[FFmpeg-soc] [Patch] Maxis EA XA decoder - GSoC Task
Michael Niedermayer
michaelni at gmx.at
Fri Apr 11 16:49:25 CEST 2008
On Fri, Apr 11, 2008 at 03:41:48PM +0200, Robert Marston wrote:
[...]
> > > > > @@ -1235,6 +1243,29 @@
> > > >
> > > > > }
> > > > > }
> > > > > break;
> > > > > + case CODEC_ID_ADPCM_EA_MAXIS_XA:
> > > > > + for(channel = 0; channel < avctx->channels; channel++) {
> > > > > + for (i=0; i<2; i++)
> > > > > + coeff[channel][i] = ea_adpcm_table[(*src >> 4) +(4*i)];
> > > >
> > > > > + shift[channel] = (*src & 0x0F) + 8;
> > > > > + src++;
> > > > > + }
> > > >
> > > > > + for (count1 = 0; count1 < ((buf_size - avctx->channels) / avctx->channels) ; count1++) {
> > > > > + for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL (mono) */
> > > > > + int32_t sample;
> > > >
> > > >
> > > > > + for(channel = 0; channel < avctx->channels; channel++) {
> > > >
> > > > > + sample = ((((*(src+channel) >> i) & 0x0F) << 0x1C) >> shift[channel]);
> > > >
> > > > This looks buggy.
> > > >
> > >
> > > Where do you think the error would occur?
> >
> > on some non x86 hardware
> >
> >
>
> What exactly you referring to here? The shift operators? A problem
> with the Endianess maybe?
Its related to the shifts.
>
> >
> > >
> > > >
> > > > [...]
> > > > > +static int xa_read_packet(AVFormatContext *s,
> > > > > + AVPacket *pkt)
> > > > > +{
> > > > > + MaxisXADemuxContext *xa = s->priv_data;
> > > > > + AVStream *st = s->streams[0];
> > > > > + ByteIOContext *pb = s->pb;
> > > > > + unsigned int packet_size;
> > > > > + int ret = 0;
> > > > > +
> > > > > + if(xa->sent_bytes > xa->out_size)
> > > > > + return AVERROR(EIO);
> > > > > + /* 1 byte header and 14 bytes worth of samples * number channels per block */
> > > > > + packet_size = 15*st->codec->channels;
> > > > > +
> > > > > + ret = av_get_packet(pb, pkt, packet_size);
> > > > > + pkt->stream_index = st->index;
> > > > > +
> > > > > + xa->sent_bytes += packet_size;
> > > >
> > > >
> > > > > + pkt->pts = 90000;
> > > > > + pkt->pts *= xa->audio_frame_counter;
> > > > > + pkt->pts /= st->codec->sample_rate;
> > > > > + /* 14 Samples per channel */
> > > > > + xa->audio_frame_counter += 14;
> > > >
> > > > Still wrong
> > > >
> > > > [...]
> > >
> > > Corrected?
> >
> > no
> >
>
> Am I right in saying the pts should be incremented by 28 *
> 90K/(1/sample_rate) * 90K ... assuming we use a 90 KHz clock? and 28
> being the number of samples per channel.
no
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
The educated differ from the uneducated as much as the living from the
dead. -- Aristotle
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