[FFmpeg-devel] [PATCH] OSQ lossless audio format support

[Paul B Mahol]

On Thu, Aug 24, 2023 at 11:00 PM James Almer <jamrial at gmail.com> wrote:

> On 8/24/2023 6:52 AM, Paul B Mahol wrote:
> > +static int osq_receive_frame(AVCodecContext *avctx, AVFrame *frame)
> > +{
> > +    OSQContext *s = avctx->priv_data;
> > +    GetBitContext *gb = &s->gb;
> > +    int ret, n;
> > +
> > +    while (s->bitstream_size < s->max_framesize) {
> > +        int size;
> > +
> > +        if (!s->pkt->data) {
> > +            ret = ff_decode_get_packet(avctx, s->pkt);
> > +            if (ret == AVERROR_EOF && s->bitstream_size > 0)
> > +                break;
> > +            if (ret < 0)
> > +                return ret;
> > +        }
> > +
> > +        size = FFMIN(s->pkt->size - s->pkt_offset, s->max_framesize -
> s->bitstream_size);
> > +        memcpy(s->bitstream + s->bitstream_size, s->pkt->data +
> s->pkt_offset, size);
> > +        s->bitstream_size += size;
> > +        s->pkt_offset += size;
> > +
> > +        if (s->pkt_offset == s->pkt->size) {
> > +            av_packet_unref(s->pkt);
> > +            s->pkt_offset = 0;
> > +        }
>
> This looks like you're assembling a packet of max_framesize bytes. You
> should instead do that in a parser, and ensure here that the packets fed
> to this decoder are <= max_framesize.
>
> > +    }
> > +
> > +    frame->nb_samples = FFMIN(s->frame_samples, s->nb_samples);
> > +    if (frame->nb_samples <= 0)
> > +        return AVERROR_EOF;
> > +
> > +    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
> > +        goto fail;
> > +
> > +    if ((ret = init_get_bits8(gb, s->bitstream, s->bitstream_size)) < 0)
> > +        goto fail;
> > +
> > +    if ((ret = osq_decode_block(avctx, frame)) < 0)
> > +        goto fail;
> > +
> > +    s->nb_samples -= frame->nb_samples;
> > +
> > +    n = get_bits_count(gb) / 8;
> > +    if (n > s->bitstream_size) {
> > +        ret = AVERROR_INVALIDDATA;
> > +        goto fail;
> > +    }
> > +
> > +    memmove(s->bitstream, &s->bitstream[n], s->bitstream_size - n);
> > +    s->bitstream_size -= n;
> > +
> > +    return 0;
> > +
> > +fail:
> > +    s->bitstream_size = 0;
> > +    s->pkt_offset = 0;
> > +    av_packet_unref(s->pkt);
> > +
> > +    return ret;
> > +}
> > +
> > +const AVInputFormat ff_osq_demuxer = {
> > +    .name           = "osq",
> > +    .long_name      = NULL_IF_CONFIG_SMALL("raw OSQ"),
> > +    .read_probe     = osq_probe,
> > +    .read_header    = osq_read_header,
> > +    .read_packet    = ff_raw_read_partial_packet,
>
> Instead of sending an arbitrarily sized packet (1024 bytes as of now),
> you should set codecpar->frame_size and propagate packets with that
> amount of bytes instead.
> A parser is still needed, though, for non seekable input (a pipe). And
> in case the decoder is fed with non lavf input.
>

Format is not seekable, packet sizes are nowhere stored in .osq files.

Think of this format like APAC/BONK/WAVARC but just no need to keep unused
bits from previous decoded data/packet.
With this fact, parser makes no sense to do.


> > +    .extensions     = "osq",
> > +    .flags          = AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH |
> AVFMT_NO_BYTE_SEEK | AVFMT_NOTIMESTAMPS,
> > +    .raw_codec_id   = AV_CODEC_ID_OSQ,
> > +    .priv_data_size = sizeof(FFRawDemuxerContext),
> > +    .priv_class     = &ff_raw_demuxer_class,
> > +};
>
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