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

[Pavel Koshevoy]

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.