[FFmpeg-devel] [PATCH] aphaser filter

[Clément Bœsch]

On Sat, Mar 30, 2013 at 09:55:29PM +0000, Paul B Mahol wrote:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
>  doc/filters.texi         |  25 +++++
>  libavfilter/Makefile     |   1 +
>  libavfilter/af_aphaser.c | 271 +++++++++++++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c |   1 +
>  4 files changed, 298 insertions(+)
>  create mode 100644 libavfilter/af_aphaser.c
> 
[...]
> +static int query_formats(AVFilterContext *ctx)
> +{
> +    AVFilterFormats *formats;
> +    AVFilterChannelLayouts *layouts;
> +    static const enum AVSampleFormat sample_fmts[] = {
> +        AV_SAMPLE_FMT_DBLP,
> +        AV_SAMPLE_FMT_NONE

The code seems to use float, s16 and s32 but they don't appear here.

[...]
> +static void generate_wave_table(int wave_type, enum AVSampleFormat sample_fmt,
> +                                void *table, int table_size,
> +                                double min, double max, double phase)
> +{
> +    uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;
> +
> +    for (i = 0; i < table_size; i++) {
> +        uint32_t point = (i + phase_offset) % table_size;
> +        double d;
> +
> +        switch (wave_type) {
> +        case WAVE_SINE:
> +            d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;
> +            break;
> +        case WAVE_TRIANGLE:
> +            d = (double)point * 2 / table_size;
> +            switch (4 * point / table_size) {
> +            case 0: d = d + 0.5; break;
> +            case 1:
> +            case 2: d = 1.5 - d; break;
> +            case 3: d = d - 1.5; break;
> +            }
> +            break;
> +        default:
> +            av_assert0(0);
> +        }
> +
> +        d  = d * (max - min) + min;
> +        switch (sample_fmt) {
> +        case AV_SAMPLE_FMT_FLT: {
> +            float *fp = (float *)table;
> +            *fp++ = (float)d;
> +            table = fp;
> +            continue; }
> +        case AV_SAMPLE_FMT_DBL: {
> +            double *dp = (double *)table;
> +            *dp++ = d;
> +            table = dp;
> +            continue; }
> +        }
> +

> +        d += d < 0 ? -0.5 : +0.5;

No compiler will complain on this '+'?

> +        switch (sample_fmt) {
> +        case AV_SAMPLE_FMT_S16: {
> +            int16_t *sp = table;
> +            *sp++ = (int16_t)d;
> +            table = sp;
> +            continue; }
> +        case AV_SAMPLE_FMT_S32: {
> +            int32_t *ip = table;
> +            *ip++ = (int32_t)d;
> +            table = ip;
> +            continue; }
> +        default:
> +            av_assert0(0);
> +        }
> +    }
> +}
> +
[...]
> +static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
> +{
> +    AudioPhaserContext *p = inlink->dst->priv;
> +    AVFilterLink *outlink = inlink->dst->outputs[0];
> +    AVFrame *out_buf;
> +    int i, c, delay_pos, modulation_pos;
> +
> +    if (av_frame_is_writable(buf)) {
> +        out_buf = buf;
> +    } else {
> +        out_buf = ff_get_audio_buffer(inlink, buf->nb_samples);
> +        if (!out_buf)
> +            return AVERROR(ENOMEM);

> +        out_buf->pts = buf->pts;

Please use the copy props function

[...]

-- 
Clément B.
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 490 bytes
Desc: not available
URL: <http://ffmpeg.org/pipermail/ffmpeg-devel/attachments/20130331/a6309341/attachment.asc>