On 26.01.2016 23:17, Paul B Mahol wrote:
> ffmpeg | branch: master | Paul B Mahol <one...@gmail.com> | Mon Jan 25
> 21:54:17 2016 +0100| [e9e623369d7ba330ab8157157e956cb71d8058b5] | committer:
> Paul B Mahol
>
> avcodec: add Ulead DV audio decoder
>
> Fixes #1564.
>
> Signed-off-by: Paul B Mahol <one...@gmail.com>
Leaving a little bit more time for review would have been nice.
> diff --git a/libavcodec/dvaudiodec.c b/libavcodec/dvaudiodec.c
> new file mode 100644
> index 0000000..84db509
> --- /dev/null
> +++ b/libavcodec/dvaudiodec.c
> @@ -0,0 +1,144 @@
[...]
> +static int decode_frame(AVCodecContext *avctx, void *data,
> + int *got_frame_ptr, AVPacket *pkt)
> +{
> + DVAudioContext *s = avctx->priv_data;
> + AVFrame *frame = data;
> + const uint8_t *src = pkt->data;
> + int16_t *dst;
> + int ret, i;
> +
> + if (pkt->size != s->block_size)
> + return AVERROR_INVALIDDATA;
> +
> + frame->nb_samples = dv_get_audio_sample_count(pkt->data + 244,
> s->is_pal);
> + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
> + return ret;
> + dst = (int16_t *)frame->data[0];
> +
> + for (i = 0; i < frame->nb_samples; i++) {
Maybe this should be 'i < frame->nb_samples / 2'?
> + const uint8_t *v = &src[s->shuffle[i]];
> +
> + if (s->is_12bit) {
> + *dst++ = dv_audio_12to16((v[0] << 4) | ((v[2] >> 4) & 0x0f));
> + *dst++ = dv_audio_12to16((v[1] << 4) | ((v[2] >> 0) & 0x0f));
> + } else {
> + *dst++ = AV_RB16(&v[0]);
Because otherwise this can cause out of bounds writes.
Best regards,
Andreas
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