[FFmpeg-devel] [PATCH] libavfilter: add atempo filter (revised patch v5)

[Michael Niedermayer]

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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