Paul B Mahol:
> +static int laf_read_header(AVFormatContext *ctx)
> +{
> + LAFContext *s = ctx->priv_data;
> + AVIOContext *pb = ctx->pb;
> + unsigned st_count, mode;
> + unsigned sample_rate;
> + int64_t duration;
> + int codec_id;
> + int quality;
> + int bpp;
> +
> + avio_skip(pb, 9);
> + if (avio_rb32(pb) != MKBETAG('H','E','A','D'))
> + return AVERROR_INVALIDDATA;
> +
> + quality = avio_r8(pb);
> + if (quality > 3)
> + return AVERROR_INVALIDDATA;
> + mode = avio_r8(pb);
> + if (mode > 1)
> + return AVERROR_INVALIDDATA;
> + st_count = avio_rl32(pb);
> + if (st_count == 0 || st_count > 1024)
I don't know whether the limit of 1024 is arbitrary or something from
some spec. If it is the latter, you should use a #define for it and also
for the size of the StreamParams array in the ctx. If it is the former,
you might just use FF_ARRAY_ELEMS(s->p) instead of 1024 here. Or a
define, as you prefer.
> + return AVERROR_INVALIDDATA;
> +
> + for (int i = 0; i < st_count; i++) {
> + StreamParams *stp = &s->p[i];
> +
> + stp->vertical = av_int2float(avio_rl32(pb));
> + stp->horizontal = av_int2float(avio_rl32(pb));
> + stp->lfe = avio_r8(pb);
> + if (stp->lfe) {
> + stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1,
> (AV_CH_LOW_FREQUENCY));
> + } else if (stp->vertical == 0.f &&
> + stp->horizontal == 0.f) {
> + stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1,
> (AV_CH_FRONT_CENTER));
> + } else if (stp->vertical == 0.f &&
> + stp->horizontal == -30.f) {
> + stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1,
> (AV_CH_FRONT_LEFT));
> + } else if (stp->vertical == 0.f &&
> + stp->horizontal == 30.f) {
> + stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1,
> (AV_CH_FRONT_RIGHT));
> + } else if (stp->vertical == 0.f &&
> + stp->horizontal == -110.f) {
> + stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1,
> (AV_CH_SIDE_LEFT));
> + } else if (stp->vertical == 0.f &&
> + stp->horizontal == 110.f) {
> + stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1,
> (AV_CH_SIDE_RIGHT));
> + } else {
> + stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
> + }
> + }
> +
> + sample_rate = avio_rl32(pb);
> + duration = avio_rl64(pb) / st_count;
> + switch (quality) {
> + case 0:
> + codec_id = AV_CODEC_ID_PCM_U8;
> + bpp = 1;
> + break;
> + case 1:
> + codec_id = AV_CODEC_ID_PCM_S16LE;
> + bpp = 2;
> + break;
> + case 2:
> + codec_id = AV_CODEC_ID_PCM_F32LE;
> + bpp = 4;
> + break;
> + case 3:
> + codec_id = AV_CODEC_ID_PCM_S24LE;
> + bpp = 3;
> + break;
> + }
> +
> + s->index = 0;
> + s->stored_index = 0;
> + s->bpp = bpp;
> + s->data = av_mallocz(st_count * sample_rate * bpp);
sample_rate is read via avio_rl32() and therefore the multiplication on
the right can overflow (it's performed in 32bits, so this can happen
even on 64bit systems). Maybe use av_calloc(sample_rate, st_count *
bpp). But you also need to ensure that sample_rate actually fits into an
int and that st_count * sample_rate * bpp performed in the avio_read()
below also fits into an int, so you should probably just ensure this here.
> + if (!s->data)
> + return AVERROR(ENOMEM);
> +
> + for (int st = 0; st < st_count; st++) {
> + StreamParams *stp = &s->p[st];
> + LAFStream *lafst;
> + AVCodecParameters *par;
> + AVStream *st = avformat_new_stream(ctx, NULL);
> + if (!st)
> + return AVERROR(ENOMEM);
> +
> + par = st->codecpar;
> + par->codec_id = codec_id;
> + par->codec_type = AVMEDIA_TYPE_AUDIO;
> + par->ch_layout.nb_channels = 1;
> + par->ch_layout = stp->layout;
> + par->sample_rate = sample_rate;
> + st->duration = duration;
> + st->priv_data = lafst = av_mallocz(sizeof(LAFStream));
lafst is set-but-unused. And given that you are already imposing a
hardcoded limit on the number of streams you could just add an array of
1024 uint8_t to your context.
> + if (!st->priv_data)
> + return AVERROR(ENOMEM);
> +
> + avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
> + }
> +
> + return 0;
> +}
> +
> +static int laf_read_packet(AVFormatContext *ctx, AVPacket *pkt)
> +{
> + AVIOContext *pb = ctx->pb;
> + LAFContext *s = ctx->priv_data;
> + AVStream *st = ctx->streams[0];
> + LAFStream *lafst = st->priv_data;
> + const int bpp = s->bpp;
> + int header_len = (ctx->nb_streams / 8) + !!(ctx->nb_streams & 7);
(ctx->nb_streams + 7) / 8.
> + int64_t pos;
> + int ret;
> +
> +again:
> + if (avio_feof(pb))
> + return AVERROR_EOF;
> +
> + pos = avio_tell(pb);
> +
> + if (s->index >= ctx->nb_streams) {
> + int cur_st = 0, st_count = 0, st_index = 0;
> +
> + for (int i = 0; i < header_len; i++) {
> + uint8_t val = avio_r8(pb);
Given that you impose a limit of 1024 for the number of streams, you can
actually put an uint8_t [128] on the stack in this loop and read all the
values at once. This would allow to remove the outer loop.
(If you used an array of uint8_t instead of the st->priv_data for
stored, you could also use that array.)
> +
> + for (int j = 0; j < 8 && cur_st < ctx->nb_streams; j++,
> cur_st++) {
> + AVStream *st = ctx->streams[st_index];
> + LAFStream *lafst = st->priv_data;
> +
> + lafst->stored = 0;
> + if (val & 1) {
> + lafst->stored = 1;
> + st_count++;
> + }
> + val >>= 1;
> + st_index++;
> + }
> + }
> +
> + s->index = s->stored_index = 0;
> + s->nb_stored = st_count;
> + if (!st_count)
> + return AVERROR_INVALIDDATA;
> + ret = avio_read(pb, s->data, st_count * st->codecpar->sample_rate *
> bpp);
> + if (ret < 0)
> + return ret;
> + }
> +
> + st = ctx->streams[s->index];
> + lafst = st->priv_data;
> + while (!lafst->stored) {
> + s->index++;
> + if (s->index >= ctx->nb_streams)
> + goto again;
> + lafst = ctx->streams[s->index]->priv_data;
> + }
> + st = ctx->streams[s->index];
> +
> + ret = av_new_packet(pkt, st->codecpar->sample_rate * bpp);
> + if (ret < 0)
> + return ret;
> +
> + for (int n = 0; n < st->codecpar->sample_rate; n++)
> + memcpy(pkt->data + n * bpp, s->data + n * s->nb_stored * bpp +
> s->stored_index * bpp, bpp);
This looks like something that can easily trigger a timeout.
> +
> + pkt->stream_index = s->index;
> + pkt->pos = pos;
If you have data from multiple streams interleaved, then the first
stream will get the position from before reading header_len bytes, but
all the other streams will get the position from after reading the
common data. IMO all packets should get the position of the common data.
> + s->index++;
> + s->stored_index++;
> +
> + return ret;
return 0 -- it is not really defined what happens in case read_packet
callbacks return positive values (it is currently ignored and some
demuxers return the size of the packet, but that is a remnant of an
earlier API) which could happen if av_new_packet() were changed to allow
to return positive values.
> +}
> +
> +static int laf_read_seek(AVFormatContext *ctx, int stream_index,
> + int64_t timestamp, int flags)
> +{
> + LAFContext *s = ctx->priv_data;
> +
> + s->stored_index = s->index = 0;
> +
> + return -1;
> +}
> +
> +const AVInputFormat ff_laf_demuxer = {
> + .name = "laf",
> + .long_name = NULL_IF_CONFIG_SMALL("LAF (Limitless Audio Format)"),
> + .priv_data_size = sizeof(LAFContext),
> + .read_probe = laf_probe,
> + .read_header = laf_read_header,
> + .read_packet = laf_read_packet,
> + .read_seek = laf_read_seek,
> + .extensions = "laf",
> + .flags = AVFMT_GENERIC_INDEX,
> +};
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