[FFmpeg-devel] [PATCH] libavfilter: add atempo filter (revised patch v5)
Michael Niedermayer
michaelni at gmx.at
Wed Jun 13 11:41:47 CEST 2012
On Tue, Jun 12, 2012 at 09:40:53PM -0600, Pavel Koshevoy wrote:
> On 6/12/2012 8:34 PM, Michael Niedermayer wrote:
> >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
> >
>
> I need to know how the coefficients are stored after R2C transform.
> Is it a simple array of (N/2 + 1) complex numbers (or N + 2
> scalars), or is it a bit more complicated -- Y[0] and Y[N/2] are
> purely real and therefore their imaginary component is not stored,
> thus requiring only N scalar values for storage. That's how FFTW
> does it.
>
> It matters because I need to calculate a product of complex numbers,
> therefore I need to know where each Re and Im component is stored.
I think we put the Y[N/2] in Y[0].im but see the code to make sure
>
> Also, is there an ffmpeg utility function for multiplying two
> vectors of complex numbers?
not that i remember
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
What does censorship reveal? It reveals fear. -- Julian Assange
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