#include <vlc_common.h>
#include <vlc_aout.h>
+#include <vlc_cpu.h>
+#include <vlc_modules.h>
+
#include "aout_internal.h"
/*****************************************************************************
audio_sample_format_t * p_format )
{
/* Retrieve user defaults. */
- int i_rate = config_GetInt( p_aout, "aout-rate" );
+ int i_rate = var_InheritInteger( p_aout, "aout-rate" );
vlc_value_t val, text;
/* kludge to avoid a fpu error when rate is 0... */
if( i_rate == 0 ) i_rate = -1;
p_aout->output.output.i_rate = i_rate;
aout_FormatPrepare( &p_aout->output.output );
- aout_lock_output_fifo( p_aout );
-
/* Find the best output plug-in. */
p_aout->output.p_module = module_need( p_aout, "audio output", "$aout", false );
if ( p_aout->output.p_module == NULL )
{
msg_Err( p_aout, "no suitable audio output module" );
- aout_unlock_output_fifo( p_aout );
return -1;
}
aout_FormatPrepare( &p_aout->output.output );
+ aout_lock_output_fifo( p_aout );
+
/* Prepare FIFO. */
aout_FifoInit( p_aout, &p_aout->output.fifo,
p_aout->output.output.i_rate );
aout_FormatPrint( p_aout, "output", &p_aout->output.output );
/* Calculate the resulting mixer output format. */
- memcpy( &p_aout->mixer.mixer, &p_aout->output.output,
- sizeof(audio_sample_format_t) );
+ p_aout->mixer_format = p_aout->output.output;
if ( !AOUT_FMT_NON_LINEAR(&p_aout->output.output) )
{
/* Non-S/PDIF mixer only deals with float32 or fixed32. */
- p_aout->mixer.mixer.i_format
- = (vlc_CPU() & CPU_CAPABILITY_FPU) ?
- VLC_CODEC_FL32 : VLC_CODEC_FI32;
- aout_FormatPrepare( &p_aout->mixer.mixer );
+ p_aout->mixer_format.i_format
+ = HAVE_FPU ? VLC_CODEC_FL32 : VLC_CODEC_FI32;
+ aout_FormatPrepare( &p_aout->mixer_format );
}
else
{
- p_aout->mixer.mixer.i_format = p_format->i_format;
+ p_aout->mixer_format.i_format = p_format->i_format;
}
- aout_FormatPrint( p_aout, "mixer", &p_aout->mixer.mixer );
+ aout_FormatPrint( p_aout, "mixer", &p_aout->mixer_format );
/* Create filters. */
p_aout->output.i_nb_filters = 0;
if ( aout_FiltersCreatePipeline( p_aout, p_aout->output.pp_filters,
&p_aout->output.i_nb_filters,
- &p_aout->mixer.mixer,
+ &p_aout->mixer_format,
&p_aout->output.output ) < 0 )
{
msg_Err( p_aout, "couldn't create audio output pipeline" );
}
/* Prepare hints for the buffer allocator. */
- p_aout->mixer.output_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
- p_aout->mixer.output_alloc.i_bytes_per_sec
- = p_aout->mixer.mixer.i_bytes_per_frame
- * p_aout->mixer.mixer.i_rate
- / p_aout->mixer.mixer.i_frame_length;
+ p_aout->mixer_allocation.b_alloc = true;
+ p_aout->mixer_allocation.i_bytes_per_sec
+ = p_aout->mixer_format.i_bytes_per_frame
+ * p_aout->mixer_format.i_rate
+ / p_aout->mixer_format.i_frame_length;
aout_FiltersHintBuffers( p_aout, p_aout->output.pp_filters,
p_aout->output.i_nb_filters,
- &p_aout->mixer.output_alloc );
+ &p_aout->mixer_allocation );
p_aout->output.b_error = 0;
return 0;
*****************************************************************************/
void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
{
- aout_FiltersPlay( p_aout, p_aout->output.pp_filters,
- p_aout->output.i_nb_filters,
+ aout_FiltersPlay( p_aout->output.pp_filters, p_aout->output.i_nb_filters,
&p_buffer );
- if( p_buffer->i_nb_bytes == 0 )
+ if( !p_buffer )
+ return;
+ if( p_buffer->i_buffer == 0 )
{
- aout_BufferFree( p_buffer );
+ block_Release( p_buffer );
return;
}
/* Drop the audio sample if the audio output is really late.
* In the case of b_can_sleek, we don't use a resampler so we need to be
* a lot more severe. */
- while ( p_buffer && p_buffer->start_date <
+ while ( p_buffer && p_buffer->i_pts <
(b_can_sleek ? start_date : mdate()) - AOUT_PTS_TOLERANCE )
{
msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), "
- "trashing %"PRId64"us", mdate() - p_buffer->start_date,
- p_buffer->end_date - p_buffer->start_date );
+ "trashing %"PRId64"us", mdate() - p_buffer->i_pts,
+ p_buffer->i_length );
p_buffer = p_buffer->p_next;
aout_BufferFree( p_aout->output.fifo.p_first );
p_aout->output.fifo.p_first = p_buffer;
/* Here we suppose that all buffers have the same duration - this is
* generally true, and anyway if it's wrong it won't be a disaster.
*/
- if ( p_buffer->start_date > start_date
- + (p_buffer->end_date - p_buffer->start_date) )
+ if ( p_buffer->i_pts > start_date + p_buffer->i_length )
/*
* + AOUT_PTS_TOLERANCE )
* There is no reason to want that, it just worsen the scheduling of
* --Gibalou
*/
{
- const mtime_t i_delta = p_buffer->start_date - start_date;
+ const mtime_t i_delta = p_buffer->i_pts - start_date;
aout_unlock_output_fifo( p_aout );
if ( !p_aout->output.b_starving )
p_aout->output.b_starving = 0;
+ p_aout->output.fifo.p_first = p_buffer->p_next;
+ if ( p_buffer->p_next == NULL )
+ {
+ p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
+ }
+
if ( !b_can_sleek &&
- ( (p_buffer->start_date - start_date > AOUT_PTS_TOLERANCE)
- || (start_date - p_buffer->start_date > AOUT_PTS_TOLERANCE) ) )
+ ( (p_buffer->i_pts - start_date > AOUT_PTS_TOLERANCE)
+ || (start_date - p_buffer->i_pts > AOUT_PTS_TOLERANCE) ) )
{
/* Try to compensate the drift by doing some resampling. */
int i;
- mtime_t difference = start_date - p_buffer->start_date;
+ mtime_t difference = start_date - p_buffer->i_pts;
msg_Warn( p_aout, "output date isn't PTS date, requesting "
"resampling (%"PRId64")", difference );
+ aout_FifoMoveDates( p_aout, &p_aout->output.fifo, difference );
+ aout_unlock_output_fifo( p_aout );
+
aout_lock_input_fifos( p_aout );
for ( i = 0; i < p_aout->i_nb_inputs; i++ )
{
- aout_fifo_t * p_fifo = &p_aout->pp_inputs[i]->fifo;
+ aout_fifo_t * p_fifo = &p_aout->pp_inputs[i]->mixer.fifo;
aout_FifoMoveDates( p_aout, p_fifo, difference );
}
-
- aout_FifoMoveDates( p_aout, &p_aout->output.fifo, difference );
aout_unlock_input_fifos( p_aout );
}
+ else
+ aout_unlock_output_fifo( p_aout );
- p_aout->output.fifo.p_first = p_buffer->p_next;
- if ( p_buffer->p_next == NULL )
- {
- p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
- }
-
- aout_unlock_output_fifo( p_aout );
return p_buffer;
}
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