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vlc-2-2
Commit ef6c706b authored by Christophe Massiot's avatar Christophe Massiot
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* Makefile.old: Fixed BeOS compilation. 

* ALL: Fewer overhead when resampling.
parent dca61730
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Showing with 72 additions and 63 deletions
Makefile.old
View file @ ef6c706b
......@@ -61,9 +61,9 @@ LIBVLC_OBJ += src/misc/win32_specific.o
RESOURCE_OBJ := share/vlc_win32_rc.o
endif
LIBVLC_OBJ += $(CPP_OBJ) $(M_OBJ) $(BUILTIN_OBJ)
C_OBJ := $(VLC_OBJ) $(LIBVLC_OBJ) $(BUILTIN_OBJ)
C_OBJ := $(VLC_OBJ) $(LIBVLC_OBJ)
LIBVLC_OBJ += $(CPP_OBJ) $(M_OBJ) $(BUILTIN_OBJ)
VLC_OBJ += $(RESOURCE_OBJ)
......
include/aout_internal.h
View file @ ef6c706b
......@@ -2,7 +2,7 @@
* aout_internal.h : internal defines for audio output
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: aout_internal.h,v 1.21 2002/09/26 22:40:18 massiot Exp $
* $Id: aout_internal.h,v 1.22 2002/09/27 23:38:04 massiot Exp $
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
......@@ -144,6 +144,10 @@ struct aout_input_t
aout_filter_t * pp_filters[AOUT_MAX_FILTERS];
int i_nb_filters;
/* resamplers */
aout_filter_t * pp_resamplers[AOUT_MAX_FILTERS];
int i_nb_resamplers;
aout_fifo_t fifo;
/* Mixer information */
......
include/vlc_config.h
View file @ ef6c706b
......@@ -147,6 +147,9 @@
* time, without resampling */
#define AOUT_PTS_TOLERANCE (mtime_t)(.02*CLOCK_FREQ)
/* Max acceptable resampling (in %) */
#define AOUT_MAX_RESAMPLING 10
/*****************************************************************************
* Video configuration
*****************************************************************************/
......
modules/audio_filter/converter/s16tofloat32swab.c
View file @ ef6c706b
......@@ -3,7 +3,7 @@
* with endianness change
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: s16tofloat32swab.c,v 1.4 2002/09/26 22:40:19 massiot Exp $
* $Id: s16tofloat32swab.c,v 1.5 2002/09/27 23:38:04 massiot Exp $
*
* Authors: Samuel Hocevar <sam@zoy.org>
* Henri Fallon <henri@videolan.org>
......@@ -103,7 +103,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
#ifdef HAVE_SWAB
s16 * p_swabbed = alloca( i * sizeof(s16) );
swab( p_in_buf->p_buffer, p_swabbed, i * sizeof(s16) );
swab( p_in_buf->p_buffer, (void *)p_swabbed, i * sizeof(s16) );
p_in = p_swabbed + i - 1;
#else
byte_t p_tmp[2];
......
src/audio_output/input.c
View file @ ef6c706b
......@@ -2,7 +2,7 @@
* input.c : internal management of input streams for the audio output
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: input.c,v 1.13 2002/09/26 22:40:25 massiot Exp $
* $Id: input.c,v 1.14 2002/09/27 23:38:04 massiot Exp $
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
......@@ -41,14 +41,19 @@
*****************************************************************************/
int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
{
audio_sample_format_t intermediate_format;
/* Prepare FIFO. */
aout_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate );
p_input->p_first_byte_to_mix = NULL;
/* Create filters. */
memcpy( &intermediate_format, &p_aout->mixer.mixer,
sizeof(audio_sample_format_t) );
intermediate_format.i_rate = p_input->input.i_rate;
if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
&p_input->i_nb_filters, &p_input->input,
&p_aout->mixer.mixer ) < 0 )
&intermediate_format ) < 0 )
{
msg_Err( p_aout, "couldn't set an input pipeline" );
......@@ -58,10 +63,37 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
return -1;
}
/* Create resamplers. */
intermediate_format.i_rate = (p_input->input.i_rate
* (100 + AOUT_MAX_RESAMPLING)) / 100;
if ( intermediate_format.i_rate == p_aout->mixer.mixer.i_rate )
{
/* Just in case... */
intermediate_format.i_rate++;
}
if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_resamplers,
&p_input->i_nb_resamplers,
&intermediate_format,
&p_aout->mixer.mixer ) < 0 )
{
msg_Err( p_aout, "couldn't set a resampler pipeline" );
aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
p_input->i_nb_filters );
aout_FifoDestroy( p_aout, &p_input->fifo );
p_input->b_error = 1;
return -1;
}
/* Prepare hints for the buffer allocator. */
p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
p_input->input_alloc.i_bytes_per_sec = -1;
aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_input->input_alloc );
aout_FiltersHintBuffers( p_aout, p_input->pp_filters,
p_input->i_nb_filters,
&p_input->input_alloc );
......@@ -91,6 +123,8 @@ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input )
aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
p_input->i_nb_filters );
aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers );
aout_FifoDestroy( p_aout, &p_input->fifo );
return 0;
......@@ -140,6 +174,10 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
if ( start_date == 0 ) start_date = p_buffer->start_date;
/* Run pre-filters. */
aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
&p_buffer );
if ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE
|| start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE )
{
......@@ -150,81 +188,45 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
* synchronization
* Solution : resample the buffer to avoid a scratch.
*/
audio_sample_format_t new_input;
int i_ratio, i_nb_filters;
int i_ratio;
mtime_t old_duration;
aout_filter_t * pp_filters[AOUT_MAX_FILTERS];
aout_buffer_t * p_new_buffer;
aout_alloc_t dummy_alloc;
mtime_t drift = p_buffer->start_date - start_date;
msg_Warn( p_aout, "buffer is %lld %s, resampling",
drift > 0 ? drift : -drift,
drift > 0 ? "in advance" : "late" );
memcpy( &new_input, &p_input->input,
sizeof(audio_sample_format_t) );
old_duration = p_buffer->end_date - p_buffer->start_date;
duration = p_buffer->end_date - start_date;
i_ratio = duration * 100 / old_duration;
i_ratio = (duration * 100) / old_duration;
/* If the ratio is too != 100, the sound quality will be awful. */
if ( i_ratio < 66 /* % */ )
if ( i_ratio < 100 - AOUT_MAX_RESAMPLING /* % */ )
{
duration = old_duration * 66 / 100;
duration = (old_duration * (100 - AOUT_MAX_RESAMPLING)) / 100;
}
if ( i_ratio > 150 /* % */ )
if ( i_ratio > 100 + AOUT_MAX_RESAMPLING /* % */ )
{
duration = old_duration * 150 / 100;
duration = (old_duration * (100 + AOUT_MAX_RESAMPLING)) / 100;
}
new_input.i_rate = new_input.i_rate * old_duration / duration;
aout_FormatPrepare( &new_input );
if ( aout_FiltersCreatePipeline( p_aout, pp_filters,
&i_nb_filters, &new_input,
&p_aout->mixer.mixer ) < 0 )
{
msg_Err( p_aout, "couldn't set an input pipeline for resampling" );
vlc_mutex_lock( &p_aout->mixer_lock );
aout_FifoSet( p_aout, &p_input->fifo, 0 );
vlc_mutex_unlock( &p_aout->mixer_lock );
aout_BufferFree( p_buffer );
vlc_mutex_unlock( &p_input->lock );
return 0;
}
dummy_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
dummy_alloc.i_bytes_per_sec = -1;
aout_FiltersHintBuffers( p_aout, pp_filters, i_nb_filters,
&dummy_alloc );
dummy_alloc.i_bytes_per_sec = __MAX(
dummy_alloc.i_bytes_per_sec,
new_input.i_bytes_per_frame
* new_input.i_rate
/ new_input.i_frame_length );
dummy_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
aout_BufferAlloc( &dummy_alloc, duration, NULL, p_new_buffer );
memcpy( p_new_buffer->p_buffer, p_buffer->p_buffer,
p_buffer->i_nb_bytes );
p_new_buffer->i_nb_samples = p_buffer->i_nb_samples;
p_new_buffer->i_nb_bytes = p_buffer->i_nb_bytes;
aout_BufferFree( p_buffer );
p_buffer = p_new_buffer;
p_input->pp_resamplers[0]->input.i_rate
= (p_input->input.i_rate * old_duration) / duration;
aout_FiltersPlay( p_aout, pp_filters, i_nb_filters,
aout_FiltersPlay( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_buffer );
aout_FiltersDestroyPipeline( p_aout, pp_filters,
i_nb_filters );
}
else
{
/* No resampling needed (except maybe the one imposed by the
* output). */
duration = p_buffer->end_date - p_buffer->start_date;
aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
&p_buffer );
if ( p_input->input.i_rate != p_aout->mixer.mixer.i_rate )
{
/* Standard resampling is needed ! */
p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
aout_FiltersPlay( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_buffer );
}
}
/* Adding the start date will be managed by aout_FifoPush(). */
......
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