Пример #1
0
/*****************************************************************************
 * DoWork: convert a buffer
 *****************************************************************************/
static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
                    aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
{
    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
    float *p_out = (float *)p_out_buf->p_buffer;

    int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
    int i_in_nb = p_in_buf->i_nb_samples;
    int i_in, i_out = 0;
    unsigned int i_out_rate;
    double d_factor, d_scale_factor, d_old_scale_factor;
    int i_filter_wing;

    if( p_sys->b_filter2 )
        i_out_rate = p_filter->output.i_rate;
    else
        i_out_rate = p_aout->mixer_format.i_rate;

    /* Check if we really need to run the resampler */
    if( i_out_rate == p_filter->input.i_rate )
    {
        if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
            p_sys->i_old_wing &&
            p_in_buf->i_size >=
              p_in_buf->i_nb_bytes + p_sys->i_old_wing *
              p_filter->input.i_bytes_per_frame )
        {
            /* output the whole thing with the samples from last time */
            memmove( ((float *)(p_in_buf->p_buffer)) +
                     i_nb_channels * p_sys->i_old_wing,
                     p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
            memcpy( p_in_buf->p_buffer, p_sys->p_buf +
                    i_nb_channels * p_sys->i_old_wing,
                    p_sys->i_old_wing *
                    p_filter->input.i_bytes_per_frame );

            p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
                p_sys->i_old_wing;

            p_out_buf->start_date = date_Get( &p_sys->end_date );
            p_out_buf->end_date =
                date_Increment( &p_sys->end_date,
                                p_out_buf->i_nb_samples );

            p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
                p_filter->input.i_bytes_per_frame;
        }
        p_filter->b_continuity = false;
        p_sys->i_old_wing = 0;
        return;
    }

    if( !p_filter->b_continuity )
    {
        /* Continuity in sound samples has been broken, we'd better reset
         * everything. */
        p_filter->b_continuity = true;
        p_sys->i_remainder = 0;
        date_Init( &p_sys->end_date, i_out_rate, 1 );
        date_Set( &p_sys->end_date, p_in_buf->start_date );
        p_sys->i_old_rate   = p_filter->input.i_rate;
        p_sys->d_old_factor = 1;
        p_sys->i_old_wing   = 0;
    }

#if 0
    msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
             p_sys->i_old_rate, p_sys->d_old_factor,
             p_sys->i_old_wing, i_in_nb );
#endif

    /* Prepare the source buffer */
    i_in_nb += (p_sys->i_old_wing * 2);

    float p_in_orig[i_in_nb * p_filter->input.i_bytes_per_frame / 4],
         *p_in = p_in_orig;

    /* Copy all our samples in p_in */
    if( p_sys->i_old_wing )
    {
        vlc_memcpy( p_in, p_sys->p_buf,
                    p_sys->i_old_wing * 2 *
                      p_filter->input.i_bytes_per_frame );
    }
    vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
                p_in_buf->p_buffer,
                p_in_buf->i_nb_samples * p_filter->input.i_bytes_per_frame );

    /* Make sure the output buffer is reset */
    memset( p_out, 0, p_out_buf->i_size );

    /* Calculate the new length of the filter wing */
    d_factor = (double)i_out_rate / p_filter->input.i_rate;
    i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;

    /* Account for increased filter gain when using factors less than 1 */
    d_old_scale_factor = SMALL_FILTER_SCALE *
        p_sys->d_old_factor + 0.5;
    d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;

    /* Apply the old rate until we have enough samples for the new one */
    i_in = p_sys->i_old_wing;
    p_in += p_sys->i_old_wing * i_nb_channels;
    for( ; i_in < i_filter_wing &&
           (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
    {
        if( p_sys->d_old_factor == 1 )
        {
            /* Just copy the samples */
            memcpy( p_out, p_in,
                    p_filter->input.i_bytes_per_frame );
            p_in += i_nb_channels;
            p_out += i_nb_channels;
            i_out++;
            continue;
        }

        while( p_sys->i_remainder < p_filter->output.i_rate )
        {

            if( p_sys->d_old_factor >= 1 )
            {
                /* FilterFloatUP() is faster if we can use it */

                /* Perform left-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->output.i_rate,
                               -1, i_nb_channels );
                /* Perform right-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->output.i_rate -
                               p_sys->i_remainder,
                               p_filter->output.i_rate,
                               1, i_nb_channels );

#if 0
                /* Normalize for unity filter gain */
                for( i = 0; i < i_nb_channels; i++ )
                {
                    *(p_out+i) *= d_old_scale_factor;
                }
#endif

                /* Sanity check */
                if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
                    <= (unsigned int)i_out+1 )
                {
                    p_out += i_nb_channels;
                    i_out++;
                    p_sys->i_remainder += p_filter->input.i_rate;
                    break;
                }
            }
            else
            {
                /* Perform left-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->output.i_rate, p_filter->input.i_rate,
                               -1, i_nb_channels );
                /* Perform right-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->output.i_rate -
                               p_sys->i_remainder,
                               p_filter->output.i_rate, p_filter->input.i_rate,
                               1, i_nb_channels );
            }

            p_out += i_nb_channels;
            i_out++;

            p_sys->i_remainder += p_filter->input.i_rate;
        }

        p_in += i_nb_channels;
        p_sys->i_remainder -= p_filter->output.i_rate;
    }

    /* Apply the new rate for the rest of the samples */
    if( i_in < i_in_nb - i_filter_wing )
    {
        p_sys->i_old_rate   = p_filter->input.i_rate;
        p_sys->d_old_factor = d_factor;
        p_sys->i_old_wing   = i_filter_wing;
    }
    for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
    {
        while( p_sys->i_remainder < p_filter->output.i_rate )
        {

            if( d_factor >= 1 )
            {
                /* FilterFloatUP() is faster if we can use it */

                /* Perform left-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->output.i_rate,
                               -1, i_nb_channels );

                /* Perform right-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->output.i_rate -
                               p_sys->i_remainder,
                               p_filter->output.i_rate,
                               1, i_nb_channels );

#if 0
                /* Normalize for unity filter gain */
                for( int i = 0; i < i_nb_channels; i++ )
                {
                    *(p_out+i) *= d_old_scale_factor;
                }
#endif
                /* Sanity check */
                if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
                    <= (unsigned int)i_out+1 )
                {
                    p_out += i_nb_channels;
                    i_out++;
                    p_sys->i_remainder += p_filter->input.i_rate;
                    break;
                }
            }
            else
            {
                /* Perform left-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->output.i_rate, p_filter->input.i_rate,
                               -1, i_nb_channels );
                /* Perform right-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->output.i_rate -
                               p_sys->i_remainder,
                               p_filter->output.i_rate, p_filter->input.i_rate,
                               1, i_nb_channels );
            }

            p_out += i_nb_channels;
            i_out++;

            p_sys->i_remainder += p_filter->input.i_rate;
        }

        p_in += i_nb_channels;
        p_sys->i_remainder -= p_filter->output.i_rate;
    }

    /* Buffer i_filter_wing * 2 samples for next time */
    if( p_sys->i_old_wing )
    {
        memcpy( p_sys->p_buf,
                p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
                i_nb_channels, (2 * p_sys->i_old_wing) *
                p_filter->input.i_bytes_per_frame );
    }

#if 0
    msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
             i_out * p_filter->input.i_bytes_per_frame );
#endif

    /* Finalize aout buffer */
    p_out_buf->i_nb_samples = i_out;
    p_out_buf->start_date = date_Get( &p_sys->end_date );
    p_out_buf->end_date = date_Increment( &p_sys->end_date,
                                          p_out_buf->i_nb_samples );

    p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
        i_nb_channels * sizeof(int32_t);

}
Пример #2
0
static void ResampleFloat( filter_t *p_filter,
                           block_t **pp_out_buf,  size_t *pi_out,
                           float **pp_in,
                           int i_in, int i_in_end,
                           double d_factor, bool b_factor_old,
                           int i_nb_channels, int i_bytes_per_frame )
{
    filter_sys_t *p_sys = p_filter->p_sys;

    float *p_in = *pp_in;
    size_t i_out = *pi_out;
    float *p_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;

    for( ; i_in < i_in_end; i_in++ )
    {
        if( b_factor_old && d_factor == 1 )
        {
            if( ReallocBuffer( pp_out_buf, &p_out,
                               i_out, i_nb_channels, i_bytes_per_frame ) )
                return;
            /* Just copy the samples */
            memcpy( p_out, p_in, i_bytes_per_frame );
            p_in += i_nb_channels;
            p_out += i_nb_channels;
            i_out++;
            continue;
        }

        while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
        {
            if( ReallocBuffer( pp_out_buf, &p_out,
                               i_out, i_nb_channels, i_bytes_per_frame ) )
                return;

            if( d_factor >= 1 )
            {
                /* FilterFloatUP() is faster if we can use it */

                /* Perform left-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate,
                               -1, i_nb_channels );
                /* Perform right-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->fmt_out.audio.i_rate -
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate,
                               1, i_nb_channels );

#if 0
                /* Normalize for unity filter gain */
                for( i = 0; i < i_nb_channels; i++ )
                {
                    *(p_out+i) *= d_old_scale_factor;
                }
#endif
            }
            else
            {
                /* Perform left-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
                               -1, i_nb_channels );
                /* Perform right-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->fmt_out.audio.i_rate -
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
                               1, i_nb_channels );
            }

            p_out += i_nb_channels;
            i_out++;

            p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
        }

        p_in += i_nb_channels;
        p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
    }

    *pp_in  = p_in;
    *pi_out = i_out;
}
Пример #3
0
/*****************************************************************************
 * Resample: convert a buffer
 *****************************************************************************/
static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
{
    if( !p_in_buf || !p_in_buf->i_nb_samples )
    {
        if( p_in_buf )
            block_Release( p_in_buf );
        return NULL;
    }

    filter_sys_t *p_sys = p_filter->p_sys;
    unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
    int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );

    /* Check if we really need to run the resampler */
    if( i_out_rate == p_filter->fmt_in.audio.i_rate )
    {
        if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) &&
            p_sys->i_old_wing )
        {
            /* output the whole thing with the samples from last time */
            p_in_buf = block_Realloc( p_in_buf,
                p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
                p_in_buf->i_buffer );
            if( !p_in_buf )
                return NULL;
            memcpy( p_in_buf->p_buffer, p_sys->p_buf +
                    i_nb_channels * p_sys->i_old_wing,
                    p_sys->i_old_wing *
                    p_filter->fmt_in.audio.i_bytes_per_frame );

            p_in_buf->i_nb_samples += p_sys->i_old_wing;

            p_in_buf->i_pts = date_Get( &p_sys->end_date );
            p_in_buf->i_length =
                date_Increment( &p_sys->end_date,
                                p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
        }
        p_sys->i_old_wing = 0;
        p_sys->b_first = true;
        return p_in_buf;
    }

    unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
                                 p_filter->fmt_out.audio.i_bitspersample / 8;
    size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples *
              p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) )
            + p_filter->p_sys->i_buf_size;
    block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
    if( !p_out_buf )
        return NULL;
    float *p_out = (float *)p_out_buf->p_buffer;

    if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
    {
        /* Continuity in sound samples has been broken, we'd better reset
         * everything. */
        p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
        p_sys->i_remainder = 0;
        date_Init( &p_sys->end_date, i_out_rate, 1 );
        date_Set( &p_sys->end_date, p_in_buf->i_pts );
        p_sys->d_old_factor = 1;
        p_sys->i_old_wing   = 0;
        p_sys->b_first = false;
    }

    int i_in_nb = p_in_buf->i_nb_samples;
    int i_in, i_out = 0;
    double d_factor, d_scale_factor, d_old_scale_factor;
    int i_filter_wing;

#if 0
    msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
             p_sys->i_old_rate, p_sys->d_old_factor,
             p_sys->i_old_wing, i_in_nb );
#endif

    /* Prepare the source buffer */
    i_in_nb += (p_sys->i_old_wing * 2);

    float p_in_orig[i_in_nb * p_filter->fmt_in.audio.i_bytes_per_frame / 4],
         *p_in = p_in_orig;

    /* Copy all our samples in p_in */
    if( p_sys->i_old_wing )
    {
        vlc_memcpy( p_in, p_sys->p_buf,
                    p_sys->i_old_wing * 2 *
                      p_filter->fmt_in.audio.i_bytes_per_frame );
    }
    /* XXX: why i_nb_channels instead of i_bytes_per_frame??? */
    vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
                p_in_buf->p_buffer,
                p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame );
    block_Release( p_in_buf );

    /* Make sure the output buffer is reset */
    memset( p_out, 0, p_out_buf->i_buffer );

    /* Calculate the new length of the filter wing */
    d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
    i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;

    /* Account for increased filter gain when using factors less than 1 */
    d_old_scale_factor = SMALL_FILTER_SCALE *
        p_sys->d_old_factor + 0.5;
    d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;

    /* Apply the old rate until we have enough samples for the new one */
    i_in = p_sys->i_old_wing;
    p_in += p_sys->i_old_wing * i_nb_channels;
    for( ; i_in < i_filter_wing &&
           (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
    {
        if( p_sys->d_old_factor == 1 )
        {
            /* Just copy the samples */
            memcpy( p_out, p_in,
                    p_filter->fmt_in.audio.i_bytes_per_frame );
            p_in += i_nb_channels;
            p_out += i_nb_channels;
            i_out++;
            continue;
        }

        while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
        {

            if( p_sys->d_old_factor >= 1 )
            {
                /* FilterFloatUP() is faster if we can use it */

                /* Perform left-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate,
                               -1, i_nb_channels );
                /* Perform right-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->fmt_out.audio.i_rate -
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate,
                               1, i_nb_channels );

#if 0
                /* Normalize for unity filter gain */
                for( i = 0; i < i_nb_channels; i++ )
                {
                    *(p_out+i) *= d_old_scale_factor;
                }
#endif

                /* Sanity check */
                if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
                    <= (unsigned int)i_out+1 )
                {
                    p_out += i_nb_channels;
                    i_out++;
                    p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
                    break;
                }
            }
            else
            {
                /* Perform left-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
                               -1, i_nb_channels );
                /* Perform right-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->fmt_out.audio.i_rate -
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
                               1, i_nb_channels );
            }

            p_out += i_nb_channels;
            i_out++;

            p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
        }

        p_in += i_nb_channels;
        p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
    }

    /* Apply the new rate for the rest of the samples */
    if( i_in < i_in_nb - i_filter_wing )
    {
        p_sys->d_old_factor = d_factor;
        p_sys->i_old_wing   = i_filter_wing;
    }
    for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
    {
        while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
        {

            if( d_factor >= 1 )
            {
                /* FilterFloatUP() is faster if we can use it */

                /* Perform left-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate,
                               -1, i_nb_channels );

                /* Perform right-wing inner product */
                FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->fmt_out.audio.i_rate -
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate,
                               1, i_nb_channels );

#if 0
                /* Normalize for unity filter gain */
                for( int i = 0; i < i_nb_channels; i++ )
                {
                    *(p_out+i) *= d_old_scale_factor;
                }
#endif
                /* Sanity check */
                if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
                    <= (unsigned int)i_out+1 )
                {
                    p_out += i_nb_channels;
                    i_out++;
                    p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
                    break;
                }
            }
            else
            {
                /* Perform left-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in, p_out,
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
                               -1, i_nb_channels );
                /* Perform right-wing inner product */
                FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
                               SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
                               p_filter->fmt_out.audio.i_rate -
                               p_sys->i_remainder,
                               p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
                               1, i_nb_channels );
            }

            p_out += i_nb_channels;
            i_out++;

            p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
        }

        p_in += i_nb_channels;
        p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
    }

    /* Finalize aout buffer */
    p_out_buf->i_nb_samples = i_out;
    p_out_buf->i_pts = date_Get( &p_sys->end_date );
    p_out_buf->i_length = date_Increment( &p_sys->end_date,
                                  p_out_buf->i_nb_samples ) - p_out_buf->i_pts;

    p_out_buf->i_buffer = p_out_buf->i_nb_samples *
        i_nb_channels * sizeof(int32_t);

    /* Buffer i_filter_wing * 2 samples for next time */
    if( p_sys->i_old_wing )
    {
        size_t newsize = p_sys->i_old_wing * 2
                         * p_filter->fmt_in.audio.i_bytes_per_frame;
        if( newsize > p_sys->i_buf_size )
        {
            free( p_sys->p_buf );
            p_sys->p_buf = malloc( newsize );
            if( p_sys->p_buf != NULL )
                p_sys->i_buf_size = newsize;
            else
            {
                p_sys->i_buf_size = p_sys->i_old_wing = 0; /* oops! */
                return p_out_buf;
            }
        }
        memcpy( p_sys->p_buf,
                p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
                i_nb_channels, (2 * p_sys->i_old_wing) *
                p_filter->fmt_in.audio.i_bytes_per_frame );
    }

#if 0
    msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
             i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
#endif

    return p_out_buf;
}