[FFmpeg-devel] [PATCH] aphaser filter

[Paul B Mahol]

On 3/31/13, Clement Boesch <ubitux at gmail.com> wrote:
> 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.

Because they are unrelated to table generation.
I will try to add support for other sample formats but can't promise anything.

>
> [...]
>> +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 '+'?

Changed.

>
>> +        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

Done.

>
> [...]
>
> --
> Clement B.
>