/*****************************************************************************
* output.c : internal management of output streams for the audio output
*****************************************************************************
* Copyright (C) 2002-2004 the VideoLAN team
* $Id$
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "vlc_common.h"
#include "vlc_aout.h"
#include "aout_internal.h"
/*****************************************************************************
* aout_OutputNew : allocate a new output and rework the filter pipeline
*****************************************************************************
* This function is entered with the mixer lock.
*****************************************************************************/
int aout_OutputNew( aout_instance_t * p_aout,
audio_sample_format_t * p_format )
{
/* Retrieve user defaults. */
int i_rate = config_GetInt( 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;
memcpy( &p_aout->output.output, p_format, sizeof(audio_sample_format_t) );
if ( 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", 0);
if ( p_aout->output.p_module == NULL )
{
msg_Err( p_aout, "no suitable audio output module" );
aout_unlock_output_fifo( p_aout );
return -1;
}
if ( var_Type( p_aout, "audio-channels" ) ==
(VLC_VAR_INTEGER | VLC_VAR_HASCHOICE) )
{
/* The user may have selected a different channels configuration. */
var_Get( p_aout, "audio-channels", &val );
if ( val.i_int == AOUT_VAR_CHAN_RSTEREO )
{
p_aout->output.output.i_original_channels |=
AOUT_CHAN_REVERSESTEREO;
}
else if ( val.i_int == AOUT_VAR_CHAN_STEREO )
{
p_aout->output.output.i_original_channels =
AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
}
else if ( val.i_int == AOUT_VAR_CHAN_LEFT )
{
p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT;
}
else if ( val.i_int == AOUT_VAR_CHAN_RIGHT )
{
p_aout->output.output.i_original_channels = AOUT_CHAN_RIGHT;
}
else if ( val.i_int == AOUT_VAR_CHAN_DOLBYS )
{
p_aout->output.output.i_original_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_DOLBYSTEREO;
}
}
else if ( p_aout->output.output.i_physical_channels == AOUT_CHAN_CENTER
&& (p_aout->output.output.i_original_channels
& AOUT_CHAN_PHYSMASK) == (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT) )
{
/* Mono - create the audio-channels variable. */
var_Create( p_aout, "audio-channels",
VLC_VAR_INTEGER | VLC_VAR_HASCHOICE );
text.psz_string = _("Audio Channels");
var_Change( p_aout, "audio-channels", VLC_VAR_SETTEXT, &text, NULL );
val.i_int = AOUT_VAR_CHAN_STEREO; text.psz_string = _("Stereo");
var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
val.i_int = AOUT_VAR_CHAN_LEFT; text.psz_string = _("Left");
var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
val.i_int = AOUT_VAR_CHAN_RIGHT; text.psz_string = _("Right");
var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DUALMONO )
{
/* Go directly to the left channel. */
p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT;
val.i_int = AOUT_VAR_CHAN_LEFT;
var_Set( p_aout, "audio-channels", val );
}
var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart,
NULL );
}
else if ( p_aout->output.output.i_physical_channels ==
(AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)
&& (p_aout->output.output.i_original_channels &
(AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)) )
{
/* Stereo - create the audio-channels variable. */
var_Create( p_aout, "audio-channels",
VLC_VAR_INTEGER | VLC_VAR_HASCHOICE );
text.psz_string = _("Audio Channels");
var_Change( p_aout, "audio-channels", VLC_VAR_SETTEXT, &text, NULL );
if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DOLBYSTEREO )
{
val.i_int = AOUT_VAR_CHAN_DOLBYS;
text.psz_string = _("Dolby Surround");
}
else
{
val.i_int = AOUT_VAR_CHAN_STEREO;
text.psz_string = _("Stereo");
}
var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
val.i_int = AOUT_VAR_CHAN_LEFT; text.psz_string = _("Left");
var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
val.i_int = AOUT_VAR_CHAN_RIGHT; text.psz_string = _("Right");
var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
val.i_int = AOUT_VAR_CHAN_RSTEREO; text.psz_string=_("Reverse stereo");
var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DUALMONO )
{
/* Go directly to the left channel. */
p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT;
val.i_int = AOUT_VAR_CHAN_LEFT;
var_Set( p_aout, "audio-channels", val );
}
var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart,
NULL );
}
val.b_bool = true;
var_Set( p_aout, "intf-change", val );
aout_FormatPrepare( &p_aout->output.output );
/* Prepare FIFO. */
aout_FifoInit( p_aout, &p_aout->output.fifo,
p_aout->output.output.i_rate );
aout_unlock_output_fifo( p_aout );
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) );
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_FOURCC('f','l','3','2') :
VLC_FOURCC('f','i','3','2');
aout_FormatPrepare( &p_aout->mixer.mixer );
}
else
{
p_aout->mixer.mixer.i_format = p_format->i_format;
}
aout_FormatPrint( p_aout, "mixer", &p_aout->mixer.mixer );
/* 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->output.output ) < 0 )
{
msg_Err( p_aout, "couldn't create audio output pipeline" );
module_Unneed( p_aout, p_aout->output.p_module );
return -1;
}
/* 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;
aout_FiltersHintBuffers( p_aout, p_aout->output.pp_filters,
p_aout->output.i_nb_filters,
&p_aout->mixer.output_alloc );
p_aout->output.b_error = 0;
return 0;
}
/*****************************************************************************
* aout_OutputDelete : delete the output
*****************************************************************************
* This function is entered with the mixer lock.
*****************************************************************************/
void aout_OutputDelete( aout_instance_t * p_aout )
{
if ( p_aout->output.b_error )
{
return;
}
module_Unneed( p_aout, p_aout->output.p_module );
aout_FiltersDestroyPipeline( p_aout, p_aout->output.pp_filters,
p_aout->output.i_nb_filters );
aout_lock_output_fifo( p_aout );
aout_FifoDestroy( p_aout, &p_aout->output.fifo );
aout_unlock_output_fifo( p_aout );
p_aout->output.b_error = true;
}
/*****************************************************************************
* aout_OutputPlay : play a buffer
*****************************************************************************
* This function is entered with the mixer lock.
*****************************************************************************/
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,
&p_buffer );
if( p_buffer->i_nb_bytes == 0 )
{
aout_BufferFree( p_buffer );
return;
}
aout_lock_output_fifo( p_aout );
aout_FifoPush( p_aout, &p_aout->output.fifo, p_buffer );
p_aout->output.pf_play( p_aout );
aout_unlock_output_fifo( p_aout );
}
/*****************************************************************************
* aout_OutputNextBuffer : give the audio output plug-in the right buffer
*****************************************************************************
* If b_can_sleek is 1, the aout core functions won't try to resample
* new buffers to catch up - that is we suppose that the output plug-in can
* compensate it by itself. S/PDIF outputs should always set b_can_sleek = 1.
* This function is entered with no lock at all :-).
*****************************************************************************/
aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
mtime_t start_date,
bool b_can_sleek )
{
aout_buffer_t * p_buffer;
aout_lock_output_fifo( p_aout );
p_buffer = p_aout->output.fifo.p_first;
/* 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 <
(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 );
p_buffer = p_buffer->p_next;
aout_BufferFree( p_aout->output.fifo.p_first );
p_aout->output.fifo.p_first = p_buffer;
}
if ( p_buffer == NULL )
{
p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
#if 0 /* This is bad because the audio output might just be trying to fill
* in its internal buffers. And anyway, it's up to the audio output
* to deal with this kind of starvation. */
/* Set date to 0, to allow the mixer to send a new buffer ASAP */
aout_FifoSet( p_aout, &p_aout->output.fifo, 0 );
if ( !p_aout->output.b_starving )
msg_Dbg( p_aout,
"audio output is starving (no input), playing silence" );
p_aout->output.b_starving = 1;
#endif
aout_unlock_output_fifo( p_aout );
return NULL;
}
/* 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) )
/*
* + AOUT_PTS_TOLERANCE )
* There is no reason to want that, it just worsen the scheduling of
* an audio sample after an output starvation (ie. on start or on resume)
* --Gibalou
*/
{
const mtime_t i_delta = p_buffer->start_date - start_date;
aout_unlock_output_fifo( p_aout );
if ( !p_aout->output.b_starving )
msg_Dbg( p_aout, "audio output is starving (%"PRId64"), "
"playing silence", i_delta );
p_aout->output.b_starving = 1;
return NULL;
}
p_aout->output.b_starving = 0;
if ( !b_can_sleek &&
( (p_buffer->start_date - start_date > AOUT_PTS_TOLERANCE)
|| (start_date - p_buffer->start_date > AOUT_PTS_TOLERANCE) ) )
{
/* Try to compensate the drift by doing some resampling. */
int i;
mtime_t difference = start_date - p_buffer->start_date;
msg_Warn( p_aout, "output date isn't PTS date, requesting "
"resampling (%"PRId64")", difference );
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_FifoMoveDates( p_aout, p_fifo, difference );
}
aout_FifoMoveDates( p_aout, &p_aout->output.fifo, difference );
aout_unlock_input_fifos( 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;
}
