static void CloseFilter ( vlc_object_t * );
static block_t *Convert( filter_t *p_filter, block_t *p_block );
-static void stereo_mono_downmix( aout_instance_t *, aout_filter_t *,
- aout_buffer_t *, aout_buffer_t * );
-static unsigned int stereo_to_mono( int16_t *, int16_t *, unsigned int );
-static void silence_channel( aout_instance_t *, aout_filter_t *,
- aout_buffer_t *, aout_buffer_t * );
+static unsigned int stereo_to_mono( aout_instance_t *, aout_filter_t *,
+ aout_buffer_t *, aout_buffer_t * );
/*****************************************************************************
* Local structures
*****************************************************************************/
struct filter_sys_t
{
- vlc_bool_t b_block_channel;
int i_nb_channels; /* number of float32 per sample */
unsigned int i_channel_selected;
int i_bitspersample;
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, "invalid format" );
+ msg_Err( p_this, "filter discarded (invalid format)" );
return -1;
}
p_sys->i_channel_selected =
(unsigned int) var_GetInteger( p_this, MONO_CFG "mono-channel" );
- /* temporarily force channel silence */
- p_sys->b_block_channel = VLC_TRUE;
- if( p_sys->b_block_channel )
- {
- p_filter->fmt_out.audio.i_physical_channels =
+#if 0
+ p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
+#endif
+ p_filter->fmt_out.audio.i_physical_channels =
(AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
- }
- else
- p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
- p_filter->pf_audio_filter = Convert;
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_out.audio) );
+ p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
+ 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,
aout_buffer_t in_buf, out_buf;
block_t *p_out = NULL;
int i_out_size;
+ unsigned int i_samples;
if( !p_block || !p_block->i_samples )
{
}
i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
- p_filter->p_sys->i_nb_channels;
+ aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
if( !p_out )
return NULL;
}
- p_out->i_samples = p_block->i_samples;
+ 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;
in_buf.i_nb_samples = p_block->i_samples;
#if 0
- if( in_buf.i_nb_bytes != (p_filter->p_sys->i_bitspersample/8) * in_buf.i_nb_samples )
+ 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 alligned" );
-/* if( in_buf.i_nb_bytes > (p_filter->p_sys->i_bitspersample/8) * in_buf.i_nb_samples)
- in_buf.i_nb_bytes = (p_filter->p_sys->i_bitspersample/8) * in_buf.i_nb_samples;
- else
- //in_buf*/
+ msg_Err( p_filter, "input buffer is not word aligned" );
+ /* Fix output buffer to be word aligned */
}
#endif
out_buf.i_nb_bytes = p_out->i_buffer;
out_buf.i_nb_samples = p_out->i_samples;
- stereo_mono_downmix( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
+ i_samples = stereo_to_mono( (aout_instance_t *)p_filter, &aout_filter,
+ &out_buf, &in_buf );
p_out->i_buffer = out_buf.i_nb_bytes;
p_out->i_samples = out_buf.i_nb_samples;
return p_out;
}
-static void stereo_mono_downmix( 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;
-
- if( p_sys->b_block_channel )
- {
- silence_channel( p_aout, p_filter, p_out_buf, p_in_buf );
- }
- else
- {
- unsigned int i_samples;
-
- i_samples = stereo_to_mono( (int16_t *)p_out_buf->p_buffer, (int16_t *)p_in_buf->p_buffer,
- p_out_buf->i_nb_samples );
- }
-
- p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
-}
-
-/* silence_channel - play silence on all channels except the selected one.
+/* stereo_to_mono - mix 2 channels (left,right) into one and play silence on
+ * all other channels.
*/
-static void silence_channel( aout_instance_t * p_aout, aout_filter_t * p_filter,
- aout_buffer_t *p_out_buf, aout_buffer_t *p_in_buf )
+static unsigned int stereo_to_mono( aout_instance_t * p_aout, 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;
- unsigned int n = 0;
int16_t *p_in, *p_out;
+ unsigned int n;
- p_in = (int16_t *)p_in_buf->p_buffer;
- p_out = (int16_t *)p_out_buf->p_buffer;
+ p_in = (int16_t *) p_input->p_buffer;
+ p_out = (int16_t *) p_output->p_buffer;
- for( n = 0; n < p_in_buf->i_nb_samples * p_sys->i_nb_channels; n++ )
+ for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
{
if( (n%p_sys->i_nb_channels) == p_sys->i_channel_selected )
{
- p_out[n] = p_in[n];
+ p_out[n] = (p_in[n] + p_in[n+1]) >> 1;
}
else
{
p_out[n] = 0x0;
}
}
-}
-
-/* stereo_to_mono() function is from ffmpeg file libavcodec/resample.c
- * Copyright (c) 2000 Fabrice Bellard.
- */
-static unsigned int stereo_to_mono( int16_t *p_output, int16_t *p_input,
- unsigned int i_samples )
-{
- int16_t *p, *q;
- unsigned int n = i_samples;
-
- p = p_input;
- q = p_output;
-
- while (n >= 4) {
- q[0] = (p[0] + p[1]) >> 1;
- q[1] = (p[2] + p[3]) >> 1;
- q[2] = (p[4] + p[5]) >> 1;
- q[3] = (p[6] + p[7]) >> 1;
- q += 4;
- p += 8;
- n -= 4;
- }
- while (n > 0) {
- q[0] = (p[0] + p[1]) >> 1;
- q++;
- p += 2;
- n--;
- }
return n;
}