[FFmpeg-devel] [PATCH] libavfilter: add atempo filter (revised patch v5)
Pavel Koshevoy
pkoshevoy at gmail.com
Wed Jun 13 05:40:53 CEST 2012
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.
Also, is there an ffmpeg utility function for multiplying two vectors of
complex numbers?
Thank you,
Pavel.
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