Commit b5ab4930 authored by Christophe Massiot's avatar Christophe Massiot
Browse files

* Finally fixed the segfault when resampling.

* Reactivated the A/52 demux.
* Wrote a real full-featured float32 mixer.
parent c02a9ef4
......@@ -445,7 +445,7 @@ dnl
BUILTINS="${BUILTINS}"
PLUGINS="${PLUGINS} misc/dummy/dummy misc/null"
PLUGINS="${PLUGINS} control/rc/rc misc/logger/logger access/file misc/memcpy/memcpy"
PLUGINS="${PLUGINS} demux/mpeg/es demux/mpeg/audio demux/mpeg/mpeg_system demux/mpeg/ps demux/mpeg/ts"
PLUGINS="${PLUGINS} demux/mpeg/es demux/mpeg/audio demux/mpeg/mpeg_system demux/mpeg/ps demux/mpeg/ts demux/a52sys"
PLUGINS="${PLUGINS} codec/mpeg_video/idct/idct codec/mpeg_video/idct/idctclassic codec/mpeg_video/motion/motion codec/mpeg_video/mpeg_video codec/spudec/spudec codec/spdif codec/mpeg_audio/mpeg_audio"
PLUGINS="${PLUGINS} codec/a52old/imdct/imdct codec/a52old/downmix/downmix codec/a52old/a52old"
#PLUGINS="${PLUGINS} codec/lpcm/lpcm"
......@@ -453,7 +453,7 @@ PLUGINS="${PLUGINS} video_filter/deinterlace/deinterlace video_filter/invert vid
PLUGINS="${PLUGINS} audio_filter/converter/float32tos16 audio_filter/converter/float32tos8 audio_filter/converter/float32tou16 audio_filter/converter/float32tou8 audio_filter/converter/a52tospdif audio_filter/converter/fixed32tofloat32 audio_filter/converter/fixed32tos16 audio_filter/converter/s16tofloat32"
PLUGINS="${PLUGINS} audio_filter/resampler/trivial audio_filter/resampler/ugly"
PLUGINS="${PLUGINS} audio_filter/channel_mixer/trivial"
PLUGINS="${PLUGINS} audio_mixer/trivial audio_mixer/spdif"
PLUGINS="${PLUGINS} audio_mixer/float32 audio_mixer/trivial audio_mixer/spdif"
PLUGINS="${PLUGINS} audio_output/file"
#PLUGINS="${PLUGINS} visualization/scope/scope"
PLUGINS="${PLUGINS} video_chroma/i420_rgb video_chroma/i420_yuy2 video_chroma/i422_yuy2 video_chroma/i420_ymga"
......
......@@ -2,7 +2,7 @@
* trivial.c : trivial channel mixer plug-in (drops unwanted channels)
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: trivial.c,v 1.1 2002/08/21 22:41:59 massiot Exp $
* $Id: trivial.c,v 1.2 2002/08/28 22:25:38 massiot Exp $
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
......@@ -87,7 +87,7 @@ static void SparseCopy( s32 * p_dest, const s32 * p_src, size_t i_len,
int i_output_stride, int i_input_stride )
{
int i;
for ( i = 0; i < i_len; i++ )
for ( i = i_len; i--; )
{
int j;
for ( j = 0; j < i_output_stride; j++ )
......
......@@ -2,7 +2,7 @@
* ugly.c : ugly resampler (changes pitch)
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: ugly.c,v 1.1 2002/08/24 20:22:34 sam Exp $
* $Id: ugly.c,v 1.2 2002/08/28 22:25:38 massiot Exp $
*
* Authors: Samuel Hocevar <sam@zoy.org>
*
......@@ -91,8 +91,9 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
for( i_chan = p_filter->input.i_channels ; i_chan ; )
{
i_chan--;
*p_out++ = p_in[i_chan];
p_out[i_chan] = p_in[i_chan];
}
p_out += p_filter->input.i_channels;
i_remainder += p_filter->input.i_rate;
while( i_remainder >= p_filter->output.i_rate )
......
trivial_SOURCES = trivial.c
float32_SOURCES = float32.c
spdif_SOURCES = spdif.c
/*****************************************************************************
* float32.c : precise float32 audio mixer implementation
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: float32.c,v 1.1 2002/08/28 22:25:38 massiot Exp $
*
* 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., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#include <errno.h>
#include <stdlib.h> /* malloc(), free() */
#include <string.h>
#include <vlc/vlc.h>
#include "audio_output.h"
#include "aout_internal.h"
/*****************************************************************************
* Local prototypes
*****************************************************************************/
static int Create ( vlc_object_t * );
static void DoWork ( aout_instance_t *, aout_buffer_t * );
/*****************************************************************************
* Module descriptor
*****************************************************************************/
vlc_module_begin();
set_description( _("float32 audio mixer module") );
set_capability( "audio mixer", 10 );
set_callbacks( Create, NULL );
vlc_module_end();
/*****************************************************************************
* Create: allocate trivial mixer
*****************************************************************************/
static int Create( vlc_object_t *p_this )
{
aout_instance_t * p_aout = (aout_instance_t *)p_this;
if ( p_aout->mixer.mixer.i_format != AOUT_FMT_FLOAT32 )
{
return -1;
}
if ( p_aout->i_nb_inputs == 1 )
{
/* Tell the trivial mixer to go for it. */
return -1;
}
p_aout->mixer.pf_do_work = DoWork;
return 0;
}
/*****************************************************************************
* ScaleWords: prepare input words for averaging
*****************************************************************************/
static void ScaleWords( float * p_out, const float * p_in, size_t i_nb_words,
int i_nb_inputs )
{
int i;
for ( i = i_nb_words; i--; )
{
*p_out++ = *p_in++ / i_nb_inputs;
}
}
/*****************************************************************************
* MeanWords: average input words
*****************************************************************************/
static void MeanWords( float * p_out, const float * p_in, size_t i_nb_words,
int i_nb_inputs )
{
int i;
for ( i = i_nb_words; i--; )
{
*p_out++ += *p_in++ / i_nb_inputs;
}
}
/*****************************************************************************
* DoWork: mix a new output buffer
*****************************************************************************
* Terminology : in this function a word designates a single float32, eg.
* a stereo sample is consituted of two words.
*****************************************************************************/
static void DoWork( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
{
int i_nb_inputs = p_aout->i_nb_inputs;
int i_input;
for ( i_input = 0; i_input < i_nb_inputs; i_input++ )
{
int i_nb_words = p_buffer->i_nb_samples
* p_aout->mixer.mixer.i_channels;
aout_input_t * p_input = p_aout->pp_inputs[i_input];
float * p_out = (float *)p_buffer->p_buffer;
float * p_in = (float *)p_input->p_first_byte_to_mix;
for ( ; ; )
{
ptrdiff_t i_available_words = (
(float *)p_input->fifo.p_first->p_buffer - p_in)
+ p_input->fifo.p_first->i_nb_samples
* p_aout->mixer.mixer.i_channels;
if ( i_available_words < i_nb_words )
{
aout_buffer_t * p_old_buffer;
if ( i_available_words > 0 )
{
if ( !i_input )
{
ScaleWords( p_out, p_in, i_available_words,
i_nb_inputs );
}
else
{
MeanWords( p_out, p_in, i_available_words,
i_nb_inputs );
}
}
i_nb_words -= i_available_words;
p_out += i_available_words;
/* Next buffer */
p_old_buffer = aout_FifoPop( p_aout, &p_input->fifo );
aout_BufferFree( p_old_buffer );
if ( p_input->fifo.p_first == NULL )
{
msg_Err( p_aout, "internal amix error" );
return;
}
p_in = (float *)p_input->fifo.p_first->p_buffer;
}
else
{
if ( i_nb_words > 0 )
{
if ( !i_input )
{
ScaleWords( p_out, p_in, i_nb_words, i_nb_inputs );
}
else
{
MeanWords( p_out, p_in, i_nb_words, i_nb_inputs );
}
}
p_input->p_first_byte_to_mix = (void *)(p_in
+ i_nb_words);
break;
}
}
}
}
......@@ -2,7 +2,7 @@
* generic.c: MPEG audio decoder
*****************************************************************************
* Copyright (C) 1999-2001 VideoLAN
* $Id: generic.c,v 1.3 2002/08/26 23:00:22 massiot Exp $
* $Id: generic.c,v 1.4 2002/08/28 22:25:38 massiot Exp $
*
* Authors: Michel Kaempf <maxx@via.ecp.fr>
* Michel Lespinasse <walken@via.ecp.fr>
......@@ -71,7 +71,7 @@ int adec_SyncFrame( adec_thread_t * p_adec, adec_sync_info_t * p_sync_info )
u32 header;
int index;
int * bit_rates;
const int * bit_rates;
int sample_rate;
int bit_rate;
int frame_size;
......
a52sys_SOURCES = a52sys.c
/*****************************************************************************
* a52_system.c : A52 input module for vlc
* a52sys.c : A/52 input module for vlc
*****************************************************************************
* Copyright (C) 2001 VideoLAN
* $Id: demux.c,v 1.1 2002/08/04 17:23:42 sam Exp $
* $Id: a52sys.c,v 1.1 2002/08/28 22:25:38 massiot Exp $
*
* Authors: Arnaud de Bossoreille de Ribou <bozo@via.ecp.fr>
*
......@@ -50,9 +50,9 @@ static int Demux ( input_thread_t * );
*****************************************************************************/
vlc_module_begin();
set_description( "A52 demuxer" );
set_capability( "demux", 150 );
set_capability( "demux", 155 );
set_callbacks( Init, NULL );
add_shortcut( "a52sys" );
add_shortcut( "a52" );
vlc_module_end();
/*****************************************************************************
......@@ -84,7 +84,7 @@ static int Init( vlc_object_t * p_this )
if( *p_peek != 0x0b || *(p_peek + 1) != 0x77 )
{
if( *p_input->psz_demux && !strncmp( p_input->psz_demux, "a52sys", 3 ) )
if( *p_input->psz_demux && !strncmp( p_input->psz_demux, "a52", 3 ) )
{
/* User forced */
msg_Err( p_input, "this doesn't look like an a52 stream, continuing" );
......@@ -128,11 +128,16 @@ static int Demux( input_thread_t * p_input )
pes_packet_t * p_pes;
data_packet_t * p_data;
if( p_fifo == NULL )
{
return -1;
}
i_read = input_SplitBuffer( p_input, &p_data, A52_PACKET_SIZE );
if ( i_read <= 0 )
{
return( i_read );
return i_read;
}
p_pes = input_NewPES( p_input->p_method_data );
......@@ -169,6 +174,6 @@ static int Demux( input_thread_t * p_input )
input_DecodePES( p_fifo, p_pes );
return( 1 );
return 1;
}
......@@ -2,7 +2,7 @@
* filters.c : audio output filters management
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: filters.c,v 1.6 2002/08/19 21:54:37 massiot Exp $
* $Id: filters.c,v 1.7 2002/08/28 22:25:39 massiot Exp $
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
......@@ -316,6 +316,7 @@ void aout_FiltersHintBuffers( aout_instance_t * p_aout,
p_first_alloc->i_bytes_per_sec = __MAX(
p_first_alloc->i_bytes_per_sec,
i_input_size );
p_filter->output_alloc.i_alloc_type = AOUT_ALLOC_NONE;
}
else
{
......@@ -344,7 +345,7 @@ void aout_FiltersPlay( aout_instance_t * p_aout,
aout_BufferAlloc( &p_filter->output_alloc,
(mtime_t)(*pp_input_buffer)->i_nb_samples * 1000000
/ p_filter->output.i_rate, *pp_input_buffer,
/ p_filter->input.i_rate, *pp_input_buffer,
p_output_buffer );
if ( p_output_buffer == NULL )
{
......
......@@ -2,7 +2,7 @@
* input.c : internal management of input streams for the audio output
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: input.c,v 1.9 2002/08/26 23:00:23 massiot Exp $
* $Id: input.c,v 1.10 2002/08/28 22:25:39 massiot Exp $
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
......@@ -268,7 +268,7 @@ void 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_output;
audio_sample_format_t new_input;
int i_ratio, i_nb_filters;
mtime_t old_duration;
aout_filter_t * pp_filters[AOUT_MAX_FILTERS];
......@@ -279,7 +279,7 @@ void aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
msg_Warn( p_aout, "buffer is %lld %s, resampling",
drift > 0 ? drift : -drift,
drift > 0 ? "in advance" : "late" );
memcpy( &new_output, &p_aout->mixer.mixer,
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;
......@@ -293,11 +293,12 @@ void aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
{
duration = old_duration * 150 / 100;
}
new_output.i_rate = new_output.i_rate * duration / old_duration;
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, &p_input->input,
&new_output ) < 0 )
&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 );
......@@ -318,16 +319,17 @@ void aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
&dummy_alloc );
dummy_alloc.i_bytes_per_sec = __MAX(
dummy_alloc.i_bytes_per_sec,
p_input->input.i_bytes_per_frame
* p_input->input.i_rate
/ p_input->input.i_frame_length );
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, old_duration, NULL, p_new_buffer );
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;
......
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