mono.c

VLC Player Source Code

}

/*****************************************************************************
 * OpenFilter
 *****************************************************************************/
static int OpenFilter( vlc_object_t *p_this )
{
    filter_t * p_filter = (filter_t *)p_this;
    filter_sys_t *p_sys = NULL;

    if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
    {
        /*msg_Dbg( p_filter, "filter discarded (incompatible format)" );*/
        return VLC_EGENERIC;
    }

    if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
        (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
    {
        /*msg_Err( p_this, "filter discarded (invalid format)" );*/
        return VLC_EGENERIC;
    }

    if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
        (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
        (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
        (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
        (p_filter->fmt_in.audio.i_bitspersample !=
                                    p_filter->fmt_out.audio.i_bitspersample))
    {
        /*msg_Err( p_this, "couldn't load mono filter" );*/
        return VLC_EGENERIC;
    }

    /* Allocate the memory needed to store the module's structure */
    p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
    if( p_sys == NULL )
        return VLC_EGENERIC;

    var_Create( p_this, MONO_CFG "downmix",
                VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
    p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );

    var_Create( p_this, MONO_CFG "channel",
                VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
    p_sys->i_channel_selected =
            (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );

    if( p_sys->b_downmix )
    {
        msg_Dbg( p_this, "using stereo to mono downmix" );
        p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
        p_filter->fmt_out.audio.i_channels = 1;
    }
    else
    {
        msg_Dbg( p_this, "using pseudo mono" );
        p_filter->fmt_out.audio.i_physical_channels =
                            (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
        p_filter->fmt_out.audio.i_channels = 2;
    }

    p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
    p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;

    p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
    p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;

    p_sys->i_overflow_buffer_size = 0;
    p_sys->p_overflow_buffer = NULL;
    p_sys->i_nb_atomic_operations = 0;
    p_sys->p_atomic_operations = NULL;

    if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
              aout_FormatNbChannels( &p_filter->fmt_in.audio ),
              p_filter->fmt_in.audio.i_physical_channels,
              p_filter->fmt_in.audio.i_rate ) < 0 )
    {
        var_Destroy( p_this, MONO_CFG "channel" );
        var_Destroy( p_this, MONO_CFG "downmix" );
        free( p_sys );
        return VLC_EGENERIC;
    }

    p_filter->pf_audio_filter = Convert;

    msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
             (char *)&p_filter->fmt_in.i_codec,
             (char *)&p_filter->fmt_out.i_codec,
             p_filter->fmt_in.audio.i_physical_channels,
             p_filter->fmt_out.audio.i_physical_channels,
             p_filter->fmt_in.audio.i_bitspersample,
             p_filter->fmt_out.audio.i_bitspersample );

    return VLC_SUCCESS;
}

/*****************************************************************************
 * CloseFilter
 *****************************************************************************/
static void CloseFilter( vlc_object_t *p_this)
{
    filter_t *p_filter = (filter_t *) p_this;
    filter_sys_t *p_sys = p_filter->p_sys;

    var_Destroy( p_this, MONO_CFG "channel" );
    var_Destroy( p_this, MONO_CFG "downmix" );
    free( p_sys->p_atomic_operations );
    free( p_sys->p_overflow_buffer );
    free( p_sys );
}

/*****************************************************************************
 * Convert
 *****************************************************************************/
static block_t *Convert( filter_t *p_filter, block_t *p_block )
{
    aout_filter_t aout_filter;
    aout_buffer_t in_buf, out_buf;
    block_t *p_out = NULL;
    unsigned int i_samples;
    int i_out_size;

    if( !p_block || !p_block->i_samples )
    {
        if( p_block )
            block_Release( p_block );
        return NULL;
    }

    i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
                 aout_FormatNbChannels( &(p_filter->fmt_out.audio) );

    p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
    if( !p_out )
    {
        msg_Warn( p_filter, "can't get output buffer" );
        block_Release( p_block );
        return NULL;
    }
    p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
                       aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
    p_out->i_dts = p_block->i_dts;
    p_out->i_pts = p_block->i_pts;
    p_out->i_length = p_block->i_length;

    aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
    aout_filter.input = p_filter->fmt_in.audio;
    aout_filter.input.i_format = p_filter->fmt_in.i_codec;
    aout_filter.output = p_filter->fmt_out.audio;
    aout_filter.output.i_format = p_filter->fmt_out.i_codec;

    in_buf.p_buffer = p_block->p_buffer;
    in_buf.i_nb_bytes = p_block->i_buffer;
    in_buf.i_nb_samples = p_block->i_samples;

#if 0
    unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
                             aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
    if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
    {
        msg_Err( p_filter, "input buffer is not word aligned" );
        /* Fix output buffer to be word aligned */
    }
#endif

    out_buf.p_buffer = p_out->p_buffer;
    out_buf.i_nb_bytes = p_out->i_buffer;
    out_buf.i_nb_samples = p_out->i_samples;

    memset( p_out->p_buffer, 0, i_out_size );
    if( p_filter->p_sys->b_downmix )
    {
        stereo2mono_downmix( &aout_filter, &in_buf, &out_buf );
        i_samples = mono( &aout_filter, &out_buf, &in_buf );
    }
    else
    {
        i_samples = stereo_to_mono( &aout_filter, &out_buf, &in_buf );
    }

    p_out->i_buffer = out_buf.i_nb_bytes;
    p_out->i_samples = out_buf.i_nb_samples;

    block_Release( p_block );
    return p_out;
}

/* stereo2mono_downmix - stereo channels into one mono channel.
 * Code taken from modules/audio_filter/channel_mixer/headphone.c
 * converted from float into int16_t based downmix
 * Written by Boris Dorès <[e-mail]>
 */
static void stereo2mono_downmix( 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;

    int i_input_nb = aout_FormatNbChannels( &p_filter->input );
    int i_output_nb = aout_FormatNbChannels( &p_filter->output );

    int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
    uint8_t * p_out;
    uint8_t * p_overflow;
    uint8_t * p_slide;

    size_t i_overflow_size;     /* in bytes */
    size_t i_out_size;          /* in bytes */

    unsigned int i, j;

    int i_source_channel_offset;
    int i_dest_channel_offset;
    unsigned int i_delay;
    double d_amplitude_factor;

    /* out buffer characterisitcs */
    p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
    p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
    p_out = p_out_buf->p_buffer;
    i_out_size = p_out_buf->i_nb_bytes;

    if( p_sys != NULL )
    {
        /* Slide the overflow buffer */
        p_overflow = p_sys->p_overflow_buffer;
        i_overflow_size = p_sys->i_overflow_buffer_size;

        if ( i_out_size > i_overflow_size )
            memcpy( p_out, p_overflow, i_overflow_size );
        else
            memcpy( p_out, p_overflow, i_out_size );

        p_slide = p_sys->p_overflow_buffer;
        while( p_slide < p_overflow + i_overflow_size )
        {
            if( p_slide + i_out_size < p_overflow + i_overflow_size )
            {
                memset( p_slide, 0, i_out_size );
                if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
                    memcpy( p_slide, p_slide + i_out_size, i_out_size );
                else
                    memcpy( p_slide, p_slide + i_out_size,
                            p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
            }
            else
            {
                memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
            }
            p_slide += i_out_size;
        }

        /* apply the atomic operations */
        for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
        {
            /* shorter variable names */
            i_source_channel_offset
                = p_sys->p_atomic_operations[i].i_source_channel_offset;
            i_dest_channel_offset
                = p_sys->p_atomic_operations[i].i_dest_channel_offset;
            i_delay = p_sys->p_atomic_operations[i].i_delay;
            d_amplitude_factor
                = p_sys->p_atomic_operations[i].d_amplitude_factor;

            if( p_out_buf->i_nb_samples > i_delay )
            {
                /* current buffer coefficients */
                for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
                {
                    ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
                        += p_in[ j * i_input_nb + i_source_channel_offset ]
                           * d_amplitude_factor;
                }

                /* overflow buffer coefficients */
                for( j = 0; j < i_delay; j++ )
                {
                    ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
                        += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
                           * i_input_nb + i_source_channel_offset ]
                           * d_amplitude_factor;
                }
            }
            else
            {
                /* overflow buffer coefficients only */
                for( j = 0; j < p_out_buf->i_nb_samples; j++ )
                {
                    ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
                        * i_output_nb + i_dest_channel_offset ]
                        += p_in[ j * i_input_nb + i_source_channel_offset ]
                           * d_amplitude_factor;
                }
            }
        }
    }
    else
    {
        memset( p_out, 0, i_out_size );
    }
}

/* Simple stereo to mono mixing. */
static unsigned int mono( aout_filter_t *p_filter,
                          aout_buffer_t *p_output, aout_buffer_t *p_input )
{
    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
    int16_t *p_in, *p_out;
    unsigned int n = 0, r = 0;

    p_in = (int16_t *) p_input->p_buffer;
    p_out = (int16_t *) p_output->p_buffer;

    while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
    {
        p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
        r++;
        n += 2;
    }
    return r;
}

/* Simple stereo to mono mixing. */
static unsigned int stereo_to_mono( aout_filter_t *p_filter,
                                    aout_buffer_t *p_output, aout_buffer_t *p_input )
{
    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
    int16_t *p_in, *p_out;
    unsigned int n;

    p_in = (int16_t *) p_input->p_buffer;
    p_out = (int16_t *) p_output->p_buffer;

    for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
    {
        /* Fake real mono. */
        if( p_sys->i_channel_selected == -1)
        {
            p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
            n++;
        }
        else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
        {
            p_out[n] = p_out[n+1] = p_in[n];
        }
    }
    return n;
}