Commit 1d0b15b5 authored by Jean-Paul Saman's avatar Jean-Paul Saman
Browse files

Resample audio channels when doing real stereo to mono.

parent 1e85ac0b
......@@ -26,6 +26,7 @@
*****************************************************************************/
#include <stdlib.h> /* malloc(), free() */
#include <string.h>
#include <math.h> /* sqrt */
#ifdef HAVE_STDINT_H
# include <stdint.h> /* int16_t .. */
......@@ -54,19 +55,41 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block );
static unsigned int stereo_to_mono( aout_instance_t *, aout_filter_t *,
aout_buffer_t *, aout_buffer_t * );
static void stereo2mono_downmix( aout_instance_t *, aout_filter_t *,
aout_buffer_t *, aout_buffer_t * );
/*****************************************************************************
* Local structures
*****************************************************************************/
struct atomic_operation_t
{
int i_source_channel_offset;
int i_dest_channel_offset;
unsigned int i_delay;/* in sample unit */
double d_amplitude_factor;
};
struct filter_sys_t
{
int i_nb_channels; /* number of float32 per sample */
vlc_bool_t b_downmix;
int i_nb_channels; /* number of int16_t per sample */
int i_channel_selected;
int i_bitspersample;
size_t i_overflow_buffer_size;/* in bytes */
byte_t * p_overflow_buffer;
unsigned int i_nb_atomic_operations;
struct atomic_operation_t * p_atomic_operations;
};
#define MONO_DOWNMIX_TEXT ("Use downmix algorithme.")
#define MONO_DOWNMIX_LONGTEXT ("This option selects a stereo to mono " \
"downmix algorithm that is used in the headphone channel mixer. It" \
"gives the effect of standing in a room full of speakers." )
#define MONO_CHANNEL_TEXT ("Select channel to keep")
#define MONO_CHANNEL_LONGTEXT ("This option silcences all other channels " \
#define MONO_CHANNEL_LONGTEXT ("This option silences all other channels " \
"except the selected channel. Choose one from (0=left, 1=right " \
"2=rear left, 3=rear right, 4=center, 5=left front)")
......@@ -88,7 +111,10 @@ vlc_module_begin();
set_description( _("Audio filter for stereo to mono conversion") );
set_capability( "audio filter2", 5 );
add_integer( MONO_CFG "mono-channel", -1, NULL, MONO_CHANNEL_TEXT, MONO_CHANNEL_LONGTEXT, VLC_FALSE );
add_bool( MONO_CFG "mono-downmix", VLC_FALSE, NULL, MONO_DOWNMIX_TEXT,
MONO_DOWNMIX_LONGTEXT, VLC_FALSE );
add_integer( MONO_CFG "mono-channel", -1, NULL, MONO_CHANNEL_TEXT,
MONO_CHANNEL_LONGTEXT, VLC_FALSE );
change_integer_list( pi_pos_values, ppsz_pos_descriptions, 0 );
set_category( CAT_AUDIO );
......@@ -97,6 +123,238 @@ vlc_module_begin();
set_shortname( "Mono" );
vlc_module_end();
/* Init() and ComputeChannelOperations() -
* Code taken from modules/audio_filter/channel_mixer/headphone.c
* converted from float into int16_t based downmix
* Written by Boris Dorès <babal@via.ecp.fr>
*/
/*****************************************************************************
* Init: initialize internal data structures
* and computes the needed atomic operations
*****************************************************************************/
/* x and z represent the coordinates of the virtual speaker
* relatively to the center of the listener's head, measured in meters :
*
* left right
*Z
*-
*a head
*x
*i
*s
* rear left rear right
*
* x-axis
* */
static void ComputeChannelOperations( struct filter_sys_t * p_data,
unsigned int i_rate, unsigned int i_next_atomic_operation,
int i_source_channel_offset, double d_x, double d_z,
double d_compensation_length, double d_channel_amplitude_factor )
{
double d_c = 340; /*sound celerity (unit: m/s)*/
double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
/* Left ear */
p_data->p_atomic_operations[i_next_atomic_operation]
.i_source_channel_offset = i_source_channel_offset;
p_data->p_atomic_operations[i_next_atomic_operation]
.i_dest_channel_offset = 0;/* left */
p_data->p_atomic_operations[i_next_atomic_operation]
.i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
/ d_c * i_rate - d_compensation_delay );
if( d_x < 0 )
{
p_data->p_atomic_operations[i_next_atomic_operation]
.d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
}
else if( d_x > 0 )
{
p_data->p_atomic_operations[i_next_atomic_operation]
.d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
}
else
{
p_data->p_atomic_operations[i_next_atomic_operation]
.d_amplitude_factor = d_channel_amplitude_factor / 2;
}
/* Right ear */
p_data->p_atomic_operations[i_next_atomic_operation + 1]
.i_source_channel_offset = i_source_channel_offset;
p_data->p_atomic_operations[i_next_atomic_operation + 1]
.i_dest_channel_offset = 1;/* right */
p_data->p_atomic_operations[i_next_atomic_operation + 1]
.i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
/ d_c * i_rate - d_compensation_delay );
if( d_x < 0 )
{
p_data->p_atomic_operations[i_next_atomic_operation + 1]
.d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
}
else if( d_x > 0 )
{
p_data->p_atomic_operations[i_next_atomic_operation + 1]
.d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
}
else
{
p_data->p_atomic_operations[i_next_atomic_operation + 1]
.d_amplitude_factor = d_channel_amplitude_factor / 2;
}
}
static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
unsigned int i_nb_channels, uint32_t i_physical_channels,
unsigned int i_rate )
{
double d_x = config_GetInt( p_this, "headphone-dim" );
double d_z = d_x;
double d_z_rear = -d_x/3;
double d_min = 0;
unsigned int i_next_atomic_operation;
int i_source_channel_offset;
unsigned int i;
if( p_data == NULL )
{
msg_Dbg( p_this, "passing a null pointer as argument" );
return 0;
}
if( config_GetInt( p_this, "headphone-compensate" ) )
{
/* minimal distance to any speaker */
if( i_physical_channels & AOUT_CHAN_REARCENTER )
{
d_min = d_z_rear;
}
else
{
d_min = d_z;
}
}
/* Number of elementary operations */
p_data->i_nb_atomic_operations = i_nb_channels * 2;
if( i_physical_channels & AOUT_CHAN_CENTER )
{
p_data->i_nb_atomic_operations += 2;
}
p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
* p_data->i_nb_atomic_operations );
if( p_data->p_atomic_operations == NULL )
{
msg_Err( p_this, "out of memory" );
return -1;
}
/* For each virtual speaker, computes elementary wave propagation time
* to each ear */
i_next_atomic_operation = 0;
i_source_channel_offset = 0;
if( i_physical_channels & AOUT_CHAN_LEFT )
{
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, -d_x , d_z , d_min , 2.0 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
if( i_physical_channels & AOUT_CHAN_RIGHT )
{
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, d_x , d_z , d_min , 2.0 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
{
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, -d_x , 0 , d_min , 1.5 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
{
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, d_x , 0 , d_min , 1.5 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
if( i_physical_channels & AOUT_CHAN_REARLEFT )
{
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
if( i_physical_channels & AOUT_CHAN_REARRIGHT )
{
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
if( i_physical_channels & AOUT_CHAN_REARCENTER )
{
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, 0 , -d_z , d_min , 1.5 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
if( i_physical_channels & AOUT_CHAN_CENTER )
{
/* having two center channels increases the spatialization effect */
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
i_next_atomic_operation += 2;
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
if( i_physical_channels & AOUT_CHAN_LFE )
{
ComputeChannelOperations( p_data , i_rate
, i_next_atomic_operation , i_source_channel_offset
, 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
i_next_atomic_operation += 2;
i_source_channel_offset++;
}
/* Initialize the overflow buffer
* we need it because the process induce a delay in the samples */
p_data->i_overflow_buffer_size = 0;
for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
{
if( p_data->i_overflow_buffer_size
< p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
{
p_data->i_overflow_buffer_size
= p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
}
}
p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
if( p_data->p_atomic_operations == NULL )
{
msg_Err( p_this, "out of memory" );
return -1;
}
memset( p_data->p_overflow_buffer, 0 , p_data->i_overflow_buffer_size );
/* end */
return 0;
}
/*****************************************************************************
* OpenFilter
*****************************************************************************/
......@@ -135,23 +393,45 @@ static int OpenFilter( vlc_object_t *p_this )
return VLC_EGENERIC;
}
var_Create( p_this, MONO_CFG "mono-downmix",
VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "mono-downmix" );
var_Create( p_this, MONO_CFG "mono-channel",
VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
p_sys->i_channel_selected =
(unsigned int) var_GetInteger( p_this, MONO_CFG "mono-channel" );
#if 0
p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
#else
p_filter->fmt_out.audio.i_physical_channels =
if( p_sys->b_downmix )
{
p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
}
else
{
p_filter->fmt_out.audio.i_physical_channels =
(AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
#endif
}
p_filter->fmt_out.audio.i_channels = aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
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_in.audio) );
p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
p_sys->i_overflow_buffer_size = 0;
p_sys->p_overflow_buffer = NULL;
p_sys->i_nb_atomic_operations = 0;
p_sys->p_atomic_operations = NULL;
if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
aout_FormatNbChannels( &p_filter->fmt_in.audio ),
p_filter->fmt_in.audio.i_physical_channels,
p_filter->fmt_in.audio.i_rate ) < 0 )
{
return VLC_EGENERIC;
}
p_filter->pf_audio_filter = Convert;
msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
......@@ -174,6 +454,7 @@ static void CloseFilter( vlc_object_t *p_this)
filter_sys_t *p_sys = p_filter->p_sys;
var_Destroy( p_this, MONO_CFG "mono-channel" );
var_Destroy( p_this, MONO_CFG "mono-downmix" );
free( p_sys );
}
......@@ -185,8 +466,8 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block )
aout_filter_t aout_filter;
aout_buffer_t in_buf, out_buf;
block_t *p_out = NULL;
int i_out_size;
unsigned int i_samples;
int i_out_size;
if( !p_block || !p_block->i_samples )
{
......@@ -205,9 +486,8 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block )
p_block->pf_release( p_block );
return NULL;
}
p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
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;
......@@ -236,8 +516,17 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block )
out_buf.i_nb_bytes = p_out->i_buffer;
out_buf.i_nb_samples = p_out->i_samples;
i_samples = stereo_to_mono( (aout_instance_t *)p_filter, &aout_filter,
&out_buf, &in_buf );
if( p_filter->p_sys->b_downmix )
{
memset( out_buf.p_buffer, 0, i_out_size );
stereo2mono_downmix( (aout_instance_t *)p_filter, &aout_filter,
&in_buf, &out_buf );
}
else
{
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;
......@@ -246,9 +535,122 @@ static block_t *Convert( filter_t *p_filter, block_t *p_block )
return p_out;
}
/* stereo_to_mono - mix 2 channels (left,right) into one and play silence on
* all other channels.
/* stereo2mono_downmix - stereo channels into one mono channel.
* Code taken from modules/audio_filter/channel_mixer/headphone.c
* converted from float into int16_t based downmix
* Written by Boris Dorès <babal@via.ecp.fr>
*/
static void stereo2mono_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;
int i_input_nb = aout_FormatNbChannels( &p_filter->input );
int i_output_nb = aout_FormatNbChannels( &p_filter->output );
int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
byte_t * p_out;
byte_t * p_overflow;
byte_t * p_slide;
size_t i_overflow_size; /* in bytes */
size_t i_out_size; /* in bytes */
unsigned int i, j;
int i_source_channel_offset;
int i_dest_channel_offset;
unsigned int i_delay;
double d_amplitude_factor;
/* out buffer characterisitcs */
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
p_out = p_out_buf->p_buffer;
i_out_size = p_out_buf->i_nb_bytes;
if( p_sys != NULL )
{
/* Slide the overflow buffer */
p_overflow = p_sys->p_overflow_buffer;
i_overflow_size = p_sys->i_overflow_buffer_size;
memset( p_out, 0, i_out_size );
if ( i_out_size > i_overflow_size )
memcpy( p_out, p_overflow, i_overflow_size );
else
memcpy( p_out, p_overflow, i_out_size );
p_slide = p_sys->p_overflow_buffer;
while( p_slide < p_overflow + i_overflow_size )
{
if( p_slide + i_out_size < p_overflow + i_overflow_size )
{
memset( p_slide, 0, i_out_size );
if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
memcpy( p_slide, p_slide + i_out_size, i_out_size );
else
memcpy( p_slide, p_slide + i_out_size,
p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
}
else
{
memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
}
p_slide += i_out_size;
}
/* apply the atomic operations */
for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
{
/* shorter variable names */
i_source_channel_offset
= p_sys->p_atomic_operations[i].i_source_channel_offset;
i_dest_channel_offset
= p_sys->p_atomic_operations[i].i_dest_channel_offset;
i_delay = p_sys->p_atomic_operations[i].i_delay;
d_amplitude_factor
= p_sys->p_atomic_operations[i].d_amplitude_factor;
if( p_out_buf->i_nb_samples > i_delay )
{
/* current buffer coefficients */
for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
{
((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
+= p_in[ j * i_input_nb + i_source_channel_offset ]
* d_amplitude_factor;
}
/* overflow buffer coefficients */
for( j = 0; j < i_delay; j++ )
{
((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
+= p_in[ (p_out_buf->i_nb_samples - i_delay + j)
* i_input_nb + i_source_channel_offset ]
* d_amplitude_factor;
}
}
else
{
/* overflow buffer coefficients only */
for( j = 0; j < p_out_buf->i_nb_samples; j++ )
{
((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
* i_output_nb + i_dest_channel_offset ]
+= p_in[ j * i_input_nb + i_source_channel_offset ]
* d_amplitude_factor;
}
}
}
}
else
{
memset( p_out, 0, i_out_size );
}
}
/* Simple stereo to mono mixing. */
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 )
{
......
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