p_filter->p_owner->p_input = p_input;
/* request format */
- memcpy( &p_filter->input, &chain_output_format,
+ memcpy( &p_filter->fmt_in.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
- memcpy( &p_filter->output, &chain_output_format,
+ memcpy( &p_filter->fmt_out.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
if ( p_filter->p_module == NULL )
{
/* if the filter requested a special format, retry */
- if ( !( AOUT_FMTS_IDENTICAL( &p_filter->input,
+ if ( !( AOUT_FMTS_IDENTICAL( &p_filter->fmt_in.audio,
&chain_input_format )
- && AOUT_FMTS_IDENTICAL( &p_filter->output,
+ && AOUT_FMTS_IDENTICAL( &p_filter->fmt_out.audio,
&chain_output_format ) ) )
{
- aout_FormatPrepare( &p_filter->input );
- aout_FormatPrepare( &p_filter->output );
+ aout_FormatPrepare( &p_filter->fmt_in.audio );
+ aout_FormatPrepare( &p_filter->fmt_out.audio );
p_filter->p_module = module_need( p_filter,
"audio filter",
psz_parser, true );
/* try visual filters */
else
{
- memcpy( &p_filter->input, &chain_output_format,
+ memcpy( &p_filter->fmt_in.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
- memcpy( &p_filter->output, &chain_output_format,
+ memcpy( &p_filter->fmt_out.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
p_filter->p_module = module_need( p_filter,
"visualization",
}
/* complete the filter chain if necessary */
- if ( !AOUT_FMTS_IDENTICAL( &chain_input_format, &p_filter->input ) )
+ if ( !AOUT_FMTS_IDENTICAL( &chain_input_format,
+ &p_filter->fmt_in.audio ) )
{
if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
&p_input->i_nb_filters,
&chain_input_format,
- &p_filter->input ) < 0 )
+ &p_filter->fmt_in.audio ) < 0 )
{
msg_Err( p_aout, "cannot add user filter %s (skipped)",
psz_parser );
/* success */
p_filter->b_continuity = false;
p_input->pp_filters[p_input->i_nb_filters++] = p_filter;
- memcpy( &chain_input_format, &p_filter->output,
+ memcpy( &chain_input_format, &p_filter->fmt_out.audio,
sizeof( audio_sample_format_t ) );
if( i_visual == 0 ) /* scaletempo */
p_input->input_alloc.b_alloc = true;
/* Setup the initial rate of the resampler */
- p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+ p_input->pp_resamplers[0]->fmt_in.audio.i_rate = p_input->input.i_rate;
}
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
/* Handle input rate change, but keep drift correction */
if( i_input_rate != p_input->i_last_input_rate )
{
- unsigned int * const pi_rate = &p_input->p_playback_rate_filter->input.i_rate;
+ unsigned int * const pi_rate = &p_input->p_playback_rate_filter->fmt_in.audio.i_rate;
#define F(r,ir) ( INPUT_RATE_DEFAULT * (r) / (ir) )
const int i_delta = *pi_rate - F(p_input->input.i_rate,p_input->i_last_input_rate);
*pi_rate = F(p_input->input.i_rate + i_delta, i_input_rate);
if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
{
- p_input->pp_resamplers[0]->input.i_rate += 2; /* Hz */
+ p_input->pp_resamplers[0]->fmt_in.audio.i_rate += 2; /* Hz */
}
else
{
- p_input->pp_resamplers[0]->input.i_rate -= 2; /* Hz */
+ p_input->pp_resamplers[0]->fmt_in.audio.i_rate -= 2; /* Hz */
}
/* Check if everything is back to normal, in which case we can stop the
(p_input->pp_resamplers[0] == p_input->p_playback_rate_filter)
? INPUT_RATE_DEFAULT * p_input->input.i_rate / i_input_rate
: p_input->input.i_rate;
- if( p_input->pp_resamplers[0]->input.i_rate == i_nominal_rate )
+ if( p_input->pp_resamplers[0]->fmt_in.audio.i_rate == i_nominal_rate )
{
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
msg_Warn( p_aout, "resampling stopped after %"PRIi64" usec "
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
if( p_input->i_nb_resamplers != 0 )
{
- p_input->pp_resamplers[0]->input.i_rate =
+ p_input->pp_resamplers[0]->fmt_in.audio.i_rate =
( p_input->pp_resamplers[0] == p_input->p_playback_rate_filter )
? INPUT_RATE_DEFAULT * p_input->input.i_rate / p_input->i_last_input_rate
: p_input->input.i_rate;