Commit 38feb73c authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont

Ugly resampler: implement all linear formats

parent 095526d1
......@@ -59,17 +59,14 @@ static int Create( vlc_object_t *p_this )
{
filter_t * p_filter = (filter_t *)p_this;
if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
|| p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
|| p_filter->fmt_in.audio.i_physical_channels
!= p_filter->fmt_out.audio.i_physical_channels
|| p_filter->fmt_in.audio.i_original_channels
!= p_filter->fmt_out.audio.i_original_channels
|| (p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32
&& p_filter->fmt_in.audio.i_format != VLC_CODEC_FI32) )
{
if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
|| p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
|| p_filter->fmt_in.audio.i_physical_channels
!= p_filter->fmt_out.audio.i_physical_channels
|| p_filter->fmt_in.audio.i_original_channels
!= p_filter->fmt_out.audio.i_original_channels
|| AOUT_FMT_NON_LINEAR( &p_filter->fmt_in.audio ) )
return VLC_EGENERIC;
}
p_filter->pf_audio_filter = DoWork;
return VLC_SUCCESS;
......@@ -87,12 +84,12 @@ static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
block_t *p_out_buf = p_in_buf;
unsigned int i_out_nb = p_in_buf->i_nb_samples
* p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate;
const unsigned int i_sample_bytes =
aout_FormatNbChannels( &p_filter->fmt_in.audio ) * sizeof(int32_t);
const unsigned framesize = (p_filter->fmt_in.audio.i_bitspersample / 8)
* aout_FormatNbChannels( &p_filter->fmt_in.audio );
if( p_filter->fmt_out.audio.i_rate > p_filter->fmt_in.audio.i_rate )
{
p_out_buf = block_Alloc( i_out_nb * i_sample_bytes );
p_out_buf = block_Alloc( i_out_nb * framesize );
if( !p_out_buf )
goto out;
}
......@@ -102,7 +99,7 @@ static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
unsigned int i_remainder = 0;
p_out_buf->i_nb_samples = i_out_nb;
p_out_buf->i_buffer = i_out_nb * i_sample_bytes;
p_out_buf->i_buffer = i_out_nb * framesize;
p_out_buf->i_pts = p_in_buf->i_pts;
p_out_buf->i_length = p_out_buf->i_nb_samples *
1000000 / p_filter->fmt_out.audio.i_rate;
......@@ -110,14 +107,14 @@ static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
while( i_out_nb )
{
if( p_out != p_in )
memcpy( p_out, p_in, i_sample_bytes );
p_out += i_sample_bytes;
memcpy( p_out, p_in, framesize );
p_out += framesize;
i_out_nb--;
i_remainder += p_filter->fmt_in.audio.i_rate;
while( i_remainder >= p_filter->fmt_out.audio.i_rate )
{
p_in += i_sample_bytes;
p_in += framesize;
i_remainder -= p_filter->fmt_out.audio.i_rate;
}
}
......
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