int n = 0; n < st->codecpar->sample_rate; n++)
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AV_WN32(pkt->data + n * 4, AV_RN32(s->data + n * s->nb_stored * 4 + s->stored_index * 4));
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break;
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}
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pkt->stream_index = s->index;
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pkt->pos = pos;
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s->index++;
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s->stored_index++;
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return 0;
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}
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static int laf_read_seek(AVFormatContext *ctx, int stream_index,
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int64_t timestamp, int flags)
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{
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LAFContext *s = ctx->priv_data;
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s->stored_index = s->index = s->nb_stored = 0;
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return -1;
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}
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const AVInputFormat ff_laf_demuxer = {
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.name = "laf",
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.long_name = NULL_IF_CONFIG_SMALL("LAF (Limitless Audio Format)"),
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.priv_data_size = sizeof(LAFContext),
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.read_probe = laf_probe,
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.read_header = laf_read_header,
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.read_packet = laf_read_packet,
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.read_seek = laf_read_seek,
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.extensions = "laf",
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.flags = AVFMT_GENERIC_INDEX,
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};
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