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videolan
vlc-2-2
Commits
c57ef1df
Commit
c57ef1df
authored
May 12, 2011
by
Jean-Paul Saman
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access/alsa.c: Todays capture cards support more than one format for audio capturing.
(cherry picked from commit
fb46b4a1
)
parent
171bdb14
Changes
1
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81 additions
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7 deletions
+81
-7
modules/access/alsa.c
modules/access/alsa.c
+81
-7
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modules/access/alsa.c
View file @
c57ef1df
...
@@ -44,6 +44,8 @@
...
@@ -44,6 +44,8 @@
#include <vlc_access.h>
#include <vlc_access.h>
#include <vlc_demux.h>
#include <vlc_demux.h>
#include <vlc_input.h>
#include <vlc_input.h>
#include <vlc_fourcc.h>
#include <vlc_aout.h>
#include <unistd.h>
#include <unistd.h>
#include <sys/ioctl.h>
#include <sys/ioctl.h>
...
@@ -68,6 +70,10 @@ static void DemuxClose( vlc_object_t * );
...
@@ -68,6 +70,10 @@ static void DemuxClose( vlc_object_t * );
#define STEREO_LONGTEXT N_( \
#define STEREO_LONGTEXT N_( \
"Capture the audio stream in stereo." )
"Capture the audio stream in stereo." )
#define FORMAT_TEXT N_( "Capture format (default s16l)" )
#define FORMAT_LONGTEXT N_( \
"Capture format of audio stream." )
#define SAMPLERATE_TEXT N_( "Samplerate" )
#define SAMPLERATE_TEXT N_( "Samplerate" )
#define SAMPLERATE_LONGTEXT N_( \
#define SAMPLERATE_LONGTEXT N_( \
"Samplerate of the captured audio stream, in Hz (eg: 11025, 22050, 44100, 48000)" )
"Samplerate of the captured audio stream, in Hz (eg: 11025, 22050, 44100, 48000)" )
...
@@ -80,6 +86,23 @@ static void DemuxClose( vlc_object_t * );
...
@@ -80,6 +86,23 @@ static void DemuxClose( vlc_object_t * );
#define ALSA_DEFAULT "hw"
#define ALSA_DEFAULT "hw"
#define CFG_PREFIX "alsa-"
#define CFG_PREFIX "alsa-"
static
const
char
*
const
ppsz_fourcc
[]
=
{
"u8"
,
"s8"
,
"gsm"
,
"u16l"
,
"s16l"
,
"u16b"
,
"s16b"
,
"u24l"
,
"s24l"
,
"u24b"
,
"s24b"
,
"u32l"
,
"s32l"
,
"u32b"
,
"s32b"
,
"f32l"
,
"f32b"
,
"f64l"
,
"f64b"
};
static
const
char
*
const
ppsz_fourcc_text
[]
=
{
N_
(
"PCM U8"
),
N_
(
"PCM S8"
),
N_
(
"GSM Audio"
),
N_
(
"PCM U16 LE"
),
N_
(
"PCM S16 LE"
),
N_
(
"PCM U16 BE"
),
N_
(
"PCM S16 BE"
),
N_
(
"PCM U24 LE"
),
N_
(
"PCM S24 LE"
),
N_
(
"PCM U24 BE"
),
N_
(
"PCM S24 BE"
),
N_
(
"PCM U32 LE"
),
N_
(
"PCM S32 LE"
),
N_
(
"PCM U32 BE"
),
N_
(
"PCM S32 BE"
),
N_
(
"PCM F32 LE"
),
N_
(
"PCM F32 BE"
),
N_
(
"PCM F64 LE"
),
N_
(
"PCM F64 BE"
)
};
vlc_module_begin
()
vlc_module_begin
()
set_shortname
(
N_
(
"ALSA"
)
)
set_shortname
(
N_
(
"ALSA"
)
)
set_description
(
N_
(
"ALSA audio capture input"
)
)
set_description
(
N_
(
"ALSA audio capture input"
)
)
...
@@ -93,6 +116,9 @@ vlc_module_begin()
...
@@ -93,6 +116,9 @@ vlc_module_begin()
add_bool
(
CFG_PREFIX
"stereo"
,
true
,
STEREO_TEXT
,
STEREO_LONGTEXT
,
add_bool
(
CFG_PREFIX
"stereo"
,
true
,
STEREO_TEXT
,
STEREO_LONGTEXT
,
true
)
true
)
add_string
(
CFG_PREFIX
"format"
,
"s16l"
,
FORMAT_TEXT
,
FORMAT_LONGTEXT
,
true
)
change_string_list
(
ppsz_fourcc
,
ppsz_fourcc_text
,
0
)
add_integer
(
CFG_PREFIX
"samplerate"
,
48000
,
SAMPLERATE_TEXT
,
add_integer
(
CFG_PREFIX
"samplerate"
,
48000
,
SAMPLERATE_TEXT
,
SAMPLERATE_LONGTEXT
,
true
)
SAMPLERATE_LONGTEXT
,
true
)
vlc_module_end
()
vlc_module_end
()
...
@@ -116,6 +142,7 @@ struct demux_sys_t
...
@@ -116,6 +142,7 @@ struct demux_sys_t
/* Audio */
/* Audio */
unsigned
int
i_sample_rate
;
unsigned
int
i_sample_rate
;
bool
b_stereo
;
bool
b_stereo
;
vlc_fourcc_t
i_format
;
size_t
i_max_frame_size
;
size_t
i_max_frame_size
;
block_t
*
p_block
;
block_t
*
p_block
;
es_out_id_t
*
p_es
;
es_out_id_t
*
p_es
;
...
@@ -245,6 +272,10 @@ static int DemuxOpen( vlc_object_t *p_this )
...
@@ -245,6 +272,10 @@ static int DemuxOpen( vlc_object_t *p_this )
p_sys
->
p_block
=
NULL
;
p_sys
->
p_block
=
NULL
;
p_sys
->
i_next_demux_date
=
-
1
;
p_sys
->
i_next_demux_date
=
-
1
;
char
*
psz_format
=
var_InheritString
(
p_demux
,
CFG_PREFIX
"format"
);
p_sys
->
i_format
=
vlc_fourcc_GetCodecFromString
(
AUDIO_ES
,
psz_format
);
free
(
psz_format
);
const
char
*
psz_device
=
NULL
;
const
char
*
psz_device
=
NULL
;
if
(
p_demux
->
psz_location
&&
*
p_demux
->
psz_location
)
if
(
p_demux
->
psz_location
&&
*
p_demux
->
psz_location
)
psz_device
=
p_demux
->
psz_location
;
psz_device
=
p_demux
->
psz_location
;
...
@@ -434,6 +465,46 @@ static block_t* GrabAudio( demux_t *p_demux )
...
@@ -434,6 +465,46 @@ static block_t* GrabAudio( demux_t *p_demux )
return
p_block
;
return
p_block
;
}
}
static
snd_pcm_format_t
GetAlsaPCMFormat
(
demux_t
*
p_demux
,
const
vlc_fourcc_t
i_format
)
{
demux_sys_t
*
p_sys
=
p_demux
->
p_sys
;
switch
(
i_format
)
{
case
VLC_CODEC_U8
:
return
SND_PCM_FORMAT_U8
;
case
VLC_CODEC_S8
:
return
SND_PCM_FORMAT_S8
;
case
VLC_CODEC_GSM
:
return
SND_PCM_FORMAT_GSM
;
case
VLC_CODEC_U16L
:
return
SND_PCM_FORMAT_U16_LE
;
case
VLC_CODEC_S16L
:
return
SND_PCM_FORMAT_S16_LE
;
case
VLC_CODEC_U16B
:
return
SND_PCM_FORMAT_U16_BE
;
case
VLC_CODEC_S16B
:
return
SND_PCM_FORMAT_S16_BE
;
case
VLC_CODEC_U24L
:
return
SND_PCM_FORMAT_U24_3LE
;
case
VLC_CODEC_S24L
:
return
SND_PCM_FORMAT_S24_3LE
;
case
VLC_CODEC_U24B
:
return
SND_PCM_FORMAT_U24_3BE
;
case
VLC_CODEC_S24B
:
return
SND_PCM_FORMAT_S24_3BE
;
case
VLC_CODEC_U32L
:
return
SND_PCM_FORMAT_U32_LE
;
case
VLC_CODEC_U32B
:
return
SND_PCM_FORMAT_U32_BE
;
case
VLC_CODEC_S32L
:
return
SND_PCM_FORMAT_S32_LE
;
case
VLC_CODEC_S32B
:
return
SND_PCM_FORMAT_S32_BE
;
case
VLC_CODEC_F32L
:
return
SND_PCM_FORMAT_FLOAT_LE
;
case
VLC_CODEC_F32B
:
return
SND_PCM_FORMAT_FLOAT_BE
;
case
VLC_CODEC_F64L
:
return
SND_PCM_FORMAT_FLOAT64_LE
;
case
VLC_CODEC_F64B
:
return
SND_PCM_FORMAT_FLOAT64_BE
;
default:
msg_Err
(
p_demux
,
"ALSA: unsupported sample format '%s' falling back to 's16l'"
,
(
const
char
*
)
&
i_format
);
p_sys
->
i_format
=
VLC_CODEC_S16L
;
}
return
SND_PCM_FORMAT_S16_LE
;
}
/*****************************************************************************
/*****************************************************************************
* OpenAudioDev: open and set up the audio device and probe for capabilities
* OpenAudioDev: open and set up the audio device and probe for capabilities
*****************************************************************************/
*****************************************************************************/
...
@@ -442,6 +513,7 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
...
@@ -442,6 +513,7 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
demux_sys_t
*
p_sys
=
p_demux
->
p_sys
;
demux_sys_t
*
p_sys
=
p_demux
->
p_sys
;
p_sys
->
p_alsa_pcm
=
NULL
;
p_sys
->
p_alsa_pcm
=
NULL
;
snd_pcm_hw_params_t
*
p_hw_params
=
NULL
;
snd_pcm_hw_params_t
*
p_hw_params
=
NULL
;
snd_pcm_format_t
i_alsa_pcm_format
;
snd_pcm_uframes_t
buffer_size
;
snd_pcm_uframes_t
buffer_size
;
snd_pcm_uframes_t
chunk_size
;
snd_pcm_uframes_t
chunk_size
;
...
@@ -489,8 +561,9 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
...
@@ -489,8 +561,9 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
goto
adev_fail
;
goto
adev_fail
;
}
}
/* Set 16 bit little endian */
/* Set capture format, default is signed 16 bit little endian */
if
(
(
i_err
=
snd_pcm_hw_params_set_format
(
p_sys
->
p_alsa_pcm
,
p_hw_params
,
SND_PCM_FORMAT_S16_LE
)
)
<
0
)
i_alsa_pcm_format
=
GetAlsaPCMFormat
(
p_demux
,
p_sys
->
i_format
);
if
(
(
i_err
=
snd_pcm_hw_params_set_format
(
p_sys
->
p_alsa_pcm
,
p_hw_params
,
i_alsa_pcm_format
)
)
<
0
)
{
{
msg_Err
(
p_demux
,
"ALSA: cannot set sample format (%s)"
,
msg_Err
(
p_demux
,
"ALSA: cannot set sample format (%s)"
,
snd_strerror
(
i_err
)
);
snd_strerror
(
i_err
)
);
...
@@ -572,7 +645,7 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
...
@@ -572,7 +645,7 @@ static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
goto
adev_fail
;
goto
adev_fail
;
}
}
int
bits_per_sample
=
snd_pcm_format_physical_width
(
SND_PCM_FORMAT_S16_LE
);
int
bits_per_sample
=
snd_pcm_format_physical_width
(
i_alsa_pcm_format
);
int
bits_per_frame
=
bits_per_sample
*
channels
;
int
bits_per_frame
=
bits_per_sample
*
channels
;
p_sys
->
i_alsa_chunk_size
=
chunk_size
;
p_sys
->
i_alsa_chunk_size
=
chunk_size
;
...
@@ -611,16 +684,17 @@ static int OpenAudioDev( demux_t *p_demux, const char *psz_device )
...
@@ -611,16 +684,17 @@ static int OpenAudioDev( demux_t *p_demux, const char *psz_device )
if
(
OpenAudioDevAlsa
(
p_demux
,
psz_device
)
!=
VLC_SUCCESS
)
if
(
OpenAudioDevAlsa
(
p_demux
,
psz_device
)
!=
VLC_SUCCESS
)
return
VLC_EGENERIC
;
return
VLC_EGENERIC
;
msg_Dbg
(
p_demux
,
"opened adev=`%s' %s %dHz"
,
msg_Dbg
(
p_demux
,
"opened adev=`%s' %s %dHz
codec '%s'
"
,
psz_device
,
p_sys
->
b_stereo
?
"stereo"
:
"mono"
,
psz_device
,
p_sys
->
b_stereo
?
"stereo"
:
"mono"
,
p_sys
->
i_sample_rate
);
p_sys
->
i_sample_rate
,
vlc_fourcc_GetDescription
(
AUDIO_ES
,
p_sys
->
i_format
)
);
es_format_t
fmt
;
es_format_t
fmt
;
es_format_Init
(
&
fmt
,
AUDIO_ES
,
VLC_FOURCC
(
'a'
,
'r'
,
'a'
,
'w'
)
);
es_format_Init
(
&
fmt
,
AUDIO_ES
,
p_sys
->
i_format
);
fmt
.
audio
.
i_channels
=
p_sys
->
b_stereo
?
2
:
1
;
fmt
.
audio
.
i_channels
=
p_sys
->
b_stereo
?
2
:
1
;
fmt
.
audio
.
i_rate
=
p_sys
->
i_sample_rate
;
fmt
.
audio
.
i_rate
=
p_sys
->
i_sample_rate
;
fmt
.
audio
.
i_bitspersample
=
16
;
fmt
.
audio
.
i_bitspersample
=
aout_BitsPerSample
(
p_sys
->
i_format
)
;
fmt
.
audio
.
i_blockalign
=
fmt
.
audio
.
i_channels
*
fmt
.
audio
.
i_bitspersample
/
8
;
fmt
.
audio
.
i_blockalign
=
fmt
.
audio
.
i_channels
*
fmt
.
audio
.
i_bitspersample
/
8
;
fmt
.
i_bitrate
=
fmt
.
audio
.
i_channels
*
fmt
.
audio
.
i_rate
*
fmt
.
audio
.
i_bitspersample
;
fmt
.
i_bitrate
=
fmt
.
audio
.
i_channels
*
fmt
.
audio
.
i_rate
*
fmt
.
audio
.
i_bitspersample
;
...
...
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