[FFmpeg-devel] [PATCH] libavfilter: add atempo filter (revised patch v5)
Michael Niedermayer
michaelni at gmx.at
Wed Jun 13 04:34:11 CEST 2012
On Tue, Jun 12, 2012 at 07:19:54PM -0600, Pavel Koshevoy wrote:
> On 06/12/2012 05:07 PM, Michael Niedermayer wrote:
> >On Mon, Jun 11, 2012 at 09:18:02PM -0600, Pavel Koshevoy wrote:
>
> [...]
>
> >>+/**
> >>+ * Prepare filter for processing audio data of given format,
> >>+ * sample rate and number of channels.
> >>+ */
> >>+static int yae_reset(ATempoContext *atempo,
> >>+ enum AVSampleFormat format,
> >>+ int sample_rate,
> >>+ int channels)
> >>+{
> >>+ const int sample_size = av_get_bytes_per_sample(format);
> >>+ uint32_t nlevels = 0;
> >>+ uint32_t pot;
> >>+ int i;
> >>+
> >>+ atempo->format = format;
> >>+ atempo->channels = channels;
> >>+ atempo->stride = sample_size * channels;
> >>+
> >>+ // pick a segment window size:
> >>+ atempo->window = sample_rate / 24;
> >>+
> >>+ // adjust window size to be a power-of-two integer:
> >>+ nlevels = av_log2(atempo->window);
> >>+ pot = 1<< nlevels;
> >>+ av_assert0(pot<= atempo->window);
> >>+
> >>+ if (pot< atempo->window) {
> >>+ atempo->window = pot * 2;
> >>+ nlevels++;
> >>+ }
> >>+
> >>+ // initialize audio fragment buffers:
> >>+ REALLOC_OR_FAIL(atempo->frag[0].data,
> >>+ atempo->window * atempo->stride);
> >>+
> >>+ REALLOC_OR_FAIL(atempo->frag[1].data,
> >>+ atempo->window * atempo->stride);
> >>+
> >>+ REALLOC_OR_FAIL(atempo->frag[0].xdat,
> >>+ atempo->window * 2 * sizeof(FFTComplex));
> >>+
> >>+ REALLOC_OR_FAIL(atempo->frag[1].xdat,
> >>+ atempo->window * 2 * sizeof(FFTComplex));
> >>+
> >>+ // initialize FFT contexts:
> >>+ av_fft_end(atempo->fft_forward);
> >>+ av_fft_end(atempo->fft_inverse);
> >maybe this should call uninit()
> >
> >also if something in this fails then the resuslting state is a mess
> >some arrays one size some others another some memleaks and the ffts
> >might even end with a double free i think
>
> I'll look into that, thank you.
>
> >
> >
> >>+
> >>+ atempo->fft_forward = av_fft_init(nlevels + 1, 0);
> >>+ if (!atempo->fft_forward) {
> >>+ return AVERROR(ENOMEM);
> >>+ }
> >>+
> >>+ atempo->fft_inverse = av_fft_init(nlevels + 1, 1);
> >>+ if (!atempo->fft_inverse) {
> >>+ return AVERROR(ENOMEM);
> >>+ }
> >by using a RDFT you can cut the computations needed down by a factor
> >of 2
>
> My DSP experience is limited. What is the procedure for computing
> cross correlation of two signals using rDFT?
basically same procedure, its just half the real scalar values the
fft deals with thus 2x as fast
[...]
> >
> >ill leave further review& applying to stefano
> >
> >if you want to maintain this filter once its
> >in git, then you should get a public git clone of ffmpeg (for example
> >on github) so you can conveniently maintain it. And once you have
> >some changes in your git repository with which you are happy just
> >ask me to merge them ...
> >
>
> I am not sure if it's really worth it for just one file. Why
> shouldn't I keep a private branch as I do now and simply send a
> patch when I am ready?
whichever way you prefer ...
but i think git is more convenient, easier to collaborate with others,
easier to update than a patch, and easier for random users to checkout
and try/test
having a git clone doesnt mean you cant send a patch ...
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
I have often repented speaking, but never of holding my tongue.
-- Xenocrates
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